Olivier CALVANO
2011-Apr-03 16:01 UTC
[asterisk-users] Asterisk 1.6 => No sound/voice when i redirect the call
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
Mark Murawski
2011-Apr-03 16:25 UTC
[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/${NUMAPPEL},180,rtU(start-audio)) context start-audio { s => { Playback(silence/1); } } The above might help... What it does is plays an audio track on the callee's channel (SIP/MyOperator-xxxx) before bridging the audio. On 04/03/11 12:01, Olivier CALVANO wrote:> Hi > > i use this into my extension : > > > exten => _00339xxxxxxxx,1,Set(foo=${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