[asterisk-users] Set Call type in dial plan

Sammy Govind govoiper at gmail.com
Tue Jan 3 03:26:58 CST 2012


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> wrote:

> this is what my SIP Invite message when I make Video call
>
> INVITE sip:6500 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>;tag=1857098215
> To: <sip:6500 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 ?
> --
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