We have an issue with a customer where when calls are sent to one of their offices as soon as the call is answered the call fails. We are performing remote bridging and switching the audio from the server which initiated the call to our switch which is on the same network. After the call is answered we switch the audio which is accepted fine but we then send the following packet and get a SIP/488 response from the far end. This packet seems to be updating the version for the o= session id which is fair enough. Its sending the c= connection data but not the m=audio line which appears to be what the remote end is complaining about.
Can anyone with a bit more knowledge about the SDP standard tell me if what asterisk is doing is correct? Or if it must be a bug with our customers equipment? Thanks Gareth U 2013/09/27 11:04:55.352854 88.x.x.25:5060 -> 213.x.x.24:5060 INVITE sip:[email protected]:54900 SIP/2.0. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713. Route:<sip:213.x.x.24;lr=on;ftag=as691af817;did=ecd.c2dc96e6>. Max-Forwards: 70. From:<sip:[email protected]>;tag=as691af817. To:<sip:[email protected]>;tag=ee7a6c7cad57f096i1. Contact:<sip:[email protected]:5060>. Call-ID: [email protected]:5060. CSeq: 104 INVITE. User-Agent: Asterisk PBX 11.2-cert2. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH. Supported: replaces, timer. Content-Type: application/sdp. Content-Length: 110. . v=0. o=root 716216031 716216033 IN IP4 88.x.x.35. s=Asterisk PBX 11.2-cert2. c=IN IP4 88.x.x.35. t=0 0. # U 2013/09/27 11:04:55.365458 213.x.x.24:5060 -> 88.x.x.25:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP 88.x.x.25:5060;branch=z9hG4bK62215713;rport=5060. From:<sip:[email protected]>;tag=as691af817. To:<sip:[email protected]>;tag=ee7a6c7cad57f096i1. Call-ID: [email protected]:5060. CSeq: 104 INVITE. Server: OpenSIPS (1.5.3-notls (x86_64/linux)). Content-Length: 0. . # U 2013/09/27 11:04:55.431674 213.x.x.24:5060 -> 88.x.x.25:5060 SIP/2.0 488 Not Acceptable Here. To:<sip:[email protected]>;tag=ee7a6c7cad57f096i1. From:<sip:[email protected]>;tag=as691af817. Call-ID: [email protected]:5060. CSeq: 104 INVITE. Via: SIP/2.0/UDP 88.x.x.25:5060;rport=5060;received=88.x.x.25;branch=z9hG4bK62215713. Record-Route:<sip:213.x.x.24;lr=on;ftag=as691af817>. Contact: "freespeech"<sip:[email protected]:54900>. Warning: 304 spa "Media type not available". Server: Cisco/SPA303-7.5.4. Content-Length: 0. -- _____________________________________________________________________ -- 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
