Hello List, i need to be able to record the call transferred from iax extension to sip extension when i call the sip extension from the IAX extension i can record the call without any issue but when i receive a call from customer in IAX and i transfer this call to SIP client the conversation between customer and IAX client is recorded but the conversation between customer and sip extension is not recorded any help please thanks and regards extensions.conf exten => 223,1,Answer() exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0)) exten => 223,3,Dial(SIP/223) exten => 223,4,Hangup() -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110507/56430afa/attachment.htm>
On Sat, 7 May 2011, salaheddine elharit wrote:> Hello List, > > i need to be able to record the call transferred from iax extension to sip > extension > > when i call the sip extension from the IAX extension i can record the call > without any issue > > but when i receive a call from customer in IAX and i transfer this call to > SIP client > > the conversation between customer and IAX client is recorded but the > conversation between customer and sip extension is not recorded > > any help please > > thanks and regardsIt's it's a transfer, then try this:> > extensions.conf > exten => 223,1,Answer() > exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))Add: exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)> exten => 223,3,Dial(SIP/223) > exten => 223,4,Hangup()And do use 'n' in your dialplan, rather than numbers. It makes adding/deleting lines easy. Gordon
On Sat, 2011-05-07 at 17:04 +0000, salaheddine elharit wrote:> Hello List, > > i need to be able to record the call transferred from iax extension > to sip extension > > when i call the sip extension from the IAX extension i can record the > call without any issue > > but when i receive a call from customer in IAX and i transfer this > call to SIP client > > the conversation between customer and IAX client is recorded but the > conversation between customer and sip extension is not recorded > > any help please > > thanks and regards > > extensions.conf > exten => 223,1,Answer() > exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0)) > exten => 223,3,Dial(SIP/223) > exten => 223,4,Hangup() > -- > __Look into the audiohook_inherit function https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez Prats Director de Tecnolog?a +52-55-91169161 ext 2001 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/d0f4b842/attachment.pgp>
ok thanks i will test this solution 2011/5/10 Carlos Chavez <cursor at telecomabmex.com>> On Sat, 2011-05-07 at 17:04 +0000, salaheddine elharit wrote: > > Hello List, > > > > i need to be able to record the call transferred from iax extension > > to sip extension > > > > when i call the sip extension from the IAX extension i can record the > > call without any issue > > > > but when i receive a call from customer in IAX and i transfer this > > call to SIP client > > > > the conversation between customer and IAX client is recorded but the > > conversation between customer and sip extension is not recorded > > > > any help please > > > > thanks and regards > > > > extensions.conf > > exten => 223,1,Answer() > > exten => 223,2,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0)) > > exten => 223,3,Dial(SIP/223) > > exten => 223,4,Hangup() > > -- > > __ > > Look into the audiohook_inherit function > > https://wiki.asterisk.org/wiki/display/AST/Function_AUDIOHOOK_INHERIT > > -- > Telecomunicaciones Abiertas de M?xico S.A. de C.V. > Carlos Ch?vez Prats > Director de Tecnolog?a > +52-55-91169161 ext 2001 > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110510/6d0cb3d2/attachment.htm>