Re: [asterisk-users] LDAPget or something else?
Hello David, * Klaverstyn, David C [EMAIL PROTECTED] [09-05-07 09:40]: We are currently using LDAPget 1.0rc6 with Asterisk 1.2.x. I see that there is LDAPget 2.0rc1 for Asterisk 1.4.x. I was wondering if there was something better. Are people using LDAPget or something else? I have ported LDAPget 2.0 to FreeBSD, works fine for me with asterisk 1.4. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug no. 8680 (billsec is 0 even when the call is answered) in Asterisk 1.4.2
We recently installed Asterisk 1.4.2 Tried to make calls using the Originate command (Asterisk Manager Interface) All of the calls have zero billsec in the CDR. Stumbled upon this: http://bugs.digium.com/view.php?id=8680 so I guess the fix is not yet in 1.4.2. Is this fixed in 1.4.3/1.4.4? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?
Hi, Everything was working fine on this 10 phone office, but for last few weeks they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. What are the possible causes for this to happen? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] Audio going blank for a few seconds and then comesback. What could be the reason?
Hi Zeeshan, Ethernet Network (or Switch) congestion ? QoS not realy effective ? Too high CPU load in Asterisk the server ? Who knows... You must check during a default. Good kuck ! Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Zeeshan Zakaria Envoyé : mercredi 9 mai 2007 12:02 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : [asterisk-users] Audio going blank for a few seconds and then comesback. What could be the reason? Hi, Everything was working fine on this 10 phone office, but for last few weeks they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. What are the possible causes for this to happen? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing Volume
Jadrien Wauthier wrote: Does anyone know how to adjust the volume of the ringing application? I have done a lot of internet searching and have not found much. You cannot do this in Asterisk. Some SIP phones might allow you to do so by setting an option on the phone, but you would have to ask the company that makes that specific phone how to do that. If Asterisk generates the audio, then it seems that there would be a source file that I could edit if nothing else. I looked at the app_dial.c, but I didn't see anything. Maybe I over looked something. If I lower the volume on the phone, then all audio on the phone would be lower. I am just interested in lowering the volume of the ringing. Basically, rings from the pstn is at one level, and the rings from Asterisk are at another level. I need to normalize the Asterisk volume. Thank you so much for your help with this. Jad Jad, Are you referring to the ring back (progress tones) when you call out? I have the same issue. Depending on the type of interface you have to the PSTN, you could try raising the inbound gain from the PSTN to match that of asterisk. Bob... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The purpose of DUNDi
Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Thanks for that. Will get a spare * box. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] app_txfax, app_rxfax
That is not true regarding voice / fax detection with iaxmodem. If you are running zaptel, then let it do the fax detection and have the iaxmodems called from the fax context. Craig -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Wednesday, 9 May 2007 12:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Microsoft CRM Asterisk
Hi Calvis, We have develop with MsCRM and Asterisk, if you still interested we are very pleasure to help you.ç Frank Bobbio +34 932289310 www.icr.es Barcelona Spain [Asterisk-Users] Microsoft CRM Asterisk calvis calvis at itechgroup.com mailto:asterisk-users%40lists.digium.com?Subject=%5BAsterisk-Users%5D%20Microsoft%20CRM%20%26%20AsteriskIn-Reply-To=448076B2.4050601%40howardsilvan.com Fri Jun 2 11:02:30 MST 2006 * Previous message: [Asterisk-Users] Prices of g729 codec * Next message: [Asterisk-Users] Microsoft CRM Asterisk * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] _ Has anyone done any integration with Asterisk Microsoft Dynamics CRM? I just wanted to check with the list before I pursue a project with the above integration. In addition, if anyone would be interested in such an integration let me know, and I will keep you posted on the results. Thanks, Charles Alvis Internet Technology Group, Inc. Redmond,WA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
Any field return on this ? Our last field trial of HPEC concluded we shouldn't use it at all, due to audio clipping. Is it now fixed ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 45
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
Just the sake of curiosity, how many sites (or user) did you interconnect using DUNDi ? Regards 2007/5/9, Bruce Reeves [EMAIL PROTECTED]: I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send SIP Re-invite.
8 maj 2007 kl. 15.40 skrev Joshua Colp: Rohan Hathiwala wrote: Hi, I need asterisk to instruct the other side to send RTP to a conference server running on a different machine. The conference server does not understand SIP so I cannot use the SIP REFER method. I have another question. Suppose when processing a SIP INVITE we want to use asterisk only for call control and let another server handle the RTP is there a clean way to do this in asterisk. Regards, Rohan Hathiwala. Asterisk/chan_sip wasn't designed to be able to do this. You're going to end up modifying things... potentially a lot. If the conference server does SIP though you can just dial it, make sure canreinvite is set to yes, and audio should go direct. ...psst... There's a patch in the bug tracker that I believe is what you want. Please test and review, add your comments. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Replaces header
I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk, 3-8396 is removed from the call path but no call back happens and the PSTN telephone just gets disconnected. The console log showing the invite from our SER proxy is listed below. Does anyone have any thoughts on what might be happening? Thanks,Steve [May 9 08:42:42] DEBUG[18512]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 128.91.56.38 [May 9 08:42:42] DEBUG[18512]: chan_sip.c:15336 sip_devicestate: Checking device state for peer 128.91.56.38 [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: chan_sip.c:3481 sip_answer: SIP answering channel: SIP/128.91.56.38-09c6e8f0 [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6452 transmit_response_with_sdp: Setting framing from config on incoming call [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6220 add_sdp: ** Our capability: 0x4 (ulaw) Video flag: True [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6221 add_sdp: ** Our prefcodec: 0x0 (nothing) [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6352 add_sdp: -- Done with adding codecs to SDP [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: channel.c:2381 ast_internal_timing_enabled: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) [May 9 08:42:42] DEBUG[20530]: chan_sip.c:6397 add_sdp: Done building SDP. Settling with this capability: 0x4 (ulaw) [May 9 08:42:42] DEBUG[18512]: channel.c:1026 channel_find_locked: Avoiding initial deadlock for channel '0xa29dd20' [May 9 08:42:42] DEBUG[20530]: pbx.c:1795 pbx_extension_helper: Launching 'Wait' [May 9 08:42:42] DEBUG[18512]: devicestate.c:287 do_state_change: Changing state for SIP/128.91.56.38 - state 2 (In use) [May 9 08:42:42] DEBUG[18518]: chan_sip.c:4393 find_call: = Found Their Call ID: [EMAIL PROTECTED] Their Tag 479EE6C-1A45 Our tag: as33bbca55 [May 9 08:42:42] DEBUG[18518]: chan_sip.c:14725 handle_request: Received ACK (6) - Command in SIP ACK [May 9 08:42:42] DEBUG[18518]: chan_sip.c:2107 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 101: Match Not Found [May 9 08:42:43] DEBUG[20530]: pbx.c:1795 pbx_extension_helper: Launching 'Set' [May 9 08:42:43] WARNING[20530]: pbx.c:1783 pbx_extension_helper: No application 'SIPGetHeader' for extension (default, 700, 4) [May 9 08:42:43] DEBUG[20530]: pbx.c:2393 __ast_pbx_run: Spawn extension (default,700,4) exited non-zero on 'SIP/128.91.56.38-09c6e8f0' [May 9 08:42:43] DEBUG[20530]: channel.c:1693 ast_hangup: Hanging up channel 'SIP/128.91.56.38-09c6e8f0' [May 9 08:42:43] DEBUG[20530]: chan_sip.c:3330 sip_hangup: Hangup call SIP/128.91.56.38-09c6e8f0, SIP callid [EMAIL PROTECTED]) [May 9 08:42:43] DEBUG[20530]: devicestate.c:303 __ast_device_state_changed_literal: Notification of state change to be queued on device/channel SIP/128.91.56.38-09c6e8f0 [May 9 08:42:43] DEBUG[18512]: devicestate.c:161 ast_device_state: No provider found, checking channel drivers for SIP - 128.91.56.38 [May 9 08:42:43] DEBUG[18512]: chan_sip.c:15336 sip_devicestate: Checking device state for peer 128.91.56.38 [May 9 08:42:43] DEBUG[18512]: devicestate.c:287 do_state_change: Changing state for SIP/128.91.56.38 - state 1 (Not in use) [May 9 08:42:43] DEBUG[18518]: chan_sip.c:4393 find_call: = Found Their Call ID: [EMAIL PROTECTED] Their Tag 479EE6C-1A45 Our tag: as33bbca55 [May 9 08:42:43] DEBUG[18518]: chan_sip.c:2107 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK octothorpe*CLI exit ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?
