Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi very thanks, that's work bye olivier 2011/4/3 Mark Murawski markm-li...@intellasoft.net: I gave you the syntax in ael format, if you want to use extensions.conf you'll have to use the syntax that's applicable, which is: [start-audio] exten = s,1,Playback(silence/1) On 04/03/11 14:14, Olivier CALVANO wrote: Hi Mark Thanks for your answer, but i am new in asterisk ;=) the context start-audio ... i put it into the extension.conf ? because i have a error: [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config: ==!!== Unknown directive: s at line 135 -- IGNORING!!! thanks for your help olivier 2011/4/3 Mark Murawskimarkm-li...@intellasoft.net: In that situation, I've had to do a pickup macro that kind of primes the audio. Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s = { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel (SIP/MyOperator-) before bridging the audio. On 04/03/11 12:01, Olivier CALVANO wrote: Hi i use this into my extension : exten = _00339,1,Set(foo=${SIP_HEADER(To)}) exten = _00339,2,Set(cut1=${CUT(foo,:,2)}) exten = _00339,3,Set(CLI=${CUT(cut1,,1)}) exten = _00339,4,Set(toexten=${CUT(CLI,@,1)}) exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten = _00339,6,AGI(Ddi-Network.agi,${toexten}) exten = _00339,7,Set(CALLERPRES()=prohib_not_screened) exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten = _00339,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xx secret=x When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the 00339xxx.., the call are correct, asterisk call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- -- _ -- 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
[asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi i use this into my extension : exten = _00339,1,Set(foo=${SIP_HEADER(To)}) exten = _00339,2,Set(cut1=${CUT(foo,:,2)}) exten = _00339,3,Set(CLI=${CUT(cut1,,1)}) exten = _00339,4,Set(toexten=${CUT(CLI,@,1)}) exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten = _00339,6,AGI(Ddi-Network.agi,${toexten}) exten = _00339,7,Set(CALLERPRES()=prohib_not_screened) exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten = _00339,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xx secret=x When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the 00339xxx.., the call are correct, asterisk call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- _ -- 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
Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
In that situation, I've had to do a pickup macro that kind of primes the audio. Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s = { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel (SIP/MyOperator-) before bridging the audio. On 04/03/11 12:01, Olivier CALVANO wrote: Hi i use this into my extension : exten = _00339,1,Set(foo=${SIP_HEADER(To)}) exten = _00339,2,Set(cut1=${CUT(foo,:,2)}) exten = _00339,3,Set(CLI=${CUT(cut1,,1)}) exten = _00339,4,Set(toexten=${CUT(CLI,@,1)}) exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten = _00339,6,AGI(Ddi-Network.agi,${toexten}) exten = _00339,7,Set(CALLERPRES()=prohib_not_screened) exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten = _00339,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xx secret=x When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the 00339xxx.., the call are correct, asterisk call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- _ -- 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
Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
Hi Mark Thanks for your answer, but i am new in asterisk ;=) the context start-audio ... i put it into the extension.conf ? because i have a error: [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config: ==!!== Unknown directive: s at line 135 -- IGNORING!!! thanks for your help olivier 2011/4/3 Mark Murawski markm-li...@intellasoft.net: In that situation, I've had to do a pickup macro that kind of primes the audio. Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s = { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel (SIP/MyOperator-) before bridging the audio. On 04/03/11 12:01, Olivier CALVANO wrote: Hi i use this into my extension : exten = _00339,1,Set(foo=${SIP_HEADER(To)}) exten = _00339,2,Set(cut1=${CUT(foo,:,2)}) exten = _00339,3,Set(CLI=${CUT(cut1,,1)}) exten = _00339,4,Set(toexten=${CUT(CLI,@,1)}) exten = _00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten = _00339,6,AGI(Ddi-Network.agi,${toexten}) exten = _00339,7,Set(CALLERPRES()=prohib_not_screened) exten = _00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten = _00339,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xx secret=x When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the 00339xxx.., the call are correct, asterisk call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- _ -- 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
Re: [asterisk-users] Asterisk 1.6 = No sound/voice when i redirect the call
I gave you the syntax in ael format, if you want to use extensions.conf you'll have to use the syntax that's applicable, which is: [start-audio] exten = s,1,Playback(silence/1) On 04/03/11 14:14, Olivier CALVANO wrote: Hi Mark Thanks for your answer, but i am new in asterisk ;=) the context start-audio ... i put it into the extension.conf ? because i have a error: [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 134 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No '=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf [Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config: ==!!== Unknown directive: s at line 135 -- IGNORING!!! thanks for your help olivier 2011/4/3 Mark Murawskimarkm-li...@intellasoft.net: In that situation, I've had to do a pickup macro that kind of primes the audio. Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s = { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel (SIP/MyOperator-) before bridging the audio. On 04/03/11 12:01, Olivier CALVANO wrote: Hi i use this into my extension : exten =_00339,1,Set(foo=${SIP_HEADER(To)}) exten =_00339,2,Set(cut1=${CUT(foo,:,2)}) exten =_00339,3,Set(CLI=${CUT(cut1,,1)}) exten =_00339,4,Set(toexten=${CUT(CLI,@,1)}) exten =_00339,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) exten =_00339,6,AGI(Ddi-Network.agi,${toexten}) exten =_00339,7,Set(CALLERPRES()=prohib_not_screened) exten =_00339,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) exten =_00339,9,Hangup and i have in sip.conf: [MyOperator] type=peer host=host-of-my-operator qualify=yes dtmf=rfc2833 nat=no canreinvite=no canredirect=yes insecure=port,invite dtmfmode=rfc2833 disallow=all allow=g729 allow=alaw allow=g723 defaultuser=0033xx secret=x When i call directly from [MyOperator], no probleme i have sound/Voice but when a customer call to the 00339xxx.., the call are correct, asterisk call to my standard SIP/MyOperator/${NUMAPPEL} but no sound/voice (i receive the call without problems, only sound off) anyone have a idea of this problems ? bye Olivier -- -- _ -- 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