Greetings List.
I Have a small test server and i'm facing a small issue.
i have setup two SIP PEERS and they are able to do Video calls.
now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the
codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW
CHANNELS shows only the Codec i set in the dialplan.
is it possible to avoid this problem?
Asterisk version
1.8.11.0
SIP.CONF
=======
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
[TK1000]
type=friend
secret=0jCiOdT81P
videosupport=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
context=DER-TEST
canreinvite=yes
disallow=all
allow=ulaw,alaw,gsm,h263,h263p
EXTENSIONS.CONF
[DER-TEST]
;exten => _.,1,NoCDR()
exten => _.,1,Set(SIP_CODEC=alaw)
exten => _.,2,Set(SIP_CODEC_OUTBOUND=gsm)
;exten => _.,2,Set(SIP_CODEC_INBOUND=gsm)
exten => _.,n,DIAL(SIP/TK${EXTEN})
exten => h,1,Hangup()
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
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