thats excatly what I want, can u plz give me the command, I want to choose only ulow ________________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [govoi...@gmail.com] Sent: Tuesday, January 03, 2012 3:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call type in dial plan
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<mailto:fkha...@iconnecths.com>> wrote: this is what my SIP Invite message when I make Video call INVITE sip:6500@192.168.21.102<mailto:sip%3A6500@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<mailto:sip%3A6097@192.168.21.102>>;tag=1857098215 To: <sip:6500@192.168.21.102<mailto:sip%3A6500@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
