No compatible codecs, not accepting this offer! – after upgrading to 1.8.11

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Asterisk Users 4 Comments

Hi,

I’ve upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don’t work anymore. I
don’t know why!…

This is the SDP portion that comes in the INVITE messages of calls through
that trunk (let’s say, whose endpoint has the IP x.x.x.x, purposely
omitted). Nothing seems to be wrong with that to me:
v=0
o=CSM 0 1 IN IP4 x.x.x.x
s=Acme
c=IN IP4 x.x.x.x
t=0 0
m=audio 22152 RTP/AVP 8 0 18 4 101
a=rtpmap:101 telephone-event/8000

And here’s the debugging:
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP
to Off
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP v=0… UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP o=CSM 0 1 IN IP4 x.x.x.x… UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP s=Acme… UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport:
Splitting ‘x.x.x.x’ into…
[May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport:
…host ‘x.x.x.x’ and port ”.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP c=IN IP4 x.x.x.x… OK.
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing
session-level SDP t=0 0… UNSUPPORTED.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 18 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 4 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:537
ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on
0x416e25b0
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:9110 process_sdp: Processing
media-level (audio) SDP a=rtpmap:101 telephone-event/8000… OK.
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640
ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0
[May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible
codecs, not accepting this offer!

Any help?

Thanks,
Ricardo.

4 thoughts on - 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.

  • Do a “sip show peer PEERNAME” and check the codecs allowed for that specific peer.

  • 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

  • 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.