On 5/9/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. I had this same problem happening every 8 minutes. It ended up being a DSL issue at the DSLAM. You need to ping a server for a while until you get the period where there is the cutoff to check if you are not losing connectivity like I was. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
There are nine sites, 10 servers. While it is not a huge deployment by some standards, it was simplified with DUNDi. On 5/9/07, Olivier [EMAIL PROTECTED] wrote: Just the sake of curiosity, how many sites (or user) did you interconnect using DUNDi ? Regards 2007/5/9, Bruce Reeves [EMAIL PROTECTED] : I use DUNDi in this way, I have several remote sites and a MPLS network connecting the sites. I have each sites asterisk box looking at 2 DUNDi peers and those 2 central peers can query all sites. I don't have a lot of phones or people moving between sites, but I did not want to have to setup a IAX connection for every site on every server. I like the ability for DUNDi to determine which server to talk to and then configure the dial string for that call. This made my configuration easier to expand as I deployed new sites. I simply added the new peer to my central servers and configured the new site server and I could call between sites. While DUNDi's original intent was more for least cost routing or zero cost routing, I think it provides an excellent means to scale a network of asterisk systems. On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
Alex, Thanks for the linking to JR's article. That was my source for setting up DUNDi also. On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote: Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf ). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Reeves Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem when PABX call to Asterisk by Unicall
Hi all, I have an Asterisk server connected in a PABX (TELEDATA) by channel Unicall (MFC/R2).. I`m having problem when somebody call from PABX to Asterisk.. Eg: When somebody dial 1234, I received 113344 in the Asterisk CLI... If somebody can help me... or already saw this... Everton Goularth Uberlandia - MG - Brazil ___ Yahoo! Mail - Sempre a melhor opção para você! Experimente já e veja as novidades. http://br.yahoo.com/mailbeta/tudonovo/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk 1.2 and UDP packet numbering on bridgedchannels (for jitter buffering)?
Damon Estep wrote: http://www.asterisk.org/node/48317 does a nice job of explaining the 1.4 jitter buffer, however it raised a question in my mind. In 1.2 (and also 1.4), when asterisk bridges 2 SIP channels, are the UDP RTP packets renumbered on transmit, or is the original sequence number preserved in the UDP header? A comment is made on the referenced blog that jitter buffering is best implemented at the endpoint, where additional jitter will not be added by another IP link. This is logical thinking, but only possible if the bridging function in Asterisk preserves the source call leg UDP packet numbering in the terminating call LEG UDP RTP packet stream. If the effect of the Asterisk SIP to SIP bridge is such that the UDP headers are re-created on transmit it is likely that the packet sequencing is the order in which Asterisk transmitted the packets, which is may not be the order in which the original source UA transmitted them due to jitter in the IP link on the first half of the bridged call. Can anyone provide an authoritative answer on how asterisk sequences UDP RTP packets on the transmit leg of a bridged SIP call (known based on actual testing or code review)? I can tell you about our extensive tests back when we were on version 1.0.X Asterisk would take in an RTP stream and then recreate a new one on exit, putting in a new Sequence Number, and new Timestamp in the RTP Header. This effectly destroys any chance of efficiently relying on jitter buffering at the endpoints. From multiple tests over the years we have come to rely on the best jitter buffer we could devise in Asterisk regarding SIP-SIP channels. That is we loop the call out to a ZAP channel and back in, thus turning the call into SIP-ZAP-ZAP-SIP. The ZAP channels have quite good jitter buffers and they work perfectly in our configuration. Sure you eat extra T1 channels but we have not choice. Most of our customers are overseas and the jitter is quite high. [Damon Estep] I can see how bridging sip to sip via a zap channel would fix minor jitter issues, since the zap timers are very accurate, however I cannot see how this would correct out of order packets like a true jitter buffer does (without the use of a jitter buffer on the sip-zap bridge). Seems like it would be much simpler and more effective to force sip-sip bridge jitter buffering with jbforce=yes (1.4) At any rate, thanks for the information on the new sequence number in the asterisk sip-sip bridge in 1.0.x. have you done any testing in 1.2 or 1.4 to confirm this is still the case? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP peer / Maximum retries exceeded on transmission
We're actually getting two invites and schedules retransmit of both, which is bad. One retransmit is stopped and the other one keeps going, regardless of the ACKs that keep coming in. Needs to be fixed. Believe I have fixed this in 1.4 svn, please test. /O - Patch Index: channels/chan_sip.c === --- channels/chan_sip.c (revision 63252) +++ channels/chan_sip.c (working copy) @@ -13643,8 +13643,7 @@ } /* Respond to normal re-invite */ if (sendok) - transmit_response_with_sdp(p, 200 OK, req, XMIT_CRITICAL); - + transmit_response_with_sdp(p, 200 OK, req, ast_test_flag(req, SIP_PKT_IGNORE) ? XMIT_UNRELIABLE : XMIT_CRITICAL); } p-invitestate = INV_TERMINATED; break; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: RE: Digital Phones
Hi; Well, I understood now that Nortel has some digital phones that can be used with astrisk, but the question: what are the card models that should be installed on Asterisk server? Digium? What these models? Regards Bilal Ghayad Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] The 'h' extension problem
Hi all, There is a problem with my dialplan. here is the dialplan: exten= 123,1,Dial(SIP/U1,,Ttg) exten= 123,2,Hangup exten= h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling from out of our asterisk system). I understand if U1 hangsup then there is no channel to execute h extension, but is it possible to execute the h exten even then. i want the h extension to excute everytime. how can i do this. i have used the g flag in dial which tell asterisk to execute remaining extensions even after hangup but its not doing in the above described case. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] fax receiving
Hello everybody, I am receiving faxes and I don`t know how to receive, is there any posibility to receive it on amail account?¿ in the console the message is this: May 9 15:47:44 NOTICE[2618]: chan_zap.c:3703 zt_handle_dtmfup: Fax detected, but no fax extension -- SIP/101-0819b4f8 answered Zap/1-1 thanks to all___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Send SIP Re-invite.
Hi, Could you kindly send me that patch or give me the link to it. I am not familiar with the bug tracker. Regards, Rohan Hathiwala. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Wednesday, May 09, 2007 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Send SIP Re-invite. 8 maj 2007 kl. 15.40 skrev Joshua Colp: Rohan Hathiwala wrote: Hi, I need asterisk to instruct the other side to send RTP to a conference server running on a different machine. The conference server does not understand SIP so I cannot use the SIP REFER method. I have another question. Suppose when processing a SIP INVITE we want to use asterisk only for call control and let another server handle the RTP is there a clean way to do this in asterisk. Regards, Rohan Hathiwala. Asterisk/chan_sip wasn't designed to be able to do this. You're going to end up modifying things... potentially a lot. If the conference server does SIP though you can just dial it, make sure canreinvite is set to yes, and audio should go direct. ...psst... There's a patch in the bug tracker that I believe is what you want. Please test and review, add your comments. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER == This e-mail may contain privileged and confidential information which is the property of Persistent Systems Pvt. Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Pvt. Ltd. does not accept any liability for virus infected mails. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?
Zeeshan Zakaria wrote: Hi, Everything was working fine on this 10 phone office, but for last few weeks they are complaining that audio goes blank for a few seconds during the conversation, and then comes back on. It goes blank for both parties. What are the possible causes for this to happen? Disable CDP in the boot menu of the Polycom phones. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The 'h' extension problem
Rizwan Hisham wrote: exten = 123,1,Dial(SIP/U1,,Ttg) exten = 123,2,AGI(onhangup.pl) exten = 123,3,Hangup exten = h,1,DeadAGI(onhangup.pl) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems witch SPA3102.
