[asterisk-users] Set Call type in dial plan

[Faraj Khasib]

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 ?