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 =
:31, Henrik Westerberg wrote:
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
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 ${CC_DIALSTRING})
meetme room...
yves
Am 14.03.2013 08:43, schrieb Henrik Westerberg:
Hi,
The idea was to record an ongoing call by three party bridging on the mobile
phone.
Well my problem was to halt execution of the Dialplan so the server would not
hang up the call. And I don´t want the server to say anything
-Commercial Discussion
asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com,
Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@ain.se
Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI
Hi,
so if your are ok with the way you solved part 1... alright, lets
is not fully written _immediately_
after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be
interrupted, if the call is hungup...
but as you did not show your agi... these are just hints..
regards,
yves
Am 07.03.2013 16:21, schrieb Henrik Westerberg
in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
I am developing a call recording
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[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 header like this:
exten = _S,n,SipSetHeader
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/${EXTEN},60,…)
: 001501cde8f3$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:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent
#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
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
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
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 stop playin
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
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