<div>Hi,</div><div>Sorry for late reply. Hope you've already found out something about it.</div><div><br></div>What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk. <div>
See these pages:</div><div><a href="http://www.voip-info.org/wiki/view/Asterisk+variables">http://www.voip-info.org/wiki/view/Asterisk+variables</a> </div><div><a href="https://issues.asterisk.org/view.php?id=13243">https://issues.asterisk.org/view.php?id=13243</a> </div>
<div><br></div><div>Regards,</div><div>Sammy</div><div><br><br><div class="gmail_quote">On Tue, Jan 3, 2012 at 2:31 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">thats excatly what I want, can u plz give me the command, I want to choose only ulow<br>
________________________________________<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Sammy Govind [<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>]<br>
Sent: Tuesday, January 03, 2012 3:26 AM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: Re: [asterisk-users] Set Call type in dial plan<br>
<div class="im"><br>
Hi,<br>
<br>
For such call you just need to select the outbound codec before the dial() app.<br>
<br>
choose the audio-only codecs and thus no video codec strings will be exchanged in that call.<br>
<br>
--<br>
Regards,<br>
Sammy<br>
<br>
</div><div class="im">On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib <<a href="mailto:fkhasib@iconnecths.com">fkhasib@iconnecths.com</a><mailto:<a href="mailto:fkhasib@iconnecths.com">fkhasib@iconnecths.com</a>>> wrote:<br>
this is what my SIP Invite message when I make Video call<br>
<br>
</div>INVITE <a href="mailto:sip%3A6500@192.168.21.102">sip:6500@192.168.21.102</a><mailto:<a href="mailto:sip%253A6500@192.168.21.102">sip%3A6500@192.168.21.102</a>> SIP/2.0<br>
<div class="im">Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport<br>
</div>From: <<a href="mailto:sip%3A6097@192.168.21.102">sip:6097@192.168.21.102</a><mailto:<a href="mailto:sip%253A6097@192.168.21.102">sip%3A6097@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><mailto:<a href="mailto:sip%253A6500@192.168.21.102">sip%3A6500@192.168.21.102</a>>><br>
<div class="HOEnZb"><div class="h5">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>
--<br>
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<br>
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</div></div></blockquote></div><br></div>