Hi, For such call you just need to select the outbound codec before the dial() app.
choose the audio-only codecs and thus no video codec strings will be exchanged in that call. -- Regards, Sammy On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib <fkha...@iconnecths.com> wrote: > this is what my SIP Invite message when I make Video call > > INVITE sip:6500@192.168.21.102 SIP/2.0 > Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport > From: <sip:6097@192.168.21.102>;tag=1857098215 > To: <sip:6500@192.168.21.102> > Contact: <sip:6097@192.168.21.193:52933 > ;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" > Call-ID: b9453704-d76a-b8ce-3247-c999abff7395 > CSeq: 324677463 INVITE > Content-Type: application/sdp > Content-Length: 588 > Max-Forwards: 70 > Route: <sip:192.168.21.102:5060;lr;transport=udp> > Accept-Contact: > *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel" > P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel > Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, > REFER > Privacy: none > P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000 > User-Agent: Medcor > Supported: 100rel > > v=0 > o=doubango 1983 678901 IN IP4 192.168.21.193 > s=- > c=IN IP4 192.168.21.193 > t=0 0 > m=audio 36372 RTP/AVP 8 0 9 101 > a=ptime:20 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:9 G722/8000/1 > a=rtpmap:101 telephone-event/8000/1 > a=fmtp:101 0-15 > m=video 59296 RTP/AVP 125 106 121 103 > a=rtpmap:125 VP8/90000 > a=fmtp:125 CIF=2;QCIF=2;SQCIF=2 > a=rtpmap:106 H264/90000 > a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; > max-mbps=11880 > a=rtpmap:121 MP4V-ES/90000 > a=fmtp:121 profile-level-id=3 > a=rtpmap:103 H263-1998/90000 > a=fmtp:103 CIF=2;QCIF=2;SQCIF=2 > > when I make Audio call requests I dont have the video part .... but at > receiver since two clients can make video call they have Asterisks adds the > Video Part in request sent to receiver,I dont want that part added , how I > can delete it ? > -- > _____________________________________________________________________ > -- 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