[asterisk-users] Problem with Caller ID when receiving hidden number in via DAHDI and redirecting out via SIP

Hi, We have a system with both ISDN trunks and SIP. We receive incoming calls on both but always dial out via SIP. When dialing out the caller id is set like this: exten => _X.,1,Set(CALLERID(num)=${CC_ORIGNUM}) exten => _X.,n,Set(CALLERID(name)=${CC_ORIGNAME}) exten => _X.,n,Dial(${CC_DIALSTRIN

Re: [asterisk-users] executing the h extension at the real hangup of the call

13 september 2013 13:53 Till: Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>> Ämne: Re: [asterisk-users] executing the h extension at the real hangup of the call On 13/09/13 12:31, Henrik Westerberg wrote: Hi, I am running Asterisk 11.3 with

[asterisk-users] executing the h extension at the real hangup of the call

Hi, I am running Asterisk 11.3 with both SIP and ISDN. When dialing out (always over SIP) I want to keep track of who answered and of the length of the call. [outgoing-dev2] exten => h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished) exten => _X.,1,NoOp(Will send call to ${CC_DIALSTRING}) e

Re: [asterisk-users] Recording with MixMonitor and AGI

mailto:asterisk-users@lists.digium.com>> Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, the "music" heard by MoH is configurable... so if you want silence... But "hold" could e.g. also be done by transferring a caller into a dynamic meetme room... yves

Re: [asterisk-users] Recording with MixMonitor and AGI

ers Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com>>, Henrik Westerberg mailto:henrik.westerb...@ain.se>> Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again

Re: [asterisk-users] Recording with MixMonitor and AGI

n h priority. However, you have to use >DeadAgi in h extension. As your channel already hangup, it can not >run on AGI. > >Hope it will help you. > >Regards, > >Bharat Lalcheta > >On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg > wrote: >> Hi, >> >> I

Re: [asterisk-users] Recording with MixMonitor and AGI

)? One obstacle is, that the recorded file is not fully written _immediately_ after stopmixmonitor or hangup... this has to be taken care of and depending on your agi... it might be interrupted, if the call is hungup... but as you did not show your agi... these are just hints.. regards, yves A

[asterisk-users] Recording with MixMonitor and AGI

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-r

Re: [asterisk-users] Dialing out and recording

gt;[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik >Westerberg >Sent: Wednesday, January 02, 2013 3:20 PM >To: asterisk-users@lists.digium.com >Subject: Re: [asterisk-users] Dialing out and recording > >#2 works for me on Asterisk 1.8.12 when setting the head

Re: [asterisk-users] Dialing out and recording

#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-boun...@lists.digium.com >[mailto:asterisk-users-

Re: [asterisk-users] Dialing out and recording

>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-boun...@lists.digium.com >[mailto:asteris

[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,…) exte

Re: [asterisk-users] async agi question

Thanks, I was not familiar with this application. /Henrik Kevin P. Fleming skrev: Henrik Westerberg wrote: Yes, this works good for me. A StopIO feature would of course be cleaner but this certainly does the trick. The ExternalIVR interface, while not quite as

Re: [asterisk-users] async agi question

e can probably write something in res_agi.c Moy On Fri, Dec 5, 2008 at 3:01 AM, Henrik Westerberg <[EMAIL PROTECTED]> wrote: Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to

[asterisk-users] async agi question

Hi, I am developing asterisk support for our application using the Async AGI and Asterisk-Java. One thing I haven't been able to implement is how to stop playing a sound. Something similar to StopIO for Dialogic GlobalCall or DivaStopSending for Eicon. Is there any way to achieve this today which

[asterisk-users] pri rdnis found as Facility but not set

Hi, I'm running asterisk with a PRI. But I can't get hold of the rdnis number. When running pri debug I can see the true rdnis number as Facility, the number 703289840 as shown below. Is it possible to get hold of this value in some way from extensions.conf? Or is it necessary to modify the sourc