Re: [asterisk-users] Asterisk 1.6 and Outbound SIP Proxy configuration
Update: No luck with versions 1.6 and 1.8.7 I had to revert back to 1.4 which worked with no problem. Probably if I have some time, I will do more testing with version 1.8.7 to see what the difference is and what changes need to be made for this kind of setup to work in 1.8.7 Joseph On Mon, Aug 6, 2012 at 10:59 AM, Joseph Begumisa j.begum...@gmail.comwrote: Hello, Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below: -- [trunk1] defaultuser=x...@sip.provider.com fromuser= fromdomain=sip.provider.com type=peer secret=a outboundproxy=10.10.10.10 ;(replaced actual ip) nat=no host=sip.provider.com dtmfmode=auto disallow=all context=from-internal canreinvite=no allow=g729 trustrpid=yes sendrpid=yes register = x...@sip.provider.com:a@10.10.10.10:5060 -- With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call: -- [2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup ' sip.provider.com' [2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: sip.provider.com, on peer trunk1, removing peer -- I have done some research on this issue but not been able to find anything conclusive on why this would happen. I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this. Any pointers? Thanks. Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 and Outbound SIP Proxy configuration
Hello, Using asterisk 1.6 as sip client to register with sip provider and terminate calls through them. SIP Provider has provided sip proxy and sip server details. The problem is that the sip server FQDN does not resolve on the internet. So I can only presume that the SIP proxy knows how to reach the sip server. Asterisk 1.6 seems to have a problem with this. This is my config below: -- [trunk1] defaultuser=x...@sip.provider.com fromuser= fromdomain=sip.provider.com type=peer secret=a outboundproxy=10.10.10.10 ;(replaced actual ip) nat=no host=sip.provider.com dtmfmode=auto disallow=all context=from-internal canreinvite=no allow=g729 trustrpid=yes sendrpid=yes register = x...@sip.provider.com:a@10.10.10.10:5060 -- With the above config, I can register with the providers sip proxy, however, the error below is observed in the logs concerning the host when I try to make a call: -- [2012-08-02 23:37:31] WARNING[26155] acl.c: Unable to lookup ' sip.provider.com' [2012-08-02 23:37:31] ERROR[26155] chan_sip.c: srvlookup failed for host: sip.provider.com, on peer trunk1, removing peer -- I have done some research on this issue but not been able to find anything conclusive on why this would happen. I tested the sip details provided with a different sip client (actually an IP phone) and was able to register and send / receive calls with no problem. The problem just seems to be somewhere in my asterisk client configuration or a known bug with the version of asterisk I am using for this. Any pointers? Thanks. Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Determine When Call Is Picked Up In Queue
Hi, I have a situation where a call comes in to my asterisk server, goes through an IVR and is then handed off to another asterisk server where it enters a queue waiting for an agent to answer the call. (I do not control the second asterisk server). Is there a way for me to know when the call is actually picked up on the second asterisk server? I have a billing application that needs to start billing when the call is actually answered by an agent. Thanks a lot. Best Regards, Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
And not to mention the need for power over ethernet switches to avoid having many power adpaters lying all over. Don't get me wrong, I'm for IP Phones, however, in this specific scenario that I have, getting an FXS to SIP gateway with 24 ports makes more sense. Thanks for all the pointers. Best Regards, Joseph On Tue, Mar 30, 2010 at 8:39 AM, Andrew Latham lath...@gmail.com wrote: And to add to this, analog is useful for its distance when running wall phones in a large warehouse setting... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Mar 30, 2010 at 11:29 AM, Darrick Hartman dhart...@djhsolutions.com wrote: Sometimes you need to look at the cost to pull new wire too, not just the cost of the phones. There are a few cases where the channel banks + analog phones make sense, especially when the analog devices are already there. Sent from my BlackBerry® wireless device from U.S. Cellular -Original Message- From: hin lee hi...@yahoo.com Date: Tue, 30 Mar 2010 08:25:19 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Thanks for the feedback. Btw, I meant SIP / IAX gateway. I'll take a look at the suggestions. Best Regards, Joseph On Sun, Mar 28, 2010 at 7:28 PM, Steve Edwards asterisk@sedwards.comwrote: On Sun, 28 Mar 2010, Joseph Begumisa wrote: Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Adtran channel banks are a great trailing edge technology. You can get them off Ebay for pennies on the original dollar and they are built like a tank. (voip gateway is not very specific. If you meant SIP or IAX, you might want to specify which.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Hi, Can anyone recommend a 24 fxs port voip gateway that has worked well with asterisk? I have a couple of analog handsets that I want to hookup to my asterisk server? Any tested and tried product recommendations are welcome. Thanks. Best Regards, Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card - Update
