Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI(did.php)
exten = 3130XXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
Update,
My first question solved already.
There was an error on my agi script.
But second problem still valid.
On Tuesday, March 01, 2011 11:04:50 am Oguzhan Kayhan wrote:
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming
Subject: [asterisk-users] two questions regarding incoming call
Hello,
I want to make an agi script to match incoming DIDs with usernames.
I tried to do such entry in incoming trunk.
[DID_diddw]
include = from-didww
[from-didww]
exten = 3130XXX,1,AGI(did.php)
exten = 3130XXX,n,DIAL(SIP
Thanks to all who responded.
Jim.
Alex Robar wrote:
Jim,
There are SourceForge.net forums for [EMAIL PROTECTED] where you'll
probably find better answers to your AAH questions. They are located
here: https://sourceforge.net/forum/?group_id=123387
First question, is there a forum for [EMAIL PROTECTED] specific questions?
I've asked what must have been questions about [EMAIL PROTECTED] here and
gotten some indication they weren't welcome.
Second, does anyone know what files need to be backed up? I don't need
to back up the entire
, modules, sip.conf, etc.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Lynch
Sent: Friday, May 26, 2006 10:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
First question
Jim,There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387
In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule,
You might try these sites:
http://sourceforge.net/forum/forum.php?forum_id=420324 Backup has
been discussed many times here. Unfortunately, the SF forums suck in
terms of searching.
http://www.freepbx.org/
http://aussievoip.com.au/wiki/index.php?page=FreePBX
1) What do these two notices mean?
Oct 4 09:34:30 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
198.22.67.70, request '[EMAIL PROTECTED]' does not
exist
Oct 4 09:34:51 NOTICE[49584]: chan_iax2.c:5777
socket_read: Rejected connect attempt from
66.234.228.170,
: Tuesday, August 02, 2005 2:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Two questions about Asterisk Call Center
Hi:
I am new at Asterisk. Does anyone know how to define the call routing based
on DNIS as our conventional ACD to route a call in Asterisk? Second, how do
I collect
Hi:
I am new at Asterisk. Does anyone know how to define the call routing based on
DNIS as our conventional ACD to route a call in Asterisk? Second, how do I
collect Wrap-Up code for agents in Asterisk?
Many thanks.
Tielin
___
Asterisk-Users
Ok my first question is I have seen messages about a patch for asterisk so
that I can do auto answer on these phones. I found the message in the
archives but I do not have that message as an email still, so I do not have
the attachment. Can anyone tell me where to get it? Also on this phone how
What is the purpose of the beeping?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joel Duffield
Sent: Saturday, May 07, 2005 12:59 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] two questions about the Sipura 841?
Ok my first question
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jim Sturtevant
Sent: Saturday, May 07, 2005 4:25 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] two questions about the Sipura 841?
What is the purpose of the beeping?
-Original
Christopher L. Wade schrieb:
Please understand that Digium (Mark) is the reason * exists.
That's clear to me.
Selling a $10 winmodem for $80 is the reason Digium exists.
But the difference between $10 and $80 is really to much for me for just
playing around like I'm doing with asterisk. If I would
On Tuesday 07 December 2004 04:36, Erick Perez wrote:
Hi people,
question one
i see that asterisk is now in 1.x release. having tried it in the past
i want to know if i can use a voice modem as an outgoing line.
i know in the past that was not possible/supported so im just asking
in case
I see that the 100p is a modem with an Ambient chipset. Why does it
sell for 80$ in some places? i can get Ambient pci modem down here for
9 dollars. Any difference?
On Tue, 7 Dec 2004 10:58:55 +, Jon Lawrence [EMAIL PROTECTED] wrote:
On Tuesday 07 December 2004 04:36, Erick Perez wrote:
Erick Perez wrote:
I see that the 100p is a modem with an Ambient chipset. Why does it
sell for 80$ in some places? i can get Ambient pci modem down here for
9 dollars. Any difference?
Because Digium is selling support plus the modem, not just the modem.
-Chris
--
Christopher L. Wade
Christopher L. Wade schrieb:
Because Digium is selling support plus the modem, not just the modem.
But when you don't need the support?
Bye!
Michael
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Michael Vogel wrote:
Christopher L. Wade schrieb:
Because Digium is selling support plus the modem, not just the modem.
But when you don't need the support?
Bye!
Michael
Exactly. Choose the level of support you want from Digium and/or the
list. Historically, Digium equipment gets support from
Hello Christopher L. Wade,
I must say I 100% agree. I bought a cheap $10 because it is just for fun to
test it out. If I put it in my work I will definatley go with the digium true
hardware to support the company that put out the software for free. You can
pick up their other hardware at
Hi people,
question one
i see that asterisk is now in 1.x release. having tried it in the past
i want to know if i can use a voice modem as an outgoing line.
i know in the past that was not possible/supported so im just asking
in case the option is now available.
question two
im planing to use
Hi,
I'm a lab manager / supervisor at our labs. We've had Asterisk in
use for over a year directly hooked to the PSTN - a no brainer for
configuration (although I had to fix some ATT specific things in libpri).
Right now I have two big challenges. One is to hook our box up lineside to
a
We're getting ready to ditch our hosted virtual PBX for an Asterisk
solution and I've got a couple questions that probably come from
experience; we're looking to host two PBXs on the same box if the PBXs
are identified by different DID numbers, and I was curious if:
1. Is it possible to have
On Mon, 2 Jun 2003, Jason Smith wrote:
We're getting ready to ditch our hosted virtual PBX for an Asterisk
YAY! more power to ya...
1. Is it possible to have duplicate extensions between the two PBXs?
Eg: 555-x100 and 555-x100 on same * server
don't see any reason why
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