Re: [sipx-users] Dial Plan Issue
Is the call an intra or inter domain call? On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote: The 12xx gateway has a gateway setup for 37xx and the appropriate dial-plan. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:26 AM does the other system also have a gateway from your system? if not, it might not pass a check which shows it is an allowed call in that manner. On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.org wrote: All, Having a newly found problem where we have assigned 12XX to a location, server IP is 172.16.20.8. From our main office at 192.168.1.8, calls destined to 12XX seem to route to that server only. Dialplan says 12 and 2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4 server throws a 500 internal server error. Sipx-trace is attached. Any suggestions or help is appreciated and to make sure it wasn't 12XX messing it up, I created a dialplan of 2300 to dial to 1250 at the distant end and still get a 500 internal server error. I put the dial plans all the way at the top and still no change, its like its trying to route it within its self. Maybe a second set of eyes will catch something I'm missing? Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Dial Plan Issue
Thats not a valid answer. Your answer should be in the form of either intra or inter. (Inside the same sipdomain--intra, or between two different sip domains--inter) Can you call from a UA to the same destination in the same format? ie.e. 1234@ip? Is the dialplan entry CUSTOM or SITE TO SITE (should be site to site using sipdomain name) On Tue, Apr 24, 2012 at 10:48 AM, Aaron Pursell aar...@esgw.org wrote: Yes Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM Is the call an intra or inter domain call? On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote: The 12xx gateway has a gateway setup for 37xx and the appropriate dial-plan. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:26 AM does the other system also have a gateway from your system? if not, it might not pass a check which shows it is an allowed call in that manner. On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.org wrote: All, Having a newly found problem where we have assigned 12XX to a location, server IP is 172.16.20.8. From our main office at 192.168.1.8, calls destined to 12XX seem to route to that server only. Dialplan says 12 and 2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4 server throws a 500 internal server error. Sipx-trace is attached. Any suggestions or help is appreciated and to make sure it wasn't 12XX messing it up, I created a dialplan of 2300 to dial to 1250 at the distant end and still get a 500 internal server error. I put the dial plans all the way at the top and still no change, its like its trying to route it within its self. Maybe a second set of eyes will catch something I'm missing? Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Dial Plan Issue
did you try calling from a UA at one end in the same form? 1234@ipaddress) to reach the foreign system? On Tue, Apr 24, 2012 at 11:06 AM, Aaron Pursell aar...@esgw.org wrote: Inter, its a custom but I've tried it every way I can think of. So back to the drawing board. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:55 AM Thats not a valid answer. Your answer should be in the form of either intra or inter. (Inside the same sipdomain--intra, or between two different sip domains--inter) Can you call from a UA to the same destination in the same format? ie.e. 1234@ip? Is the dialplan entry CUSTOM or SITE TO SITE (should be site to site using sipdomain name) On Tue, Apr 24, 2012 at 10:48 AM, Aaron Pursell aar...@esgw.org wrote: Yes Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM Is the call an intra or inter domain call? On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote: The 12xx gateway has a gateway setup for 37xx and the appropriate dial-plan. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:26 AM does the other system also have a gateway from your system? if not, it might not pass a check which shows it is an allowed call in that manner. On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.org wrote: All, Having a newly found problem where we have assigned 12XX to a location, server IP is 172.16.20.8. From our main office at 192.168.1.8, calls destined to 12XX seem to route to that server only. Dialplan says 12 and 2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4 server throws a 500 internal server error. Sipx-trace is attached. Any suggestions or help is appreciated and to make sure it wasn't 12XX messing it up, I created a dialplan of 2300 to dial to 1250 at the distant end and still get a 500 internal server error. I put the dial plans all the way at the top and still no change, its like its trying to route it within its self. Maybe a second set of eyes will catch something I'm missing? Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http
Re: [sipx-users] Fax Service Query Question
its in the alias file, see: /var/sipxdata/sipdb/alias.xml see also your last query about this on March 21 for the same information. See also http://track.sipfoundry.org/browse/XX-10127 On Tue, Apr 24, 2012 at 11:06 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: I am in the need to query the sipXecs system of the fax Service DID/Alias numbers that I have entered for my users. If I use search I am unable to locate the user account with the DID/Alias assignment. The information is not available in the export report as well. ** ** If I grep the validusers.xml file that information is not listed. ** ** Does anyone have other suggestions? ** ** Thanks Everyone ** ** Rob -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Dial Plan Issue
is it an internal (vpn or locally connected) call? if it is, do the subnets on each side reflect a local subnet (to not invoke media relay)? i am thinking if each system can see the other system and look it up via dns and resolve the SRV to the local address or use a simple gateway method (not as desirable, IMO), you probably forgot to add the subnets for the other systems at the other end in each system... if so, thats an oopsie on you. On Tue, Apr 24, 2012 at 11:12 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: did you try calling from a UA at one end in the same form? 1234@ipaddress) to reach the foreign system? On Tue, Apr 24, 2012 at 11:06 AM, Aaron Pursell aar...@esgw.org wrote: Inter, its a custom but I've tried it every way I can think of. So back to the drawing board. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:55 AM Thats not a valid answer. Your answer should be in the form of either intra or inter. (Inside the same sipdomain--intra, or between two different sip domains--inter) Can you call from a UA to the same destination in the same format? ie.e. 1234@ip? Is the dialplan entry CUSTOM or SITE TO SITE (should be site to site using sipdomain name) On Tue, Apr 24, 2012 at 10:48 AM, Aaron Pursell aar...@esgw.org wrote: Yes Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM Is the call an intra or inter domain call? On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote: The 12xx gateway has a gateway setup for 37xx and the appropriate dial-plan. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:26 AM does the other system also have a gateway from your system? if not, it might not pass a check which shows it is an allowed call in that manner. On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.org wrote: All, Having a newly found problem where we have assigned 12XX to a location, server IP is 172.16.20.8. From our main office at 192.168.1.8, calls destined to 12XX seem to route to that server only. Dialplan says 12 and 2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4 server throws a 500 internal server error. Sipx-trace is attached. Any suggestions or help is appreciated and to make sure it wasn't 12XX messing it up, I created a dialplan of 2300 to dial to 1250 at the distant end and still get a 500 internal server error. I put the dial plans all the way at the top and still no change, its like its trying to route it within its self. Maybe a second set of eyes will catch something I'm missing? Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users
Re: [sipx-users] Dial Plan Issue
Can you duplex care if from a non-dialplan issue? Phone dials 123@IP If not then maybe the con is not actually allowing the traffic. I have ipsec Vpn connections but then I have to allow the traffic (filter) to actually use it. Can sipx1 trace route to sipx2, vice versa. On Apr 24, 2012 12:02 PM, Aaron Pursell aar...@esgw.org wrote: VPN. We use standard internal subnets 172.16 and 192.168 for all networks. All the subnets are there, but it is what it is, the call on the 192.168.1.x network doesn't even leave our sip server even though the dial plan says shoot it over to 172.16.20.8. I'm sure its an mistake on our end (obviously) however, I don't think I'll find any more insight here than what has already been given. All I know is the server the calls originate from show a 500 internal server error when dialing, where as when you dial one of the other VPN sites, you see it route it over to the other site error free. There is no difference in the cisco configs between the sites other than information relevant to their link. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 9:29 AM is it an internal (vpn or locally connected) call? if it is, do the subnets on each side reflect a local subnet (to not invoke media relay)? i am thinking if each system can see the other system and look it up via dns and resolve the SRV to the local address or use a simple gateway method (not as desirable, IMO), you probably forgot to add the subnets for the other systems at the other end in each system... if so, thats an oopsie on you. On Tue, Apr 24, 2012 at 11:12 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: did you try calling from a UA at one end in the same form? 1234@ipaddress) to reach the foreign system? On Tue, Apr 24, 2012 at 11:06 AM, Aaron Pursell aar...@esgw.org wrote: Inter, its a custom but I've tried it every way I can think of. So back to the drawing board. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:55 AM Thats not a valid answer. Your answer should be in the form of either intra or inter. (Inside the same sipdomain--intra, or between two different sip domains--inter) Can you call from a UA to the same destination in the same format? ie.e. 1234@ip? Is the dialplan entry CUSTOM or SITE TO SITE (should be site to site using sipdomain name) On Tue, Apr 24, 2012 at 10:48 AM, Aaron Pursell aar...@esgw.org wrote: Yes Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM Is the call an intra or inter domain call? On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.orgwrote: The 12xx gateway has a gateway setup for 37xx and the appropriate dial-plan. Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:26 AM does the other system also have a gateway from your system? if not, it might not pass a check which shows it is an allowed call in that manner. On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.orgwrote: All, Having a newly found problem where we have assigned 12XX to a location, server IP is 172.16.20.8. From our main office at 192.168.1.8, calls destined to 12XX seem to route to that server only. Dialplan says 12 and 2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4 server throws a 500 internal server error. Sipx-trace is attached. Any suggestions or help is appreciated and to make sure it wasn't 12XX messing it up, I created a dialplan of 2300 to dial to 1250 at the distant end and still get a 500 internal server error. I put the dial plans all the way at the top and still no change, its like its trying to route it within its self. Maybe a second set of eyes will catch something I'm missing? Aaron Pursell Network Systems Administration Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http
Re: [sipx-users] Fwd: Grandstream GXW4104
I would use the model 4524 instead of 4114. On Apr 24, 2012 2:48 PM, Bryan Anderson branderso...@msn.com wrote: Sorry I sent this yesterday from the wrong email account. Since I tried sending that though, It looks like we might be going with a patton. I was looking at the SN4114/JO/EUI? Is that a good model to go with for a 4port FXO or is there any issues with this one? Thanks, Bryan Thank you for your responses. It was not my choice to go with the grandstream, it was what was given to me for the office. I will attach a sip trace for a call where I called in, and wasn't able to be transferred from the front desk. I have noticed that when the calls stop transferring the gateway's web interface shows no calls on the lines, but the telnet interface show's all 8 channels use, but it is a 4 line FXO. The gateway isn't releasing the channels, and sends a No channels available message. I have sent this same trace to grandstream and am awaiting a reply. Not all calls get stuck just some, like my cell phone, but I don't know the other numbers. I did notice some past posts of people using this gateway previously but none of them have said anything as to if they still use it or not. I realize this is not an optimal gateway but have been told to do everything I can and have a clear reason why it doesn't work before they will go with a Patton. I did notice the Soundpoint 331's are using the 3.3.3 firmware which I believe I have been seeing some concerns about, these arrived last week with that firmware installed. -Bryan Anderson On Mon, Apr 23, 2012 at 2:00 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: Really the best thing you can do is put your log with sipx (proxy) to debug, and grab whatever best level of detail/logging you can from your gateway. I don't think this happens with others and people probably arent answering you because either it doesnt work well for them or the MFR simply doesnt provide an adequate sip stack or support. If you see something in the logs, post it here, but you need to discern WHERE the BYE is coming from. Since the RTP is established between the UA (phone) and the gateway, sipx is mostly out of the picture except recording the BYE to cut the CRD record. This is why it is important to use a good network infrastructure along with the gateway and handset, of course. There are a couple of easy gateways to use: AudioCodes and Patton. For less detailed configuration options and ease of configuration a lot of people choose Audiocodes. (not me). Good luck. 2012/4/23 Nitin Mirchandani nitin_mirchand...@hotmail.com I have one suggestion for you - Dont use Grandstream. I dont know which stack they use - But be it gateway or phone - Its simply unstable (gave up trying) -- Date: Mon, 23 Apr 2012 11:54:14 -0700 From: branderso...@msn.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Grandstream GXW4104 Could Problem number two be caused by incorrect Refresher, or timer settings? If so, what should they be? On the gateway: *Session Expiration: * (in seconds. default 180 seconds) * Min-SE: * (in seconds. default and minimum 90 seconds) * Caller Request Timer: * Yes No (Request for timer when making outbound calls) *Callee Request Timer: * Yes No (When caller supports timer but did not request one) * Force Timer: * Yes No (Use timer even when remote party does not support) *UAC Specify Refresher: * UAC UAS Omit (Recommended) * UAS Specify Refresher: * UAC UAS (When UAC did not specify refresher tag) -Bryan Anderson On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson branderso...@msn.comwrote: I have been having issues with a new Grandstream GXW4104 fxo gateway and was wondering if anyone could help. We have 4 pstn lines from qwest going into the gateway. All calls go to an Auto Attendant when answered. the two problems we have experienced are: 1) After about 1-1.5 hours the call hit the Auto Attendant but wont transfer out. Some dials and extension they just get dead air. (this is fixed by rebooting the gateway.) 2) The external uses (either some one who called it, or some one we have called) stop hearing audio, but we can still here them. This happens anywhere from 1-10 minutes into the call. sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1) Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4 Boot--1.1.3.2 The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331 -Bryan Anderson ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users
Re: [sipx-users] Fwd: Grandstream GXW4104
(4500 series have two ethernet portsand are typically more flexible in the event a basic sip connection has to be used from it (i.e. some people use them to connect one ethernet port to a sip provider as it has a nat function and bring in trunks for non-sipx related stuff too). Unless you have a need for more than 4 FXO ports, the 4500 series is the way to go (IMO). On Tue, Apr 24, 2012 at 3:07 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: I would use the model 4524 instead of 4114. On Apr 24, 2012 2:48 PM, Bryan Anderson branderso...@msn.com wrote: Sorry I sent this yesterday from the wrong email account. Since I tried sending that though, It looks like we might be going with a patton. I was looking at the SN4114/JO/EUI? Is that a good model to go with for a 4port FXO or is there any issues with this one? Thanks, Bryan Thank you for your responses. It was not my choice to go with the grandstream, it was what was given to me for the office. I will attach a sip trace for a call where I called in, and wasn't able to be transferred from the front desk. I have noticed that when the calls stop transferring the gateway's web interface shows no calls on the lines, but the telnet interface show's all 8 channels use, but it is a 4 line FXO. The gateway isn't releasing the channels, and sends a No channels available message. I have sent this same trace to grandstream and am awaiting a reply. Not all calls get stuck just some, like my cell phone, but I don't know the other numbers. I did notice some past posts of people using this gateway previously but none of them have said anything as to if they still use it or not. I realize this is not an optimal gateway but have been told to do everything I can and have a clear reason why it doesn't work before they will go with a Patton. I did notice the Soundpoint 331's are using the 3.3.3 firmware which I believe I have been seeing some concerns about, these arrived last week with that firmware installed. -Bryan Anderson On Mon, Apr 23, 2012 at 2:00 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: Really the best thing you can do is put your log with sipx (proxy) to debug, and grab whatever best level of detail/logging you can from your gateway. I don't think this happens with others and people probably arent answering you because either it doesnt work well for them or the MFR simply doesnt provide an adequate sip stack or support. If you see something in the logs, post it here, but you need to discern WHERE the BYE is coming from. Since the RTP is established between the UA (phone) and the gateway, sipx is mostly out of the picture except recording the BYE to cut the CRD record. This is why it is important to use a good network infrastructure along with the gateway and handset, of course. There are a couple of easy gateways to use: AudioCodes and Patton. For less detailed configuration options and ease of configuration a lot of people choose Audiocodes. (not me). Good luck. 2012/4/23 Nitin Mirchandani nitin_mirchand...@hotmail.com I have one suggestion for you - Dont use Grandstream. I dont know which stack they use - But be it gateway or phone - Its simply unstable (gave up trying) -- Date: Mon, 23 Apr 2012 11:54:14 -0700 From: branderso...@msn.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Grandstream GXW4104 Could Problem number two be caused by incorrect Refresher, or timer settings? If so, what should they be? On the gateway: *Session Expiration: * (in seconds. default 180 seconds) * Min-SE: * (in seconds. default and minimum 90 seconds) * Caller Request Timer: * Yes No (Request for timer when making outbound calls) *Callee Request Timer: * Yes No (When caller supports timer but did not request one) * Force Timer: * Yes No (Use timer even when remote party does not support) *UAC Specify Refresher: * UAC UAS Omit (Recommended) * UAS Specify Refresher: * UAC UAS (When UAC did not specify refresher tag) -Bryan Anderson On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson branderso...@msn.comwrote: I have been having issues with a new Grandstream GXW4104 fxo gateway and was wondering if anyone could help. We have 4 pstn lines from qwest going into the gateway. All calls go to an Auto Attendant when answered. the two problems we have experienced are: 1) After about 1-1.5 hours the call hit the Auto Attendant but wont transfer out. Some dials and extension they just get dead air. (this is fixed by rebooting the gateway.) 2) The external uses (either some one who called it, or some one we have called) stop hearing audio, but we can still here them. This happens anywhere from 1-10 minutes into the call. sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1) Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4
[Assp-test] assp-white , assp-spam 1.9.6.8(0.0.04)
Hello when I send request to assp-white@ I receive an email with an empty body message. I have seen the problem also using assp-spam@ . Thank you Graziano -- Live Security Virtual Conference Exclusive live event will cover all the ways today's security and threat landscape has changed and how IT managers can respond. Discussions will include endpoint security, mobile security and the latest in malware threats. http://www.accelacomm.com/jaw/sfrnl04242012/114/50122263/ ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
Re: [sipx-dev] Paging Group
Does yum info sip page Return anything? Is this a clean install or update? On Apr 23, 2012 8:03 AM, Kumaran thiru.venkateshwa...@ttplservices.com wrote: Douglas Hubler wrote: On Mon, Apr 23, 2012 at 6:11 AM, Kumaran thiru.venkateshwa...@ttplservices.com wrote: I'm getting following message while installing sipxpage after yum clean all Setting up Install Process No package sipxpage available. Error: Nothing to do try this first yum clean all I'm not sure what the default cache expire time is of yum. I tried yum clean all first then tried to install sipxpage... Regards, Kumaran T ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-users] Grandstream GXW4104
Really the best thing you can do is put your log with sipx (proxy) to debug, and grab whatever best level of detail/logging you can from your gateway. I don't think this happens with others and people probably arent answering you because either it doesnt work well for them or the MFR simply doesnt provide an adequate sip stack or support. If you see something in the logs, post it here, but you need to discern WHERE the BYE is coming from. Since the RTP is established between the UA (phone) and the gateway, sipx is mostly out of the picture except recording the BYE to cut the CRD record. This is why it is important to use a good network infrastructure along with the gateway and handset, of course. There are a couple of easy gateways to use: AudioCodes and Patton. For less detailed configuration options and ease of configuration a lot of people choose Audiocodes. (not me). Good luck. 2012/4/23 Nitin Mirchandani nitin_mirchand...@hotmail.com I have one suggestion for you - Dont use Grandstream. I dont know which stack they use - But be it gateway or phone - Its simply unstable (gave up trying) -- Date: Mon, 23 Apr 2012 11:54:14 -0700 From: branderso...@msn.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Grandstream GXW4104 Could Problem number two be caused by incorrect Refresher, or timer settings? If so, what should they be? On the gateway: *Session Expiration: * (in seconds. default 180 seconds) * Min-SE: * (in seconds. default and minimum 90 seconds) * Caller Request Timer: * Yes No (Request for timer when making outbound calls) *Callee Request Timer: * Yes No (When caller supports timer but did not request one) * Force Timer: * Yes No (Use timer even when remote party does not support) *UAC Specify Refresher: * UAC UAS Omit (Recommended) * UAS Specify Refresher: * UAC UAS (When UAC did not specify refresher tag) -Bryan Anderson On Wed, Apr 18, 2012 at 10:37 AM, Bryan Anderson branderso...@msn.comwrote: I have been having issues with a new Grandstream GXW4104 fxo gateway and was wondering if anyone could help. We have 4 pstn lines from qwest going into the gateway. All calls go to an Auto Attendant when answered. the two problems we have experienced are: 1) After about 1-1.5 hours the call hit the Auto Attendant but wont transfer out. Some dials and extension they just get dead air. (this is fixed by rebooting the gateway.) 2) The external uses (either some one who called it, or some one we have called) stop hearing audio, but we can still here them. This happens anywhere from 1-10 minutes into the call. sipXecs (4.4.0- 2011-08-14EDT19:47:12 domU-12-31-39-04-D4-A1) Grandstream firmware: Program--1.3.4.13Loader--1.1.3.4Boot--1.1.3.2 The phones are 1 Polycom IP 550 and 7 Polycom SoundPoint IP 331 -Bryan Anderson ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[Assp-test] UseTrapToCollect