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Wednesday, October 24, 2007 9:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card We had issues with TE110p cards in Dell 860's, but TE120p's fixed the problem. PaulH It is now 1 week since I replaced the TE110P with the TE120P in the Dell Poweredge 1950 and I have not had any problems. The TE120P seems to have resolved the earlier problem I had. Joseph ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compatibility Issues with dell poweredge 195 and TE110P card
Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. Yes, I have had this problem with a dell PE1650, 1850, SC1400, and PE650. I have a TE410P that does it. It may not be wise, but I just ignore the orange blinking LCD display (or light, depending on the model). I did try reseating the card, and it works for a few weeks, and then goes back to the same old thing. Yes, that happened too. Digium has graciously offered to send me a TE120P with the Digium VoiceBus technology which I will test out on the Dell 1950. Will post my findings thereafter. Joseph. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compatibility Issues with dell poweredge 1950 and TE110P card
Has anyone had any compatibility issues with a TE110P card installed on a Dell Poweredge 1950? I noted the following error on the LCD display of the Dell Poweredge 1950: E1711 PCI PErr Slot 1 E171F PCIE Fatal Error B0 D4 F0. The Dell hardware owners manual states that it means the system BIOS has reported a PCI parity error on a component that resides in PCI configuration space at bus 0, device 4, function 0 and advises that the PCI expansion card be removed and reseated. Any suggestions on what exactly might be causing this are welcome. Thanks. Joseph ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Next, try doing this in your dialplan and see if it works: exten = ,1,SIPAddHeader(Call-Info:\;answer-after=0) exten = ,n,Page(SIP/201SIP/202SIP/203SIP/204) The initial email mentioned that the Page command causes asterisk to reboot. Joseph ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AA50 Paging
Hi, I am curious. What version of asterisk is running on that AA50? Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kelly opal Sent: Sunday, October 14, 2007 5:46 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] AA50 Paging Hi I just got an AA50 from Digium and the paging command reboots asterisk when you use it. Digium says it is a requested feature and is of low priority. Is there any other way to page 10 Grandstream gxp2000 phones with meetme or some other command than the page command. Thanks in advance. Kelly ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
No, I am not sure whether it's still an issue with the newer series because I have not had a chance to test this with one of those models. Regards, Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Canfield Sent: Thursday, October 11, 2007 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Yes, this is true when using presence on the 601's. With presence disabled, you get no reboots at all. That's why when I realized that, I decided on the setup I mentioned below. However, you could let us know whether this is true for the 650's 550's and 330's. Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Friday, October 12, 2007 9:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk I use a mysql script to dynamically generate the page command and page about 70 phones, and I have never had a reboot problem. Sometimes there is a slight delay waiting for all the phones to join the page conference. I am using a mix of 650's, 550's, and 330's. It must only be an issue if you are using presence. Maybe I will setup presence on a couple phones and see if they reboot. Forrest Beck [EMAIL PROTECTED] http://www.shift8.biz/blog On Oct 11, 2007, at 4:34 PM, Jim Canfield wrote: Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. For example if one phone was extension 154, I added a second line key assigned extension 8154 to that phone only for the purpose of receiving pages and then added 8154 to the page group instead of 154. This way presence can still be enabled for 154 while 8154 which is not watched (and therefore will not send out those presence notifications which cause the phone to reboot) can be used to receive the pages. Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, October 10, 2007 1:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk i'm using Polycom 601 in an office of 30 handsets. I have not heard my customer complaining about phones being rebooted after page. On 10/9/07, Bill Andersen [EMAIL PROTECTED] wrote: I could not tell you in asterisknow but I use this feature with Polycom phones on all of my installs. It is very well documented in voip-info.org Do you have any problem with the Paging when there are say 20 phones in the page group? We have a IP601 that is used by the receptionist and has 2 side cars. We have to keep presence (Buddy List) enabled so the sidecar lights go on and off. However, about 1 out of 10 times the receptionist pages, her phone reboots. Polycom says it can't handle the traffic from the buddy list presence notifications. Have you seen this? Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dtmf issues on PRI and 1.4.11
