<div>Hi,</div><div><br></div>For such call you just need to select the outbound codec before the dial() app.<div><br></div><div>choose the audio-only codecs and thus no video codec strings will be exchanged in that call.</div>
<div><br></div><div>--</div><div>Regards,</div><div>Sammy<br><br><div class="gmail_quote">On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib <span dir="ltr"><<a href="mailto:fkhasib@iconnecths.com">fkhasib@iconnecths.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">this is what my SIP Invite message when I make Video call<br>
<br>
INVITE <a href="mailto:sip%3A6500@192.168.21.102">sip:6500@192.168.21.102</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport<br>
From: <<a href="mailto:sip%3A6097@192.168.21.102">sip:6097@192.168.21.102</a>>;tag=1857098215<br>
To: <<a href="mailto:sip%3A6500@192.168.21.102">sip:6500@192.168.21.102</a>><br>
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"<br>
Call-ID: b9453704-d76a-b8ce-3247-c999abff7395<br>
CSeq: 324677463 INVITE<br>
Content-Type: application/sdp<br>
Content-Length: 588<br>
Max-Forwards: 70<br>
Route: <sip:192.168.21.102:5060;lr;transport=udp><br>
Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"<br>
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel<br>
Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER<br>
Privacy: none<br>
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000<br>
User-Agent: Medcor<br>
Supported: 100rel<br>
<br>
v=0<br>
o=doubango 1983 678901 IN IP4 192.168.21.193<br>
s=-<br>
c=IN IP4 192.168.21.193<br>
t=0 0<br>
m=audio 36372 RTP/AVP 8 0 9 101<br>
a=ptime:20<br>
a=rtpmap:8 PCMA/8000/1<br>
a=rtpmap:0 PCMU/8000/1<br>
a=rtpmap:9 G722/8000/1<br>
a=rtpmap:101 telephone-event/8000/1<br>
a=fmtp:101 0-15<br>
m=video 59296 RTP/AVP 125 106 121 103<br>
a=rtpmap:125 VP8/90000<br>
a=fmtp:125 CIF=2;QCIF=2;SQCIF=2<br>
a=rtpmap:106 H264/90000<br>
a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880<br>
a=rtpmap:121 MP4V-ES/90000<br>
a=fmtp:121 profile-level-id=3<br>
a=rtpmap:103 H263-1998/90000<br>
a=fmtp:103 CIF=2;QCIF=2;SQCIF=2<br>
<br>
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 ?<br>
<div class="HOEnZb"><div class="h5">--<br>
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</div></div></blockquote></div><br></div>