Jonson Player wrote: Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. Linksys SPA3102 or Sipura SPA300 docs http://www.sipura.com/support/index.htm http://www.jmgtechnology.com.au/spa_3102_guide.pdf http://www.jmgtechnology.com.au/spa_3000_guide.pdf -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] select menu
Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten = s,1,Answer() exten = s,2,Wait(1) exten = s,3,Dial(SIP/101,30,Ttm) sorry about my english, thanks to all be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using voip software client as public address system. Low volume
Hello all. We have an asterisk working perfectly but we need a sollution for the PA system. Before Asterisk PBX we had an expensive analog PBX which plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the amplifier, into the amplifier. I've configured a pjsua with auto answer but the audio is very poor, very low volume compared to a normal audio playing (like 'aplay ttt.wav'). Is there any way to increase the volume of sip calls? Is a client side configuration, a server side or both :) Any ideas? Please, I'm going mad. Thanks in advance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
Hello ! For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require voltage on the line (although they don't use it to powerup and it just draws a few mil amps) As for PRI never tested, i would be interested to know how your test goes Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gavin Henry Sent: Wednesday, May 09, 2007 2:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Thanks for that. Will get a spare * box. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits
I wonder if the your hardware is doing the actual DTMF detecting. What hardware are you using? I'm using the TE205P and I believe that the DTMF detection is being done in the software. Remi Steve Davies wrote: On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote: When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it and regenerates it. Sensitive machines like auto attendants pick up both the brief end user generated tone as well as the full length asterisk generated tone and ultimately perceive each digit twice. Is anyone else experiencing this? I have reproduced this in an environment * with one asterisk server that is both the feature server and the media gateway, and is timing off of network T1s * with two servers, one feature server (timing off of ztdummy) and one media gateway (timing off of network T1s) using IAX as the inter asterisk protocol It is pretty easy to reproduce: -Dial a PSTN number(like your cell) from a sip phone using inband DTMF, and configured in asterisk sip.conf with dtmfmode=inband. -Answer the PSTN end. -Press and hold a digit on the sip phone. On the PSTN phone you will hear a very brief, end user generated, tone. -Let go of the digit on the sip phone. On the PSTN phone you will hear the asterisk generated tone. Can anyone else hear the brief initial tone? Any help is greatly appreciated! Yes, we have a similar issue, but do not normally use inband DTMF because SIP phones very cleanly generate rfc2833 RTP packets directly and remove this issue. On the other hand, asterisk is not alone dealing with this issue in SIP. The Linksys ATAs have exactly the same issue. Strangely, I do not have a problem receiving inband DTMF through Zaptel, which I believe uses the same DSP code for DTMF detection... Or does it? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits
I wonder if your hardware is doing the actual DTMF detecting. What hardware are you using? I'm using the TE205P and I believe that the DTMF detection is being done in the software in my case. Remi Steve Davies wrote: On 5/3/07, Ken Leland III [EMAIL PROTECTED] wrote: When dtmfmode is set to inband for SIP, and i originate a call from sip out to the PSTN, I can hear the DTMF digit twice in the audio stream. Once very briefly and once for normal duration. Our Theory: While Asterisk is parsing the DTMF, for a fraction of a second, while the end user generated DTMF is being detected, the DTMF is passed inband. Once the DTMF is detected Asterisk silences it and regenerates it. Sensitive machines like auto attendants pick up both the brief end user generated tone as well as the full length asterisk generated tone and ultimately perceive each digit twice. Is anyone else experiencing this? I have reproduced this in an environment * with one asterisk server that is both the feature server and the media gateway, and is timing off of network T1s * with two servers, one feature server (timing off of ztdummy) and one media gateway (timing off of network T1s) using IAX as the inter asterisk protocol It is pretty easy to reproduce: -Dial a PSTN number(like your cell) from a sip phone using inband DTMF, and configured in asterisk sip.conf with dtmfmode=inband. -Answer the PSTN end. -Press and hold a digit on the sip phone. On the PSTN phone you will hear a very brief, end user generated, tone. -Let go of the digit on the sip phone. On the PSTN phone you will hear the asterisk generated tone. Can anyone else hear the brief initial tone? Any help is greatly appreciated! Yes, we have a similar issue, but do not normally use inband DTMF because SIP phones very cleanly generate rfc2833 RTP packets directly and remove this issue. On the other hand, asterisk is not alone dealing with this issue in SIP. The Linksys ATAs have exactly the same issue. Strangely, I do not have a problem receiving inband DTMF through Zaptel, which I believe uses the same DSP code for DTMF detection... Or does it? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The 'h' extension problem
Please try this, exten= 123,1,Dial(SIP/U1,,Tt) exten= 123,2,Hangup exten= h,1,DEADAGI(onhangup.pl) ok? ;) Rafael Rodrigo M. Rosa. www.megavoz.com.br http://www.megavoz.com.br/ (Voip e Telemarketing) www.nsinet.com.br http://www.nsinet.com.br/ (Serviços Internet) http://www.megavoz.com.br/ Rizwan Hisham escreveu: Hi all, There is a problem with my dialplan. here is the dialplan: exten= 123,1,Dial(SIP/U1,,Ttg) exten= 123,2,Hangup exten= h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling from out of our asterisk system). I understand if U1 hangsup then there is no channel to execute h extension, but is it possible to execute the h exten even then. i want the h extension to excute everytime. how can i do this. i have used the g flag in dial which tell asterisk to execute remaining extensions even after hangup but its not doing in the above described case. -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 34, Issue 46
I will be out of the office until Monday, May 14. Please contact OWD at 800-337-3839 and ask for Client Services if you need immediate assistance. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] select menu
i suggest that you place it on a queue.. On 5/9/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: Hello everybody. I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3). if he choose 1 it will redirect to 101 extension if he choose 2 it will redirect to 102 extension if he choose 3 it will redirect to 103 extension my extensions.conf is this one: [default] exten = s,1,Answer() exten = s,2,Wait(1) exten = s,3,Dial(SIP/101,30,Ttm) sorry about my english, thanks to all be ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [asterisk-users] select menu
Hi,
My suggestion:
extensios.conf
exten = s,1,Answer()
exten = s,n,Wait(1)
exten = s,n,Read(MyVariable,TheNameOfSoudFile,1, , ,10)
exten = s,n,GotoIf($[${MyVariable}=1]?11)
exten = s,n,GotoIf($[${MyVariable}=2]?12)
exten = s,n,GotoIf($[${MyVariable}=3]?13)
exten = s,11,Dial(SIP/101,30,Ttm)
exten = s,12,Dial(SIP/102,30,Ttm)
exten = s,13,Dial(SIP/103,30,Ttm)
-- Timeout
|
Read(MyVariable,MySoudFile,1, , ,10)
|
|
- Number of characters that user has to
digit
Asterisk cmd READ
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
[]'s
Moacir O. de Souza Junior
Belo Horizonte - Minas Gerais - Brasil
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Josu Lazkano
Lete
Enviada em: quarta-feira, 9 de maio de 2007 11:49
Para: asterisk-users@lists.digium.com
Assunto: [asterisk-users] select menu
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone
calls the Asterisk play a record (in gsm) and them the caller must choose a
option (1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose 2 it will redirect to 102 extension
if he choose 3 it will redirect to 103 extension
my extensions.conf is this one:
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
sorry about my english,
thanks to all
be
___
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Re: [asterisk-users] asterisk 1.2 and UDP packet numbering on bridgedchannels (for jitter buffering)?
[Damon Estep] I can see how bridging sip to sip via a zap channel would fix minor jitter issues, since the zap timers are very accurate, however I cannot see how this would correct out of order packets like a true jitter buffer does (without the use of a jitter buffer on the sip-zap bridge). Seems like it would be much simpler and more effective to force sip-sip bridge jitter buffering with jbforce=yes (1.4) I cannot comment on 1.4 as we are still not even close to implementing it. In the case of out-of-order packets, you are correct. Our solution does not fix that. But it does fix jitter better than any other solution up to 1.2. Out-of-order packets are much harder to come by than regular 30-60ms jitter which we do find on at least 30% of international calls. At any rate, thanks for the information on the new sequence number in the asterisk sip-sip bridge in 1.0.x. have you done any testing in 1.2 or 1.4 to confirm this is still the case? I cannot remember doing testing in 1.2, but since there wasn't a readily available jitter buffer for SIP in Asterisk 1.2 we continued using our solution. When we get ready for 1.4 we will start all over again with our testing to see if the new jitter buffer is as good as what we can get with the ZAP timers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Problems continue...
SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116716 0a2a959d3d3 00102/0 unkn No Init: INVITE 10.200.26.115715 1dee947d485 00102/0 unkn No Init: INVITE 10.200.26.104704 28808764699 00102/0 unkn No Init: INVITE 10.200.26.104704 36d3e88f59c 00102/0 unkn No Init: INVITE 10.200.26.104704 0e00060800d 00102/0 unkn No Init: INVITE Second, I've gone through and basically redone my extensions.conf to have it flow much smoother and clearer. I thought for sure my problem was coming from a loop somewhere in extensions.conf, but I'm now certain my extensions.conf is fine (but I'm glad I redid it, much easier to follow now). Third, I removed 'qualify=yes' from my sip.conf. I had read where people were having SIP channel lockups with this enabled, I again thought I had found the problem...but alas...In addition I had seen someone suggest setting REINVITE=NO, in addition to CANREINVITE=NO...no good. Fourth, I downgraded all my GXP-2000's to the latest released version of the software (1.1.1.14), some were on a newer version that I'm not sure where it came from (1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be downgraded), as those apparently had lock up issues as well...again thought I had found the problem... Fifth, I installed the latest SVN of 1.4 last night in hopes it was a known issue that had been fixednope We don't have a very complicated setup at all. The server is running CentOS 4, it has two TDM-400 cards with 6 FXS 2 FXO. We have about 25 GXP-2000 phones. My dialplan is nice and clean now. If no one has any further suggestions I'm to the point of opening a bug report with digium. I've read a ton on other people who have had this problem and followed the fixes for those people, but I can't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no [EMAIL PROTECTED] call-limit=9 allowsubscribe=yes Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fax receiving
As usual, it is worth searching the WiKi for answers to this sort of question: http://www.voip-info.org/wiki/view/Asterisk+Fax+to+email This is not the only answer. Regards, Steve On 5/9/07, Josu Lazkano Lete [EMAIL PROTECTED] wrote: Hello everybody, I am receiving faxes and I don`t know how to receive, is there any posibility to receive it on amail account?¿ in the console the message is this: May 9 15:47:44 NOTICE[2618]: chan_zap.c:3703 zt_handle_dtmfup: Fax detected, but no fax extension -- SIP/101-0819b4f8 answered Zap/1-1 thanks to all ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Problems continue...
Ken, I have similar problems every now and then on one of my asterisk boxes. I'm also running CentOS4 on that box. I've found that doing a sip reload when in that state results in something along : Last reload not yet finished (can't remember the exact wording) We're using cisco 7960's here. The ONLY time I've seen this happening is when I reload everything VIA freepbx. It used to do it every time I reloaded. I read somewhere that this was a result of DNS queries not being done in a timely fashion - So I went and replaced all the host statement in my trunk with IP addresses and now it doesn't do it very often at all. I don't know if this is your problem at all but it might be worth a shot. Replace any host names with IP addresses in sip.conf and anywhere else. Failing that and if you're still pulling your hair out at the end of the week ( I know how it is), I would really consider re-installing the box (I'm using centos5 now on this server I'm configuring currently) and starting from scratch. I know it sounds like a cop out but that's what I would do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] select menu
this is, just in case your expecting a volume of calls
exten = ,1,Goto(contexts,s,1)
[context]
exten = s,1,Answer()
exten = s,2,Background(support)
exten = 1,1,Goto(context1,s,1)
exten = 2,1,Goto(context2,s,1)
exten = 3,1,Goto(context3,s,1)
exten = 4,1,Goto(context4,s,1)
exten = i,1,Goto(main-menu,s,1)
exten = t,1,Playback(silence/1)
On 5/9/07, Moacir O. de Souza Junior - Personalsoft Sistemas Ltda.
[EMAIL PROTECTED] wrote:
Hi,
My suggestion:
extensios.conf
exten = s,1,Answer()
exten = s,n,Wait(1)
exten = s,n,Read(MyVariable,TheNameOfSoudFile,1, , ,10)
exten = s,n,GotoIf($[${MyVariable}=1]?11)
exten = s,n,GotoIf($[${MyVariable}=2]?12)
exten = s,n,GotoIf($[${MyVariable}=3]?13)
exten = s,11,Dial(SIP/101,30,Ttm)
exten = s,12,Dial(SIP/102,30,Ttm)
exten = s,13,Dial(SIP/103,30,Ttm)
-- Timeout
|
Read(MyVariable,MySoudFile,1, , ,10)
|
|
- Number of characters that user has
to
digit
Asterisk cmd READ
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
[]'s
Moacir O. de Souza Junior
Belo Horizonte - Minas Gerais - Brasil
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de Josu Lazkano
Lete
Enviada em: quarta-feira, 9 de maio de 2007 11:49
Para: asterisk-users@lists.digium.com
Assunto: [asterisk-users] select menu
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone
calls the Asterisk play a record (in gsm) and them the caller must choose
a
option (1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose2 it will redirect to 102 extension
if he choose3 it will redirect to 103 extension
my extensions.conf is this one:
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
sorry about my english,
thanks to all
be
___
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Re: [asterisk-users] select menu
My suggestion:
[your incoming context]
#answer the phone
exten = s,1,Answer()
#playback recording but also accept extensions
exten = s,2,Background(your_gsm_recording)
#wait for caller to dial extension
exten = s,3,WaitExten(10)
#if they haven't hit an extension yet, play the message again
exten = s,4,Background(your_gsm_recording)
#give them one more chance
exten = s,5,WaitExten(10)
#send them to a default extension...maybe they have rotary phone
exten = s,6,Dial(SIP/101|30|tm)
#if all else fails, hangup
exten = s,7,Hangup()
# dynamic extension which makes 1=101, 2=102, etc.
exten = _X,1,Dial(SIP/10${EXTEN}|30|tm)
*
Adam Moffett
Plexicomm, LLC
[EMAIL PROTECTED]
ph: 866-759-4678x104
*
Josu Lazkano Lete wrote:
Hello everybody.
I want to make a menu with the incoming calls, I want that when someone calls the Asterisk play a record (in gsm) and them the caller must choose a option (1,2 or 3).
if he choose 1 it will redirect to 101 extension
if he choose 2 it will redirect to 102 extension
if he choose 3 it will redirect to 103 extension
my extensions.conf is this one:
[default]
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Dial(SIP/101,30,Ttm)
sorry about my english,
thanks to all
be
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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RE: [asterisk-users] SIP Problems continue...
That was in my list of things I've done, but failed to mention :). I never have used DNS on this box, but for verification I removed DNS servers and verified all addresses were IP's (which they were). There is no DNS active on this box at all. There's also no freepbx, just straight Asterisk. As for your comment on starting fresh, if I get no further help by this weekend that'll be my fun little weekend project. Thanks for the info. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Wednesday, May 09, 2007 10:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... Ken, I have similar problems every now and then on one of my asterisk boxes. I'm also running CentOS4 on that box. I've found that doing a sip reload when in that state results in something along : Last reload not yet finished (can't remember the exact wording) We're using cisco 7960's here. The ONLY time I've seen this happening is when I reload everything VIA freepbx. It used to do it every time I reloaded. I read somewhere that this was a result of DNS queries not being done in a timely fashion - So I went and replaced all the host statement in my trunk with IP addresses and now it doesn't do it very often at all. I don't know if this is your problem at all but it might be worth a shot. Replace any host names with IP addresses in sip.conf and anywhere else. Failing that and if you're still pulling your hair out at the end of the week ( I know how it is), I would really consider re-installing the box (I'm using centos5 now on this server I'm configuring currently) and starting from scratch. I know it sounds like a cop out but that's what I would do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Problems continue...
