global_s Posted July 20, 2012 Report Share Posted July 20, 2012 Hi, In my pbx I tried changing the codec priority in system settings, allowing only g711u and g711a. When I did a test call from an extension (who has the same codec enabled and many more) I receive error 415 Unsupported media type. I then removed the changes I made, but the error remains. I tried to reboot the pbx with no luck Any help? I attach here the log from a test call INVITE sip:8877@176.50.240.2 SIP/2.0 Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2> Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 8 INVITE Contact: <sip:43402@85.50.240.22:53137> Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: <sip:43402@176.50.240.2> Content-Length: 394 v=0 o=- 1452070351 0 IN IP4 192.168.1.5 s=SIPPER for PhonerLite c=IN IP4 192.168.1.5 t=0 0 m=audio 53139 RTP/AVP 0 8 2 3 97 110 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [5] 2012/07/20 11:00:24: SIP Tx udp:85.50.240.22:53137: SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport=53137 From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 8 INVITE Content-Length: 0 [5] 2012/07/20 11:00:24: SIP Tx udp:85.50.240.22:53137: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport=53137 From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 8 INVITE User-Agent: snomONE/4.5.0.1075 Delta Aurigids WWW-Authenticate: Digest realm="176.50.240.2",nonce="6ae674e91ce208ae882342ff67954e2a",domain="sip:8877@176.50.240.2",algorithm=MD5 Content-Length: 0 [5] 2012/07/20 11:00:24: SIP Rx udp:85.50.240.22:53137: ACK sip:8877@176.50.240.2 SIP/2.0 Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK80b87a1db7d0e1118ab1001e9030b32e;rport From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 8 ACK Max-Forwards: 70 Content-Length: 0 [5] 2012/07/20 11:00:24: SIP Rx udp:85.50.240.22:53137: INVITE sip:8877@176.50.240.2 SIP/2.0 Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2> Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 9 INVITE Contact: <sip:43402@85.50.240.22:53137> Authorization: Digest username="43402", realm="176.50.240.2", nonce="6ae674e91ce208ae882342ff67954e2a", uri="sip:8877@176.50.240.2", response="d47ab92ac330ba427d2b0770a3f555a1", algorithm=MD5 Content-Type: application/sdp Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE Max-Forwards: 70 Supported: 100rel, replaces, from-change User-Agent: SIPPER for PhonerLite P-Preferred-Identity: <sip:43402@176.50.240.2> Content-Length: 394 v=0 o=- 1452070351 0 IN IP4 192.168.1.5 s=SIPPER for PhonerLite c=IN IP4 192.168.1.5 t=0 0 m=audio 53139 RTP/AVP 0 8 2 3 97 110 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [5] 2012/07/20 11:00:24: SIP Tx udp:85.50.240.22:53137: SIP/2.0 100 Trying Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport=53137 From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 9 INVITE Content-Length: 0 [5] 2012/07/20 11:00:24: SIP Tx udp:85.50.240.22:53137: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport=53137 From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 9 INVITE Contact: <sip:43402@176.50.240.2:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snomONE/4.5.0.1075 Delta Aurigids Content-Length: 0 [5] 2012/07/20 11:00:24: SIP Rx udp:85.50.240.22:53137: ACK sip:8877@176.50.240.2 SIP/2.0 Via: SIP/2.0/UDP 85.50.240.22:53137;branch=z9hG4bK803f0e21b7d0e1118ab1001e9030b32e;rport From: "43402" <sip:43402@176.50.240.2>;tag=3448809797 To: <sip:8877@176.50.240.2>;tag=c0c8306a6c Call-ID: 80B87A1D-B7D0-E111-8AB0-001E9030B32E@85.50.240.22 CSeq: 9 ACK Authorization: Digest username="43402", realm="176.50.240.2", nonce="6ae674e91ce208ae882342ff67954e2a", uri="sip:8877@176.50.240.2", response="d47ab92ac330ba427d2b0770a3f555a1", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia support Posted July 20, 2012 Report Share Posted July 20, 2012 Check the "SIPPER for PhonerLite" codec and see if it's using a different one then what is supported on the pbx.. Quote Link to comment Share on other sites More sharing options...
global_s Posted July 20, 2012 Author Report Share Posted July 20, 2012 Check the "SIPPER for PhonerLite" codec and see if it's using a different one then what is supported on the pbx.. a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 speex/16000 a=rtpmap:9 G722/8000 these are the one configured SmonOne has all the default codec enabled, which should comprehend g711u,g711a Why do I receive the error? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted July 20, 2012 Report Share Posted July 20, 2012 Hmm. Do you also see the traffic on the other side of the B2BUA? Maybe the codec negotiation runs into a problem on the other leg, and the PBX just relay the message "415 Unsupported Media Type". It can also happen if the PBX proposes SRTP, but does not get the SRTP key back. Quote Link to comment Share on other sites More sharing options...
global_s Posted July 20, 2012 Author Report Share Posted July 20, 2012 Hmm. Do you also see the traffic on the other side of the B2BUA? Maybe the codec negotiation runs into a problem on the other leg, and the PBX just relay the message "415 Unsupported Media Type". It can also happen if the PBX proposes SRTP, but does not get the SRTP key back. Fixed the issue, but don't know the cause how I fixed: removed supported codec from Domain --> Settings --> Port Save Reboot pbx insert supported codec in Domain --> settings --> Port Save Reboot pbx I honestly don't know if all the steps were required. But at least I have the pbx working again. Quote Link to comment Share on other sites More sharing options...
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