Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Do a sip show peer PEERNAME and check the codecs allowed for that specific peer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ricardo Carvalho Sent: Wednesday, May 09, 2012 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11 That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what peer is being selected. Andres Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.uk mailto:asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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] No compatible codecs, not accepting this offer! - after upgrading to 1.8.11
Problem SOLVED. You'r right, this is a problem of codec mismatching. Activating sip debug i can see it: Capabilities: us - 0x802 (gsm|g726), peer - audio=0x10d (g723|ulaw|alaw|g729) [May 9 17:16:37] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! I solved the problem thanks to your help! Since that SIP trunk isn't authenticated, i just receive calls in the default context that is set in sip.conf, and so, I don't set the codecs to be used. I discovered that the problem was that i had one other peer defined in sip.conf that had the same IP address set, so it was shuffling asterisk some how. Funny that the same configuration wasn't a problem in asterisk 1.4, but in this 1.8 it caused this problem. Thank you onde again, Regards, Ricardo. On Wed, May 9, 2012 at 5:10 PM, Andres and...@telesip.net wrote: On 5/9/2012 11:56 AM, Ricardo Carvalho wrote: That's weird, because it's negotiated with success the codec ulaw for outbound calls through the same SIP trunk. My guess is the incoming call is not being matched with the peer you are expecting. Do a sip debug and watch the output to see what peer is being selected. Andres Besides, ulaw and alaw shows up when i do core show codecs audio in the asterisk CLI, and there exists both codec_ulaw.so and codec_alaw.so modules under the path /usr/lib/asterisk/modules/ I don't get it!... More ideas? Thanks, Ricardo. On Wed, May 9, 2012 at 3:32 PM, A J Stiles asterisk_l...@earthshod.co.ukmailto: asterisk_l...@earthshod.co.uk wrote: On Wednesday 09 May 2012, Ricardo Carvalho wrote: [May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer! Any help? Are you sure you compiled all the codecs you need? What happens if you run `make menuselect` in both the 1.4 source tree and in the 1.8 source tree, side-by-side in tabs of the same terminal window? You need at least GSM, A-law and micro-law. (The above is my preferred method of building a configuration like an existing one. No doubt someone will weigh in with a better way of doing it.) -- AJS Answers come *after* questions. -- _ -- 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