whats the asterisk version your using? On 5/10/07, Ken Williams [EMAIL PROTECTED] wrote: SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116716 0a2a959d3d3 00102/0 unkn No Init: INVITE 10.200.26.115715 1dee947d485 00102/0 unkn No Init: INVITE 10.200.26.104704 28808764699 00102/0 unkn No Init: INVITE 10.200.26.104704 36d3e88f59c 00102/0 unkn No Init: INVITE 10.200.26.104704 0e00060800d 00102/0 unkn No Init: INVITE Second, I've gone through and basically redone my extensions.conf to have it flow much smoother and clearer. I thought for sure my problem was coming from a loop somewhere in extensions.conf, but I'm now certain my extensions.conf is fine (but I'm glad I redid it, much easier to follow now). Third, I removed 'qualify=yes' from my sip.conf. I had read where people were having SIP channel lockups with this enabled, I again thought I had found the problem...but alas...In addition I had seen someone suggest setting REINVITE=NO, in addition to CANREINVITE=NO...no good. Fourth, I downgraded all my GXP-2000's to the latest released version of the software (1.1.1.14), some were on a newer version that I'm not sure where it came from (1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be downgraded), as those apparently had lock up issues as well...again thought I had found the problem... Fifth, I installed the latest SVN of 1.4 last night in hopes it was a known issue that had been fixednope We don't have a very complicated setup at all. The server is running CentOS 4, it has two TDM-400 cards with 6 FXS 2 FXO. We have about 25 GXP-2000 phones. My dialplan is nice and clean now. If no one has any further suggestions I'm to the point of opening a bug report with digium. I've read a ton on other people who have had this problem and followed the fixes for those people, but I can't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no [EMAIL PROTECTED] call-limit=9 allowsubscribe=yes Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with festival facing problem
Cheikhou DIAW wrote: hi List, i've been trying to get festival work on my 1.4.4 *box for the last 3days, i've used the tutorial on this page http://www.voip-info.org/wiki-Asterisk+Festival+installation http://www.voip-info.org/wiki-Asterisk+Festival+installation with exactly the same line in my dialplan just to make a test I would recommend looking at Cepstal. The voices are cheap, like $30(US) per voice per server. The quality is great and there are a number of third party libraries that abstract the use of the Cepstral Swift engine. Or you can just use the system command to create the sound files to play. http://www.cepstral.com I wrote a free AGI application for swift a while back that should work ok for you as well: http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper There is another one, a direct add-on for Asterisk, that will play the cepstral voices directly from a stream instead of first creating a file. I forgot the name/url though. Hopefully someone will jump in and provide that. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] additional volume added to sound on CONSOLE/dsp
Is there anyway to add additional volume gain the console/dsp port? I have used the mixer settings to set my volume on the soundcard to like 80 percent (I have even gone higher). However I still need some additional volume when speaking to the console/dsp. With SOX I can do a -v X on files and this helps when I play the file over the console/dsp. However, how can I add additional gain to live voice? Is there some way to put sox in the middle of the sound going to console/dsp? THanks for the suggestions. Jerry ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about Asterisk 1.4 depoyment.
Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can I either fix or work around this? Is Asterisk 1.4.2 ready for production deployment? Regards, Vietnhi -- Vietnhi Phuvan Senior Systems Engineer SPECIAL APPLIED INTELLIGENCE 2620 Jackson Ave, LIC, NY 11101 800.511.9818 [Tauk*] x2000 718.576.1404 [fax] - progress for hire - http://www.specialai.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HPEC audio clipping
If you contact Digium tech support directly they will provide you with the previous version of the echo canceler until the fix is made to the current version. Matthew Fredrickson On May 9, 2007, at 7:27 AM, Olivier wrote: Any field return on this ? Our last field trial of HPEC concluded we shouldn't use it at all, due to audio clipping. Is it now fixed ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
bilal ghayyad wrote: Hi; Well, I understood now that Nortel has some digital phones that can be used with astrisk, but the question: what are the card models that should be installed on Asterisk server? Digium? What these models? None. There are no Nortel digital phones that work with Asterisk. As I understand it, they MAY have some SIP phones, but I suspect they use a Nortel variant of SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Problems continue...
Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of franco escalona Sent: Wednesday, May 09, 2007 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... whats the asterisk version your using? On 5/10/07, Ken Williams [EMAIL PROTECTED] wrote: SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116716 0a2a959d3d3 00102/0 unkn No Init: INVITE 10.200.26.115715 1dee947d485 00102/0 unkn No Init: INVITE 10.200.26.104704 28808764699 00102/0 unkn No Init: INVITE 10.200.26.104704 36d3e88f59c 00102/0 unkn No Init: INVITE 10.200.26.104704 0e00060800d 00102/0 unkn No Init: INVITE Second, I've gone through and basically redone my extensions.conf to have it flow much smoother and clearer. I thought for sure my problem was coming from a loop somewhere in extensions.conf, but I'm now certain my extensions.conf is fine (but I'm glad I redid it, much easier to follow now). Third, I removed 'qualify=yes' from my sip.conf. I had read where people were having SIP channel lockups with this enabled, I again thought I had found the problem...but alas...In addition I had seen someone suggest setting REINVITE=NO, in addition to CANREINVITE=NO...no good. Fourth, I downgraded all my GXP-2000's to the latest released version of the software (1.1.1.14), some were on a newer version that I'm not sure where it came from (1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be downgraded), as those apparently had lock up issues as well...again thought I had found the problem... Fifth, I installed the latest SVN of 1.4 last night in hopes it was a known issue that had been fixednope We don't have a very complicated setup at all. The server is running CentOS 4, it has two TDM-400 cards with 6 FXS 2 FXO. We have about 25 GXP-2000 phones. My dialplan is nice and clean now. If no one has any further suggestions I'm to the point of opening a bug report with digium. I've read a ton on other people who have had this problem and followed the fixes for those people, but I can't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no [EMAIL PROTECTED] call-limit=9 allowsubscribe=yes Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] List of telemarketers??
Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone would prefer to access the dbase. Ritesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mobile Number to Mobile carrier mapping
Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc.
On 09/05/07, Stelios Koroneos [EMAIL PROTECTED] wrote: Hello ! For isdn BRI, the zaphfc in NT mode works fine. Some equipment might require voltage on the line (although they don't use it to powerup and it just draws a few mil amps) As for PRI never tested, i would be interested to know how your test goes I'll report back on making the first * server as the peer/provider etc. The card is arrving tomorrow. Thanks. Stelios S. Koroneos Digital OPSiS - Embedded Intelligence http://www.digital-opsis.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gavin Henry Sent: Wednesday, May 09, 2007 2:09 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. On 09/05/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi Gavin, A second Asterisk server replacing the provider (best way), or doing a loop between two different ISDN ports on a same card (worst way) must help you. Thanks for that. Will get a spare * box. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Gavin Henry Envoyé : mercredi 9 mai 2007 09:40 À : asterisk-users@lists.digium.com Objet : [asterisk-users] Testing ISDN T1/E1 Bri and Pri etc. Hi All, Can anyone recommend any test kit that you can hook up your Pri/Bri cards to without having actual ISDN in your office. For example testing an * system before it goes to a clients office. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Problems continue...
