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/${EXTEN},60,?) exten => _X.,n,Agi(agi://localhost/aj.agi?action=??..) I have looked through all arguments of Dial but haven't found any way to continue having a connected call between the caller and the callee and have AGI control of it. Is there a way to do this or do I have to use G() and connect the both ends to AGI separately and then bridging them before recording the call? Thanks for help. Regards, Henrik -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/3c12d0f5/attachment.htm>
Put the AGI call in a macro context and add M(macro) to your Dial string. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henrik Westerberg Sent: Wednesday, January 02, 2013 8:02 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Dialing out and recording Hi, I am using asterisk via AGI and want to be able to record a call. The scenario is: 1. A call comes in 2. The call is redirected to a mobile number via a local extension and ChannelRedirect 3. The local extension looks like something this: exten => _X.,1,Dial(SIP/${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/ce6b7c57/attachment.htm>
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. > >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: ><http://lists.digium.com/pipermail/asterisk-users/attachments/20130102/459 >43b1f/attachment-0001.htm> > >------------------------------
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