Hello in DoPenaltyMakeTraps description there is a reference to UseTrapToCollect but there is no trace of UseTrapToCollect in ASSP . If UseTrapToCollect is also set they will work like spamaddresses and collect the mails. If set to 'use for validation' all entries regardless of their frequency will be used to validate incoming addresses. Note: LocalAddresses_Flat or DoLDAP or DoVRFY must be enabled. Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
Re: [sipx-users] Trunk to Trunk Transfer
The issue is with the trunk connection. Re-read my post. On Apr 21, 2012 8:34 PM, Tommy Laino tomla...@gmail.com wrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: camgknjuupybrra7jvsgxx7mopvvxzva9em-pcgx0l_wamww...@mail.gmail.com X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67744 Message-ID: 108a0.4f935...@forum.sipfoundry.org OK Tony I tried forwarding to an extension that is not forwarded and I get the same result. So to update my issue, every transfer from the AA fails. Internal and remote extension and external numbers. -- Tommy Laino Dome Technologies ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Fax Problems (Big Time)
faxing into sipx only support t.38 and your audiocodes fax settings have g711u. you need to change it to t.38. On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Has anyone had problems with faxing into sipXecs? We are having huge problems with faxes being delivered to the users emails as zero page tiff’s. It is not an every fax occurrence however it is at a rate that I am no longer able to rely upon the faxing delivery capabilities of sipXecs. I am trying to determine if the problems is with the AudioCodes, PRI Provider or sipXecs. ** ** Just wondering if there is a known issue. I am running (4.4.0- 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with an AudioCodes Mediant 1000 SIP Gateway. ** ** Listed below is my current configuration settings on the AudioCodes! ** ** *AudioCodes Information* Version ID: 6.00A.032.003 DSP Type: 2 DSP Software Version: 6008 DSP Software Name: 624AE3 Flash Version: 208 ** ** AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings ** ** *General Settings* Fax Transport Mode: RelayEnable Caller ID Transport Type: Mute Caller ID Type: Standard Bellcore V.21 Modem Transport Type: Disable V.22 Modem Transport Type: Disable V.23 Modem Transport Type: Disable V.32 Modem Transport Type: Disable V.34 Modem Transport Type: Enable Bypass Fax CNG Mode: Disable CNG Detector Mode: Disable ** ** *Fax Relay Settings* Fax Relay Redundancy Depth: 0 Fax Relay Enhanced Redundancy Depth: 4 Fax Relay ECM Enable: Enable Fax Relay Max Rate (bps): 9600bps ** ** *Bypass Settings* Fax/Modem Bypass Coder Type: G711Mulaw Fax/Modem Bypass Packing Factor: 1 Fax Bypass Output Gain: 0 Modem Bypass Output Gain: 0 ** ** Any suggestions from the techs would be very welcomed! ** ** Thanks Everyone, ** ** Rob Schroeder -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bria for iphone/ipod touch registration
Please understand that sipxecs is meant for enterprise deployments. If it were meant for smaller deployments as its market segment things would probably be designed very differently, hence the prerequisite reading and architectural understanding. It is not aimed at users either, it's aimed at densely populated environments with a well-rounded IT staff. It is not for everybody. With some basic's learned though, it's easy enough to deploy if you fully understand the concepts, but sidestepping fundementals does noone any service. On Fri, Apr 20, 2012 at 10:36 AM, Stiles Watson wat...@datatek-net.comwrote: Tony, The original intent of the email was to offer some constructive criticism and explain a little of my confusion. I never intended to make it personal. I realize that my never let go of that bone comment made it personal, and for that, I apologize. You guys have a superior product and I hope it continues to improve and continues to be used by more companies. Thank you, Tony, and everyone else, for your help. Stiles On 04/19/2012 01:31 PM, Tony Graziano wrote: On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wat...@datatek-net.comwrote: On 04/18/2012 06:15 PM, Tony Graziano wrote: You shouldn't invite violence. I have been known to clobber people with iPads. Why am I not surprised by this? Um that was a cartoon reference dude, you take things wayyy to seriously. Obviously you don't get out too much. I think the wiki could be clearer, but really I think you are the only one to make the leap from Bria is bria is bria is bria. It ain't, but sipfoundry doesn't manuafcture it either. Bria has undergone numerous name changes in the last 3 or four years. I look for the names to change yet again, because why leave something the same? I am not the only one who has been confused on this point. I have all the emails from this list for the past year and a half or more. There have been several people who have been confused about this. I know sipfoundry does not manufacturer Bria, but you are saying that it is supported by your software and therefore it is to your advantage to help your users/customers use your product with whatever different versions exist of the third party software you say you are interoperable with. Thanks for making my point about not taking a users view of things. Wrong, the wiki was changed by multiple editors to leave no chance for stiles.user error.. It clearly stated Bria 3.x, but the ASSUMPTION you had was all BRIA is BRIA. Whatever. It got changed to quell your issue. You didn't like the clarity of the wiki, so now it's idiot proffed (maybe) and you still want to complain about it. You guys do realize that the people that use your product are your customers, right? I know no money changes hands, but you are creating this for people to use, correct? You want companies big AND small to embrace your product, correct? There are a number of potential users/customers who have left your product because they could not get the help they thought they needed. Its a very friendly environment. The wiki is there for a reason. There are many people who try to do things that LACK THE SKILLSET and/or FAIL TO READ OR UNDERSTAND what the best approach is to installing/configuring a system and then expect wayyy to much in having community members send them the wiki pages and explain the basic concepts. There are skills required as a prerequisite. Give people more information than they need? That's been done, numerous times. When they can't follow directions the first few times, you have to beat them over the head with an iPad and chant DNS DNS DNS. I love how you never let go of that bone! You are right, DNS is very important and I have some things which still need to be configured. The problem is that I do not think the DNS wiki page is perfectly clear and I have been give contradictory info on this forum - again, its free support offered by the sipX community which includes old hats, newbies, developers and users so that happens. I was told that sipX sets up everything correctly and that I do not need external DNS SRV records, but that was not correct. It makes perfect sense now that I would need them. I think it was correct in the context or your related question(s) at that time, but clearly you want to pick yet another bone. Bringing this up out of context is just sour grapes and not really fair. Understand then fix your DNS and stop ranting to the masses who already get that. Ahh, just Understand. It is easy to wave that wand when it is something you are very familiar with. You might be one of those very gifted people who instantly understands everything the first time they see it. I'm happy for you - really. However, on the whole, God has gifted different people with different abilities. The thing which is easy for one, may very very difficult for another. DNS, DNS, DNS! What
Re: [sipx-users] Fax Problems (Big Time)
Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways for use with t.38 in the wiki. Perhaps someone else would be so kind as to do the same for an AC PRI gateway instead of keeping it a big secret and make sure it is also documented on the wiki. Try: Configuration Protocol Configuration Protocol Definition SIP General Parameters Fax Signaling Method T.38 Relay Detect Fax on Answer Tone Initiate T.38 on Preamble SIP Transport Type UDP I don't know what the other settings are for Detect Fax on Answer Tone, you really should address this before logging and tracing, because it should have fax set for t.38 on the Mediant FIRST. For sanity, when you do find Nirvana, please post it here for someone to update the wiki if you don't have a wiki account. On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Tony: ** ** I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile Definitions – IP Profile Settings - Gateway Parameters – Fax Signaling Method. ** ** ** ** ** ** *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano *Sent:* Friday, April 20, 2012 9:29 AM *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] Fax Problems (Big Time) ** ** faxing into sipx only support t.38 and your audiocodes fax settings have g711u. you need to change it to t.38. On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Has anyone had problems with faxing into sipXecs? We are having huge problems with faxes being delivered to the users emails as zero page tiff’s. It is not an every fax occurrence however it is at a rate that I am no longer able to rely upon the faxing delivery capabilities of sipXecs. I am trying to determine if the problems is with the AudioCodes, PRI Provider or sipXecs. Just wondering if there is a known issue. I am running (4.4.0- 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with an AudioCodes Mediant 1000 SIP Gateway. Listed below is my current configuration settings on the AudioCodes! *AudioCodes Information* Version ID: 6.00A.032.003 DSP Type: 2 DSP Software Version: 6008 DSP Software Name: 624AE3 Flash Version: 208 AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings *General Settings* Fax Transport Mode: RelayEnable Caller ID Transport Type: Mute Caller ID Type: Standard Bellcore V.21 Modem Transport Type: Disable V.22 Modem Transport Type: Disable V.23 Modem Transport Type: Disable V.32 Modem Transport Type: Disable V.34 Modem Transport Type: Enable Bypass Fax CNG Mode: Disable CNG Detector Mode: Disable *Fax Relay Settings* Fax Relay Redundancy Depth: 0 Fax Relay Enhanced Redundancy Depth: 4 Fax Relay ECM Enable: Enable Fax Relay Max Rate (bps): 9600bps *Bypass Settings* Fax/Modem Bypass Coder Type: G711Mulaw Fax/Modem Bypass Packing Factor: 1 Fax Bypass Output Gain: 0 Modem Bypass Output Gain: 0 Any suggestions from the techs would be very welcomed! Thanks Everyone, Rob Schroeder ** ** -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ** ** -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net ** ** Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain
Re: [sipx-users] Fax Problems (Big Time)
did you do that? if so, did it fix it? On Fri, Apr 20, 2012 at 11:53 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Tony, ** ** Thank you for the suggestion and for your help. ** ** Rob ** ** *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano *Sent:* Friday, April 20, 2012 11:00 AM *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] Fax Problems (Big Time) ** ** Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways for use with t.38 in the wiki. Perhaps someone else would be so kind as to do the same for an AC PRI gateway instead of keeping it a big secret and make sure it is also documented on the wiki. ** ** Try: ** ** Configuration Protocol Configuration Protocol Definition SIP General Parameters Fax Signaling Method T.38 Relay Detect Fax on Answer Tone Initiate T.38 on Preamble SIP Transport Type UDP ** ** I don't know what the other settings are for Detect Fax on Answer Tone, you really should address this before logging and tracing, because it should have fax set for t.38 on the Mediant FIRST. ** ** For sanity, when you do find Nirvana, please post it here for someone to update the wiki if you don't have a wiki account. ** ** On Fri, Apr 20, 2012 at 9:50 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Tony: I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile Definitions – IP Profile Settings - Gateway Parameters – Fax Signaling Method. *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano *Sent:* Friday, April 20, 2012 9:29 AM *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] Fax Problems (Big Time) faxing into sipx only support t.38 and your audiocodes fax settings have g711u. you need to change it to t.38. On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: Has anyone had problems with faxing into sipXecs? We are having huge problems with faxes being delivered to the users emails as zero page tiff’s. It is not an every fax occurrence however it is at a rate that I am no longer able to rely upon the faxing delivery capabilities of sipXecs. I am trying to determine if the problems is with the AudioCodes, PRI Provider or sipXecs. Just wondering if there is a known issue. I am running (4.4.0- 2012-04-13EDT09:44:58 ip-10-72-10-163). I am also using three PRI circuits with an AudioCodes Mediant 1000 SIP Gateway. Listed below is my current configuration settings on the AudioCodes! *AudioCodes Information* Version ID: 6.00A.032.003 DSP Type: 2 DSP Software Version: 6008 DSP Software Name: 624AE3 Flash Version: 208 AudioCodes Configuration – Media Settings – Fax/Modem/CID Settings *General Settings* Fax Transport Mode: RelayEnable Caller ID Transport Type: Mute Caller ID Type: Standard Bellcore V.21 Modem Transport Type: Disable V.22 Modem Transport Type: Disable V.23 Modem Transport Type: Disable V.32 Modem Transport Type: Disable V.34 Modem Transport Type: Enable Bypass Fax CNG Mode: Disable CNG Detector Mode: Disable *Fax Relay Settings* Fax Relay Redundancy Depth: 0 Fax Relay Enhanced Redundancy Depth: 4 Fax Relay ECM Enable: Enable Fax Relay Max Rate (bps): 9600bps *Bypass Settings* Fax/Modem Bypass Coder Type: G711Mulaw Fax/Modem Bypass Packing Factor: 1 Fax Bypass Output Gain: 0 Modem Bypass Output Gain: 0 Any suggestions from the techs would be very welcomed! Thanks Everyone, Rob Schroeder -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile
Re: [sipx-users] Trunk to Trunk Transfer
Please test the following. Calling from PSTN to AA and try to transfer to an extension that is not forwarded. You have to ensure your ITSP is sending the INVITE to port 5080 in order for any transfers to succeed. If not, what Gerald said... how to get around that is to create a dialplan (56+10 digits sends 10 digits to a specific other gateway), which can accomplish calls coming from one provider but the forward dialplan sends it out through another. That should fix you. On Fri, Apr 20, 2012 at 1:08 PM, Tommy Laino tomla...@gmail.com wrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67729 Message-ID: 10891.4f919...@forum.sipfoundry.org I have 3 IP trunks on my test system. I am trying to have an option from the auto attendant transfer to a cell phone. If i do it from a local or remote polycom phone it works fine. Once an external call comes into a trunk and it chooses the option the caller gets disconnected. Anything that I might be missing. -- Tommy Laino Dome Technologies ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] SharedAppearanceAgent does not match the software version (4.4.0).
have you compared the rpm versions? On Fri, Apr 20, 2012 at 1:49 PM, Kyle Haefner kyle.haef...@colostate.eduwrote: Hi Users, I'm running Sipx 4.4 in a three node cluster with the most recent updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163. In one of the redundant nodes when I get this: sudo sipxproc | grep SharedAppearanceAgent SharedAppearanceAgent=Disabled, In the other redundant node i get this: sudo sipxproc | grep SharedAppearanceAgent SharedAppearanceAgent=ConfigurationMismatch, sudo sipxproc -m SharedAppearanceAgent [version.mismatch: software '4.4.0' != config ''] So the question is, Is SharedAppearanceAgent supposed to be running on redundant nodes According to this it is supposed to be running. http://wiki.sipfoundry.org/display/sipXecs/Roles,+Services+and+Processes The next question is if it is supposed to be running, why does it come up with this error? I've sent profiles many times, restarted sipx and even rebooted the server. Any ideas? Thanks! Kyle -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haef...@colostate.edu ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Generate CSR Question
I seem to recall the script may need to be or was it already modified to handle 2048 bit certificates? Besides that I think it had to be done manually AND noone updated the wiki or the list as to whether or not it worked. On Fri, Apr 20, 2012 at 4:34 PM, Michael Picher mpic...@ezuce.com wrote: did you check the wiki? On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder robert.schroe...@memberfirstmortgage.com wrote: How do I change the configuration for the certificates area to generate a 2048 bit key instead of a 1024? I have changed the openssl.cnf file in /etc/pki/tls/ location and selected the generate button and still no 2048 key is generated. ** ** I am sure this is an educational issue on my part. ** ** Yes I have searched the wiki site. ** ** Thanks everyone, ** ** Rob -- NOTICE: This electronic mail message and any content within it are intended exclusively for the individual(s) or entities to which it is addressed. The message, together with any attachments and all other content, may contain confidential and/or privileged information. Any unauthorized review, use, print, save, copy, disclosure or distribution is strictly prohibited. If you have received this message in error, please immediately advise the sender by reply email and delete all copies. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Generate CSR Question
. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-dev] sipxecs -4.6 : The message played when VM is disabled
I do think there is room for improvement here. If you do not have voicemail permissions, a brief audio message stating, This account is not enabled for messages., would certainly be more desirable. I do think playing the normal greetings as you indicated will lead to user-confusion and churn too much time figuring it out, since it is misleading. I would suggest a JIRA. On Thu, Apr 19, 2012 at 6:40 AM, Chitra chitra...@ttplservices.com wrote: Hi All, We are working on Sipxecs-4.6 18th Apr build. Please check the below given scenario: VM is disabled for user 202 1.Call 101 from user 202 phone 2.It will play the message as : Welcome your call has been answered by an automated communication system Enter your personal identification number, and then press pound. To login as a different user, press pound. and when we enter the PIN and press #, it should play the message as : “That personal identification number is not valid. Enter your personal identification number, and then press pound. To login as a different user, press pound” should be played for 3 times and then “Thank you, Goodbye” plays and gets disconnected. But when we enter the PIN and press #, it is playing the message as : “Record your name then press pound” Then a beep should be heard. This record message should be played when we enable Attendant directory permission and disable Voicemail permission. So could anybody please let us know if this is fine or do we need to raise a trivial bug for this. Thanks Regards, Chitra ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] sipxecs -4.6 : The message played when VM is disabled
I think extension is not valid is misleading too. That's why I suggested an improvment. It's not a high priority, but at some point when the UI starts to incorporate fax and other features, some recoding will need to be done and that would be a good time to inject any changes (like the time sucking Please wait while your call is transferred audio prompt... can be shortened consierably and a lot of users have asked how to remove it. So I'll consider a more encompassing improvement request to bundle this all together. On Thu, Apr 19, 2012 at 7:22 AM, George Niculae geo...@ezuce.com wrote: On Thu, Apr 19, 2012 at 2:20 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: I do think there is room for improvement here. If you do not have voicemail permissions, a brief audio message stating, This account is not enabled for messages., would certainly be more desirable. I do think playing the normal greetings as you indicated will lead to user-confusion and churn too much time figuring it out, since it is misleading. I would suggest a JIRA. What if we play an existing prompt as this extension is not valid? Recording a new greeting could take some time. George ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] Where I can download SFTF
Yes I think he wants the interopserver though I am not sure if it handles rfc4475 stuff. On Apr 19, 2012 1:47 PM, Michael Picher mpic...@ezuce.com wrote: Was this what you were looking for? http://interop.sipfoundry.org ? I don't think that works either at the moment... I'm looking at bringing a new one on-line though. Mike On Thu, Apr 19, 2012 at 10:02 AM, Derrick Ding dd...@aastra.com wrote: Hi All, ** ** I think this is not a new question. However I didn't find a good answer for that. The weblink https://scm.sipfoundry.org/rep/sftf/ is invalid now ** ** I am a user of SipX, and I also want to use SFTF to test RFC4475 on my phone. Any one can tell me where I can download SFTF or any other tools for this test? ** ** Thanks a lot. ** ** Derrick ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] Where I can download SFTF
Ha. I expect it to say SIP MESSAGE TORTURE TEST. That's pretty unique a lot of test servers don;t actually offer that! On Thu, Apr 19, 2012 at 4:47 PM, Michael Picher mpic...@ezuce.com wrote: it did... test # 7. i don't think this box works at all anymore though. thanks, mike On Thu, Apr 19, 2012 at 4:02 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: Yes I think he wants the interopserver though I am not sure if it handles rfc4475 stuff. On Apr 19, 2012 1:47 PM, Michael Picher mpic...@ezuce.com wrote: Was this what you were looking for? http://interop.sipfoundry.org ? I don't think that works either at the moment... I'm looking at bringing a new one on-line though. Mike On Thu, Apr 19, 2012 at 10:02 AM, Derrick Ding dd...@aastra.com wrote: Hi All, ** ** I think this is not a new question. However I didn't find a good answer for that. The weblink https://scm.sipfoundry.org/rep/sftf/ is invalid now ** ** I am a user of SipX, and I also want to use SFTF to test RFC4475 on my phone. Any one can tell me where I can download SFTF or any other tools for this test? ** ** Thanks a lot. ** ** Derrick ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
[sipx-users] Please review and vote if you feel it is worthy, comments on the JIRA are welcome :: FAX UI improvements
http://track.sipfoundry.org/browse/XX-10120 -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Please review and vote if you feel it is worthy, comments on the JIRA are welcome :: Send Faxes via UI or email
http://track.sipfoundry.org/browse/XX-10121 -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Basic IVR changes :: Please vote/comment in JIRA as you feel inclined.
