thanks for your response the call is going to IAX(1000), i have i DID Number when the customer call this number 0520XXXXXX the call is goint to agent IAX. in my dialplan i have exten => 223,1,MixMonitor(blah.wav) exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 223,n,Dial(SIP/223)
and in extensions.conf i have exten => 223,1,MixMonitor(sip_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten => 223,n,Dial(SIP/${EXTEN},,KkTt) exten => 223,n,Hangup(); thanks and regards 2011/6/16 Leif Madsen <leif.mad...@asteriskdocs.org> > On 16/06/11 07:36 AM, salaheddine elharit wrote: > >> hello list, >> >> i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in >> order to record the conversation >> >> but when i receive an inbound call from customer in IAX(1000) and i want >> to transfer the call to other phone SIP(223) >> the conversation between customer and IAX is recorded but the >> conversation between customer and sip is not recorded >> > > Is the call coming from IAX(1000) or going to IAX(1000)? Note that when you > transfer calls around and are using MixMonitor() (or any recording) that you > have to think of the recording as being associated with the incoming > channel, and the recording should essentially follow it around. > > So if you have a call coming in like this: > > ITSP --> Asterisk --> Dialplan --> Mixmonitor --> Dial(SIP/1000) > > Then the MixMonitor() is associated with the channel created when the call > came in from the ITSP. If that channel is then transferred, the recording > should follow it around. > > Can you elaborate a bit more on the call flow and show the console output? > > -- > Leif Madsen > http://www.oreilly.com/catalog/asterisk > > -- > _____________________________________________________________________ > -- 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