Hi, By the sip trace is very difficult to tell because the SIP messages are fine. Try to enable all codec, and if possible copy and paste your asterisk sip configuration for this peer.
Enviado do meu telefone Android usando o Symantec TouchDown (www.symantec.com) -----Original Message----- From: Technical Support [[email protected]] Received: sexta-feira, 21 ago 2015, 19:46 To: [email protected] [[email protected]] Subject: [asterisk-users] Incoming calls get 488 error I got a new SNOM M65 which works fine for outgoing calls, but incoming calls never ring at the handset. I captured the SIP traffic and see that my M65 is replying with an "488 not acceptable here". From what I read this is usually codec related but both asterisk and the M65 are set for ulaw as first choice. I have a SIP trace below. Can someone suggest why the 488 is being generated? ----------------------------------- Received from udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (1198 bytes) INVITE sip:[email protected];line=14994 SIP/2.0 Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" <sip:[email protected]>;tag=as7b616c8d To: <sip:[email protected];line=14994> Contact: <sip:[email protected]:5060> Call-ID: [email protected]:5060 CSeq: 102 INVITE User-Agent: FPBX-2.11.0(11.10.2) Date: Fri, 21 Aug 2015 22:37:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 606 v=0 o=root 1678280845 1678280845 IN IP4 192.168.253.4 s=Asterisk PBX 11.10.2 c=IN IP4 192.168.253.4 b=CT:384 t=0 0 m=audio 18090 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=video 12226 RTP/AVP 99 98 34 31 a=rtpmap:99 H264/90000 a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 a=rtpmap:98 H263-1998/90000 a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:34 H263/90000 a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:31 H261/90000 a=sendrecv Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (280 bytes) SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a From: "test user" <sip:[email protected]>;tag=as7b616c8d To: <sip:[email protected];line=14994> Call-ID: [email protected]:5060 CSeq: 102 INVITE Content-Length: 0 Sent to udp:192.168.253.4:5060 at 21/08/2015 18:37:00 (441 bytes) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a Max-Forwards: 70 From: "test user" <sip:[email protected]>;tag=as7b616c8d To: <sip:[email protected];line=14994>;tag=ld65q Call-ID: [email protected]:5060 CSeq: 102 INVITE Contact: <sip:[email protected];line=14994> User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 00000; HW=255) 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 -- _____________________________________________________________________ -- 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