http://track.sipfoundry.org/browse/XX-10122 Relates to error message handling when voicemail is not enabled for the user and they try to check it. Also relates to being able to change the IVR message for the announced transfer process or the ability to skip it by admin option. -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Ekiga
Then I suggest you fix your DNS or set the account appropriately. I just configured EKIGA in 1 minute on ym LAN, it works. Name: whatever registrar: sipx-hostname User: subscriber, ie. 200 Authentication user: subscriber, ie. 200 Password: sipx user sip password, not PIN My domain has an alias of the hostname in my system to get less functional UA's to work. If this DOES NOT WORK, then you really need to take it up with the Ekiga project I think. 2012/4/17 Simon Brûlé sbr...@360-innovations.com I was asking because where I am working they have an Asterisk base system at the moment and the Ekiga of the employe are configure with the fqdn and it's working well so I wanted to verified that it wasn't a problem on the SipXecs side. 2012/4/17 Simon Brûlé sbr...@360-innovations.com Ok thank you I am going to check with them. 2012/4/17 Michael Picher mpic...@ezuce.com i guess i'd ask ekiga if they support SRV records... On Tue, Apr 17, 2012 at 4:46 PM, Simon Brûlé sbr...@360-innovations.com wrote: Is Ekiga fully functional with SipXecs because I am trying to register with the Domain Name of my SipXecs and it doesn't work but with the IP Adresse it's working well. I got a X-lite installed on a Windows and this one is able to connect with the Domain Name without any kind of problem. They both are in the same subnet. I am using SipXecs 4.4. Thanks ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Ekiga
It does noone any good when you talk about that. It sounds like a network or PC related issue (i.e. local firewall settings, etc.). It's still an EKIGA error and it points back to your system(s). Transport error n/a Local_BadTransportAddress http://wiki.ekiga.org/index.php/Documentation You are not providing an error or other message from the sipx logs that shows the attempt (or not) and or a decline request from sipx. I think if you looked through the ekiga forums you will see there are issues changing from one sip provider to another with the same account, etc. Perhaps you can create a new account in akiga and provide meaningful logs from sipx, a packet capture or follow through on the Ekiga forums... 2012/4/19 Simon Brûlé sbr...@360-innovations.com In my System -- Domain I got the hostname of my server as an alias. The Ekiga config is the following: Name:3014 Registrar:voiptest.netappsid.local(the hostname of my server) User:3014 Authentification User:3014 Password:The generated sip password Timeout:3600 and it give me a Transport Error when I try to enable it. 2012/4/19 Tony Graziano tgrazi...@myitdepartment.net Then I suggest you fix your DNS or set the account appropriately. I just configured EKIGA in 1 minute on ym LAN, it works. Name: whatever registrar: sipx-hostname User: subscriber, ie. 200 Authentication user: subscriber, ie. 200 Password: sipx user sip password, not PIN My domain has an alias of the hostname in my system to get less functional UA's to work. If this DOES NOT WORK, then you really need to take it up with the Ekiga project I think. 2012/4/17 Simon Brûlé sbr...@360-innovations.com I was asking because where I am working they have an Asterisk base system at the moment and the Ekiga of the employe are configure with the fqdn and it's working well so I wanted to verified that it wasn't a problem on the SipXecs side. 2012/4/17 Simon Brûlé sbr...@360-innovations.com Ok thank you I am going to check with them. 2012/4/17 Michael Picher mpic...@ezuce.com i guess i'd ask ekiga if they support SRV records... On Tue, Apr 17, 2012 at 4:46 PM, Simon Brûlé sbr...@360-innovations.com wrote: Is Ekiga fully functional with SipXecs because I am trying to register with the Domain Name of my SipXecs and it doesn't work but with the IP Adresse it's working well. I got a X-lite installed on a Windows and this one is able to connect with the Domain Name without any kind of problem. They both are in the same subnet. I am using SipXecs 4.4. Thanks ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list
Re: [sipx-users] Ekiga
I dont use an ekiga account/call out account. I skip all that and just create a sip account (only). 2012/4/19 Simon Brûlé sbr...@360-innovations.com My Ekiga gave me the error that he cannot discover the network automatically so I would need to configure it manually and they gave me a page with instruction to forward port from the router (I imagine that is for people using Ekiga.net account). Can the part with Ekiga not being able to discover the network automatically could be related to my problem you think ?? 2012/4/19 Simon Brûlé sbr...@360-innovations.com In the option of ekiga i don't have a Stun Server option but when I go in the Ekiga configuration with the Configuration Editor tool on Ubuntu 11.10 i see the stun server option and it's disabled. 2012/4/19 Tony Graziano tgrazi...@myitdepartment.net perhaps you have enabled STUN (not needed with sipx) on Ekiga preferences? My version (3.27) does not have advanced options like stun but some older version probably do. On Thu, Apr 19, 2012 at 9:13 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: It does noone any good when you talk about that. It sounds like a network or PC related issue (i.e. local firewall settings, etc.). It's still an EKIGA error and it points back to your system(s). Transport errorn/a Local_BadTransportAddress http://wiki.ekiga.org/index.php/Documentation You are not providing an error or other message from the sipx logs that shows the attempt (or not) and or a decline request from sipx. I think if you looked through the ekiga forums you will see there are issues changing from one sip provider to another with the same account, etc. Perhaps you can create a new account in akiga and provide meaningful logs from sipx, a packet capture or follow through on the Ekiga forums... 2012/4/19 Simon Brûlé sbr...@360-innovations.com In my System -- Domain I got the hostname of my server as an alias. The Ekiga config is the following: Name:3014 Registrar:voiptest.netappsid.local(the hostname of my server) User:3014 Authentification User:3014 Password:The generated sip password Timeout:3600 and it give me a Transport Error when I try to enable it. 2012/4/19 Tony Graziano tgrazi...@myitdepartment.net Then I suggest you fix your DNS or set the account appropriately. I just configured EKIGA in 1 minute on ym LAN, it works. Name: whatever registrar: sipx-hostname User: subscriber, ie. 200 Authentication user: subscriber, ie. 200 Password: sipx user sip password, not PIN My domain has an alias of the hostname in my system to get less functional UA's to work. If this DOES NOT WORK, then you really need to take it up with the Ekiga project I think. 2012/4/17 Simon Brûlé sbr...@360-innovations.com I was asking because where I am working they have an Asterisk base system at the moment and the Ekiga of the employe are configure with the fqdn and it's working well so I wanted to verified that it wasn't a problem on the SipXecs side. 2012/4/17 Simon Brûlé sbr...@360-innovations.com Ok thank you I am going to check with them. 2012/4/17 Michael Picher mpic...@ezuce.com i guess i'd ask ekiga if they support SRV records... On Tue, Apr 17, 2012 at 4:46 PM, Simon Brûlé sbr...@360-innovations.com wrote: Is Ekiga fully functional with SipXecs because I am trying to register with the Domain Name of my SipXecs and it doesn't work but with the IP Adresse it's working well. I got a X-lite installed on a Windows and this one is able to connect with the Domain Name without any kind of problem. They both are in the same subnet. I am using SipXecs 4.4. Thanks ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone
Re: [sipx-users] Bria for iphone/ipod touch registration
your statements show you as out of order. Read the wiki should be first, which is what leads to a string of a dozen plus messages. Stop making the wrong points. Good luck. On Thu, Apr 19, 2012 at 12:09 PM, Stiles Watson wat...@datatek-net.comwrote: Since when have I had to set up a manually configured sip phone until now? You are still thinking like this is old hat to everyone using the system. Thank you for making my point. Even if someone has been a traditional phone guy for a long time, if he has never messed with SIP at all, how would know this until he screwed around with it for awhile, made mistakes, and gotten confused a few times, asked people questions, read manuals, wiki pages, books, etc.? On the this wiki page it gives instructions for setting up a Bria client with sipX and it specifically mentions Bria for iphone. In the instructions for starting Bria it says to use the PIN and not the SIP password. I have all the emails from this mailing list for the past year and half or more. There a several emails from people who have been confused about this same thing. Stiles On 04/18/2012 05:53 PM, Michael Picher wrote: since when would you enter a pin in a manually configured sip phone as your sip password? on bria ipad i don't need to enter an outbound proxy or auth name... but then again, my external dns is setup properly... mike On Wed, Apr 18, 2012 at 4:07 PM, Stiles Watson wat...@datatek-net.comwrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Intermittent Faxing Issues on Patton
no *codec 2 g711alaw64k rx-length 20 tx-length 20* On Thu, Apr 19, 2012 at 11:27 AM, Jesse Becker beck...@sunyulster.eduwrote: Tony, I modified my profile voip default. It didn't accept some of the command, and the configuration is now: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder dejitter-mode static fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 fax detection fax-frames modem dejitter-max-delay 60 no modem detection on-remote-fax-request * Attached is an updated debug. It still doesn't appear to be using t.38. For some reason it sitll appears to use the g711u. Do I need to add the additional commands you provided to the Patton T1 gateway? Currently it's profile shows: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 g711alaw64k rx-length 20 tx-length 20 dtmf-relay rtp flash-hook-relay rtp rtp traffic-class local-default fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames * For the test, we are faxing from a fax machine on a Verizon POTS line to a fax machine on the Patton FXS gateway. So the fax call goes out on a POTS, comes in on the PRI T1 (Patton), then to SipX, then to the FXS gateway. Also, how do your remove a line of configuration via that CLI? For example, while in profile voip default, how would I then remove a line, for example: *codec 2 g711alaw64k rx-length 20 tx-length 20 *Thanks, Jes On 04/18/2012 05:42 PM, Becker, Jesse wrote: Tony, Thank you. I will give that a shot. Jes On Apr 18, 2012 4:48 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: The profile is not negotiating t.38, it clearly shows g711u. profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder response-preferred-codec g711ulaw64k dejitter-mode static media detection-timeout 5 fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 modem dejitter-max-delay 60 no modem detection on-remote-fax-request Try that on The 4424. On Apr 18, 2012 4:23 PM, Becker, Jesse beck...@sunyulster.edu wrote: All, We are experiencing intermittent issues on our fax machines connected to a Patton SN4424. Outbound seems to work reliably, however, we have issues with incoming calls. The call get answers, and then seems to disconnect before the fax can finish. Incoming calls hit a Patton 4e1t1 device first, then SipX, then registered UA on the Patton SN4424. On both devices I have configured: fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames under the profile voip default. Attached you will find the T1 gateway config, FXS gateway config, as well as a call-control debug on an incoming fax that did not complete. I have removed all passwords and replaced the caller id prefix with NPANXX to hide full number. Did I miss something in the config ? Any assistance would be appreciated. Thanks, Jes -- Jesse Becker Technical Manager Office of Information Technology Network+ | Linux+ Certified Professional Ellucian @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu | www.sunyulster.edu Open or check the status of a ticket by visiting Helpdesk Onlinehttps://helpdesk.sunyulster.edu/ Look up answers to frequently asked questions by visiting the Knowledge Base https://kb.sunyulster.edu/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http
Re: [sipx-users] Bria for iphone/ipod touch registration
no, its not wrong, I have changed it back... Only the Bria 3.x is capable of being auto provisioned. The iPhone, iPad and Android editions do not have this feature from Counterpath. Starting Bria When launching Bria 3.x you have to provide Username, Password and Provisioning server. For Username enter: the user's extension. For Password enter: the user's voicemail PIN For Provisioning Server enter: http://ip.of.sipxecs.server:12000/cmcprov/login Bria will start, download the .ini file and after a short delay be ready to use with the system. now im starting to grumble. I think its straightforward. Please pay attention!! On Thu, Apr 19, 2012 at 12:21 PM, Philippe Laurent p...@ideos.com wrote: Looks like that part of the wiki is wrong then, I've changed to the wiki entry to show that the password used should be the SIP password. On Thu, Apr 19, 2012 at 12:09 PM, Stiles Watson wat...@datatek-net.comwrote: Since when have I had to set up a manually configured sip phone until now? You are still thinking like this is old hat to everyone using the system. Thank you for making my point. Even if someone has been a traditional phone guy for a long time, if he has never messed with SIP at all, how would know this until he screwed around with it for awhile, made mistakes, and gotten confused a few times, asked people questions, read manuals, wiki pages, books, etc.? On the this wiki page it gives instructions for setting up a Bria client with sipX and it specifically mentions Bria for iphone. In the instructions for starting Bria it says to use the PIN and not the SIP password. I have all the emails from this mailing list for the past year and half or more. There a several emails from people who have been confused about this same thing. Stiles On 04/18/2012 05:53 PM, Michael Picher wrote: since when would you enter a pin in a manually configured sip phone as your sip password? on bria ipad i don't need to enter an outbound proxy or auth name... but then again, my external dns is setup properly... mike On Wed, Apr 18, 2012 at 4:07 PM, Stiles Watson wat...@datatek-net.comwrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi
Re: [sipx-users] Intermittent Faxing Issues on Patton
No. I thought I sent you a 4960 config some time ago. The most recent one I put on the wiki does work in conjunction with the 4424 profile I already gave you though. I'm sure if it doesnt work after comparing, patton can fix with you though. On Thu, Apr 19, 2012 at 12:38 PM, Becker, Jesse beck...@sunyulster.eduwrote: Tony, I had guessed the no as I thought it would be similar to Cisco. The issue was that you have to omit the number (ie. 2) for it to work. Any ideas on why it still isn't does t.38? Do I need to make the same modifications you sent me for the 4424 to the SN4950 (T1 gateway) ? Thanks, Jes On Thu, Apr 19, 2012 at 12:19 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: no *codec 2 g711alaw64k rx-length 20 tx-length 20* On Thu, Apr 19, 2012 at 11:27 AM, Jesse Becker beck...@sunyulster.eduwrote: Tony, I modified my profile voip default. It didn't accept some of the command, and the configuration is now: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder dejitter-mode static fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 fax detection fax-frames modem dejitter-max-delay 60 no modem detection on-remote-fax-request * Attached is an updated debug. It still doesn't appear to be using t.38. For some reason it sitll appears to use the g711u. Do I need to add the additional commands you provided to the Patton T1 gateway? Currently it's profile shows: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 g711alaw64k rx-length 20 tx-length 20 dtmf-relay rtp flash-hook-relay rtp rtp traffic-class local-default fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames * For the test, we are faxing from a fax machine on a Verizon POTS line to a fax machine on the Patton FXS gateway. So the fax call goes out on a POTS, comes in on the PRI T1 (Patton), then to SipX, then to the FXS gateway. Also, how do your remove a line of configuration via that CLI? For example, while in profile voip default, how would I then remove a line, for example: *codec 2 g711alaw64k rx-length 20 tx-length 20 *Thanks, Jes On 04/18/2012 05:42 PM, Becker, Jesse wrote: Tony, Thank you. I will give that a shot. Jes On Apr 18, 2012 4:48 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: The profile is not negotiating t.38, it clearly shows g711u. profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder response-preferred-codec g711ulaw64k dejitter-mode static media detection-timeout 5 fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 modem dejitter-max-delay 60 no modem detection on-remote-fax-request Try that on The 4424. On Apr 18, 2012 4:23 PM, Becker, Jesse beck...@sunyulster.edu wrote: All, We are experiencing intermittent issues on our fax machines connected to a Patton SN4424. Outbound seems to work reliably, however, we have issues with incoming calls. The call get answers, and then seems to disconnect before the fax can finish. Incoming calls hit a Patton 4e1t1 device first, then SipX, then registered UA on the Patton SN4424. On both devices I have configured: fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames under the profile voip default. Attached you will find the T1 gateway config, FXS gateway config, as well as a call-control debug on an incoming fax that did not complete. I have removed all passwords and replaced the caller id prefix with NPANXX to hide full number. Did I miss something in the config ? Any assistance would be appreciated. Thanks, Jes -- Jesse Becker Technical Manager Office of Information Technology Network+ | Linux+ Certified Professional Ellucian @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu | www.sunyulster.edu Open or check the status of a ticket by visiting Helpdesk Onlinehttps://helpdesk.sunyulster.edu/ Look up answers to frequently asked questions by visiting the Knowledge Base https://kb.sunyulster.edu/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx
Re: [sipx-users] Bria for iphone/ipod touch registration
it also says for 3.x and earlier on the page it says 3.x is provisionable, not the mobile apps. On Thu, Apr 19, 2012 at 12:37 PM, Philippe Laurent p...@ideos.com wrote: Out of context, I now see it more clearly. It's not so clear in the wiki that the Startup section is intended for the provisioning mode. So, I've changed the section title to 'Starting Bria in provisioned mode'. Now it's way clear. On Thu, Apr 19, 2012 at 12:30 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: no, its not wrong, I have changed it back... Only the Bria 3.x is capable of being auto provisioned. The iPhone, iPad and Android editions do not have this feature from Counterpath. Starting Bria When launching Bria 3.x you have to provide Username, Password and Provisioning server. For Username enter: the user's extension. For Password enter: the user's voicemail PIN For Provisioning Server enter: http://ip.of.sipxecs.server:12000/cmcprov/login Bria will start, download the .ini file and after a short delay be ready to use with the system. now im starting to grumble. I think its straightforward. Please pay attention!! On Thu, Apr 19, 2012 at 12:21 PM, Philippe Laurent p...@ideos.com wrote: Looks like that part of the wiki is wrong then, I've changed to the wiki entry to show that the password used should be the SIP password. On Thu, Apr 19, 2012 at 12:09 PM, Stiles Watson wat...@datatek-net.comwrote: Since when have I had to set up a manually configured sip phone until now? You are still thinking like this is old hat to everyone using the system. Thank you for making my point. Even if someone has been a traditional phone guy for a long time, if he has never messed with SIP at all, how would know this until he screwed around with it for awhile, made mistakes, and gotten confused a few times, asked people questions, read manuals, wiki pages, books, etc.? On the this wiki page it gives instructions for setting up a Bria client with sipX and it specifically mentions Bria for iphone. In the instructions for starting Bria it says to use the PIN and not the SIP password. I have all the emails from this mailing list for the past year and half or more. There a several emails from people who have been confused about this same thing. Stiles On 04/18/2012 05:53 PM, Michael Picher wrote: since when would you enter a pin in a manually configured sip phone as your sip password? on bria ipad i don't need to enter an outbound proxy or auth name... but then again, my external dns is setup properly... mike On Wed, Apr 18, 2012 at 4:07 PM, Stiles Watson wat...@datatek-net.comwrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us
Re: [sipx-users] Intermittent Faxing Issues on Patton
I will send you an fxs config I use all the time offline/ On Thu, Apr 19, 2012 at 12:51 PM, Becker, Jesse beck...@sunyulster.eduwrote: Tony, My PRI config matches yours. I will get in touch with Patton to see if they can help me get this resolved. Thanks, Jes On Thu, Apr 19, 2012 at 12:42 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: No. I thought I sent you a 4960 config some time ago. The most recent one I put on the wiki does work in conjunction with the 4424 profile I already gave you though. I'm sure if it doesnt work after comparing, patton can fix with you though. On Thu, Apr 19, 2012 at 12:38 PM, Becker, Jesse beck...@sunyulster.eduwrote: Tony, I had guessed the no as I thought it would be similar to Cisco. The issue was that you have to omit the number (ie. 2) for it to work. Any ideas on why it still isn't does t.38? Do I need to make the same modifications you sent me for the 4424 to the SN4950 (T1 gateway) ? Thanks, Jes On Thu, Apr 19, 2012 at 12:19 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: no *codec 2 g711alaw64k rx-length 20 tx-length 20* On Thu, Apr 19, 2012 at 11:27 AM, Jesse Becker beck...@sunyulster.eduwrote: Tony, I modified my profile voip default. It didn't accept some of the command, and the configuration is now: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder dejitter-mode static fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 fax detection fax-frames modem dejitter-max-delay 60 no modem detection on-remote-fax-request * Attached is an updated debug. It still doesn't appear to be using t.38. For some reason it sitll appears to use the g711u. Do I need to add the additional commands you provided to the Patton T1 gateway? Currently it's profile shows: * profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 g711alaw64k rx-length 20 tx-length 20 dtmf-relay rtp flash-hook-relay rtp rtp traffic-class local-default fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames * For the test, we are faxing from a fax machine on a Verizon POTS line to a fax machine on the Patton FXS gateway. So the fax call goes out on a POTS, comes in on the PRI T1 (Patton), then to SipX, then to the FXS gateway. Also, how do your remove a line of configuration via that CLI? For example, while in profile voip default, how would I then remove a line, for example: *codec 2 g711alaw64k rx-length 20 tx-length 20 *Thanks, Jes On 04/18/2012 05:42 PM, Becker, Jesse wrote: Tony, Thank you. I will give that a shot. Jes On Apr 18, 2012 4:48 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: The profile is not negotiating t.38, it clearly shows g711u. profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder response-preferred-codec g711ulaw64k dejitter-mode static media detection-timeout 5 fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 modem dejitter-max-delay 60 no modem detection on-remote-fax-request Try that on The 4424. On Apr 18, 2012 4:23 PM, Becker, Jesse beck...@sunyulster.edu wrote: All, We are experiencing intermittent issues on our fax machines connected to a Patton SN4424. Outbound seems to work reliably, however, we have issues with incoming calls. The call get answers, and then seems to disconnect before the fax can finish. Incoming calls hit a Patton 4e1t1 device first, then SipX, then registered UA on the Patton SN4424. On both devices I have configured: fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames under the profile voip default. Attached you will find the T1 gateway config, FXS gateway config, as well as a call-control debug on an incoming fax that did not complete. I have removed all passwords and replaced the caller id prefix with NPANXX to hide full number. Did I miss something in the config ? Any assistance would be appreciated. Thanks, Jes -- Jesse Becker Technical Manager Office of Information Technology Network+ | Linux+ Certified Professional Ellucian @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu | www.sunyulster.edu Open or check the status of a ticket by visiting Helpdesk Onlinehttps://helpdesk.sunyulster.edu/ Look up answers to frequently asked questions by visiting the Knowledge Base https://kb.sunyulster.edu/ ___ sipx-users mailing list sipx-users
