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Faraj Khasib

2012-Jan-02 09:07 UTC

[asterisk-users] Set Call type in dial plan

Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib

Faraj Khasib

2012-Jan-02 20:29 UTC

head link

[asterisk-users] Set Call type in dial plan

Please help, I have tried many things I cannt make it work, when I make an audio
call it is converted by asterisk to video call request, Please how to set the
call type at extensions.conf, I tried setting the codec manually but didnt work
also... any help .. any suggest will be great
Thanx
________________________________________
From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at
lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com]
Sent: Monday, January 02, 2012 3:07 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Set Call type in dial plan

Hi All,
How to set C all type (Audio/Video) in dial plan?
Regards
Faraj Khasib
--
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Sammy Govind

2012-Jan-03 09:26 UTC

head link

[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>
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 ?
> --
> _____________________________________________________________________
> -- 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
>-------------- next part --------------
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Faraj Khasib

2012-Jan-03 09:31 UTC

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[asterisk-users] Set Call type in dial plan

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 [asterisk-users-bounces at
lists.digium.com] On Behalf Of Sammy Govind [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>> 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>
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>>;tag=1857098215
To: <sip:6500 at 192.168.21.102<mailto:sip%3A6500 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

asterisk users - Jan 2012 - Set Call type in dial plan

[asterisk-users] Set Call type in dial plan

[asterisk-users] Set Call type in dial plan

[asterisk-users] Set Call type in dial plan

[asterisk-users] Set Call type in dial plan