[asterisk-users] Set Call type in dial plan

Fri Jan 6 05:58:39 CST 2012

I already tried what u posted .... didnt work ....
but thanx for the reply :)
________________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Sammy Govind [govoiper at gmail.com]
Sent: Wednesday, January 04, 2012 11:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call type in dial plan

Hi,
Sorry for late reply. Hope you've already found out something about it.

What version of asterisk you are using, that function for choosing inbound/outbound call leg codecs is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
https://issues.asterisk.org/view.php?id=13243

Regards,
Sammy

On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib <fkhasib at iconnecths.com<mailto:fkhasib at iconnecths.com>> wrote:
thats excatly what I want, can u plz give me the command, I want to choose only ulow
________________________________________
From: asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com> [asterisk-users-bounces at lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of Sammy Govind [govoiper at gmail.com<mailto:govoiper at 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 <fkhasib at iconnecths.com<mailto:fkhasib at iconnecths.com><mailto:fkhasib at iconnecths.com<mailto:fkhasib at iconnecths.com>>> wrote:
this is what my SIP Invite message when I make Video call

INVITE sip:6500 at 192.168.21.102<mailto:sip%3A6500 at 192.168.21.102><mailto:sip%3A6500 at 192.168.21.102<mailto:sip%253A6500 at 192.168.21.102>> SIP/2.0
Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
From: <sip:6097 at 192.168.21.102<mailto:sip%3A6097 at 192.168.21.102><mailto:sip%3A6097 at 192.168.21.102<mailto:sip%253A6097 at 192.168.21.102>>>;tag=1857098215
To: <sip:6500 at 192.168.21.102<mailto:sip%3A6500 at 192.168.21.102><mailto:sip%3A6500 at 192.168.21.102<mailto:sip%253A6500 at 192.168.21.102>>>
Contact: <sip:6097 at 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