You are fighting a losing battle - you can't control the other end Ignoring ${SIP_CODEC} variable because it is not shared by both ends.
You can probably do a SIP SET DEBUG ON and see what codecs are available on the other end. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf -- Executing [6500@DLPN_DialPlan1:1] Set("SIP/6000-00000000", "SIP_CODEC=gsm ") in new stack -- Executing [6500@DLPN_DialPlan1:2] Set("SIP/6000-00000000", "SIP_CODEC_INB OUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:3] Set("SIP/6000-00000000", "SIP_CODEC_OUT BOUND=gsm") in new stack -- Executing [6500@DLPN_DialPlan1:4] Answer("SIP/6000-00000000", "") in new stack [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan 4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not shared by both ends. -- Executing [6500@DLPN_DialPlan1:5] Playback("SIP/6000-00000000", "welcome" ) in new stack [Jan 4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do es not exist in any format [Jan 4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No such file or directory [Jan 4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil e failed on SIP/6000-00000000 for welcome -- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader("SIP/6000-00000000", "emai l:fkha...@iconnecths.com") in new stack -- Executing [6500@DLPN_DialPlan1:7] MixMonitor("SIP/6000-00000000", "2012-0 1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b") in new stack -- Executing [6500@DLPN_DialPlan1:8] Queue("SIP/6000-00000000", "6500") in n ew stack -- Started music on hold, class 'default', on SIP/6000-00000000 == Begin MixMonitor Recording SIP/6000-00000000 ________________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf CLI output from call? -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf didnt work also ....:( ________________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is cumbersome; 1-n-n-n-n-n is more practical). Like this exten=6500,1,Answer exten=6500,n,Playback(welcome) exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed .... exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) or exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed .... exten=6500,n,Answer exten=6500,n,Playback(welcome) exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)}) exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H: %M:%S_%A)}_${SIP_HEADER(email)}.wav,b) exten=6500,n,Queue(${EXTEN}) -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib Sent: Wednesday, January 04, 2012 11:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Set Call Codec in extension.conf how .... can u give me a command?!.. ________________________________________ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas [da...@debsinc.com] Sent: Wednesday, January 04, 2012 11:29 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Set Call Codec in extension.conf My guess is that you should set the codec either before SIPADDHEADER or before ANSWER. -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling [ewiel...@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-boun...@lists.digium.com [mailto:asterisk-users-boun...@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-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib [fkha...@iconnecths.com] Sent: Wednesday, January 04, 2012 5:53 AM To: asterisk-users@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? 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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? 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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