1.6 and 1.8 ... I tried changing stuff on both .... when I make audio call from my client which supports both audio and video its sent to the other client as video call .....I tried settings the SIP_CODED_INBOUND and outbound also ... but no luck ________________________________________ From: [email protected] [[email protected]] On Behalf Of Eric Wieling [[email protected]] Sent: Wednesday, January 04, 2012 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf
Providing which version of Asterisk you are using might be helpful. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf anyhelp guys? I tried a lot of stuff but it doesnt work .... the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? ________________________________________ From: [email protected] [[email protected]] On Behalf Of Faraj Khasib [[email protected]] Sent: Wednesday, January 04, 2012 5:53 AM To: [email protected] Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed .... exten=6500,6,Queue(${EXTEN}) can any body help me with that? -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