Hi Jerry, Please post your Zapata.conf configuration. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Wednesday, September 19, 2007 8:37 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] dtmf issues on PRI and 1.4.11 I am missing DTMF digits on a PRI with 1.4.11 I added dtmf logging in logger.conf. I can see that if I enter 205 I dont see the 2 but all I see is 05. I have added the dsp.c patch that was recently added to bugs but that doesnt seem to help my situation. What can I do to provide more information so this can be fixed/patched. THanks, Jerry ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
Actually this problem is with a telco in the US [the setup is in the US]. I will get in touch with them to have them look into it. There is another similar setup with the same telco and there are no such problems. The only difference in the setups is that in this case, the T1 is terminated into a Cisco 2430 Integrated Access Device and then a T1 from that device terminates into the Asterisk PBX. Probably I will have them bypass the Cisco device and see whether I can replicate this again. Joseph. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Monday, September 10, 2007 7:39 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: Thanks. My results after applying the patch and recompiling are that the problem can only be replicated with calls from mobile networks. Digits like 160 entered in the digital receptionist by a caller are received by the asterisk server as 16660 sometimes. Other times it is received as 1660. Digits like 1234 are received as 1222334 etc... From fixed lines, there is no problem. Digits are received as they have been sent. Any other pointers? Hmm, that sounds like a problem with the GSM-to-PSTN gateway that the calls are passing through. Unless things are different in Uganda, I believe when a user presses a DTMF key on their mobile, it doesn't send a tone through the mobile network, but rather a start dtmf control message followed by a stop dtmf control message. When the call gets gatewayed from GSM to the PSTN network, it is the job of the gateway to generate the tones as instructed by the control protocol. (Someone please correct me if I'm wrong). So you may need to take it up with your telco. Cheers Tony Thanks a lot. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, September 09, 2007 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. Thanks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, September 05, 2007 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Try the patch at http://bugs.digium.com/view.php?id=10535 and see if it helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
Thanks. My results after applying the patch and recompiling are that the problem can only be replicated with calls from mobile networks. Digits like 160 entered in the digital receptionist by a caller are received by the asterisk server as 16660 sometimes. Other times it is received as 1660. Digits like 1234 are received as 1222334 etc... From fixed lines, there is no problem. Digits are received as they have been sent. Any other pointers? Thanks a lot. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, September 09, 2007 12:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems On 12:09, Sun 09 Sep 07, Joseph Begumisa wrote: I applied the patch, however, I'd like to know which particular files to copy after running a make. I do not wish to run make install as it will overwrite other configuration changes I have made. A make install will not overwrite any configfile. It will install the asterisk binary and the modules (thus overwriting the existing files) but configfiles will only be overwritten when you run: make samples -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Relay Problems
Thanks. Will check that out. Joseph -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Wednesday, September 05, 2007 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF Relay Problems In article [EMAIL PROTECTED], Joseph Begumisa [EMAIL PROTECTED] wrote: I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Try the patch at http://bugs.digium.com/view.php?id=10535 and see if it helps. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Relay Problems
Hi, I have a client setup where a T1 is terminated into a Cisco IAD2430 Series device which then interfaces with a Digium Wildcard TE110P card in a server running Asterisk 1.2.23. I am having a problem with the DTMF tones being passed to the Asterisk server. Wrong tones are being passed to the server especially during the digital receptionist menu selections. Setting relaxdtmf=yes does not seem to address the situation. Any pointers? Regards, Joseph ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problems with Asterisk 1.2.23 and Polycom 601
Hi, I have a polycom 601 with 3 expansion modules attached and about 40 extensions. When someone does a page all from that phone, sometimes the expansion modules reboot and sometimes the phone itself reboots. This happens randomly. I suspected it to be a problem with the presence / buddy watch feature in the directory on the phone so I turned that off for all the extensions listed in the directory and the problem has not re-occurred. However, this means the status of the other phones cannot be seen on the expansion modules of the reception phone and I am interested in that. Any pointers? I have done a couple of searches and not come up with anything concrete on resolving this problem. I am running Asterisk 1.2.23. Any help would be appreciated. Regards, Joseph ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
