Hi Matt Thanks for your response. I have tried with two GXV3175 with same result. Let me dig deep on this to find out the route cause
Sam Matthew Jordan wrote: > On Thu, Jun 13, 2013 at 12:04 PM, <resea...@businesstz.com> wrote: > >> Hi there >> >> I have asterisk 10.11.1 which seems to have problem negotiating codec. >> >> Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p >> and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, >> h263p. I have tried similar combination of codecs and SIP phone but when >> making a video call, it report "Peer doesn't provide video". It seems >> Asterisk is failing to set capability correct. Both codecs are enabled >> on >> the SIP Phones >> >> > <snip> > > The 200 OK response from the called XLite phone is declining the video > stream: > > <--- SIP read from UDP:10.10.10.129:48464 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 > Contact: <sip:1003@10.10.10.129:48464> > To: "SAM"<sip:1003@10.10.10.105>;tag=0c90cc0c > From: <sip:1004@10.10.10.105>;tag=as24914503 > Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA > CSeq: 102 INVITE > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Type: application/sdp > Supported: replaces, eventlist > User-Agent: X-Lite release 4.5.2 stamp 70142 > Content-Length: 234 > > v=0 > o=- 13015615910543193 2 IN IP4 10.10.10.129 > s=X-Lite 4 release 4.5.2 stamp 70142 > c=IN IP4 10.10.10.129 > t=0 0 > m=audio 53188 RTP/AVP 8 101 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=sendrecv > m=video 0 RTP/AVP 115 > <-------------> > --- (12 headers 10 lines) --- > Found RTP audio format 8 > Found RTP audio format 101 > Found audio description format telephone-event for ID 101 > Capabilities: us - (alaw|h263p), peer - > audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) > > Note that the port for the video stream is set to 0. > > Asterisk is doing the correct thing: it notes that the answer to its offer > declined the video stream, so it disables video for the call between the > two endpoints. > > Matt > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > -- > _____________________________________________________________________ > -- 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