I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this ...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
Regarding (2) - you can either provide a realtime query service supporting web service interface which can be consumed using virtually any programming language and it would be very easy to build an AGI script around it. the second option would be to periodically update a flat file (csv) and provide ftp access - this way you won't have to sustain the load of the realtime queries as the demand grows and the numbers can be provisioned into PBX which doesn't have public Internet access. personally I don't have a use for such a DB, but I'm willing to help on setting it up for the community if needed. Joss. On 5/9/07, Ritesh Agrawal [EMAIL PROTECTED] wrote: Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone would prefer to access the dbase. Ritesh ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile Number to Mobile carrier mapping
Try this: http://puck.nether.net/npa-nxx/ * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Ritesh Agrawal wrote: Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
Hi Bilal - Well, I understood now that Nortel has some digital phones that can be used with astrisk, but the question: what are the card models that should be installed on Asterisk server? Digium? What these models? If you use the Citel Portico gateway, you don't need any telephony card, just regular ethernet. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: RE: Digital Phones
Can you connect existing Nortel system to Asterisk through fxs/fxo? That way one could use existing infrastructure for few old phones and Asterisk for new phones and all good things which come with it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Wednesday, May 09, 2007 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: RE: Digital Phones bilal ghayyad wrote: Hi; Well, I understood now that Nortel has some digital phones that can be used with astrisk, but the question: what are the card models that should be installed on Asterisk server? Digium? What these models? None. There are no Nortel digital phones that work with Asterisk. As I understand it, they MAY have some SIP phones, but I suspect they use a Nortel variant of SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Boost Polycom IP601 headset volume
Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) Alternatively, is there a way to have Asterisk 1.4.x boost the volume to a particular SIP device for all calls? Thanks for any ideas! Alvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
Hi Ritesh - Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone would prefer to access the dbase. Wow, that's a generous offer. I like the idea of a blacklist for telemarketers. It's bound to be more effective than an RBL for spammers! One thing to note: this may end up being a non-US database. Here in the US, I've experienced great success with the www.donotcall.gov service. If you're in the US and haven't signed up for this service, I'd highly recommend it. Of course, there may be non-telemarketer calls that it would be nice to be able to block. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_txfax, app_rxfax
On 5/8/07, Kevin Collins [EMAIL PROTECTED] wrote: I modified chan_sip.c to turn on a dsp to do fax detect based on inband dtmf being selected. And when reading rtp if 'f' character shows up vector to fax extension can i have your patched chan_sip.c ? Kevin Collins -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Tuesday, May 08, 2007 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] app_txfax, app_rxfax ax. The downside of rx_fax is that you need to compile it into asterisk. The downside of iaxmodem is that (to my knowledge) you can't easilly implement an auto-answer/detect fax/voice/ auto attendant/voicemail system. The channel must be dedicated to faxing, and that's that. This may or may not be an issue for you though. The last fax setup I did was for a small 2-person office where they had an existing fax machine that answered, listened for the remote fax squawk, if it didn't get it, then it rung the phones daisy-chained to it, and if they didn't answer it went to answering machine. I implemented this in asterisk fairly easilly with rx_fax. I'm not sure if you can do that with iaxmodem. Another question along these lines : How does everyone one fax detection on a sip channel? The only thing I've found is NvFaxDetect - anyone know of anything else? thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Problems continue...
Go back to 1.2.x and see if it fixes the problem. Ken Williams wrote: Started with 1.4.1, then 1.4.2, then 1.4.4, now the latest SVN (63478). From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of franco escalona Sent: Wednesday, May 09, 2007 11:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... whats the asterisk version your using? On 5/10/07, Ken Williams [EMAIL PROTECTED] wrote: SIP channel hang ups are progressively getting worse and I'm really grasping at straws here trying to find out what the cause is. The problem start, once a week or so the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. It's now doing it multiple times a day and I fear having to go back to our old phone system if I can't find a fix in the near future. When the SIP channel locks up the only fix is to restart Asterisk. SIP RELOAD RELOAD CHAN_SIP do no good. Here's a few things I've noticed and changes I've made in hopes of making it better. First, I've currently got 71 active SIP channels when only 2 people are on the phone. This doesn't happen every time, but could be part of the cause. The 'ghost' channels are all INVITES, how do I clear these without rebooting the system? 10.200.26.116716 0a2a959d3d3 00102/0 unkn No Init: INVITE 10.200.26.115715 1dee947d485 00102/0 unkn No Init: INVITE 10.200.26.104704 28808764699 00102/0 unkn No Init: INVITE 10.200.26.104704 36d3e88f59c 00102/0 unkn No Init: INVITE 10.200.26.104704 0e00060800d 00102/0 unkn No Init: INVITE Second, I've gone through and basically redone my extensions.conf to have it flow much smoother and clearer. I thought for sure my problem was coming from a loop somewhere in extensions.conf, but I'm now certain my extensions.conf is fine (but I'm glad I redid it, much easier to follow now). Third, I removed 'qualify=yes' from my sip.conf. I had read where people were having SIP channel lockups with this enabled, I again thought I had found the problem...but alas...In addition I had seen someone suggest setting REINVITE=NO, in addition to CANREINVITE=NO...no good. Fourth, I downgraded all my GXP-2000's to the latest released version of the software (1.1.1.14), some were on a newer version that I'm not sure where it came from (1.1.2.x). I also removed the 2 phones that were on 1.1.3.x (they can't be downgraded), as those apparently had lock up issues as well...again thought I had found the problem... Fifth, I installed the latest SVN of 1.4 last night in hopes it was a known issue that had been fixednope We don't have a very complicated setup at all. The server is running CentOS 4, it has two TDM-400 cards with 6 FXS 2 FXO. We have about 25 GXP-2000 phones. My dialplan is nice and clean now. If no one has any further suggestions I'm to the point of opening a bug report with digium. I've read a ton on other people who have had this problem and followed the fixes for those people, but I can't seem to get to the bottom of it. I have multiple SIP DEBUG console logs and DEBUG/VERBOSE set to 4 logs around the time SIP stops responding. SIP.CONF: [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=gsm context=from-internal allowsubscribe=yes notifyhold=no limitonpeers=yes [701] type=friend secret=blahblah port=5060 host=dynamic dtmfmode=rfc2833 dial=SIP/701 context=from-internal canreinvite=no reinvite=no [EMAIL PROTECTED] call-limit=9 allowsubscribe=yes Thanks for any help, Ken ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP Problems continue...
I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format reinstall the entire box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this ...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
Ritesh Agrawal wrote: Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone would prefer to access the dbase. Ritesh ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Be careful what you ask for. I run what's called a brute forcer list (www.infiltrated.net/bforcers) and I started off with a few people helping me. But found it was easier for me to get it to work the way I needed it to, updated the way I needed it to be, and managed by myself. It gets difficult depending on what you're doing and it will be a thankless effort. Good luck though if you need a mirror or something let me know. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Could two Asterisk servers connect through VPN
Thanks Tim, good option. The good thing with VPN is that two Asterisk servers would have no exposure on public internet. Tielin [EMAIL PROTECTED] 05/08/07 1:45 AM On 7 May 2007, at 19:51, Gordon Henderson wrote: On Mon, 7 May 2007, Tielin Xu wrote: Hi list: Has anyone done to set up two servers in different remote offices through VPN in order to get the VoIP communication? Yes it will work, but depending on your hardware you might be better off not using the VPN and just using an IAX trunk over the public Internet (unless you're really paranoid about someone listening in) Even if you are paranoid, you can still just use IAX, set 'encryption=yes' at both ends and IAX will encrypt the calls for you. There is a bandwidth overhead, but it is probably less that that of a VPN. Note, the calling/called numbers are still passed in the clear over encrypted IAX, so you are still vulnerable to traffic analysis. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk to record CDR in DB Oracle
-Original Message- From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: 08 May 2007 09:28 To: Asterisk Users Mailing List -
Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to
record CDR in DB Oracle On 7 May 2007, at 17:27, Florian Overkamp wrote:
Hi Everton,
Everton Goularth wrote:
I had success to do my asterisk to record CDR in a databese MYSQL...
Now, I need to do it to record CDR in Oracle...
Does Anybody knows how to do this??
Every hints are welcome
There is no native Oracle driver available to my knowledge, but if
you can install an ODBC driver for Oracle, Asterisk will happily
use that.
If anyone gets this to work, especially against an oracle instance on
a separate machine,
I'd love to know how you did it. I spent a day or so failing to get
it to work, then gave up
and had a perl script written that regularly posts the new CDR
records to oracle over http(s).
Tim Panton
www.mexuar.net
www.westhawk.co.uk/
Hi all,
what about yada?
I installed it and tried to connect to a database oracle em other machine em my
network but I can`t.
This are my configuration:
cdr_yada.conf
[global]
dbstr=oracle:192.168.0.180::MY_ORACLE_USER
user=MY_ORACLE_USER
pass=MY_ORACLE_PASSWORD
queue_size=500
queue_file=/var/asterisk/cdr_yada.queue
file_playback=yes
table=cdr
query=insert into cdr (id,calldate, clid, src, dst, dcontext, channel,
dstchannel, lastapp, lastdata, duration, billsec, disposition, amaflags,
accountcode, uniqueid, userfield) values (cdrseq.nextval,
to_date('?s','-mm-dd hh24:mi:ss'), ?v, ?v, ?v, ?v, ?v, ?v, ?v, ?v, ?d, ?d,
?v, ?d, ?v, ?v, ?v)
;[userfield_parse]
enabled=yes
; userfield columns ufc0 through ufc15
;[ufc0]
; name=col1
;[ufc1]
; name=col2
But in the asterisk cli I see that it isn't connected...
asterisk*CLI cdr yada status
cdr_yada build 005: $Date: 2006-04-06 22:38:22 -0500 (Thu, 06 Apr 2006) $
Not connected for 13642d16h21m8s.
