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(${EXTEN}) can any body help me with that?
anyhelp guys? I tried a lot of stuff but it doesnt work .... the Codec for audio call only cannt be set...how I can set the call type video/audio at dail plan? ________________________________________ From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Set Call Codec in extension.conf Hi All, I am trying to set call codec at extension.conf but it doesnt work ... its like my command doesnt change anything exten=6500,1,Answer exten=6500,2,Playback(welcome) exten=6500,3,SIPAddHeader(email:${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
Providing which version of Asterisk you are using might be helpful. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com 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
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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com 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
1.6 does not support setting the outbound codec. 1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com 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
I tried also in asterisk 1.8 setting outbound variable .... but didnt work also .... https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried .... but still I get a video call ________________________________________ From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec. 1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com 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 -- _____________________________________________________________________ -- 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
My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf I tried also in asterisk 1.8 setting outbound variable .... but didnt work also .... https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables check the above ... I changed it and tried .... but still I get a video call ________________________________________ From: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] Sent: Wednesday, January 04, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 1.6 does not support setting the outbound codec. 1.8 uses different variables to set the outbound codec. See UGRADE.txt in the Asterisk source for the 1.8 information,. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling [EWieling at nyigc.com] 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: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] 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: asterisk-users-bounces at lists.digium.com [asterisk-users-bounces at lists.digium.com] On Behalf Of Faraj Khasib [fkhasib at iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users at lists.digium.com 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 -- _____________________________________________________________________ -- 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