Re: [sipx-users] Bria for iphone/ipod touch registration
On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wat...@datatek-net.comwrote: On 04/18/2012 06:15 PM, Tony Graziano wrote: You shouldn't invite violence. I have been known to clobber people with iPads. Why am I not surprised by this? Um that was a cartoon reference dude, you take things wayyy to seriously. Obviously you don't get out too much. I think the wiki could be clearer, but really I think you are the only one to make the leap from Bria is bria is bria is bria. It ain't, but sipfoundry doesn't manuafcture it either. Bria has undergone numerous name changes in the last 3 or four years. I look for the names to change yet again, because why leave something the same? I am not the only one who has been confused on this point. I have all the emails from this list for the past year and a half or more. There have been several people who have been confused about this. I know sipfoundry does not manufacturer Bria, but you are saying that it is supported by your software and therefore it is to your advantage to help your users/customers use your product with whatever different versions exist of the third party software you say you are interoperable with. Thanks for making my point about not taking a users view of things. Wrong, the wiki was changed by multiple editors to leave no chance for stiles.user error.. It clearly stated Bria 3.x, but the ASSUMPTION you had was all BRIA is BRIA. Whatever. It got changed to quell your issue. You didn't like the clarity of the wiki, so now it's idiot proffed (maybe) and you still want to complain about it. You guys do realize that the people that use your product are your customers, right? I know no money changes hands, but you are creating this for people to use, correct? You want companies big AND small to embrace your product, correct? There are a number of potential users/customers who have left your product because they could not get the help they thought they needed. Its a very friendly environment. The wiki is there for a reason. There are many people who try to do things that LACK THE SKILLSET and/or FAIL TO READ OR UNDERSTAND what the best approach is to installing/configuring a system and then expect wayyy to much in having community members send them the wiki pages and explain the basic concepts. There are skills required as a prerequisite. Give people more information than they need? That's been done, numerous times. When they can't follow directions the first few times, you have to beat them over the head with an iPad and chant DNS DNS DNS. I love how you never let go of that bone! You are right, DNS is very important and I have some things which still need to be configured. The problem is that I do not think the DNS wiki page is perfectly clear and I have been give contradictory info on this forum - again, its free support offered by the sipX community which includes old hats, newbies, developers and users so that happens. I was told that sipX sets up everything correctly and that I do not need external DNS SRV records, but that was not correct. It makes perfect sense now that I would need them. I think it was correct in the context or your related question(s) at that time, but clearly you want to pick yet another bone. Bringing this up out of context is just sour grapes and not really fair. Understand then fix your DNS and stop ranting to the masses who already get that. Ahh, just Understand. It is easy to wave that wand when it is something you are very familiar with. You might be one of those very gifted people who instantly understands everything the first time they see it. I'm happy for you - really. However, on the whole, God has gifted different people with different abilities. The thing which is easy for one, may very very difficult for another. DNS, DNS, DNS! What beautiful words! On to DNS! Dude, the whole thing is: DNS, Networking, firewalls, all play a part in any VOIP deployment. The wiki article on DNS is written for administrators. Why? Because DNS is VERY important to any environment and easy to break. It's not in the best interest of any project focusing on SIP to teach DNS 101 to admins who don't understand how DNS works. There is no real easy way to explain DNS. It's not fair to tech network 101 courses. You should bring those skills to the table as a prerequisite. You really need to stop making a spectacle of yourself. You are not getting anywhere except maybe closer to a coronary. Now get on to whatever it is you were doing. Good luck. On Wed, Apr 18, 2012 at 5:44 PM, Stiles Watson wat...@datatek-net.comwrote: I'm not sure what there is to 'get'. I thought the point of the wiki was to explain how to set things up. I was directed to the Bria page on the wiki when asked about Bria and iphone (the page even mentions iphone...). However, the info there has nothing to do with Bria on iphone. I was offering to provide clarification for others so
Re: [sipx-users] Phones disconnecting
It is open to the Internet (port 5060 forwarded at your firewall)? If so, look at your log sizes for the day prior to your issue and compare them to today (roxy and registrar logs). If they are considerable larger today it might be your were the subject a a DOS attack, inspeacting the larger logs would show signs of that. On Thu, Apr 19, 2012 at 1:26 PM, Ken Ridley k...@federico.net wrote: I posted a while back about all of the phones deregistering at the same time, and was told that it was because I only had 1 GB of ram in my server A week ago, I upgraded the hardware to this: HP ProLiant ML110 G7 664723-S01 4U Micro Tower Entry-level Server - 1 x Core i3 i3-2120 3.3GHz - 4 GB RAM - 250 GB HDD - DVD-Writer - Serial ATA/300 RAID - Gigabit Ethernet ** ** Yesterday all of the phones deregistered I restarted the services, and the phones reconnected I am running the latest stable build, using Polycom SP 501 Phones, FW 3.1.7, Boot rom 4.1.4 ** ** What do I need to do now? ** ** I would like the system to run for more than a week, without a problem ** ** Thanks for your help Ken ** ** ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Experience with Microtech FaxFinder
Do you mean multi tech? On Apr 19, 2012 2:47 PM, Todd Hodgen thod...@frontier.com wrote: Does anyone have an good or bad experience with Microtech FaxFinder Appliance, either SIP or analog version to share. I’m looking at using it for an outbound fax solution for a customer, and looking for any experiences others might have has with this product. ** ** ** ** Thanks in advance for any thoughts on it. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Nat Problem
if it is local. I have seen polycom phones act like this before. In my case: The user portion of a SIP dialog MUST match the ACK and if it does not match exactly the phone will ignore it. Without a valid ACK the phone won’t start sending RTP and the UI won’t show the call as answered. You may want to do a capture on the sipx server and look at the results with wireshark. Sounds like you may still have ALG at the gateway on the 192.168.175.0 network. -- Regards -- Gerald Drouillard Technology Architect Drouillard Associates, Inc.http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Regards -- Gerald Drouillard Technology Architect Drouillard Associates, Inc.http://www.Drouillard.biz ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Nat Problem
Most likely if that is the router sitting in front of sipx, yes. It's a prerequisite for media relay to function. 1. Server behind nat has to be enabled. 2. Support remote workers needs to be enabled (sip trunking does not need to be enabled for remote workers, media relay does). 3. NAT for server needs to have access to a public IP address by getting it via STUN server or manually entering it. 4. Firewall in front of sipx needs to be able to do symmetrical outbound NAT (AON or FULL CONE NATE, at least for the outbound NAT of the sipx internal address) and have any SIP helper turned off. If the two routers are adjacent and you can route (without NAT) you would simply do so and add the PC subnet to the intranets page in sipx ONLY IF it does not have to pass through NAT (i.e route or site to site vpn, etc.). Your router is changing the ports and hence sipx is expecting the audio to come back on a port that isnt being sent by the router. 2012/4/19 Simon Brûlé sbr...@360-innovations.com So your saying the problem may come from the router I have (Linksys E2500) because it's not doing the symmetrical Nat so the RTP is getting lost? 2012/4/19 Tony Graziano tgrazi...@myitdepartment.net If there is NAT between your PC and the sipx server then: 1. The local firewall to your PC needs to have any SIP helper or Application Layer Gateway turned off. 2. At your firewall where sipx is the SIP helper or Application Layer Gateway needs to be turned off AND the NAT type for the outbound NAT from the sipx server needs to be symmterical. Home brew and residential routers usually will not do this. 3. If you have an entry in your intranet subnets that includes the PC network number it must be listed ONLY IF IF DOES NOT GO THROUGH NAT. One of these three (or a combination of them) is keeping RTP from flowing. 2012/4/19 Simon Brûlé sbr...@360-innovations.com I used the following command on the SipXecs server *tcpdump -n -s 0 -i any -w filename.cap* and then I transfer it on my computer so I could open it with my Wireshark and have a look at it. I joined the file to this e-mail so you could take a look and tell me what you think. This one is from a call I did from the softphone to the Hardphone where this one bugged like I described earlier. Thanks. 2012/4/19 Simon Brûlé sbr...@360-innovations.com As i dig more and more in Wireshark i came to the conclusion that the Wireshark information that I just sent you is pretty much useless as I now see it. I will keep looking for some piece of information that could help. Thanks. 2012/4/19 Simon Brûlé sbr...@360-innovations.com My Computer is connected in the Lan of the company and my E2500 is connected in this Lan too. My Server SipXecs and my hardphone are on the E2500. So there is an other router between my computer and the router that have my Server connected on it. For the Wireshark part when I answer the phonecall I do from the softphone to my hardĥone those request are coming in until i close the call. 109 19.596310 192.168.175.22 192.168.175.136 SIP/SDP 1466 Status: 200 OK, with session description Status-Code: 200 [resent packet : True] [Suspected resend of frame:104] [Request Frame : 57] [Response Time (ms): 10950] followed by this one: 110 19.598495 192.168.175.136 192.168.0.1 SIP 714 Request: ACK sip:3050@192.168.0.253:5060;transport=tcp Request-Line: ACK sip:3050@192.168.0.253:5060;transport=tcp SIP/2.0 Method: ACK Request-URI: sip:3050@192.168.0.253:5060;transport=tcp [Resent Packet: False] [Request Frame: 105] [Response Time (ms): 512] All those test have been done on Wireshark on the Computer with the Softphone on it. And the 192.168.0.253 that you see is the hardphone IP adresse. 2012/4/19 Gerald Drouillard gerryl...@drouillard.ca On 4/19/2012 3:25 PM, Simon Brûlé wrote: How can I do a capture with wireshark on the SipXecs server? If you google a little you will find it. About the ALG you think that the other Router that give the DHCP to my Laptop and the Wan adresse of my router would have the Sip ALG activate? That would be the only thing inbetween your softphone and the sipx server... right? http://screenshots.portforward.com/Cisco/Linksys_E2500/Management.htm 2012/4/19 Gerald Drouillard gerryl...@drouillard.ca On 4/19/2012 2:58 PM, Simon Brûlé wrote: I added 192.168.175.0/24 to the intranet subnet and I still have the same problem. 2012/4/19 Gerald Drouillard gerryl...@drouillard.ca On 4/19/2012 2:37 PM, Simon Brûlé wrote: Hi, I know I already posted something very similiar to this problem but I haven't found a solution to it so here i am reposting my problem but with more precision this time. I have a softphone (Jitis) on a Ubuntu 11.10 installation connected to the network of the company. I have a router Linksys E2500 connected to the same network. The laptop have the adresse 192.168.175.136 giving by dhcp and the router have
[Astlinux-users] DHCP server on external interface
Hello list, is there a simple way to enable dnsmasq dhcp server on an astlinux 1.0.2 installaed on a single ethernet system? I have a machine installed with ast1.0.2 and i'm going to activate a dhcp server to provide ip addresses to our sip phones but I cannot find information. Thank graziano -- Graziano Brioschi Outland s.a.s. sede operativa: Via A. Don Rocca, 13 20030, Senago (MI) tel: 02 9948 6014 mobile: 328 8382622 email: graziano.brios...@outland.it -- U4E-- -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
Re: [sipx-users] Cordless phones
In your case I would test coverage with any app, besides counterpath, you can try the free 3cx (Android and iOS) app and others. The biggest thing you will find with wifi -- battery life/talktime (especially when received wifi signals are weak), don't hold up nearly as long as DECT. So your wifi deployment, coverage has a lot to do with battery life and talktime. On Wed, Apr 18, 2012 at 1:01 AM, Andrew Radke andrew.ra...@yuruga.com.auwrote: Hi Tony, We are looking at outdoor coverage but with a lot of trees and vegetation. Considering your response it shows that things have changed in recent years too… We do also have large wifi coverage already and are constantly increasing it. In the past it seemed that wifi was considered universally terrible. Has that changed? And are there any good smartphone apps? I guess it would be Android rather than iPhone since it is possible to get reasonable Android handsets cheaply on prepaid plans and then don't use the cellular side at all. But for those of us with existing iPhones is there any recommended apps? Regards, Andrew Radke Yuruga Nursery Pty Ltd Clonal Solutions Australia Pty Ltd PO Box 220 Walkamin Qld 4872 Phone: (07) 4093 3826 Fax: (07) 4093 3869 Email: andrew.ra...@yuruga.com.au Web: www.yuruga.com.au On 17/04/2012, at 8:04 PM, Tony Graziano wrote: You need to explain what kind of coverage you need and what kind of wireless infrastructure you have (if any). Snom makes a dect phone which also has wireless repeaters and should work fine. The battery life and talk time is very good and does not interfere with wifi at all. If you have a wifi infrastructure you could opt for an app on a smartphone. On Apr 17, 2012 12:56 AM, Andrew Radke andrew.ra...@yuruga.com.au wrote: Hi all, Just a query to see what the current thoughts are on cordless phones. We probably need 2-3 phones fairly soon that can transfer calls. It would be nice (but not immediately required) to have the phones capable of switching between multiple base stations due to the physical area to be covered. Of course this adds a lot to the price so may be judged to be uneconomical. I know this has been asked before but a lot can change with VoIP phones. Andrew Radke Yuruga Nursery Pty Ltd Clonal Solutions Australia Pty Ltd PO Box 220 Walkamin Qld 4872 Phone: (07) 4093 3826 Fax: (07) 4093 3869 Email: andrew.ra...@yuruga.com.au Web: www.yuruga.com.au ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net/ Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Where I can download SFTF
I am not aware that was ever put back up after the move. I am also not sure it was meant to do torture tests. This should be posted in the sipx-dev mailing list though for feedback, since they might have more current information. On Tue, Apr 17, 2012 at 4:45 PM, Derrick Ding dd...@aastra.com wrote: Hi All, I think this is not a new question. However I didn't find a good answer for that. The weblink https://scm.sipfoundry.org/rep/sftf/ is invalid now I am a user of SipX, and I also want to use SFTF to test RFC4475 on my phone. Any one can tell me where I can download SFTF or any other tools for this test? Thanks a lot. Derrick ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Jitsi provisionning
Has anyone run through any interop testes with it yet? On Wed, Apr 18, 2012 at 12:15 PM, cyril.constan...@gmail.com wrote: Hi, Thanks Michael for your feedback, if there is any developper who wants to move forward on it, it will be really appreciated :) as this softphone is open source and ready to be deployed into company. Have a nice day all. Best Regards -Original Message- From: Michael Picher mpic...@ezuce.com Sender: sipx-users-boun...@list.sipfoundry.org Date: Wed, 18 Apr 2012 03:03:01 To: Discussion list for users of sipXecs software sipx-users@list.sipfoundry.org Reply-To: Discussion list for users of sipXecs software sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Jitsi provisionning ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bria for iphone/ipod touch registration
it would be the sip password. DNS, DNS, DNS. If your wifi network DNS doesn't resolve the domain, you are SOL. On Wed, Apr 18, 2012 at 3:22 PM, Stiles Watson wat...@datatek-net.comwrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bria for iphone/ipod touch registration
I don't get you. Counterpath has a product for the desktop that can be centrally provisioned. Their mobile devices can't (yet). Maybe we can create a stiles.wiki. JK. I think the wiki might benefit from a mobile apps section, but it requires proper layout for and could use a simple table layout for how to configure for most products and OS's. The key here is testing functionalities and features, etc. On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bria for iphone/ipod touch registration
Account advanced works unchecked if your DNS is correct. On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Intermittent Faxing Issues on Patton
The profile is not negotiating t.38, it clearly shows g711u. profile voip default codec 1 g711ulaw64k rx-length 20 tx-length 20 codec 2 transparent-clearmode rx-length 20 tx-length 20 no dtmf-relay rtp traffic-class local-default no dtmf-mute-encoder response-preferred-codec g711ulaw64k dejitter-mode static media detection-timeout 5 fax transmission 1 relay t38-udp fax volume -13.5 fax dejitter-max-delay 60 modem dejitter-max-delay 60 no modem detection on-remote-fax-request Try that on The 4424. On Apr 18, 2012 4:23 PM, Becker, Jesse beck...@sunyulster.edu wrote: All, We are experiencing intermittent issues on our fax machines connected to a Patton SN4424. Outbound seems to work reliably, however, we have issues with incoming calls. The call get answers, and then seems to disconnect before the fax can finish. Incoming calls hit a Patton 4e1t1 device first, then SipX, then registered UA on the Patton SN4424. On both devices I have configured: fax transmission 1 relay t38-udp fax transmission 2 bypass g711ulaw64k fax detection fax-frames under the profile voip default. Attached you will find the T1 gateway config, FXS gateway config, as well as a call-control debug on an incoming fax that did not complete. I have removed all passwords and replaced the caller id prefix with NPANXX to hide full number. Did I miss something in the config ? Any assistance would be appreciated. Thanks, Jes -- Jesse Becker Technical Manager Office of Information Technology Network+ | Linux+ Certified Professional Ellucian @ SUNY Ulster 491 Cottekill Road, Stone Ridge, NY 12484 Tel 845-687-5064 | Fax 845-687-5105 beck...@sunyulster.edu | www.sunyulster.edu http://www.sunyulster.edu/ Open or check the status of a ticket by visiting Helpdesk Onlinehttps://helpdesk.sunyulster.edu/ Look up answers to frequently asked questions by visiting the Knowledge Base https://kb.sunyulster.edu/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Bria for iphone/ipod touch registration
On Wed, Apr 18, 2012 at 5:17 PM, Stiles Watson wat...@datatek-net.comwrote: Thanks. I'll look into that. I set it when it would not register. I'll remove the settings and see what happens. On DNS: I never received a straight answer on this, so I'll ask it again. IF during sipX install, I chose to have DNS installed and running on the sipX server AND IF sipX auto configs DNS correctly, THEN how could DNS not be setup correctly? I've made no changes to DNS. I've also been told that I do not need external SRV records IF I'm using the DNS server installed on the sipX server. That makes complete sense if I have no remote phones or only have remote phones which can download profiles via FTP, etc. However, in the case of the Bria iphone client, which only now have I been informed does not have the ability to provision and must be configed manually, IF I do not set the Out. Proxy manually, how would the client know what the proxy is when it is not on the office network and so there are no SRV records? Provision or not, if it is being used remote, then there need to be public DNS records. This is straightforward and laid out well in the DNS area of the wiki. I will not rehash that. It resolves it records via DNS. Tony, in a previous email you stated that Bria's mobile clients can not provision remotely. No one ever mentioned that they could not be provisioned centrally at all and when I asked about the provisioning no one responded. I understand this is free support so no one has to. That is really a Counterpath feature. Why would anyone comment on it? They don't advertise it to provision remotely because it does not have the feature. The whole point of setting the Bria/ipod combo up is to operate as a remote phone which is not behind a VPN. This was one of the solutions recommended to me on this forum. IF this is a remote phone connecting anywhere there is a wifi connection AND there are no external SRV records AND if it can not be provisioned centrally at all AND the outbound proxy is not entered manually, how is it supposed to work correctly? Called SRV records. There are some really intricate workarounds for that, but I don't suggest doing so nor do I think the mobile version is flexible enough to carry them out. Personally I think the Samsung is the best choice dollar wise if you dont need a smart phone. http://www.samsung.com/us/mobile/mp3-players/YP-G70CWY/XAA I'm not being forces to use sipX. I tried several other open source SIP solutions and found them all lacking. I'm recommending sipX to my company because I think is the best solution available. I've spent weeks trying to get everything to work correctly and I'm sure to the developers and the long time phone guys that seems ridiculous, but I'm not a phone guy and I've not been working in the guts of this system for months and years so I'm just asking for a little clarity so I can keep moving this project forward. DNS DNS DNS DNS. It's spelled D N S. If you are trying to do this outside without a VPN, you can't sidestep it. Stiles On 04/18/2012 04:32 PM, Tony Graziano wrote: Account advanced works unchecked if your DNS is correct. On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users
