Hi very thanks, that's work
bye olivier 2011/4/3 Mark Murawski <[email protected]>: > 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 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
