hello list,
i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in
order to record the conversation
i can record all the calls inbound and outbound without problem.
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
extensions.conf
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/${EXTEN},,KkTt)
exten => 223,n,Hangup();
any help please
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On 16/06/11 07:36 AM, salaheddine elharit wrote:> hello list, > > i have asterisk 1.4 with IAX and sip i have configured the MixMonitor in > order to record the conversation > > but when i receive an inbound call from customer in IAX(1000) and i want > to transfer the call to other phone SIP(223) > the conversation between customer and IAX is recorded but the > conversation between customer and sip is not recordedIs the call coming from IAX(1000) or going to IAX(1000)? Note that when you transfer calls around and are using MixMonitor() (or any recording) that you have to think of the recording as being associated with the incoming channel, and the recording should essentially follow it around. So if you have a call coming in like this: ITSP --> Asterisk --> Dialplan --> Mixmonitor --> Dial(SIP/1000) Then the MixMonitor() is associated with the channel created when the call came in from the ITSP. If that channel is then transferred, the recording should follow it around. Can you elaborate a bit more on the call flow and show the console output? -- Leif Madsen http://www.oreilly.com/catalog/asterisk
thanks for your response
the call is going to IAX(1000), i have i DID Number when the customer call
this number 0520XXXXXX the call is goint to agent
IAX. in my dialplan i have
exten => 223,1,MixMonitor(blah.wav)
exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten => 223,n,Dial(SIP/223)
and in extensions.conf i have
exten => 223,1,MixMonitor(sip_${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.madsen at 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
>
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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/${EXTEN},,KkTt) > exten => 223,n,Hangup();OK, well nothing looks obviously wrong there from what I can tell. What is your console output doing though when you do the transfer? Are you using Asterisk transfers? What version of Asterisk are you using? Leif. -- Leif Madsen http://www.oreilly.com/catalog/asterisk
i have asterisk 1.4 and also i have aheeva applicaton also installed in my server in the consol this call may be monitored or recorded best regrads 2011/6/16 Leif Madsen <leif.madsen at asteriskdocs.org>> On 16/06/11 09:20 AM, salaheddine elharit wrote: > >> thanks for your response >> >> the call is going to IAX(1000), i have i DID Number when the customer >> call this number 0520XXXXXX the call is goint to agent >> IAX. in my dialplan i have >> exten => 223,1,MixMonitor(blah.wav) >> exten => 223,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) >> exten => 223,n,Dial(SIP/223) >> >> and in extensions.conf i have >> >> >> exten => 223,1,MixMonitor(sip_${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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110616/d82464dc/attachment.htm>
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <leif.madsen at asteriskdocs.org>
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
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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 <danny at debsinc.com>> 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:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *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 <leif.madsen at asteriskdocs.org> > > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110616/f014fe0a/attachment.htm>
Since AUDIOHOOK_INHERIT is a backport from 1.8, something may be amiss in
the 1.4 IAX rendition. I assume your install would not be friendly for a
1.8 upgrade?
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Thursday, June 16, 2011 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MixMonitor
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 <danny at debsinc.com>
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <leif.madsen at asteriskdocs.org>
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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