Re: [sipx-users] Bria for iphone/ipod touch registration
You shouldn't invite violence. I have been known to clobber people with iPads. I think the wiki could be clearer, but really I think you are the only one to make the leap from Bria is bria is bria is bria. It ain't, but sipfoundry doesn't manuafcture it either. Bria has undergone numerous name changes in the last 3 or four years. I look for the names to change yet again, because why leave something the same? Give people more information than they need? That's been done, numerous times. When they can't follow directions the first few times, you have to beat them over the head with an iPad and chant DNS DNS DNS. Understand then fix your DNS and stop ranting to the masses who already get that. On Wed, Apr 18, 2012 at 5:44 PM, Stiles Watson wat...@datatek-net.comwrote: I'm not sure what there is to 'get'. I thought the point of the wiki was to explain how to set things up. I was directed to the Bria page on the wiki when asked about Bria and iphone (the page even mentions iphone...). However, the info there has nothing to do with Bria on iphone. I was offering to provide clarification for others so they are not confused when they look at the wiki page and then look at the iphone settings and see little or no overlap. You guys on the sipX team have done a great job and have produced and continue to produce a great open source product, but more times than not you guys take a programmers view of things instead of a users view. This is what make the difference between products that go on to be great and widely embraced and products which are great functionally, but are never embraced by the masses (or products which people are forced to use, but hate using). Why do you think Apple has boat loads of cash on hand in a down economy even though they do not own the lion share of either the desktop or server market? It is because they focus on the user and make it easy for them to get done what they need to get done. The iphone revolutionized the smart phone market. Why? They did not have any new wiz-bang feratures (they actually had less), but they focused on the user and made it easier to use. Everyone else has been playing catch up (yes, you can argue that certain Andriod devices are better, but they followed iphone). Explain everything, teach people, give people more info then they need I've seen so many technically good products go down the toilet because the programmers could never see things from a users perspective. They turned their noses up at them instead. And in return? The users went somewhere else and the project died. Just my ten cents. The rant is over and everyone can beat me up now. Stiles On 04/18/2012 04:31 PM, Tony Graziano wrote: I don't get you. Counterpath has a product for the desktop that can be centrally provisioned. Their mobile devices can't (yet). Maybe we can create a stiles.wiki. JK. I think the wiki might benefit from a mobile apps section, but it requires proper layout for and could use a simple table layout for how to configure for most products and OS's. The key here is testing functionalities and features, etc. On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone As I mentioned in an earlier email. I was able to get Media5Fone working on an iphone with the same settings above. Any thoughts? Stiles Stiles ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org
Re: [sipx-users] Bria for iphone/ipod touch registration
Then you are the perfect wiki maintainer for the mobile section. HA! On Apr 18, 2012 6:30 PM, Michael Picher mpic...@ezuce.com wrote: i've got like one of everything now tony :-) bria for ipad kicks bria for android tablet beta's rear-end all over the place. That being said, bria on the PC is better than bria on the mac (no RLS support on the mac version). Bria for android phone is just getting bluetooth support now (beta)... i don't know about bria for iphone but i belive it is a little further along than bria for android phone. mike On Wed, Apr 18, 2012 at 6:15 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: You shouldn't invite violence. I have been known to clobber people with iPads. I think the wiki could be clearer, but really I think you are the only one to make the leap from Bria is bria is bria is bria. It ain't, but sipfoundry doesn't manuafcture it either. Bria has undergone numerous name changes in the last 3 or four years. I look for the names to change yet again, because why leave something the same? Give people more information than they need? That's been done, numerous times. When they can't follow directions the first few times, you have to beat them over the head with an iPad and chant DNS DNS DNS. Understand then fix your DNS and stop ranting to the masses who already get that. On Wed, Apr 18, 2012 at 5:44 PM, Stiles Watson wat...@datatek-net.comwrote: I'm not sure what there is to 'get'. I thought the point of the wiki was to explain how to set things up. I was directed to the Bria page on the wiki when asked about Bria and iphone (the page even mentions iphone...). However, the info there has nothing to do with Bria on iphone. I was offering to provide clarification for others so they are not confused when they look at the wiki page and then look at the iphone settings and see little or no overlap. You guys on the sipX team have done a great job and have produced and continue to produce a great open source product, but more times than not you guys take a programmers view of things instead of a users view. This is what make the difference between products that go on to be great and widely embraced and products which are great functionally, but are never embraced by the masses (or products which people are forced to use, but hate using). Why do you think Apple has boat loads of cash on hand in a down economy even though they do not own the lion share of either the desktop or server market? It is because they focus on the user and make it easy for them to get done what they need to get done. The iphone revolutionized the smart phone market. Why? They did not have any new wiz-bang feratures (they actually had less), but they focused on the user and made it easier to use. Everyone else has been playing catch up (yes, you can argue that certain Andriod devices are better, but they followed iphone). Explain everything, teach people, give people more info then they need I've seen so many technically good products go down the toilet because the programmers could never see things from a users perspective. They turned their noses up at them instead. And in return? The users went somewhere else and the project died. Just my ten cents. The rant is over and everyone can beat me up now. Stiles On 04/18/2012 04:31 PM, Tony Graziano wrote: I don't get you. Counterpath has a product for the desktop that can be centrally provisioned. Their mobile devices can't (yet). Maybe we can create a stiles.wiki. JK. I think the wiki might benefit from a mobile apps section, but it requires proper layout for and could use a simple table layout for how to configure for most products and OS's. The key here is testing functionalities and features, etc. On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote: So can the wiki be updated to reflect the differences in Bria for iphone/ipod touch? - Use SIP password not PIN - No Provisioning server - Under Account advanced, you need to enter Out. Proxy and Auth Name I do not know what the approval process is, but if I can get a login, I'd be happy to do it. Could also enter settings for Media5-fone. Stiles On 04/18/2012 03:50 PM, Stiles Watson wrote: Not sure what the deal was, but I sent the profile AGIAN, set the password to the SIP password, not PIN and it registered. Stiles On 04/18/2012 03:22 PM, Stiles Watson wrote: I am unable to register via Bria for ipod touch. I'm connected via wifi on the office network. I get Unauthorized (401). Here are my settings: Account Name: 295 Enabled: ON Username: 295 Password: (I've tried both PIN and SIP) Domain: datatek-net.com Under Account Advanced Out. Proxy: sipx.datatek-net.com Auth Name: 295 There are no other settings to enter and there is no place to enter a Provisioning Server as mentioned on the wiki page: http://wiki.sipfoundry.org/display/sipXecs
Re: [sipx-dev] Huntgroup-call forwarding
I think this is valid if the Allow Call-Forwarding is enabled in the hunt group and the user call forward is at the same time. On Tue, Apr 17, 2012 at 8:56 AM, Kumaran thiru.venkateshwa...@ttplservices.com wrote: Hi All, Please check the following scenario: Hunt group extension=444 Initially call=201 expire=20 No response =202 expire=20 Vm option unchecked and no Fallback Destination Call forwarding Enabled Call forwarding after 30 secs to extension=205 for 202 When I called 444 from 200 201 rings for 20 secs and 202 started ringing after 20 secs... Once 202 stop ringing call ended but call not forwarded to extension 205whether its a valid behavior?Only way I can forward the call by disconnect the call on 202 when started ringing then call will be forwarded to 205 and rings remaining secs left from 202... Regards, Kumaran T ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-users] Cordless phones
You need to explain what kind of coverage you need and what kind of wireless infrastructure you have (if any). Snom makes a dect phone which also has wireless repeaters and should work fine. The battery life and talk time is very good and does not interfere with wifi at all. If you have a wifi infrastructure you could opt for an app on a smartphone. On Apr 17, 2012 12:56 AM, Andrew Radke andrew.ra...@yuruga.com.au wrote: Hi all, Just a query to see what the current thoughts are on cordless phones. We probably need 2-3 phones fairly soon that can transfer calls. It would be nice (but not immediately required) to have the phones capable of switching between multiple base stations due to the physical area to be covered. Of course this adds a lot to the price so may be judged to be uneconomical. I know this has been asked before but a lot can change with VoIP phones. Andrew Radke Yuruga Nursery Pty Ltd Clonal Solutions Australia Pty Ltd PO Box 220 Walkamin Qld 4872 Phone: (07) 4093 3826 Fax: (07) 4093 3869 Email: andrew.ra...@yuruga.com.au Web: www.yuruga.com.au ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Cordless phones
To further elaborate on the Polycom Kirk DECT line, its awesome. The small server is inexpensive, but realize there are many phones and all of them are not the same. Also realize these handsets are made in different models for different environments (healthcare, office, industrial/manufacturing). I recall some of these phones do not work with SIP in general, the firmware they are locked to is specific to a particular platform (i.e. Asterisk Only, etc.). I think Philippe owes us all a wiki page on the Kirk too. [?] On Tue, Apr 17, 2012 at 6:11 AM, Philippe Laurent p...@ideos.com wrote: We're using 32 KIRK 5020 and 6020 phones with great results. We use the 6000 KIRK server, but the 300 server should work for you (max 12 phones) and can extend with repeaters at a much lower price. On Tuesday, April 17, 2012, Andrew Radke wrote: Hi all, Just a query to see what the current thoughts are on cordless phones. We probably need 2-3 phones fairly soon that can transfer calls. It would be nice (but not immediately required) to have the phones capable of switching between multiple base stations due to the physical area to be covered. Of course this adds a lot to the price so may be judged to be uneconomical. I know this has been asked before but a lot can change with VoIP phones. Andrew Radke Yuruga Nursery Pty Ltd Clonal Solutions Australia Pty Ltd PO Box 220 Walkamin Qld 4872 Phone: (07) 4093 3826 Fax: (07) 4093 3869 Email: andrew.ra...@yuruga.com.au Web: www.yuruga.com.au ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net 330.gif___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Remote provision of Bria iphone/ipod touch
The mobile/tablet versions do not have the remote provisioning feature from Counterpath, just the desktop version. On Tue, Apr 17, 2012 at 11:39 AM, Stiles Watson wat...@datatek-net.comwrote: I have a user trying to provision a Bria softphone on a ipod touch over a remote wifi. I've created a user for him, added the bria device, assigned the user as a line on the device and sent the profile. I've instructed him to use the his ext as his user name, his PIN as his password (via instructions on http://wiki.sipfoundry.org/display/sipXecs/Counterpath+Bria+Softphone) and told him to use http://WAN_IP:12000/cmcprov/login as the provisioning server. I also sent him the SIP password in case the PIN did not work. On the firewall, I opened TCP port 12000 and forwarded it to the sipX server. Is there anything I missed? Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] site-to-site transfers
You need to explain how the dial plan rule is constructed: The correct way for this to work is with the sipdomain as the gateway address at each end. For instance: using the sip hostname or internal/external IP address is not going to assist you in transfers. Since most UA's use the sipdomain, each end must KNOW about (and be able to call through) the sipdomain. Are you using sipdomain at each end? Are the phones registering via sipdomain? If so, it should work. If not, it's all up to you. On Mon, Apr 16, 2012 at 10:13 AM, Mike Graham mike_gra...@hempfieldsd.org wrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67501 Message-ID: 107ad.4f8c2...@forum.sipfoundry.org I'm having trouble transferring calls that are placed through a site-to-site dialplan rule. The problem is exactly the same as the one described in this archive post: http://forum.sipfoundry.org/index.php?t=msggoto=51090S=cbdd98bdc0e6b26024251331be321f2d The above archive doesn't seem to have any solution, so I'm curious if anyone else has run into a similar situation or found a solution. mailto:2...@domaina.org calls mailto:5...@domainb.org. mailto:5...@domainb.org transfers to mailto:5...@domainb.org. The transfer fails. Both domainA and domainB are running on identical sipx servers all on the same private network with no firewalls in between. Mike ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
Decent FAX (t.38) capable ATA's are able able to be connected to sipx without issue. Gerald says he got t.38 outbound via hylafax so you can obviously go with that as an example. I think you might be the only one bent on Outbound fax from the desktop with Hylafax. Hard to justify integrating with a project that hasn't had a update in 19 months. Since sipx is using the FS media server to actually handle t.38 inbound, it only makes it logical to follow FS for outbound, which does not mean hylafax. Hylafax is client intensive in that you have to load their print clients on the PC's. The rest of the world is really moving towards web based services: in sipx that would be its portal. This way we don't have to worry about drivers and direct connectivity with the drivers/PC to a sipx system. t.37 would work similarly. There are simpler protocols (t.37) and integration (portal) that can leveraged to send faxes. It would also lend itself to an HA environment. Hylafax doesn't even have searchable archives since February. There were only 5 subjects with a total of 7 messages in February. Hard to justify (again) getting involved with a project that is not currently active. I, for one, would not be suggesting a Hylafax integration OVER a real sipx integration for sure. There is nothing keeping someone from setting up a hylafax server and connecting it to sipx, because sipx is open enough. I don't think the majority of corporate clients want to continue managing printer drivers and teaching people to send faxes. There are easier ways. On Mon, Apr 16, 2012 at 10:12 AM, m...@grounded.net m...@grounded.net wrote: IMHO, you can setup the fanciest outbound faxing solution on the market and 90% of your users will still print out things and walk up to a fax machine so that they get things out in the order they want and they don't trust the fax server to do it. I guess it depends on who your market it. Ours is small businesses and organizations and the majority of those are still using fax machines. We always try to show them how they could be virtualizing their methods. Inbound is a whole different ball of wax though. Well, that also means there are plenty of folks still sending out :). Anyhow, I don't care, I'm using hylafax, it would simply be nicer to have it all in one. Mike Mike On Sun, Apr 15, 2012 at 9:26 PM, m...@grounded.net m...@grounded.net wrote: So is there an existing place to vote for this? Does anyone else want to see this in sipx? On Sat, 14 Apr 2012 09:07:03 -0400, Gerald Drouillard wrote: On 4/13/2012 8:20 PM, m...@grounded.net wrote: I believe I saw a thread a while back where someone was asking about sending faxes. Some searching shows that some have asked but that there are no plans. Is this still the case or are others interested in this? Even a shared outgoing account as a 'group' would be so very welcome and would instantly eliminate our having to use additional hylafax/avantfax servers just for this function. It would be way nicer to be able to tell potential customers that everything can be done from the one system. We recently had an install that was a heavy hylafax user with usb modems. We are now using sipx for receiving faxes and hylafax for sending. http://www.drouillard.biz/blog/sipxecs-hylafax-and-t38modem/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
Then I can refer you back to the original fax tracker item and to create a feature improvement to pick up where that left off and ask for community support/votes. On Mon, Apr 16, 2012 at 10:49 AM, m...@grounded.net m...@grounded.net wrote: I think you might be the only one bent on Outbound fax from the desktop with Hylafax. Hard to justify integrating with a project that hasn't had a update in 19 months. Since sipx is using the FS media Nah, not bent on it, I just like seeing all in one solutions when ever possible. With outgoing, sipx has it all in my opinion. Right now, for those who absolutely need a physical fax, we have an ATA at their location and haven't had any problems to date. Being able to show people how they can move away from physical faxing however is something we try to do. I, for one, would not be suggesting a Hylafax integration OVER a real sipx integration for sure. I thought we were talking about a real integration and not using hylafax on the same server? Mind you, I did make a mention that it would be ok to have to use a separate server which could be part of the sipx install. No big deal, if not many are interested, then it's a moot point. Mike ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
On Mon, Apr 16, 2012 at 12:20 PM, m...@grounded.net m...@grounded.net wrote: Just wanted to point out that hylafax does have email to fax gateways, linux command line sending, and print drivers for outbound. Although it does seem like a lot of extra baggage to have to install hylafax, it does provide a solution for high outbound fax sites. Yup, it's pretty cool, basically, you just send a print job to it from remote. Well, this is called t.37, and sipx does not need hylafax in order to implement t.37, as said before. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] site-to-site transfers
Good. On Apr 16, 2012 3:07 PM, Mike Graham mike_gra...@hempfieldsd.org wrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: camgknjwtdrrrwpgtplhomg9j8jw3tj0ke0unnakfvauouqy...@mail.gmail.com X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67516 Message-ID: 107bc.4f8c6...@forum.sipfoundry.org Problem solved, at least in the lab - we're testing in production tomorrow. I needed both domains listed as Intranet Domains under System Internet Calling. After adjusting some other settings, call control was working, but there was no audio. I'm guessing sipX was trying to do some sort of translation and sending the wrong IP to the other phone. Mike ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [Assp-test] Attachment problem
The problem fixed itself after today I applied an EXIM upgrade , even if it's weird since with older ASSP versions 1.8.1.7(0.0.00) and older it was working . Graziano I tried older ASSP versions and I found this current version 1.9 does not work ... ... other tested no works ... 1.8.5.9(0.0.08) does not work 1.8.5.9(0.0.07) refuse to start 1.8.5.9(0.0.06) works 1.8.5.9 (0.0.01) works other tested all working 1.8.1.7(0.0.00) works to resume latest ASSP version which passes the email below is 1.8.5.9(0.0.06) . After 1.8.5.9(0.0.06) there is NO ASSP version which is able to pass the email below to MTA . Graziano also using noprocessing I can't find a way to pass this email to MTA . i...@jancr.com and ro...@ciam.com are local email . The email is sent using a Miva web form . If I disable ASSP , the email is processed by EXIM with no problem at all . Graziano same problem with latest 1.9.6.7(0.0.06) Graziano Hello using 1.9.6.5(0.0.05) The email below (local good email) is going in black hole , never reaches MTA and disconnects after 0 seconds (?). .dat is allowed in GoodAttach Apr-13-12 11:55:45 id-33433-00011 [Attachment] 127.0.0.1i...@jancr.com to: ro...@ciam.com info: found attachment 'orders.dat'; Apr-13-12 11:55:45 id-33433-00011 [LocalOK] 127.0.0.1i...@jancr.com to: ro...@ciam.com local - 127.0.0.1 in acceptAllMail - [Order Export] - attachment 'orders.dat' ; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1i...@jancr.com to: ro...@ciam.com finished message - received DATA size: 48.09 kByte - sent DATA size: 48.04 kByte; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1i...@jancr.com to: ro...@ciam.com disconnected (0 seconds); Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
Re: [sipx-users] HA setup with more than 2 redundant servers
