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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Of course you are disabling the video maybe also include the video protocols in
the sip_codec
-----Original Message-----
From: Tarek Sawah <tareksawah at hotmail.com>
Sender: asterisk-users-bounces at lists.digium.com
Date: Sat, 19 May 2012 17:33:57
To: Asterisk Users<asterisk-users at lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Subject: [asterisk-users] SET SIP_CODEC and Video issues
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Thank you, Any idea how? Need to be able to control the codecs in use through soem bandwidth tests. so i need to be able to set the SIP_CODEC and still be able to do Video. any suggestions?> To: asterisk-users at lists.digium.com > From: isrlgb at gmail.com > Date: Sat, 19 May 2012 20:38:22 +0000 > Subject: Re: [asterisk-users] SET SIP_CODEC and Video issues > > Of course you are disabling the video maybe also include the video protocols in the sip_codec > -----Original Message----- > From: Tarek Sawah <tareksawah at hotmail.com> > Sender: asterisk-users-bounces at lists.digium.com > Date: Sat, 19 May 2012 17:33:57 > To: Asterisk Users<asterisk-users at lists.digium.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > Subject: [asterisk-users] SET SIP_CODEC and Video issues > > -- > _____________________________________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120519/c0dc17c2/attachment.htm>