On 8/21/15 6:45 PM, Technical Support wrote:> 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.
Looks like the SNOM does not accept the video call. Maybe you should
look into why the Asterisk is trying to use video in the first
place.>
> 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:290006 at 192.168.253.20;line=14994 SIP/2.0
> Via: SIP/2.0/UDP 192.168.253.4:5060;branch=z9hG4bK4990095a
> Max-Forwards: 70
> From: "test user" <sip:230 at 192.168.253.4>;tag=as7b616c8d
> To: <sip:290006 at 192.168.253.20;line=14994>
> Contact: <sip:230 at 192.168.253.4:5060>
> Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
> 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:230 at 192.168.253.4>;tag=as7b616c8d
> To: <sip:290006 at 192.168.253.20;line=14994>
> Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
> 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:230 at 192.168.253.4>;tag=as7b616c8d
> To: <sip:290006 at 192.168.253.20;line=14994>;tag=ld65q
> Call-ID: 36334383058109cd2325341a0f18ac79 at 192.168.253.4:5060
> CSeq: 102 INVITE
> Contact: <sip:290006 at 192.168.253.20;line=14994>
> User-Agent: snomM700/03.24.0007 (MAC=0004136103FB; SER= 00000; HW=255)
> Content-Length: 0
>
>
>
--
Technical Support
http://www.cellroute.net