I would suspect ipsec would work well, we use it all the time for site to site vpn's. On Apr 14, 2012 10:46 AM, Nitin Mirchandani nitin_mirchand...@hotmail.com wrote: The only thing that can raise issue is that the sites are connected via PPTP VPN. Latency is sub 10ms. Will MS PPTP(the BEST implemetation so far - as it is transparent to all protocols) be an issue? We had issues with OpenVPn which is half ass implemetation. Does the tunnel (which sipx uses) can have issues with PPTP? I can see that sometime later the sending profile fails. Rgds Nitin -- Date: Sat, 14 Apr 2012 03:28:51 -0400 From: mpic...@ezuce.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] HA setup with more than 2 redundant servers have many customers running with 3-5 servers... your statement is incorrect. i would say that your installation is unstable with more than 2 servers. mike On Sat, Apr 14, 2012 at 3:14 AM, Nitin Mirchandani nitin_mirchand...@hotmail.com wrote: Of course I need to send profiles - But when its done, it starts failing in different ways. Each time error is different. If I switch off one server(redundant), and send profiels, it starts working again. Since its failing in different ways, and dns is correct (i have rechecked hundreds of times), I think sipx is unstable on 3 servers. -- Date: Fri, 13 Apr 2012 09:44:09 -0400 From: mpic...@ezuce.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] HA setup with more than 2 redundant servers you always need to send profiles after the server pulls down it's configuration stuff... On Fri, Apr 13, 2012 at 9:27 AM, George Niculae geo...@ezuce.com wrote: 2012/4/13 Nitin Mirchandani nitin_mirchand...@hotmail.com Hello Installed it 5 or 6 time. Now, the third server has come with service disabled Maybe - 4.4 doesnot support more than 2 servers? Send profiles to that particular server George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] HA setup with more than 2 redundant servers
You can simulate this inside but I think pptp is a poor choice in that the packets use a different IP address to be native to the network. IPSEC AND OPENVPN are more suited to point to point and can properly tested with the same network schema before rolling the systems out and over a vpn. Pptp is really REALLY a poor choice here. You might not understand but the routing and broadcast of how pptp works in a firewall advertises and routes thins differently. It might be fine for some remote user but as a site to site con it is somewhat crippling (IMO). On Apr 14, 2012 11:42 AM, Todd Hodgen thod...@frontier.com wrote: First mention of a Tunnel between the servers. Again, I’d recommend you get this working in a lab first, without a tunnel. Then add a tunnel between servers and see if it breaks things. You will likely see what is breaking it very quickly, and without guessing. ** ** *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Nitin Mirchandani *Sent:* Saturday, April 14, 2012 7:46 AM *To:* sipx-users@list.sipfoundry.org *Subject:* Re: [sipx-users] HA setup with more than 2 redundant servers*** * ** ** The only thing that can raise issue is that the sites are connected via PPTP VPN. Latency is sub 10ms. Will MS PPTP(the BEST implemetation so far - as it is transparent to all protocols) be an issue? We had issues with OpenVPn which is half ass implemetation. Does the tunnel (which sipx uses) can have issues with PPTP? I can see that sometime later the sending profile fails. Rgds Nitin -- Date: Sat, 14 Apr 2012 03:28:51 -0400 From: mpic...@ezuce.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] HA setup with more than 2 redundant servers have many customers running with 3-5 servers... ** ** your statement is incorrect. i would say that your installation is unstable with more than 2 servers. ** ** mike On Sat, Apr 14, 2012 at 3:14 AM, Nitin Mirchandani nitin_mirchand...@hotmail.com wrote: Of course I need to send profiles - But when its done, it starts failing in different ways. Each time error is different. If I switch off one server(redundant), and send profiels, it starts working again. Since its failing in different ways, and dns is correct (i have rechecked hundreds of times), I think sipx is unstable on 3 servers. -- Date: Fri, 13 Apr 2012 09:44:09 -0400 From: mpic...@ezuce.com To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] HA setup with more than 2 redundant servers you always need to send profiles after the server pulls down it's configuration stuff... On Fri, Apr 13, 2012 at 9:27 AM, George Niculae geo...@ezuce.com wrote:* *** 2012/4/13 Nitin Mirchandani nitin_mirchand...@hotmail.com Hello Installed it 5 or 6 time. Now, the third server has come with service disabled Maybe - 4.4 doesnot support more than 2 servers? ** ** ** ** Send profiles to that particular server ** ** George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ** ** -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com ** ** There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ** ** -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com ** ** There are 10 kinds of people in the world, those who understand binary and those who don't. ** ** ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- LAN/Telephony/Security
Re: [Assp-test] Attachment problem
same problem with latest 1.9.6.7(0.0.06) Graziano Hello using 1.9.6.5(0.0.05) The email below (local good email) is going in black hole , never reaches MTA and disconnects after 0 seconds (?). .dat is allowed in GoodAttach Apr-13-12 11:55:45 id-33433-00011 [Attachment] 127.0.0.1info@jancr to: ro...@ciam.com info: found attachment 'orders.dat'; Apr-13-12 11:55:45 id-33433-00011 [LocalOK] 127.0.0.1info@jancr to: ro...@ciam.com local - 127.0.0.1 in acceptAllMail - [Order Export] - attachment 'orders.dat' ; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1info@jancr to: ro...@ciam.com finished message - received DATA size: 48.09 kByte - sent DATA size: 48.04 kByte; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1info@jancr to: ro...@ciam.com disconnected (0 seconds); Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
Re: [Assp-test] Attachment problem
I tried older ASSP versions and I found this current version 1.9 does not work ... ... other tested no works ... 1.8.5.9(0.0.08) does not work 1.8.5.9(0.0.07) refuse to start 1.8.5.9(0.0.06) works 1.8.5.9 (0.0.01) works other tested all working 1.8.1.7(0.0.00) works to resume latest ASSP version which passes the email below is 1.8.5.9(0.0.06) . After 1.8.5.9(0.0.06) there is NO ASSP version which is able to pass the email below to MTA . Graziano also using noprocessing I can't find a way to pass this email to MTA . i...@jancr.com and ro...@ciam.com are local email . The email is sent using a Miva web form . If I disable ASSP , the email is processed by EXIM with no problem at all . Graziano same problem with latest 1.9.6.7(0.0.06) Graziano Hello using 1.9.6.5(0.0.05) The email below (local good email) is going in black hole , never reaches MTA and disconnects after 0 seconds (?). .dat is allowed in GoodAttach Apr-13-12 11:55:45 id-33433-00011 [Attachment] 127.0.0.1i...@jancr.com to: ro...@ciam.com info: found attachment 'orders.dat'; Apr-13-12 11:55:45 id-33433-00011 [LocalOK] 127.0.0.1i...@jancr.com to: ro...@ciam.com local - 127.0.0.1 in acceptAllMail - [Order Export] - attachment 'orders.dat' ; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1i...@jancr.comto: ro...@ciam.com finished message - received DATA size: 48.09 kByte - sent DATA size: 48.04 kByte; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1i...@jancr.comto: ro...@ciam.com disconnected (0 seconds); Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
[sipx-dev] 4.5.2 .. adding alias stops apache
I added a domain alias and then found sipxconfig to be running. All web queries were rejected. I had to manually start apache in order to regain connectivity with sipxconfig interface. Is there, or will there be, any type of watchdog made available in 4.6 so apache stays up? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
[sipx-dev] Unknown Test Result -2147483648 for all tests in 4.5.2
All tests are failing. 2012-04-13T12:22:00.89Z:162:JAVA:ERR:sipx.dev.myitdepartment.net:pool-4-thread-1::ExternalCommand:Cannot execute: preflight java.io.IOException: Cannot run program /usr/bin/preflight: java.io.IOException: error=2, No such file or directory I also realized after copying that file over all of the sipx-test-* files were missing so I copied those over as well and had some mixed results: Test Name Last Time Run Status DNS IP:Name resolver 4/13/12 8:30 AM Success SSL certificate 4/13/12 8:30 AM Warning Unknown test result: 127 Show details SELinux 4/13/12 8:30 AM Success Configuration files consistency 4/13/12 8:30 AM Success 'localhost' configuration4/13/12 8:30 AM Success 127.0.0.1 configuration 4/13/12 8:30 AM Error Invalid mapping for 127.0.0.1 The 127.0.0.1 address should map to only the names 'localhost.localdomain' and 'localhost'. Any other name for that address may cause routing or authentication errors. Remove any names from the 127.0.0.1 line in /etc/hosts except for 'localhost.localdomain' and 'localhost'. Show details Temporary directory 4/13/12 8:30 AM Success Apache HTTP server 4/13/12 8:30 AM Error Apache HTTP server verification error Show details Hostname 4/13/12 8:30 AM Warning Unknown test result: 127 Show details DHCP Test Show Detailed Help 4/13/12 8:30 AM Warning Unknown test result: 1 Show details DHCP (Option 120) Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details DNS Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details NTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details TFTP Test4/13/12 8:30 AM Warning Unknown test result: 1 Show details FTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details HTTP Test4/13/12 8:30 AM Warning Unknown test result: 1 *** Disclaimer: I used the sipx-test-* binaries from a 4.4 build to get these results. I see the system has its certificate generated. I expect the binary simply doesn't know how to interpret the result since its the binary from 4.4 that I used. Is this a known issue? Is there any other information I can provide? Is a JIRA warranted? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] Unknown Test Result -2147483648 for all tests in 4.5.2
I failed to mention that /usr/bin/preflight was not present and I used the 4.4 binary for that also to see what else was missing... Are these slated to be different in 4.6 or were they just not made available so they could be changed to reflect the changes in sipx overall? 2012-04-13T12:26:03.403000Z:171:JAVA:ERR:sipx.dev.myitdepartment.net:pool-5-thread-1::ExternalCommand:Cannot execute: sipx-test-hostname java.io.IOException: Cannot run program /usr/bin/sipx-test-hostname: java.io.IOException: error=2, No such file or directory The file is there with permissions 0755 (root:root) On Fri, Apr 13, 2012 at 8:36 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: All tests are failing. 2012-04-13T12:22:00.89Z:162:JAVA:ERR:sipx.dev.myitdepartment.net:pool-4-thread-1::ExternalCommand:Cannot execute: preflight java.io.IOException: Cannot run program /usr/bin/preflight: java.io.IOException: error=2, No such file or directory I also realized after copying that file over all of the sipx-test-* files were missing so I copied those over as well and had some mixed results: Test Name Last Time Run Status DNS IP:Name resolver 4/13/12 8:30 AM Success SSL certificate 4/13/12 8:30 AM Warning Unknown test result: 127 Show details SELinux 4/13/12 8:30 AM Success Configuration files consistency 4/13/12 8:30 AM Success 'localhost' configuration 4/13/12 8:30 AM Success 127.0.0.1 configuration 4/13/12 8:30 AM Error Invalid mapping for 127.0.0.1 The 127.0.0.1 address should map to only the names 'localhost.localdomain' and 'localhost'. Any other name for that address may cause routing or authentication errors. Remove any names from the 127.0.0.1 line in /etc/hosts except for 'localhost.localdomain' and 'localhost'. Show details Temporary directory 4/13/12 8:30 AM Success Apache HTTP server 4/13/12 8:30 AM Error Apache HTTP server verification error Show details Hostname 4/13/12 8:30 AM Warning Unknown test result: 127 Show details DHCP Test Show Detailed Help 4/13/12 8:30 AM Warning Unknown test result: 1 Show details DHCP (Option 120) Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details DNS Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details NTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details TFTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details FTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 Show details HTTP Test 4/13/12 8:30 AM Warning Unknown test result: 1 *** Disclaimer: I used the sipx-test-* binaries from a 4.4 build to get these results. I see the system has its certificate generated. I expect the binary simply doesn't know how to interpret the result since its the binary from 4.4 that I used. Is this a known issue? Is there any other information I can provide? Is a JIRA warranted? -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] Unknown Test Result -2147483648 for all tests in 4.5.2
OK, thanks! On Fri, Apr 13, 2012 at 8:42 AM, George Niculae geo...@ezuce.com wrote: On Fri, Apr 13, 2012 at 3:36 PM, Tony Graziano tgrazi...@myitdepartment.net wrote: All tests are failing. Is this a known issue? Is there any other information I can provide? Is a JIRA warranted? Yep, recorded already: http://track.sipfoundry.org/browse/XX-10069 There are some scripts in old tools package that were not ported yet, I moved just the one used in getting snapshots. That's the filter I created for 4.6: http://track.sipfoundry.org/secure/IssueNavigator.jspa?mode=hiderequestId=11011 George ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] 4.5.2 .. adding alias stops apache
(added apachectl start to startup routine manually), rebooted system, apache is up. add alias, within 30 seconds apache stops and does not restart... It appears there still should be a watchdog or monitor on apache (httpd) that calls apachectl to start if it is down... I do not see that on the issues list for 4.6. On Fri, Apr 13, 2012 at 8:30 AM, Kumaran thiru.venkateshwa...@ttplservices.com wrote: I had created a issue XX-10109 http://track.sipfoundry.org/browse/XX-10109 (Httpd service is shutdown after reboot)So please add the comments in the issue if its related to it...So it will be easy to track it down. Regards, Kumaran T Douglas Hubler wrote: On Fri, Apr 13, 2012 at 8:17 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: I added a domain alias and then found sipxconfig to be running. All web queries were rejected. I had to manually start apache in order to regain connectivity with sipxconfig interface. Is there, or will there be, any type of watchdog made available in 4.6 so apache stays up? there will be. I was planning on using the snmpd service is manage this, but haven't set it up yet. It has a config parameter to manage this for each service. What i don't know is how often it checks. cf-execd could also, but I'd rather use snmpd if i can. ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-dev] 4.5.2 .. adding alias stops apache
error_log shows [Fri Apr 13 09:07:02 2012] [notice] suEXEC mechanism enabled (wrapper: /usr/sbin/suexec) [Fri Apr 13 09:07:02 2012] [notice] Digest: generating secret for digest authentication ... [Fri Apr 13 09:07:02 2012] [notice] Digest: done [Fri Apr 13 09:07:02 2012] [notice] Apache/2.2.15 (Unix) DAV/2 mod_ssl/2.2.15 OpenSSL/1.0.0-fips configured -- resuming normal operations [Fri Apr 13 09:07:11 2012] [error] [client 127.0.0.1] Client sent malformed Host header [Fri Apr 13 09:07:29 2012] [notice] caught SIGTERM, shutting down ssl_error_log shows: [Fri Apr 13 09:07:02 2012] [warn] RSA server certificate is a CA certificate (BasicConstraints: CA == TRUE !?) [Fri Apr 13 09:07:02 2012] [warn] RSA server certificate is a CA certificate (BasicConstraints: CA == TRUE !?) The error log showing a malformed request from 127.0.0.1, is not suspicious, it's not running. I will open a JIRA. Any last comments? On Fri, Apr 13, 2012 at 9:03 AM, Douglas Hubler dhub...@ezuce.com wrote: On Fri, Apr 13, 2012 at 8:59 AM, Kumaran thiru.venkateshwa...@ttplservices.com wrote: Already httpd service is shutdown was discussed with Douglas..He told that fix will ready...So still there issue regarding it... we run chkconfig --add httpd so chkconfig --list httpd should show that it should start w/system. If that's not what you see, then the bug is that apache has a failure starting w/system for some reason we'd need to investigate. I would urge you to do a fresh install as there were a number of fixes in this area that a yum update might not fix. Either we can add our comments in the issue I created or open a new issue that apache stops after 30secs and doesn't restart... check logs in /var/log/httpd make sure it's not your system, otherwise it's fine to append to open issue ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-dev mailing list sipx-dev@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-dev/
Re: [sipx-users] MOH and Call Park (4.4 latest)
http://track.sipfoundry.org/browse/XX-10111 On Fri, Apr 13, 2012 at 4:56 AM, Joegen Baclor jbac...@ezuce.com wrote: There is a current limitation in media relay and sipxproxy where it does not proxy the media for INVITE with no SDP. This is what breaks MoH for remote workers. If someone could open a tracker, I will schedule fixing it when time permits. On 04/11/2012 04:16 PM, Michael Picher wrote: I believe this is a current bug. On Tue, Apr 10, 2012 at 10:43 PM, Jimmy dimitri_mano...@yahoo.com wrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: CAMgKNJUsAHehseitn1KObdN1bDG8Zs69nUGi7rM4D0HEgB痄@mail.gmail.comcamgknjusahehseitn1kobdn1bdg8zs69nugi7rm4d0hegb%2b...@mail.gmail.com X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67386 Message-ID: 1073a.4f84e...@forum.sipfoundry.org Is your phone located remotely or locally where the server is. If remotely I also don't get moh when placing a call on hold. When the phone is local on the same network as the server I get moh when placing the call on hold. MOH doesn't work when ip550, 650, IP670, IP321, IP335 is remotely. Running SIPXECS 4.4 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] problem with phonelogd
If it were me, I would look up the rsyslogd error in the Centos forum. I do see errors with rsyslogd in their forums as people try to compile, etc. This is not an answer, I know. If rsyslogd is happier with an older version, so be it. On Fri, Apr 13, 2012 at 2:47 AM, Ivan Pletenev i.plete...@gmail.com wrote: 1. you mean rsyslog.conf? it's very common, i didn't change it. here is it(i removed strings with # to shorten it): *.info;mail.none;authpriv.none;cron.none/var/log/messages authpriv.* /var/log/secure mail.* -/var/log/maillog cron.* /var/log/cron *.emerg * uucp,news.crit /var/log/spooler local7.*/var/log/boot.log 2. I installed sipxecs from the ISO image [root@sipx rsyslog]# uname -a Linux sipx 2.6.18-274.18.1.el5 #1 SMP Thu Feb 9 12:45:52 EST 2012 i686 i686 i386 GNU/Linux [root@sipx rsyslog]# lsb_release -a LSB Version: :core-4.0-ia32:core-4.0-noarch:graphics-4.0-ia32:graphics-4.0-noarch:printing-4.0-ia32:printing-4.0-noarch Distributor ID: CentOS Description:CentOS release 5.8 (Final) Release:5.8 Codename: Final 3. I didn't install rsyslogd specially. i just installed sipxecs and configured syslog. then i made yum update and got this error. after that i did yum reinstall rsyslog, but no changes. And the last thing i've just made: yum downgrade rsyslog. after that i have 2.0.6-1.el5 version instead of 3.22.1-7.el5 and it works now Should i leave old version or what? Ivan Pletenev ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] MOH and Call Park (4.4 latest)
To get some attention to it, votes would be appropriate. On Fri, Apr 13, 2012 at 6:52 AM, Tony Graziano tgrazi...@myitdepartment.net wrote: http://track.sipfoundry.org/browse/XX-10111 On Fri, Apr 13, 2012 at 4:56 AM, Joegen Baclor jbac...@ezuce.com wrote: There is a current limitation in media relay and sipxproxy where it does not proxy the media for INVITE with no SDP. This is what breaks MoH for remote workers. If someone could open a tracker, I will schedule fixing it when time permits. On 04/11/2012 04:16 PM, Michael Picher wrote: I believe this is a current bug. On Tue, Apr 10, 2012 at 10:43 PM, Jimmy dimitri_mano...@yahoo.comwrote: Content-Type: text/plain; charset=utf-8 Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: CAMgKNJUsAHehseitn1KObdN1bDG8Zs69nUGi7rM4D0HEgB痄@mail.gmail.comcamgknjusahehseitn1kobdn1bdg8zs69nugi7rm4d0hegb%2b...@mail.gmail.com X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 67386 Message-ID: 1073a.4f84e...@forum.sipfoundry.org Is your phone located remotely or locally where the server is. If remotely I also don't get moh when placing a call on hold. When the phone is local on the same network as the server I get moh when placing the call on hold. MOH doesn't work when ip550, 650, IP670, IP321, IP335 is remotely. Running SIPXECS 4.4 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Some questions regarding 0.0.4.5.2 and 4.6
I don't have a cheatsheet for myself yet. It now uses: cf-execd cf-execd is scheduler and reporter in cfengine client hosts. cf-monitord This process maintains state information about the client cf-serverd cf-serverd is responsible from giving out configuration files to There is no central control start/stop mechanism, nor a sipxproc cli status or management binary that I know of yet. On my lab system these are all independently started or listed services. sipxbridge sipxbridge is an SIP Trunking server sipxconfig sipxconfig is an administration server sipxfreeswitch Freeswitch fedora init script config: sipximbotsipximbot is an IM Bot subystem that uses FreeSWITCH as a media server. sipxivr sipxivr is a IVR subystem that uses FreeSWITCH as a media server sipxlogwatcher Simple Event Correlator script to filter log file entries sipxopenfire sipxopenfire is an administration server sipxproxysipxproxy is a SIP proxy for telecommunications sipxpublishersipxpublisher is a SIP MWI server for telecommunications sipxregistrarsipxregistrar is a SIP registrar for telecommunications sipxrelaysipxrelay is a media relay for telecommunications sipxsupervisor sipxsupervisor is responsible from giving out configuration files to It will also be using apachectl to start httpd in a more native fashion. On my Cetnos 6.2 box I think I had to manually start sipxconfig and a few other things but I have not done a fresh install in a while. IMO 4.5.2 is still al little rough around the edges but there are commits every day to wrinkle out the basic usability bugs. It's different for sure. On Fri, Apr 13, 2012 at 7:17 AM, Jan Fricke jan.fri...@iant.de wrote: Hi, I’m trying to install a testsystem based on 0.0.4.5.2 and got some problems. - What is the state of the 0.0.4.5.2 iso? Just installed it and can’t update because it is CentOS 5.x based and there are only repos for CentOS 6 and Fedora. - What's wrong with the way I try to install 0.0.4.5.2 on CentOS 6 from the repo? - Installed CentOS 6.2 minimal - Hostname sipx.test.local - configured static ip - disabled selinux iptables - yum update - got http://download.sipfoundry.org/pub/sipXecs/sipxecs-0.0.4.5.2-centos.repo - yum install epel-release - yum groupinstall sipxecs - reboot - sipxecs-setup Is this the first server in your cluster? [ enter 'y' or 'n' ] : y Configuring as the first server... Enter system host name without the domain name [ press enter for 'sipx' ] : Enter domain name : test.local Tip: Use 'sipx.test.local' as your SIP domain if you are setting up for the first time or if you know you are only going to setup one server. This can make configuration easier. You can always change the value later. Enter SIP domain name [ press enter for 'test.local' ] : Enter SIP realm [ press enter for 'test.local' ] : Configuring system, this may take a few minutes... done. - What now? There are no scripts to start services (/etc/init.d/sipxecs start). No sipXecs-system-setup that e.g. creates dns zones or configures dhcp. Did I miss an wiki article? Is there any documentation how to install it? - Is there a release date for SipX 4.6 and openUC 4.6? Jan _ Jan Fricke (B.Sc.) IANT - APPLIED NGN-TECHNOLOGIES Turn-Key VoIP/UC Solutions and More... Fon: +49 (5331) 6794 0 Fax: +49 (5331) 6794 499 Mail: jan.fri...@iant.de Web: www.iant.de IANT is eZuce Elite Partner for EMEA IANT is Member of GROUPLINK ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
