Andy Beak
2010-Jul-20 09:02 UTC
[asterisk-users] Call not going through and failing because "never answered"
Hi, I'm trying to use Asterisk to place Automated Voice Calls. A verbose log from Asterisk CLI taken when I place a call in the spool directory looks like this: -- Attempting call on SIP/MTN-NEW/my-number for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5 > Channel SIP/MTN-NEW-00000005 was never answered. [Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) My sip.conf looks like this: [MTN-NEW] host=192.168.34.1 disallow=all allow=ilbc allow=gsm allow=g729 allow=g723 allow=ulaw allow=g729 type=peer My SIP provider says that no traffic is picked up at their SBC or on the WAN gateway port assigned to us. I've just done a fresh reinstall of Asterisk and am using sample configurations for all other conf files. I am using an open source g729 codec and have tried shuffling the gsm up above it in case it doesn't work properly (to no avail). Can anybody help me on this? My boss is breathing down my neck and I've never worked with Asterisk before. Thanks, Andy -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 5606 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/1559b58d/attachment.bin
Zeeshan Zakaria
2010-Jul-20 09:40 UTC
[asterisk-users] Call not going through and failing because "never answered"
In your sip.conf, there is no mention of your sip provider's IP address, username and secret (password). Even if the provider doesn't have username and secret requirements, there should at least be his IP address somewhere in your sip.conf. Do they require registration? You should ask them what sip credentials you need to have on your system. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at cellsmart.co.za> wrote: Hi, I'm trying to use Asterisk to place Automated Voice Calls. A verbose log from Asterisk CLI taken when I place a call in the spool directory looks like this: -- Attempting call on SIP/MTN-NEW/my-number for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5> Channel SIP/MTN-NEW-00000005 was never answered.[Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) My sip.conf looks like this: [MTN-NEW] host=192.168.34.1 disallow=all allow=ilbc allow=gsm allow=g729 allow=g723 allow=ulaw allow=g729 type=peer My SIP provider says that no traffic is picked up at their SBC or on the WAN gateway port assigned to us. I've just done a fresh reinstall of Asterisk and am using sample configurations for all other conf files. I am using an open source g729 codec and have tried shuffling the gsm up above it in case it doesn't work properly (to no avail). Can anybody help me on this? My boss is breathing down my neck and I've never worked with Asterisk before. Thanks, Andy -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/54bdeae3/attachment.htm
Andy Beak
2010-Jul-20 14:22 UTC
[asterisk-users] Call not going through and failing because "never answered"
Hi, I set my list to subscribe to digest and I can't see how to reply to your reply without starting a new thread. There is no need for SIP username and password because the provider authenticates me on my IP address. I thought that "host=192.168.34.1" would be the sip provider IP address. At this point I don't need to accept incoming calls or place VOIP-to-VOIP. All I need to do is connect to their PBX to place a call to a cellphone. I reread all the documentation I could find and couldn't see where else in sip.conf I should set the provider IP. Thanks for your reply, Andy > In your sip.conf, there is no mention of your sip provider's IP address, username and secret (password). Even if the provider doesn't have username and secret > requirements, there should at least be his IP address somewhere in your sip.conf. Do they require registration? You should ask them what sip credentials you need > to have on your system. Zeeshan A Zakaria -- www.ilovetovoip.com <http://www.ilovetovoip.com>> On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at xxxxxxxxxxxxxxx > <mailto:andrewb at xxxxxxxxxxxxxxx>> wrote: > > Hi, > > I'm trying to use Asterisk to place Automated Voice Calls. > > A verbose log from Asterisk CLI taken when I place a call in the spool > directory looks like this: > > -- Attempting call on SIP/MTN-NEW/my-number for application > MP3Player(/myfile) (Retry 1) > == Using SIP RTP CoS mark 5 > > Channel SIP/MTN-NEW-00000005 was never answered. > [Jul 20 10:52:11] NOTICE[14580]: pbx_spool.c:339 attempt_thread: Call > failed to go through, reason (8) Congestion (circuits busy) > > My sip.conf looks like this: > > [MTN-NEW] > host=192.168.34.1 > disallow=all > allow=ilbc > allow=gsm > allow=g729 > allow=g723 > allow=ulaw > allow=g729 > type=peer > > My SIP provider says that no traffic is picked up at their SBC or on > the WAN gateway port assigned to us. > > I've just done a fresh reinstall of Asterisk and am using sample > configurations for all other conf files. I am using an open source > g729 codec and have tried shuffling the gsm up above it in case it > doesn't work properly (to no avail). > > Can anybody help me on this? My boss is breathing down my neck and > I've never worked with Asterisk before. > > Thanks, > Andy > > > -- > _____________________________________________________________________ > -- 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 byhttp://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 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 5606 bytes Desc: S/MIME Cryptographic Signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/dc0a2f3a/attachment.bin
Zeeshan Zakaria
2010-Jul-20 14:58 UTC
[asterisk-users] Call not going through and failing because "never answered"
This "host=192.168.34.1" is where you'll put your provider's IP address. Currently you are using some local address which is not your provider's IP address. Where did you get it from? Call your providrr and ask them the IP address of the server where you'll be sending your calls. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-20 10:27 AM, "Andy Beak" <andrewb at cellsmart.co.za> wrote: Hi, I set my list to subscribe to digest and I can't see how to reply to your reply without starting a new thread. There is no need for SIP username and password because the provider authenticates me on my IP address. I thought that "host=192.168.34.1" would be the sip provider IP address. At this point I don't need to accept incoming calls or place VOIP-to-VOIP. All I need to do is connect to their PBX to place a call to a cellphone. I reread all the documentation I could find and couldn't see where else in sip.conf I should set the provider IP. Thanks for your reply, Andy> In your sip.conf, there is no mention of your sip provider's IP address,username and secret (pa... www.ilovetovoip.com <http://www.ilovetovoip.com>> On 2010-07-20 5:09 AM, "Andy Beak" <andrewb at xxxxxxxxxxxxxxx <mailto:andrewb at xxxxxxxxxxxxxxx>> wr... -- _____________________________________________________________________ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.aste... -- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/e1a3eeeb/attachment.htm
Zeeshan Zakaria
2010-Jul-20 16:24 UTC
[asterisk-users] Call not going through and failing because "never answered"
You are getting congestion error message, which in your case only means failed sip communication, or no sip communication at all. Settings on your end are just fine. Can you post the Dial command from your extensions.conf? Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-07-20 12:16 PM, "Andy Beak" <andrewb at cellsmart.co.za> wrote: Hi, Thanks, I added that. I'll ask my network provider if they received these message tomorrow morning. That will narrow things down to either an Asterisk configuration or a network routing issue. There is not really a caller, I'm trying to use Asterisk as an Automated Voice Message server to dial phone numbers and play an mp3. I'm using my mobile phone to test on and it doesn't ring. Asterisk gives the following message immediately after reading the .call file from the spool directory: -- Attempting call on SIP/MTN-NEW/mynumber for application MP3Player(/myfile) (Retry 1) == Using SIP RTP CoS mark 5> Channel SIP/MTN-NEW-00000001 was never answered.[Jul 20 18:07:37] NOTICE[22259]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (8) Congestion (circuits busy) Because the phone doesn't ring and the error message appears immediately I don't think it's a timeout issue. Will reading the source for pbx_spool.c at line 339 give any clues as to what's happening or will that be a waste of time? Cheers, Andy On 20/07/2010 05:42 PM, Gareth Blades wrote:> > If you add qualify=yes to the setting in sip.con...-- _____________________________________________________________________ -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100720/6bbd0ba8/attachment.htm