0 of 500 records queued, 0 errors
queue_file is /var/asterisk/cdr_yada.queue
Somebody is working with yada?? Is my configuration wrong??
If somebody can help me I thank...
Everton Goularth
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Re: [asterisk-users] Question about Asterisk 1.4 depoyment.
Vietnhi Phuvan wrote: Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can I either fix or work around this? Is Asterisk 1.4.2 ready for production deployment? asterisk 1.4.2 is certainly not, 1.4.4 otoh is. I don't have any experience with meetme. Do you have zaptel loaded? Meetme depends on zaptel (at least ztdummy) for timing. Regards, Vietnhi -- Remco Post I didn't write all this code, and I can't even pretend that all of it makes sense. -- Glen Hattrup ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile Number to Mobile carrier mapping
This would not be valid, of course, for any number that was ported from 1 carrier to another. Adam Moffett wrote: Try this: http://puck.nether.net/npa-nxx/ * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * Ritesh Agrawal wrote: Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile Number to Mobile carrier mapping
On 5/9/07, Adam Moffett [EMAIL PROTECTED] wrote: Try this: http://puck.nether.net/npa-nxx/ This probably goes without saying, but this data is, at best, marginally useful due to LNP. -erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mobile Number to Mobile carrier mapping
Try this: http://puck.nether.net/npa-nxx/ A better one is: http://www.localcallingguide.com/lca_prefix.php Note, however, that this will show the allocation of the NXX which may no longer be the carrier handling the number if it has been ported to another carrier. AFAIK there is no public way to determine the termination carrier of a ported number. If anyone knows otherwise, I'd love to know the secret. -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 10 FXS - Channel Bank or PCI Card?
Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
Noah Miller wrote: Wow, that's a generous offer. I like the idea of a blacklist for telemarketers. It's bound to be more effective than an RBL for spammers! One thing to note: this may end up being a non-US database. Here in the US, I've experienced great success with the www.donotcall.gov service. If you're in the US and haven't signed up for this service, I'd highly recommend it. Of course, there may be non-telemarketer calls that it would be nice to be able to block. I would hope you have heard of CID spoofing? Won't stop as much as you think it will -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 echo infiltrated.net|sed 's/^/sil@/g' Wise men talk because they have something to say; fools, because they have to say something. -- Plato smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ericcson analog phone
Hi, Anybody using this Ericcson analog phone with Asterisk: Ericsson dialog 4187? I was told some functionalities like CLID will only work with an Ericsson PABX but other than that I would like to hear from anybody using this phone on a FXS port. Thanks, Jose Limeres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
On 5/9/07, Gavin Henry [EMAIL PROTECTED] wrote: Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. You could do it in one slot with Digium's TDM2400P (you would actually have to get 12 channels since they come in groups of 4). It tops out at 24 channels terminated at an amphenol connector, so you'll need a breakout box if you go this direction. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] List of telemarketers??
Thanks everyone for the responses, encouragement and offers to help. I will get started on this shortly and circle back with you guys. If someone has a starter list, it would help jump start the efforts/motivation :-) Ritesh On 5/9/07, Noah Miller [EMAIL PROTECTED] wrote: Hi Ritesh - Does anyone know if there is a known list of telemarketers? Something like http://whocalled.us/ with an easier access? We could all benefit if there was such a thing :-) If there is enough interest, I could put up a database that everyone can benefit from. I just need some suggestions on: (1) Adding new numbers based on community responses (some rule to sanity check) (2) Method that everyone would prefer to access the dbase. Wow, that's a generous offer. I like the idea of a blacklist for telemarketers. It's bound to be more effective than an RBL for spammers! One thing to note: this may end up being a non-US database. Here in the US, I've experienced great success with the www.donotcall.gov service. If you're in the US and haven't signed up for this service, I'd highly recommend it. Of course, there may be non-telemarketer calls that it would be nice to be able to block. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
Gavin - you should look at the Sangoma A4000X series cards, which only occupy a single slot and come in PCI or PCI-X versions. Cory Andrews -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: Wednesday, May 09, 2007 3:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 10 FXS - Channel Bank or PCI Card? Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p -393.html But it will be 3 PCI slots. Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
I would look into one of these: http://www.digium.com/en/products/hardware/analogcards.php quote who=Gavin Henry Hi All, What do you recommend? I was looking at: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci-express-p-393.html But it will be 3 PCI slots. -- And, did Galoka think the Ulus were too ugly to save? -Centauri ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA841 3.1.1(a) firmware file
Hello. I have a customer that needs to downgrade the firmware on their SPA841 to 3.1.1(a). I can't seem to find the firmware file. Google turned up 3.1.2-something and Linksys is taking a while to get back to me. Anyone happen to have that file lying around? Thanks, Nabeel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10 FXS - Channel Bank or PCI Card?
On May 9, 2007, at 3:45 PM, Gavin Henry wrote: http://www.voipon.co.uk/sangoma-a200-fxo-fxs-analogue-card-pci- express-p-393.html But it will be 3 PCI slots. Just to clarify in case you didn't already realize it. It doesn't actually *use* 3 PCI slots, it just occupies the physical space of 3. The board only connects to one slot, then has its own backplane that the additional daughter cards sit on. An important distinction if your concern with the use of 3 slots wasn't due to physical space, but rather was with dealing with IRQ and timing issues of having multiple slots in use. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a possiblity to check in the dialplan whether a SIP user is registred?
Hello everybody, Is there a possiblity to check in the dialplan whether a SIP user is registred? Something like : exten = user1,1,GotoIf(isRegistred(user1)? context1, context2, 1) Thanx, Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a possiblity to check in the dialplan whether a SIP user is registred?