Re: [sipx-users] Multiple line appearances
both as in on the phone menu (from keypad itself). Menus, Settings, Advanced, (pass), Admin Settings, Line Config... On Fri, Apr 13, 2012 at 10:59 AM, Stiles Watson wat...@datatek-net.com wrote: Todd, I'm trying to figure out where to do this. Are you saying you have to go to the physical phone itself or are you referring to the sipX web-interface for the device line? Regarding the sipX interface for the device if not using phone groups, under Call Handing, there is a Calls Per Line Key field which defaults to 24. Does this need to be changed to 1? After clicking on the only line assigned to this device, under Registration, I have lineKeys set to 2 and callsPerLineKey set to 1. Is there anything else which needs to be configured? Stiles On 04/12/2012 06:08 PM, Todd Hodgen wrote: You need to set this in two places on the phone. Go to the Device itself, and set it to two line appearances, and one line. Next, go to the Line for the device, and do the same thing. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson Sent: Thursday, April 12, 2012 8:58 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Multiple line appearances That is what I did. I have one line assigned to the phone and the registration settings as below. Stiles On 04/12/2012 11:54 AM, Tony Graziano wrote: You register the SAME LINE on both line appearances and set the limit to 1 for each. On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.com wrote: According to the sipX book (p 133): Most multiple line IP hardware phones allow multiple calls to a single line. This can be quite confusing for the average phone user and difficult to deal with at an answering position. To remedy this problem it is easier for the user to have multiple appearances of the same line on their telephone. Each successive call will ring on the next line appearance. I am using the recommended setup on the Polycom 335 phones, but the behaviour is not as indicated. When the second call comes in, it does not ring on the successive line, but goes straight to v-mail. I'm using sipX 4.4. In my phone group, under the Registration section, I have lineKeys set to 2 (the 335 has two line keys) and I have callsPerLineKey set to 1. After sending the profiles to the phones, they show the correct ext on each line. I can make two outgoing calls, but I can only receive one. I have also tried assigning the same ext twice to the phone. This again results in the same ext appearing on each line key as desired, but when the second call comes in, it does not ring the second line, but rings the first line. If I answer an incoming call while on another call, the first call goes to hold (this is an assumption because I hear the MOH music), but I do not see how to put the second call on hold and retrieve the first call. If I hang up the second call without picking up the first, they both get disconnected. I'd rather have the solution indicated in the book. Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Multiple line appearances
if you change the default from 24 to one for the device, there is no way I can see that this will work. The device needs to allow MORE THAN ONE call, where the limit is placed is on the line itself. I think you need to go back and UNDO the setting for 1 on the device and change it back. On Fri, Apr 13, 2012 at 12:33 PM, Stiles Watson wat...@datatek-net.com wrote: This is exactly what I've done. In the devices menu, I click on the MAC address Lines, click on the only line, then Registration and set the values of lineKeys and callsPerLineKey. Once this is done, if I go back to the devices menu and click on the line instead of the MAC address, the values are already filled in (since this is just an alternative path to the previous one). At this point I'm not using any phone groups and I've only made two other modifications to the default config for this device. First, in the devices menu, click on MAC address Call Handling, I've changed the value from the default to 1 (I get the same result if 1 or the default of 24). And second, MAC address Dial Plan, I've set a custom digitmap: [2-9]11|0T|RR9R011xxx.T|9011xxx.T|RR91R[2-9]x|RR9R1[2-9]x|91[2-9]x|RR91919R[2-9]xxT|91919[2-9]xx|*xx.T|[8]xxxT|[1]xxT These are the only mods I've made to the device and line. I then send the profile to the phone and after reboot, I verify the values, by Menu Settings Advanced password Admin Settings Line Configuration Calls/LnKey: 1 Line1 Line Keys No. Line Keys: 2 Line1 Line Keys Calls/LnKey: 1 Line2 Line Keys No. Line Keys: blank Line2 Line Keys Calls/LnKey: blank If I set the registration values in a phone group, the Line2 fields match the Line1 settings. With or without the group the result is the same behavior: Two outbound calls, one inbound. Second inbound call gets, user of extension is not available, and call goes to v-mail. There are no other required settings that I'm aware of. Again, my setup is Polycom SoundPoint 335 IP, firmware 3.2.6.0314, BootROM 4.2.2.0710, sipXecs v 4.4, no PSTN, US CANADA SIP trunk is voip.ms, International SIP Trunk is callwithus. At this point there is no DID to this ext, but there will be when system is live. All inbound calls come through Auto Attend. There are no call forwarding rules for this user. This user in not in any Hunt group. Stiles On 04/13/2012 11:28 AM, Todd Hodgen wrote: I’ve always done it from the sipXecs gui. It’s never required anything at the phone itself. When you are in the devices menu, it lists all of the devices – Mac address, and lines assigned. Click on the line assigned, set it there, and click on the mac address – assign it there. Maybe there is a better way, but this method does work for me. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson Sent: Friday, April 13, 2012 8:00 AM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Multiple line appearances Todd, I'm trying to figure out where to do this. Are you saying you have to go to the physical phone itself or are you referring to the sipX web-interface for the device line? Regarding the sipX interface for the device if not using phone groups, under Call Handing, there is a Calls Per Line Key field which defaults to 24. Does this need to be changed to 1? After clicking on the only line assigned to this device, under Registration, I have lineKeys set to 2 and callsPerLineKey set to 1. Is there anything else which needs to be configured? Stiles On 04/12/2012 06:08 PM, Todd Hodgen wrote: You need to set this in two places on the phone. Go to the Device itself, and set it to two line appearances, and one line. Next, go to the Line for the device, and do the same thing. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson Sent: Thursday, April 12, 2012 8:58 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Multiple line appearances That is what I did. I have one line assigned to the phone and the registration settings as below. Stiles On 04/12/2012 11:54 AM, Tony Graziano wrote: You register the SAME LINE on both line appearances and set the limit to 1 for each. On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.com wrote: According to the sipX book (p 133): Most multiple line IP hardware phones allow multiple calls to a single line. This can be quite confusing for the average phone user and difficult to deal with at an answering position. To remedy this problem it is easier for the user to have multiple appearances of the same line on their telephone. Each successive call will ring on the next line appearance. I am using the recommended setup on the Polycom 335 phones, but the behaviour is not as indicated. When the second call comes
Re: [sipx-users] Bug fix release update: sipXecs 4.4.0 update #16
introduced to retrieve details for owned active conference
-test added
-example: curl --digest -k -u 400:123
https://host:8443/sipxconfig/rest/my/conferencedetails/{confName}
commit 1aba6896e290df25fbbcd66e95ea077da07df8e8
Author: Mircea Carasel mirc...@ezuce.com
Date: Thu Jan 12 19:53:32 2012 +0200
XX-10005: sipXivr REST api (port 8085) cannot authenticate when
LDAP authentication is selected
-create sipXconfig RestRedirectorResource to bypass all calls to
callcontroller, cdr and ivr through sipXconfig
-improved LoginDetails rest api to contain information wether
ldap-openfire auth is enable
-updated ivr, callcontroller, cdr authenticators to validate
requests from trusted host
-test added
commit 4fa2fd2bef1e8bb38aac95e4872a8969893d3d7d
Author: admin admin@testhost.opsip.local
Date: Thu Feb 9 11:32:17 2012 +0100
XX-10034: Italian version: if there are no active participants in
a conference the system is counting 1 active
for past releases see
http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
~~
Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our Internet Fax services!
~~
--
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpd...@voice.myitdepartment.net
Helpdesk Customers: http://myhelp.myitdepartment.net
Blog: http://blog.myitdepartment.net
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
I think the way to approach this is with a feature request in the JIRA. I don't know that this is the way to go (Hylafax/Avantfax). Reasonably the user portal could be modified to be able to upload and send faxes. At the same time there is an established protocol, called t.37, to send faxes via email. I've delved into Hylafax and others and don't see they are scalable or flexible nor are they active communities to draw support from. Sipx has all the framework to do this with what it already has by building in the functionality into the user portal or by adding t.37 support to it. http://track.sipfoundry.org/browse/XX-8645 The original JIRA (above) had suggested 9 items.The core items needed are already implemented. There were a few suggestions that just didn't matter (I know, I wrote those). The core IVR items (IVR changes to tell you how many faxes and accessing stored faxes via sipxui) have not been done yet, and when they are scheduled this would be the time to suggest uploading files for sending faxes out or additng t.37 support. Freeswitch is very capable of doing either of these. In any case, a new JIRA is probably warranted and votes would matter. In the meantime, there are several simple, diskless appliances that do this today. Alternately, an ATA with t.38 support can be connected to a fax machine and it will work if the PSTN connection is able to support t.38 also (we do it all the time). If the votes don't accumulate, I don't see how it would make sense to spend the manpower to do it. To some organizations faxes are important, but certainly the volume in most organizations is declining. The dialplan stuff is easy, the portal stuff might be a little more time consuming, and so would the IVR stuff. On Fri, Apr 13, 2012 at 8:20 PM, m...@grounded.net m...@grounded.net wrote: I believe I saw a thread a while back where someone was asking about sending faxes. Some searching shows that some have asked but that there are no plans. Is this still the case or are others interested in this? Even a shared outgoing account as a 'group' would be so very welcome and would instantly eliminate our having to use additional hylafax/avantfax servers just for this function. It would be way nicer to be able to tell potential customers that everything can be done from the one system. Mike ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Ticket Received - [#577] Sending faxes
Evidently I did. Sorry about that. On Fri, Apr 13, 2012 at 8:30 PM, m...@grounded.net m...@grounded.net wrote: Tony, have you got something messed up? We would like to acknowledge that we have received your request and a ticket has been created with Ticket ID - 577. A support representative will be reviewing your request and will send you a personal response.(usually within 24 hours). To view the status of the ticket or add comments, please visit http://myhelp.myitdepartment.net/helpdesk/tickets/xxx Thank you for your patience. Sincerely, myITdepartment Support Team ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Sending faxes
On Fri, Apr 13, 2012 at 8:55 PM, m...@grounded.net m...@grounded.net wrote: I've delved into Hylafax and others and don't see they are scalable or flexible nor are they active communities to draw support from. Maintaining Hylafax/Avantfax server is a total headache. The smallest things can start problems escalating into huge ones fast. Sipx has all the framework to do this with what it already has by building in the functionality into the user portal or by adding t.37 support to it. Even if another server had to be used in cooperation with the main server in order to keep resource usage down, that would be a blessing over much else that's out there other than very costly commercial solutions. Asside from that, just being able to configure everything on one server would be wonderful. If the votes don't accumulate, I don't see how it would make sense to spend the manpower to do it. To some organizations faxes are important, but certainly the volume in most organizations is declining. True but they are still in use by many small business who don't have the same resources as larger companies who are able to eliminate paper. Many small business absolutely count on being able to send and receive faxes to this day. It would certainly open up opportunities for us in terms of selling people on sipx instead of going with a PBX replacement. Pointedly they dont have a way to use t38modem in a way that works. Never have thats why noone uses hylafax in t38 deployments. You still have to use hardware between hylafax and your t.38 switch. So if you put up a separate hylafax box us an ATA and be done with it. It cant really integrate. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[Assp-test] DomainBoxLimit
Hello today I found a localdomain (domain listed in localdomains) listed in DomainBoxLimit (probably added because the domain sent spoofed email ?). Could be avoided this behavior (having a localdomain added in DomainBoxLimit) ? Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
[Assp-test] denySMTPConnectionsFromAlways
Hello denySMTPConnectionsFrom has this behavior IP numbers and Hostnames in noPB, acceptAllMail, ispip, whiteListedIPs, noProcessingIPs, noBlockingIPs will pass. Could it be possible to have the behavior above also for denySMTPConnectionsFromAlways , or at least for ips listed in whiteListedIPs, noProcessingIPs ? Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
Re: [Assp-test] Antwort: denySMTPConnectionsFromAlways
For example the list could be populated from PenaltyExtremeStrict too which is not manual. Suppose PenaltyExtremeStrict adds an ip here , I go in noprocessing or whiteip and I add the ip, but the email will be still blocked . IMHO I think that at least noprocessing ip should skip every block also denySMTPConnectionsFromAlways . using ASSP 1.9.6.x Graziano Mit freundlichen Grüßen Could it be possible to have the behavior above also for denySMTPConnectionsFromAlways , or at least for ips listed in whiteListedIPs, noProcessingIPs ? It is possible to change the code to do this - but this makes no sense! Manually maintained list of IP's which should strictly be blocked after address verification Notice : Manualy - only you know this IP's- and strictly - do it anyway- ! There are only two option with a higher priority 'noPB' and 'noBlockingIPs ' ! Put the IP's in 'denySMTPConnectionsFrom' instead. If you need to put a list of IP's in to 'whiteListedIPs' and/or 'noProcessingIPs' and 'noBlockingIPs' and/or 'noPB' use an include file (V1/V2) for those addresses - or the group feature in V2. Thomas Von:Grazianodreamserv...@libero.it An: ASSP development mailing listassp-test@lists.sourceforge.net Datum: 13.04.2012 08:15 Betreff:[Assp-test] denySMTPConnectionsFromAlways Hello denySMTPConnectionsFrom has this behavior IP numbers and Hostnames in noPB, acceptAllMail, ispip, whiteListedIPs, noProcessingIPs, noBlockingIPs will pass. Could it be possible to have the behavior above also for denySMTPConnectionsFromAlways , or at least for ips listed in whiteListedIPs, noProcessingIPs ? Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test DISCLAIMER: *** This email and any files transmitted with it may be confidential, legally privileged and protected in law and are intended solely for the use of the individual to whom it is addressed. This email was multiple times scanned for viruses. There should be no known virus in this email! *** -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
[Assp-test] Attachment problem
Hello using 1.9.6.5(0.0.05) The email below (local good email) is going in black hole , never reaches MTA and disconnects after 0 seconds (?). .dat is allowed in GoodAttach Apr-13-12 11:55:45 id-33433-00011 [Attachment] 127.0.0.1 info@jancr to: ro...@ciam.com info: found attachment 'orders.dat'; Apr-13-12 11:55:45 id-33433-00011 [LocalOK] 127.0.0.1 info@jancr to: ro...@ciam.com local - 127.0.0.1 in acceptAllMail - [Order Export] - attachment 'orders.dat' ; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1 info@jancr to: ro...@ciam.com finished message - received DATA size: 48.09 kByte - sent DATA size: 48.04 kByte; Apr-13-12 11:55:45 id-33433-00011 127.0.0.1 info@jancr to: ro...@ciam.com disconnected (0 seconds); Thank you Graziano -- For Developers, A Lot Can Happen In A Second. Boundary is the first to Know...and Tell You. Monitor Your Applications in Ultra-Fine Resolution. Try it FREE! http://p.sf.net/sfu/Boundary-d2dvs2 ___ Assp-test mailing list Assp-test@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/assp-test
[Bug 711534] Re: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified
** Also affects: eucalyptus/2.0 Importance: Undecided Status: New ** Changed in: eucalyptus Status: Fix Committed = Invalid ** Changed in: eucalyptus Assignee: graziano obertelli (graziano.obertelli) = (unassigned) ** Changed in: eucalyptus Milestone: 2.0.4 = None ** Changed in: eucalyptus/2.0 Milestone: None = 2.0.4 ** Changed in: eucalyptus/2.0 Assignee: (unassigned) = graziano obertelli (graziano.obertelli) ** Changed in: eucalyptus/2.0 Importance: Undecided = Medium -- You received this bug notification because you are a member of Ubuntu Server Team, which is subscribed to eucalyptus in Ubuntu. https://bugs.launchpad.net/bugs/711534 Title: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified To manage notifications about this bug go to: https://bugs.launchpad.net/eucalyptus/+bug/711534/+subscriptions -- Ubuntu-server-bugs mailing list Ubuntu-server-bugs@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-server-bugs
[Bug 711534] Re: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified
after a chat on #eucalyptus-devel, this is not present (the mis-handling of /) in eucal3 (marking invalid for -devel). It is targeted for 2.0.4, but the IRC chat seems to imply that more work may be needed to understand the right behavior here. -- You received this bug notification because you are a member of Ubuntu Server Team, which is subscribed to eucalyptus in Ubuntu. https://bugs.launchpad.net/bugs/711534 Title: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified To manage notifications about this bug go to: https://bugs.launchpad.net/eucalyptus/+bug/711534/+subscriptions -- Ubuntu-server-bugs mailing list Ubuntu-server-bugs@lists.ubuntu.com Modify settings or unsubscribe at: https://lists.ubuntu.com/mailman/listinfo/ubuntu-server-bugs
[Bug 711534] Re: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified
** Also affects: eucalyptus/2.0 Importance: Undecided Status: New ** Changed in: eucalyptus Status: Fix Committed = Invalid ** Changed in: eucalyptus Assignee: graziano obertelli (graziano.obertelli) = (unassigned) ** Changed in: eucalyptus Milestone: 2.0.4 = None ** Changed in: eucalyptus/2.0 Milestone: None = 2.0.4 ** Changed in: eucalyptus/2.0 Assignee: (unassigned) = graziano obertelli (graziano.obertelli) ** Changed in: eucalyptus/2.0 Importance: Undecided = Medium -- You received this bug notification because you are a member of Ubuntu Bugs, which is subscribed to Ubuntu. https://bugs.launchpad.net/bugs/711534 Title: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified To manage notifications about this bug go to: https://bugs.launchpad.net/eucalyptus/+bug/711534/+subscriptions -- ubuntu-bugs mailing list ubuntu-bugs@lists.ubuntu.com https://lists.ubuntu.com/mailman/listinfo/ubuntu-bugs
[Bug 711534] Re: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified
after a chat on #eucalyptus-devel, this is not present (the mis-handling of /) in eucal3 (marking invalid for -devel). It is targeted for 2.0.4, but the IRC chat seems to imply that more work may be needed to understand the right behavior here. -- You received this bug notification because you are a member of Ubuntu Bugs, which is subscribed to Ubuntu. https://bugs.launchpad.net/bugs/711534 Title: euca-upload-bundle fails when connecting to Eucalyptus and a path in bucket is specified To manage notifications about this bug go to: https://bugs.launchpad.net/eucalyptus/+bug/711534/+subscriptions -- ubuntu-bugs mailing list ubuntu-bugs@lists.ubuntu.com https://lists.ubuntu.com/mailman/listinfo/ubuntu-bugs
Re: [sipx-users] Fresh Install Problem
