I have two server
Server_A(outbound call) for agent login and agent make a outbound call from
here and pass into server Server_B call
extension.conf
exten = _91XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR)
exten = _91XX.,n,hangup()
On 10 January 2014 17:12, James Wystead szilvertho...@gmail.com wrote:
Hello Folks;
I have an Asterisk server
Asterisk 11.7.0 built by root @xxx on a x86_64 running Linux on
2013-12-27 18:47:44 UTC
No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk.
Is there an API out there
Good Day, Ishfaq;
This may be a much better idea than the REST API.
Correct me if I'm wrong, but the concept is this:
You write to the database, and this gives the same result as perhaps
modifying the dialplan, sip, voicemail, etc *without* having to physically
modify the extensions.conf,
Yes, this would most likely be a better solution (REST API is in
Asterisk 12), just be careful about putting your dialplan in a Real Time
Database. You sometimes have to do things a little different if your
dialplan is in the RTDB. As well you need to make sure security is
locked down when using
Hi all,
I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind of access as securely as possible?
Thanks,
Patrick
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On 01/10/2014 08:33 PM, gm1 wrote:
On 01/10/2014 04:01 PM, Matthew Jordan wrote:
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com
wrote:
On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about
2014/1/10 Shaun Ruffell sruff...@digium.com
On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote:
2014/1/10 Shaun Ruffell sruff...@digium.com
You've configured the card to recover timing from the provider?
I'm not sure but I don't think so as I've just configured the card with:
On Mon, Jan 13, 2014 at 04:13:49PM +0100, Olivier wrote:
2014/1/10 Shaun Ruffell sruff...@digium.com
On Fri, Jan 10, 2014 at 08:34:57PM +0100, Olivier wrote:
2014/1/10 Shaun Ruffell sruff...@digium.com
You've configured the card to recover timing from the provider?
I'm not
On 01/13/2014 11:39 AM, Shaun Ruffell wrote:
If you have another board, yes, you could try. But I would recommend
checking all your cables, etc. Also, while highly unlikely, I've
heard of cases in the past where some smaller providers were
expecting to source timing from customer premise PBX
Ask your carrier to test the circuit. Often HDLC errors, especially with
modern cards, are caused by a dirty T-1 not a PBX or card issue.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Serafini
Sent:
Another option is to use an MRCP server like UniMRCP along with the Cepstral
plugin. One very nice thing about this approach is that there is less
'cepstral version' - 'asterisk version' dependency, which is a problem with
the current app_swift module (each app_swift version is designed to
Hello Everyone,
Calls that are private name private number have the following TO header:
From: Unavailable sip:aster...@server.com;tag=as120a1079.
Don't tell anyone, but we are trying to put on a We're big enough to own
the pricey softswitch look. Even though I would pick a OpenSIPS +
Asterisk
Correction, and by TO, I mean FROM header :)
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On Sat, Jan 11, 2014 at 4:56 PM, Charles Wang lazy.char...@gmail.com wrote:
Hi all,
I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc
to write CDR to my MySQL's cdr table.
After my testing, this scenario is working well.
After a long idle time, I didn't make any
On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Hi all,
I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind of access as securely as possible?
Well,
On Thu, Jan 9, 2014 at 12:01 PM, Olivier oza.4...@gmail.com wrote:
Hi,
On a Asterisk 1.8.12 system working OK for months (100k calls proceed),
users are complaining for bad audio.
My setup is:
PSTN --E1/PRI --- Asterisk --- E1/PRI--- Siemens HiPath ---E1/PRI ---
PSTN
asterisk -rx dahdi
On 14-01-14 02:36, Paul Belanger wrote:
On Mon, Jan 13, 2014 at 9:24 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Hi all,
I'm looking into adding the ability to call me at m...@mydomain.org on my
Asterisk 11 box. Does anyone have any tips or dialplan snippets to allow
this kind of
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