Re: [asterisk-users] SET SIP_CODEC and Video issues
On 05/19/2012 03:52 PM, Tarek Sawah wrote: 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? Unfortunately you can't do what you want using SIP_CODEC; if you set that variable, the formats (both audio and video) allowed on the channel are reset to whatever you specify, and that variable can only hold one format name. It seems odd though that you want to change audio codecs based on bandwidth tests, but still allow video. The video stream is going to consume vastly more bandwidth than the audio stream. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] SET SIP_CODEC and Video issues
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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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 tareksa...@hotmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Sat, 19 May 2012 17:33:57 To: Asterisk Usersasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@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
