Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,?)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have AGI
control of it. Is there a way to do this or do I have to use G() and connect the
both ends to AGI separately and then bridging them before recording the call?
Thanks for help.
Regards,
Henrik
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Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Dialing out and recording
Hi,
I am using asterisk via AGI and want to be able to record a call.
The scenario is:
1. A call comes in
2. The call is redirected to a mobile number via a local extension and
ChannelRedirect
3. The local extension looks like something this:
exten => _X.,1,Dial(SIP/${EXTEN},60,.)
exten => _X.,n,Agi(agi://localhost/aj.agi?action=....)
I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
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Thanks Danny I will try this. /Henrik> >Message: 12 >Date: Wed, 2 Jan 2013 08:17:59 -0600 >From: "Danny Nicholas" <danny at debsinc.com> >Subject: Re: [asterisk-users] Dialing out and recording >To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users at lists.digium.com> >Message-ID: <001501cde8f3$f7d2b290$e77817b0$@debsinc.com> >Content-Type: text/plain; charset="us-ascii" > >Put the AGI call in a macro context and add M(macro) to your Dial string. > > > >From: asterisk-users-bounces at lists.digium.com >[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henrik >Westerberg >Sent: Wednesday, January 02, 2013 8:02 AM >To: asterisk-users at lists.digium.com >Subject: [asterisk-users] Dialing out and recording > > > >Hi, > > > >I am using asterisk via AGI and want to be able to record a call. > >The scenario is: > >1. A call comes in >2. The call is redirected to a mobile number via a local extension and >ChannelRedirect >3. The local extension looks like something this: > >exten => _X.,1,Dial(SIP/${EXTEN},60,.) > >exten => _X.,n,Agi(agi://localhost/aj.agi?action=....) > > > >I have looked through all arguments of Dial but haven't found any way to >continue having a connected call between the caller and the callee and >have >AGI control of it. Is there a way to do this or do I have to use G() and >connect the both ends to AGI separately and then bridging them before >recording the call? > > > >Thanks for help. > > > >Regards, > > > >Henrik > >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/ce6 >b7c57/attachment-0001.htm> > >------------------------------
#2 works for me on Asterisk 1.8.12 when setting the header like this:
exten => _S,n,SipSetHeader("Diversion: " ${CALLERID(rdnis)})
I haven't been able to make it work on 1.6 yet though, has anyone else?
/Henrik
>
>
>
>
>
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Don Kelly
>Sent: Wednesday, January 02, 2013 9:32 AM
>To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Subject: Re: [asterisk-users] Dialing out and recording
>
>
>
>I have the same requirement, but it's important that the caller ID
>information from the original caller is presented to the destination and
>we
>announce the call before the "transfer" is complete. The carrier
requires
>a
>diversion header if the ANI is not one of "our" DIDs. Does someone
have
>experience with this working?
>
>--
>
>Two suggestions for you, Don. #1 if the Dial is "Private" the
>"announcement" is taken care of. #2 I'm supposing that you
could do a "SIP
>Header" command before the Dial to resolve the diversion header issue.
>
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