hi Danny thank you for your response i switched the MixMonitor and i still have the same result
any help please 2011/6/16 Danny Nicholas <[email protected]> > Try switching the Set and MixMonitor commands so the AUDIOHOOK_INHERIT will > be in effect when Mixmonitor starts > > exten => 223,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) > exten => 223,n,MixMonitor(blah.wav) > exten => 223,n,Dial(SIP/223) > > *From:* [email protected] [mailto: > [email protected]] *On Behalf Of *salaheddine > elharit > *Sent:* Thursday, June 16, 2011 9:13 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] MixMonitor > > > > i have asterisk 1.4 and also i have aheeva applicaton also installed in my > server > in the consol this call may be monitored or recorded > > best regrads > > > > 2011/6/16 Leif Madsen <[email protected]> > > On 16/06/11 09:20 AM, salaheddine elharit wrote: > > 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(); > > > > OK, well nothing looks obviously wrong there from what I can tell. > > What is your console output doing though when you do the transfer? Are you > using Asterisk transfers? What version of Asterisk are you using? > > Leif. > > > > -- > 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 >
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