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 <[email protected]>: > In that situation, I've had to do a pickup macro that kind of "primes" the > audio. > > Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio)) > > context start-audio { > s => { > Playback(silence/1); > } > } > > The above might help... What it does is plays an audio track on the callee's > channel (SIP/MyOperator-xxxx) before bridging the audio. > > > On 04/03/11 12:01, Olivier CALVANO wrote: >> >> Hi >> >> i use this into my extension : >> >> >> exten => _00339xxxxxxxx,1,Set(foo=${SIP_HEADER(To)}) >> exten => _00339xxxxxxxx,2,Set(cut1=${CUT(foo,:,2)}) >> exten => _00339xxxxxxxx,3,Set(CLI=${CUT(cut1,>,1)}) >> exten => _00339xxxxxxxx,4,Set(toexten=${CUT(CLI,@,1)}) >> exten => _00339xxxxxxxx,5,Noop(ORIGINAL NUMBER : [ ${toexten} ]) >> exten => _00339xxxxxxxx,6,AGI(Ddi-Network.agi,${toexten}) >> exten => _00339xxxxxxxx,7,Set(CALLERPRES()=prohib_not_screened) >> exten => _00339xxxxxxxx,8,Dial(SIP/MyOperator/${NUMAPPEL},180,rt) >> exten => _00339xxxxxxxx,9,Hangup >> >> >> and i have in sip.conf: >> >> >> [MyOperator] >> type=peer >> host=host-of-my-operator >> qualify=yes >> dtmf=rfc2833 >> nat=no >> canreinvite=no >> canredirect=yes >> insecure=port,invite >> dtmfmode=rfc2833 >> disallow=all >> allow=g729 >> allow=alaw >> allow=g723 >> defaultuser=0033xxxxxx >> secret=xxxxx >> >> >> >> When i call directly from [MyOperator], no probleme i have sound/Voice >> but when a customer call to the "00339xxx..", the call are correct, >> asterisk >> call to my standard "SIP/MyOperator/${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
