<div>Hi,</div><div><br></div><div>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!...</div>
<div><br></div><div>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:</div>
<div>v=0<br>o=CSM 0 1 IN IP4 x.x.x.x</div><div>s=Acme<br>c=IN IP4 x.x.x.x</div><div>t=0 0<br>m=audio 22152 RTP/AVP 8 0 18 4 101<br>a=rtpmap:101 telephone-event/8000</div><div><br></div><div>And here's the debugging:</div>
<div>[May 8 17:45:30] DEBUG[6444]: chan_sip.c:5092 do_setnat: Setting NAT on RTP to Off<br>[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP v=0... UNSUPPORTED.<br>[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.<br>
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP s=Acme... UNSUPPORTED.<br>[May 8 17:45:30] DEBUG[6444]: netsock2.c:134 ast_sockaddr_split_hostport: Splitting 'x.x.x.x' into...<br>
[May 8 17:45:30] DEBUG[6444]: netsock2.c:188 ast_sockaddr_split_hostport: ...host 'x.x.x.x' and port ''.<br>[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.<br>
[May 8 17:45:30] DEBUG[6444]: chan_sip.c:8891 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED.<br>[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<br>
[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<br>[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<br>
[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<br>[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<br>
[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.<br>[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 0 on 0x416e25b0<br>
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 4 on 0x416e25b0<br>[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 8 on 0x416e25b0<br>
[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 18 on 0x416e25b0<br>[May 8 17:45:30] DEBUG[6444]: rtp_engine.c:640 ast_rtp_codecs_payload_formats: Incorporating payload 101 on 0x416e25b0<br>
[May 8 17:45:30] NOTICE[6444]: chan_sip.c:9188 process_sdp: No compatible codecs, not accepting this offer!</div><div><br></div><div><br></div><div>Any help?</div><div><br></div><div>Thanks,</div><div>Ricardo.</div>