ChanAvail() [EMAIL PROTECTED] wrote: Hello everybody, Is there a possiblity to check in the dialplan whether a SIP user is registred? Something like : exten = user1,1,GotoIf(isRegistred(user1)? context1, context2, 1) Thanx, Kalle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MINNESOTA: Twin Cities Asterisk Users Group - Saturday May 12th 2007 - 11:30am
There will be a Twin Cities Asterisk Users Group meeting this Saturday, May 12th, at 11:30 'til about 1:30 at the Atacomm Corporate Offices at 7365 Kirkwood Court N., Suite 350, Maple Grove, Minnesota 55369. Although there is no formal program scheduled, we'll chat about Asterisk applications including interesting dial plans including one enabling Asterisk systems to keep an eye (ear?) on each other to warn of computer, power or network outages. We may develop a dial plan to call Mom to wish her a happy Mothers' Day. And another to call 1-800 FLOWERS to get us out of the dog house. The meeting will be sponsored by CT Magic featuring the usual pizza and pop. Feel free to let me know if you'd like to arrange something different to eat. You're welcome to bring something to give away as a door prize. We need programs for June and August (there is no user group meeting in July). We need sponsors for these meetings, too. Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is sometimes posted online: http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group +Agenda ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
Robert Augustyn wrote: Can you connect existing Nortel system to Asterisk through fxs/fxo? That way one could use existing infrastructure for few old phones and Asterisk for new phones and all good things which come with it? No. They are digital phones and use proprietary Nortel signalling. The too bad thing about Nortel is that, when all your infrastructure is Nortel, it's pretty solid, reliable stuff, and in its day it was also pretty amazing. There's a reason why their PBX hardware was the most widely deployed in North America. Every technology has its day, though, and Nortel has been milking its contribution since 1975. The new IP stuff is unimpressive; I considered getting BCM certification once but when I looked at the equipment costs, I just shuddered. No sane business operator would pay those prices, and most of the insane ones are already in jail. Your best bet is to sell the sets you already have and replace them with appropriate IP hardware (you might even consider Aastra, which inherited most of Nortel's conventional telephony portfolio and has done great things with it); you can still get good money for used Nortel digital sets; many people are perfectly happy with their systems and I expect they'll be around for some time to come. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The 'h' extension problem
Rizwan Hisham wrote: Hi all, There is a problem with my dialplan. here is the dialplan: exten= 123,1,Dial(SIP/U1,,Ttg) exten= 123,2,Hangup exten= h,1,AGI(onhangup.pl) The problem is whenever U1 is called or calls someone, if U1 hangsup the call then the h extension is NOT executed. but if the other person hangsup the call, then the h extension is executed (assuming that the other person is calling from out of our asterisk system). I understand if U1 hangsup then there is no channel to execute h extension, but is it possible to execute the h exten even then. You cannot use AGI() on an inactive channel. Use DeadAGI() instead. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Boost Polycom IP601 headset volume
Alvin Austin wrote: Hi everyone, I have a user that needs a little extra volume on his Polycom IP 601 phone set for all calls (beyond what the volume control currently offers). Is there a provisioning setting for this anywhere? (I'd like to avoid a separate amplifier between the phone and handset if possible.) If you read the SIP administrator's guide plus any addenda for the current firmware, you can see the parameters for setting the headset gain. Different phones have different keys, which is why you'll need to look at the docs. In short, this can definitely be tweaked in the phone. You can set handset, headset, and chassis gain settings; these go over and above the manual control available on the phone's panel. It should be noted that Polycom actually recommends against using headsets without amplifiers. This is a common complaint with Polycom users. You'll probably be better off with an amplifier. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: RE: Digital Phones
Stephen, I understand that these sets are digital but what about connecting Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel? Would that work? Robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Bosch Sent: Wednesday, May 09, 2007 8:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: RE: Digital Phones Robert Augustyn wrote: Can you connect existing Nortel system to Asterisk through fxs/fxo? That way one could use existing infrastructure for few old phones and Asterisk for new phones and all good things which come with it? No. They are digital phones and use proprietary Nortel signalling. The too bad thing about Nortel is that, when all your infrastructure is Nortel, it's pretty solid, reliable stuff, and in its day it was also pretty amazing. There's a reason why their PBX hardware was the most widely deployed in North America. Every technology has its day, though, and Nortel has been milking its contribution since 1975. The new IP stuff is unimpressive; I considered getting BCM certification once but when I looked at the equipment costs, I just shuddered. No sane business operator would pay those prices, and most of the insane ones are already in jail. Your best bet is to sell the sets you already have and replace them with appropriate IP hardware (you might even consider Aastra, which inherited most of Nortel's conventional telephony portfolio and has done great things with it); you can still get good money for used Nortel digital sets; many people are perfectly happy with their systems and I expect they'll be around for some time to come. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?
Its a PRI, no VoIP trunks, so no DSL. This happens only in the office, where phones are connected through the same switch on which data flows for the Internet traffic. But this started happening only few weeks ago. Is there any way that I can check if its the switch or the router? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Audio going blank for a few seconds and then comes back. What could be the reason?
I have Grandstream and Aastra phones. It happens on both of them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote: I understand that these sets are digital but what about connecting Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel? Yes you can do that; I have. No you don't want to; it doesn't work worth a shit. You lose so many features, you are constantly putzing around with it, and it never works as good as you'll hope. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
Robert Augustyn wrote: Stephen, I understand that these sets are digital but what about connecting Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel? If you leave the Nortel PBX in the picture, I see no reason why that wouldn't work. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
How about if both ServerA and ServerB houses extensions 500 throught 699. Such that users can dynamically register Server A or Server B. Can we use DUNDi to implement such network? On 5/9/07, Alex Robar [EMAIL PROTECTED] wrote: Hi Ronaldo, Yes, you can use DUNDi for this. DUNDi simply advertises routes that a given server can terminate to its peers. As a very simple example, if ServerA houses extensions 500 through 599 and ServerB houses extensions 600 through 699, ServerA would advertise that it can terminate 5XX, and ServerB would advertise that it can terminate 6XX. When any peer in your DUNDi cloud requests how to terminate extension 502, ServerA will return a route to itself that will allow that call to be made. There's a nice article on the Texas AUG site about setting up DUNDi with dynamic extensions ( http://txaug.net/storage/users/3/3/images/17/Using%20DUNDi%20with%20a%20Cluster%20of%20Asterisk%20Servers.pdf ). Cheers, Alex Robar On 5/9/07, Ronaldo [EMAIL PROTECTED] wrote: Hi all, I'm planning to deploy many Asterisk servers for remote sites connected through IAX. Behind each server, there will be many sip clients connected. A sip client from one site must be able to make calls for the other sip clients connected to the other remote Asterisk servers. I've heard that DUNDi is a good option in order for each Asterisk server to locate the right (or the best) routes for the sip clients. Is DUNDi really used for that? Thanks in advance ... Ronaldo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: RE: Digital Phones
Andrew Kohlsmith wrote: On Wednesday 09 May 2007 8:26 pm, Robert Augustyn wrote: I understand that these sets are digital but what about connecting Asterisk fxs to Nortel fxo and keep sets connected to existing Nortel? Yes you can do that; I have. No you don't want to; it doesn't work worth a shit. You lose so many features, you are constantly putzing around with it, and it never works as good as you'll hope. That's kinda what I figured, but I don't have any personal experience with it so I didn't want to say anything. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trixbox drops call after running AGI script
Hey, I'm hoping somebody knows the answer to this.
The script works fine on the old Trixbox 1.0 but have recently upgraded
(just testing in VMWare) to Trixbox 2.2
What happens is Trixbox will drop the call after I call the AGI command
in my dial plan.
I first of generate a call file to call the user, then connect them to
an extension in the dial plan
[voice-report]
exten = 1,1,Answer()
exten = 1,2,AGI(call-logger.php|${userid})
exten = 1,3,Playback(custom/welcome)
exten = 1,4,Playback(custom/report06)
exten = 1,5,HangUp()
exten = h,1,DeadAGI(call-cleanup.php|${userid}|CHANUNAVAIL})
exten = failed,1,DeadAGI(error-logger.php|${userid})
call-logger.php works perfectly but its just after call-logger that
Trixbox will just terminate the call.
it still runs the other scrips (error-logger.php) so its not like the
service crashes.
I notice after a entire reboot of Trixbox the script runs fine for one
call then it mucks up again.
All the php files do is log the passed variables into a database so they
don't interact with
Trixbox in any other way.
Any help on the matter would be greatly appreciated.
Kind regards,
Allan..
___
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RE: [asterisk-users] SIP Problems continue...
A small way to make little easy, I dont know it people are ok to that, try integrating freepbx asterisk so you know what the sip configs should look like when things are all well. Things might stop working if there is a bug or change in configs. -- Deepak Ken Williams [EMAIL PROTECTED] wrote: I mean that SIP phones cannot answer incoming calls or make outgoing calls. When a call comes in on ZAP, it actually rings all the phones like normal, but when you try to answer no one is there. In addition, when you try to dial out you eventually get a message on the phones saying unable to communicate with the server. So there is some traffic still traveling on the SIP channel (the server's dialing extensions from an incoming ZAP call) but no further communication...almost as if it's a one way street of communication. The server can send data out on SIP but isn't receiving any. As for your issue, we haven't really had that (thankfully), so I don't think you're heading down the horrible spot we're in right now. Tonight I'm going to remove all aspects of Asterisk and reinstall fresh, if that fails I'll format reinstall the entire box. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett Sent: Wednesday, May 09, 2007 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Problems continue... I also get the mysterious SIP INVITE channels. 10.101.2.204 xxx 748e8b0a625 00102/0 unkn No Init: INVITE And I also am running 1.4.4 on CentOS4. Is that a pattern or just coincidence? The other symptom you mention is this ...the SIP phones couldn't communicate with the server, though there was no error message on the server and everything appeared fine on the server. Do you mean no calls in or out until you reboot? I don't have that thankfully, but I do have a guy telling me that incoming audio just goes away for a few seconds at a time. He says also that it sometimes goes away for long enough time that he was mistaking it for a dropped call. But if he waits long enough it pretty generally always comes back. I have consistent solid network performance from the asterisk server to the ATA (and believe me, I've looked very hard for a network problem), and I don't know what to look at next. Incidentally, the guy hasn't called me since I rebooted last week. Is this similar to how your situation started? * Adam Moffett Plexicomm, LLC [EMAIL PROTECTED] ph: 866-759-4678x104 * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Got a question? Someone out there knows the answer. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