sipx does not support multiple network adapters yet. If it were me, I would reinstall with a single activated NIC and go from there. Give it a static IP and configure the subnets and/or your firewall to provide access to the phone you want to connect. 2012/4/12 Simon Brûlé sbr...@360-innovations.com Thank you for the answer. First, The record line sipxecs.netappsid.voip. INA 192.168.177.1 is in it I just didn't put the entire file in my description of the problem (my bad).Now, I failed to provide some information that as I think of it should be important. I installed SipXecs on a Vm that have 2 network adapters, One with the adresse 192.168.177.1 that give the DHCP for the hardphone and a second that is connected to the LAN with an adresse of 192.168.175.170 that he got from dhcp. The nslookup on the server adresse 192.168.175.170 give me this.(The computer used to do it is in the LAN with the adresse 192.168.175.136). server 192.168.175.170 Default server: 192.168.175.170 Address: 192.168.175.170#53 set type=srv _sip._udp.netappsid.voip Server: 192.168.175.170 Address: 192.168.175.170#53 _sip._udp.netappsid.voip service = 1 0 5060 sipxecs.netappsid.voip. and the 192.168.177.1 : server 192.168.177.1 Default server: 192.168.177.1 Address: 192.168.177.1#53 set type=srv _sip._udp.netappsid.voip ;; connection timed out; no servers could be reached 2012/4/11 Tony Graziano tgrazi...@myitdepartment.net IF sipx was told to be the DNS server it should setup the zone file to be AUTHORITIVE for the zone. Until another device (softphone or hardware based phone) tries to connect to it, it should be happy. If it is acting as an AUTHORITIVE DNS SERVER for its own ZONE (which would be the sipdomain you used in setting it up), it still needs a forwarding DNS server, whether that is inside or outside. From a PC, MAC or LINUX machine on its network, its very easy to verify if the DNS records are there: i.e. Windows PC nslookup server 1.2.3.4 (where 1.2.3.4 is the local IP of the sipx server) set type=srv _sip._udp.sipdomain.tld (where sipdomain.tld is your sipdmomain used during the setup, like mydomain.com) It should find AND return the records like this: set type=srv _sip._udp.mydomain.com Server: [10.255.252.64] Address: 10.255.252.64 _sip._udp. mydomain.com SRV service location: priority = 1 weight = 0 port = 5060 svr hostname = pbx.mydomain.com mydomain.comnameserver = pbx.mydomain.com mydomain.cominternet address = 10.255.252.64 If it is still not resolving when queried directly from the sipx server in this fashion, you have a problem THERE that you need to address first. What you FAILED to provide: SIP route to SIPXCHANGE_DOMAIN_NAME 'netappsid.voip' is not to my IP address: 192.168.177.1 The A record in your ZONE FILE should be there. sipxecs.netappsid.voip. INA 192.168.177.1 Why is that? Fix that. 2012/4/11 Simon Brûlé sbr...@360-innovations.com Than you for the quick answer. When I instaleed it I said that I had no DNS in my network and I wanted my SipXecs server to be a DNS so it is install ( i have all the config file) I probably just need to configure them like they should. When your are saying the record i assume you are talking about the SRV record and if I am right they are already configure. Here are the line in the domain.zone file : ; SRV record for domain SIP TCP netappsid.voip ; priority: 1 weight: 0 port: 5060 server: sipxecs.netappsid.voip ; _sip._tcp.netappsid.voip. IN SRV 1 0 5060 sipxecs.netappsid.voip. ; SRV record for domain SIP UDP netappsid.voip ; priority: 1 weight: 0 port: 5060 server: sipxecs.netappsid.voip ; _sip._udp.netappsid.voip. IN SRV 1 0 5060 sipxecs.netappsid.voip. ; SRV record for domain SIP TLS netappsid.voip ; priority: 1 weight: 0 port: 5061 server: sipxecs.netappsid.voip ; _sip._tls.netappsid.voip. IN SRV 1 0 5061 sipxecs.netappsid.voip. ; SRV record for domain SIPS TCP netappsid.voip ; priority: 1 weight: 0 port: 5061 server: sipxecs.netappsid.voip ; _sips._tcp.netappsid.voip. IN SRV 1 0 5061 sipxecs.netappsid.voip. Maybe this detail can help you help me :) I can ping my server with his Ip adresse but I can't ping or nslookup the domain or the fqhn the answer is unknown. Is the problem coming from the server mabe the router ?? Thank you. 2012/4/11 Josh Patten jpat...@ezuce.com Yes, you didnt set up DNS. DNS is a requirement for the system to run. Re-run sipxecs-setup-system and select the option to enable the DNS server, or configure the records yourself: http://wiki.sipfoundry.org/display/sipXecs/DNS+Concepts+for+sipXecs On Wed, Apr 11, 2012 at 3:25 PM, Simon Brûlé sbr...@360-innovations.com wrote: Hi, I just finished installing SipXecs on a new VM and when I go to the Service
Re: [sipx-users] Fresh Install Problem
That will work. If you are going to use trunking you will have firewall work to do though, but that is expected. On Apr 12, 2012 9:27 AM, Simon Brûlé sbr...@360-innovations.com wrote: So the best way to install SipXecs is to give it only 1 Network Adapter on my VM and put it on the same switch/router as the Hardphone for exemple and everything is going to be fine? 2012/4/12 Tony Graziano tgrazi...@myitdepartment.net sipx does not support multiple network adapters yet. If it were me, I would reinstall with a single activated NIC and go from there. Give it a static IP and configure the subnets and/or your firewall to provide access to the phone you want to connect. 2012/4/12 Simon Brûlé sbr...@360-innovations.com Thank you for the answer. First, The record line sipxecs.netappsid.voip. INA 192.168.177.1 is in it I just didn't put the entire file in my description of the problem (my bad).Now, I failed to provide some information that as I think of it should be important. I installed SipXecs on a Vm that have 2 network adapters, One with the adresse 192.168.177.1 that give the DHCP for the hardphone and a second that is connected to the LAN with an adresse of 192.168.175.170 that he got from dhcp. The nslookup on the server adresse 192.168.175.170 give me this.(The computer used to do it is in the LAN with the adresse 192.168.175.136). server 192.168.175.170 Default server: 192.168.175.170 Address: 192.168.175.170#53 set type=srv _sip._udp.netappsid.voip Server: 192.168.175.170 Address: 192.168.175.170#53 _sip._udp.netappsid.voip service = 1 0 5060 sipxecs.netappsid.voip. and the 192.168.177.1 : server 192.168.177.1 Default server: 192.168.177.1 Address: 192.168.177.1#53 set type=srv _sip._udp.netappsid.voip ;; connection timed out; no servers could be reached 2012/4/11 Tony Graziano tgrazi...@myitdepartment.net IF sipx was told to be the DNS server it should setup the zone file to be AUTHORITIVE for the zone. Until another device (softphone or hardware based phone) tries to connect to it, it should be happy. If it is acting as an AUTHORITIVE DNS SERVER for its own ZONE (which would be the sipdomain you used in setting it up), it still needs a forwarding DNS server, whether that is inside or outside. From a PC, MAC or LINUX machine on its network, its very easy to verify if the DNS records are there: i.e. Windows PC nslookup server 1.2.3.4 (where 1.2.3.4 is the local IP of the sipx server) set type=srv _sip._udp.sipdomain.tld (where sipdomain.tld is your sipdmomain used during the setup, like mydomain.com) It should find AND return the records like this: set type=srv _sip._udp.mydomain.com Server: [10.255.252.64] Address: 10.255.252.64 _sip._udp. mydomain.com SRV service location: priority = 1 weight = 0 port = 5060 svr hostname = pbx.mydomain.com mydomain.comnameserver = pbx.mydomain.com mydomain.cominternet address = 10.255.252.64 If it is still not resolving when queried directly from the sipx server in this fashion, you have a problem THERE that you need to address first. What you FAILED to provide: SIP route to SIPXCHANGE_DOMAIN_NAME 'netappsid.voip' is not to my IP address: 192.168.177.1 The A record in your ZONE FILE should be there. sipxecs.netappsid.voip. INA 192.168.177.1 Why is that? Fix that. 2012/4/11 Simon Brûlé sbr...@360-innovations.com Than you for the quick answer. When I instaleed it I said that I had no DNS in my network and I wanted my SipXecs server to be a DNS so it is install ( i have all the config file) I probably just need to configure them like they should. When your are saying the record i assume you are talking about the SRV record and if I am right they are already configure. Here are the line in the domain.zone file : ; SRV record for domain SIP TCP netappsid.voip ; priority: 1 weight: 0 port: 5060 server: sipxecs.netappsid.voip ; _sip._tcp.netappsid.voip. IN SRV 1 0 5060 sipxecs.netappsid.voip. ; SRV record for domain SIP UDP netappsid.voip ; priority: 1 weight: 0 port: 5060 server: sipxecs.netappsid.voip ; _sip._udp.netappsid.voip. IN SRV 1 0 5060 sipxecs.netappsid.voip. ; SRV record for domain SIP TLS netappsid.voip ; priority: 1 weight: 0 port: 5061 server: sipxecs.netappsid.voip ; _sip._tls.netappsid.voip. IN SRV 1 0 5061 sipxecs.netappsid.voip. ; SRV record for domain SIPS TCP netappsid.voip ; priority: 1 weight: 0 port: 5061 server: sipxecs.netappsid.voip ; _sips._tcp.netappsid.voip. IN SRV 1 0 5061 sipxecs.netappsid.voip. Maybe this detail can help you help me :) I can ping my server with his Ip adresse but I can't ping or nslookup the domain or the fqhn the answer is unknown. Is the problem coming from the server mabe the router ?? Thank you. 2012/4/11 Josh Patten jpat
Re: [sipx-users] problem with phonelogd
OK, as I re-read this... The actual issue is with rsyslogd. The subject of the thread made me assume something else. Have you looked at your conf file for it? What is the OS? How do did you install, from RPM or did you compile onboard? On Thu, Apr 12, 2012 at 11:12 AM, Ivan Pletenev i.plete...@gmail.comwrote: i can't remember exactly but it was the same 4.4 but from the last year. as i can understand it's not a symlink: [root@sipx ~]# service phonelogd start rsyslog runtime error(-2066): could not load module '/lib/rsyslog/lmnet.so', dlopen: /lib/rsyslog/lmnet.so: cannot open shared object file: No such file or directory [try http://www.rsyslog.com/e/2066 ] Error during class init for object 'conf' - failing... rsyslogd initializiation failed - global classes could not be initialized. Did you do a make install? Suggested action: run rsyslogd with -d -n options to see what exactly fails. rsyslogd run failed with error -2066 (see rsyslog.h or try http://www.rsyslog.com/e/2066 to learn what that number means) [root@sipx rsyslog]# ls -la /lib/rsyslog/ total 176 drwxr-xr-x 2 root root 4096 Apr 11 19:25 . drwxr-xr-x 15 root root 4096 Apr 4 04:02 .. -rwxr-xr-x 1 root root 10220 Feb 22 18:14 imfile.so -rwxr-xr-x 1 root root 22668 Feb 22 18:14 imklog.so -rwxr-xr-x 1 root root 4420 Feb 22 18:14 immark.so -rwxr-xr-x 1 root root 6896 Feb 22 18:14 imtcp.so -rwxr-xr-x 1 root root 7840 Feb 22 18:14 imudp.so -rwxr-xr-x 1 root root 10320 Feb 22 18:14 imuxsock.so -rwxr-xr-x 1 root root 18968 Feb 22 18:14 lmnet.so -rwxr-xr-x 1 root root 11964 Feb 22 18:14 lmnetstrms.so -rwxr-xr-x 1 root root 16828 Feb 22 18:14 lmnsd_ptcp.so -rwxr-xr-x 1 root root 4100 Feb 22 18:14 lmregexp.so -rwxr-xr-x 1 root root 7076 Feb 22 18:14 lmtcpclt.so -rwxr-xr-x 1 root root 16344 Feb 22 18:14 lmtcpsrv.so -rwxr-xr-x 1 root root 6116 Feb 22 18:14 omtesting.so --- Ivan Pletenev ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Multiple line appearances
You register the SAME LINE on both line appearances and set the limit to 1 for each. On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.comwrote: According to the sipX book (p 133): Most multiple line IP hardware phones allow multiple calls to a single line. This can be quite confusing for the average phone user and difficult to deal with at an answering position. To remedy this problem it is easier for the user to have multiple appearances of the same line on their telephone. Each successive call will ring on the next line appearance. I am using the recommended setup on the Polycom 335 phones, but the behaviour is not as indicated. When the second call comes in, it does not ring on the successive line, but goes straight to v-mail. I'm using sipX 4.4. In my phone group, under the Registration section, I have lineKeys set to 2 (the 335 has two line keys) and I have callsPerLineKey set to 1. After sending the profiles to the phones, they show the correct ext on each line. I can make two outgoing calls, but I can only receive one. I have also tried assigning the same ext twice to the phone. This again results in the same ext appearing on each line key as desired, but when the second call comes in, it does not ring the second line, but rings the first line. If I answer an incoming call while on another call, the first call goes to hold (this is an assumption because I hear the MOH music), but I do not see how to put the second call on hold and retrieve the first call. If I hang up the second call without picking up the first, they both get disconnected. I'd rather have the solution indicated in the book. Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Multiple line appearances
you should also indicate what firmware version your phones are using. On Thu, Apr 12, 2012 at 12:58 PM, Stiles Watson wat...@datatek-net.comwrote: I'm either not communicating my settings clearly or I'm not understanding what I'm being told to do. I have one and only one line assigned to the phone and I have lineKeys set to 2 and callsPerLineKey set to 1. I set the profiles and after the phone reboots, I verified that these are the setting which are on the phone. Stiles On 04/12/2012 12:42 PM, Michael Picher wrote: no, you don't want to register another line to the phone what you want to do is have 1 call per line and 2 line appearances... should be a setting on that same page where you limit it to 1 call per line key. On Thu, Apr 12, 2012 at 11:57 AM, Stiles Watson wat...@datatek-net.comwrote: That is what I did. I have one line assigned to the phone and the registration settings as below. Stiles On 04/12/2012 11:54 AM, Tony Graziano wrote: You register the SAME LINE on both line appearances and set the limit to 1 for each. On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.comwrote: According to the sipX book (p 133): Most multiple line IP hardware phones allow multiple calls to a single line. This can be quite confusing for the average phone user and difficult to deal with at an answering position. To remedy this problem it is easier for the user to have multiple appearances of the same line on their telephone. Each successive call will ring on the next line appearance. I am using the recommended setup on the Polycom 335 phones, but the behaviour is not as indicated. When the second call comes in, it does not ring on the successive line, but goes straight to v-mail. I'm using sipX 4.4. In my phone group, under the Registration section, I have lineKeys set to 2 (the 335 has two line keys) and I have callsPerLineKey set to 1. After sending the profiles to the phones, they show the correct ext on each line. I can make two outgoing calls, but I can only receive one. I have also tried assigning the same ext twice to the phone. This again results in the same ext appearing on each line key as desired, but when the second call comes in, it does not ring the second line, but rings the first line. If I answer an incoming call while on another call, the first call goes to hold (this is an assumption because I hear the MOH music), but I do not see how to put the second call on hold and retrieve the first call. If I hang up the second call without picking up the first, they both get disconnected. I'd rather have the solution indicated in the book. Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
Re: [sipx-users] Multiple line appearances
If you are using any EFK features in the 335 all bets are probably off too... On Thu, Apr 12, 2012 at 1:41 PM, Michael Picher mpic...@ezuce.com wrote: That should work fine, it has on every 650 i've ever tried it on. I've honestly never ever tried it on a 335 however. I generally use this feature for operator types... and operator types don't use 335's. Mike On Thu, Apr 12, 2012 at 12:58 PM, Stiles Watson wat...@datatek-net.comwrote: I'm either not communicating my settings clearly or I'm not understanding what I'm being told to do. I have one and only one line assigned to the phone and I have lineKeys set to 2 and callsPerLineKey set to 1. I set the profiles and after the phone reboots, I verified that these are the setting which are on the phone. Stiles On 04/12/2012 12:42 PM, Michael Picher wrote: no, you don't want to register another line to the phone what you want to do is have 1 call per line and 2 line appearances... should be a setting on that same page where you limit it to 1 call per line key. On Thu, Apr 12, 2012 at 11:57 AM, Stiles Watson wat...@datatek-net.comwrote: That is what I did. I have one line assigned to the phone and the registration settings as below. Stiles On 04/12/2012 11:54 AM, Tony Graziano wrote: You register the SAME LINE on both line appearances and set the limit to 1 for each. On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.comwrote: According to the sipX book (p 133): Most multiple line IP hardware phones allow multiple calls to a single line. This can be quite confusing for the average phone user and difficult to deal with at an answering position. To remedy this problem it is easier for the user to have multiple appearances of the same line on their telephone. Each successive call will ring on the next line appearance. I am using the recommended setup on the Polycom 335 phones, but the behaviour is not as indicated. When the second call comes in, it does not ring on the successive line, but goes straight to v-mail. I'm using sipX 4.4. In my phone group, under the Registration section, I have lineKeys set to 2 (the 335 has two line keys) and I have callsPerLineKey set to 1. After sending the profiles to the phones, they show the correct ext on each line. I can make two outgoing calls, but I can only receive one. I have also tried assigning the same ext twice to the phone. This again results in the same ext appearing on each line key as desired, but when the second call comes in, it does not ring the second line, but rings the first line. If I answer an incoming call while on another call, the first call goes to hold (this is an assumption because I hear the MOH music), but I do not see how to put the second call on hold and retrieve the first call. If I hang up the second call without picking up the first, they both get disconnected. I'd rather have the solution indicated in the book. Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com
Re: [sipx-users] voicemail greeting does not change
if it does not change via the webgui, sipxconfig should be throwing and error in its log. you should capture that error and open a jira with it. /var/log/sipxpbx/sipxconfig.log On Wed, Apr 11, 2012 at 8:06 AM, Henry Dogger h.dog...@telecats.nl wrote: But this is strange, since I recorded the new greeting via the sipxecs voicemail menu…. The setting in the webinterface doesn’t change either. Strange thing is, when I change the greeting using the web gui, it does work, but changing via the voicemail menu does not… ** ** *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Michael Picher *Sent:* dinsdag 10 april 2012 19:13 *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] voicemail greeting does not change ** ** usually either wav file format or file permissions... On Tue, Apr 10, 2012 at 11:23 AM, Henry Dogger h.dog...@telecats.nl wrote: Hi all, When I try to change my voicemail greeting using the voicemail menu options build in sipx/freeswitch I can’t change my greeting. I do not get a warning of error, and there is also no warning or error in the logging or on the freeswitch cli to be found… Is this a known problem, if yes is there a fix? Kind regards, Henry Dogger Telecats BV ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ** ** -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com ** ** There are 10 kinds of people in the world, those who understand binary and those who don't. ** ** ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] voicemail greeting does not change
actually you are saying there is no logging for the actio using the IVR menu, even in the sipxivr log file? if so, a jira should still be opened showing the erroneous successful set from the sipxivr log. Normally that would happen if the user box has a permissions issue (owner/group or rights) from a manual restore or moving a wav file manually. Since it works in one interface and not the other, it is more likely an internal process. If it does not work via IVR, please provide IVR log with the success message and put it in a JIRA. I seem to recall this and thought it had been fixed a while back. Are you on a current version? Have you checked the tracker for this issue? On Wed, Apr 11, 2012 at 8:53 AM, Henry Dogger h.dog...@telecats.nl wrote: Well what I mean is this: I change using voicemail menu (option 5) then I change voicemail greeting under option 3. This does not work, no errors are in any logging… I also notice that the option in the webgui is not changed to what I chose in the voicemail menu (as it is supposed to be…) When I change the setting in web gui, it does change my greeting, so no problems there…. And sipxconfig would not be the place to look for errors I suppose… If I could find a error message, I would open a jira with it, but it seems that there is no error reported, but still I can’t change my voicemail greeting via the phone. ** ** *From:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano *Sent:* woensdag 11 april 2012 14:43 *To:* Discussion list for users of sipXecs software *Subject:* Re: [sipx-users] voicemail greeting does not change ** ** if it does not change via the webgui, sipxconfig should be throwing and error in its log. you should capture that error and open a jira with it.** ** ** ** /var/log/sipxpbx/sipxconfig.log On Wed, Apr 11, 2012 at 8:06 AM, Henry Dogger h.dog...@telecats.nl wrote: But this is strange, since I recorded the new greeting via the sipxecs voicemail menu…. The setting in the webinterface doesn’t change either. Strange thing is, when I change the greeting using the web gui, it does work, but changing via the voicemail menu does not… ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] sipX feature codes
Realize you are asking for a digital system to return a BUSY signal when calling a user with a digital handset. Yes you can dumb down the phone (with some) to handle a single call at a time). Since the phones can handle more than 1 call, have voicemail and such, it really centers on the reasoning behind trying to do this. We normally see incoming calls in our environments hit a hunt group to handle manually live person answered calls during business hours. In a call center environment, you would use other tools, such as the upcoming OpenACD integration. Start by understanding what the system has and publishing that to your users. I find trying to get every bit of functionality isn't always prudent right away EVEN if it is possible. Users need to grasp the basic differences when migrating from a KEY system. Fortunately you can put the handsets side-by-side for an acclimation period. On Wed, Apr 11, 2012 at 12:12 PM, Stiles Watson wat...@datatek-net.comwrote: Just want to know where the differences are so I can communicate those to the users. If I'm the sipX admin, I need to know more about the system than anyone else in the company. If I'm asked a question, I need to be able to answer it. Thanks for the feedback. Stiles On 04/11/2012 12:07 PM, Michael Picher wrote: Those are the only feature codes... Other features are phone specific. Did you want a SIP Communications System or a TDM PBX? ;-) Mike On Wed, Apr 11, 2012 at 10:40 AM, Stiles Watson wat...@datatek-net.comwrote: There is a sample Quick reference guide in the book, but it only mentions *78 for directed call pickup and *4 for picking up a parked call. If you read through the TUI section in chapter 8, it mentions the following: Directed Call Pickup: *78 + ext Pickup a Parked Call: *4 + park orbit Intercom: *76 + ext Paging Groups: *77 + paging group number ACD sign in and out: *86 *88 respectively Are there any others? There are in-conference commands and v-mail menu options also, but specifically I was looking for feature codes. The main reason I was asking is that moving from a standard PBX w/analog phones to sipXecs and digital phones some features are moved around and I'm looking to ease the transition or at least make a cross-reference. Users are used to doing everything via the phone's keypad. Now some items are moved to the web portal (call forwarding) and some to a menu on the phone (speed dial). I'm trying to think about user training. Stiles On 04/10/2012 07:59 PM, Todd Hodgen wrote: The book has a sample user guide that has all the features in a nice summary, including the feature codes if I recall correctly. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Stiles Watson Sent: Tuesday, April 10, 2012 3:55 PM To: Discussion list for users of sipXecs software Subject: [sipx-users] sipX feature codes Is there a central list of all the feature codes in sipXecs? I've searched feature codes in both the wiki and the book, but can not find a list. Stiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- Michael Picher, Director of Technical Services eZuce, Inc. 300 Brickstone Square Suite 201 Andover, MA. 01810 O.978-296-1005 X2015 M.207-956-0262 @mpicher http://twitter.com/mpicher www.ezuce.com There are 10 kinds of people in the world, those who understand binary and those who don't. ___ sipx-users mailing listsipx-us...@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx
