In my situation AMI is not an option. When somebdy puts a call on hold, on
asterisk console I can see messages like Started music on hold, class
'default', on SIP/ and Started music on hold, class 'default', on
SIP/. I guess the only way in my scenerio is to modify
Hi,
I am using asterisk version 1.4.22.1 on a centos 5.2 machine.
Is there any way to run a script somebody puts the call on/off hold ? The
script must be run both on hold and off hold.
Best ragards.
Idris
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You can try Adit 600 which is distributed by Alliance.
-Original Message-
From: Jan Marek [mailto:[EMAIL PROTECTED]
Sent: Thursday, August 30, 2007 9:20 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Channel banks for E1
Hello all,
please, can anyone advertise me some
Do you have any extension in default context of your extensions.conf
file to accept incoming calls ?
It must be something like;
exten = 12345678,1,Answer()
exten = 12345678,2,Playback(Welcome)
...
12345678 = The DID number you are calling to reach E1
Idris
-Original Message-
From:
In general section of sip.conf you can bind sip service to multiple ip
addresses. If you setup routing successfully you can send the call
received one of ip address through other ip addresses of asterisk. All
you have to do is to setup routing the right way. In this conf asterisk
can be used both
You can find detailed info about command Transfer at
http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer .
_
From: Lucian Romi [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 19, 2007 2:55 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to config SIP blind
Hello,
Is there a way to find out which party hanged up the call. Generally
this is reported as Local disconnet/Remote disconnect in callcenter
environments.
Thanks.
Idris
Information and Communication Technologies Manager
Vodatech
___
: [asterisk-users] Hangup Party
On Tue, 12 Dec 2006 15:27:06 +0200
Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
Is there a way to find out which party hanged up the call. Generally
this is reported as Local disconnet/Remote disconnect in callcenter
environments.
This is already written
Hi,
I want to run a script when users puts other party on hold. Script may
be anything. Perl, Agi ...
Is there anyway to do this ?
Idris
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asterisk-users mailing list
To UNSUBSCRIBE
Michael,
After many months of search we decided to develop an in-house solution
for such kind of needs. For a month our solution is in production and
does everything you mentioned below. Asterisk's built-in call queue does
not provide many of the features necessary for large organizations.
Idris
We've been using cisco 2600 gateways with asterisk for a year and
everything works fine. IOS 12.2 is installed in gateways.
-Original Message-
From: Tijl Van den Broeck [mailto:[EMAIL PROTECTED]
Sent: Monday, October 02, 2006 10:30 AM
To: Asterisk Users Mailing List - Non-Commercial
You should look for dialplan (extensions.conf) commands. They include
everything you mansion. Examples include mysql queries. Maybe you spend
sometime for Sybase connections.
-Original Message-
From: Dirk Enrique Seiffert [mailto:[EMAIL PROTECTED]
Sent: Friday, July 07, 2006 4:16 PM
To:
Check this on :
http://www.digium.com/en/docs/misc/compatibility_notes.php
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 05, 2006 11:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] HP Proliant server?
Has anyone had any
I use Asus Barebone server's that can handle 120 Zap -- SIP calls. You
can find additional info on http://www.snc.com.tr/ractory_rx1ba-n.asp
which is a Asus distributor in Turkey. My server's details are;
2 X Xeon 3.0 HT Supported
2 GB RAM
2 X 250 Sata (RAID 1)
Onboard Ethernet, VGA
1U Size
I use TE411P in this server.
-Original Message-
From: Idris AVCI
Sent: Wednesday, July 05, 2006 1:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Intel E7220 chipset?
I use Asus Barebone server's that can handle 120 Zap -- SIP calls. You
Asterisk logs very detailed information in /var/log/asterisk/queue_log
file including abandoned calls. You can import this log to mysql with a
simple perl script running periodically.
-Original Message-
From: Michael Konietzny [mailto:[EMAIL PROTECTED]
Sent: Friday, June 30, 2006 11:44
You must create an extension in dialplan and use this extension in your
originate script. So you can do whatever you want in that extension.
-Original Message-
From: lokotes [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 22, 2006 1:30 PM
To: asterisk-users@lists.digium.com
Subject:
Hi,
You can try ChanSpy http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy.
Idris
From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006
12:23 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Monitor
a particular SIP call
Hi Steve,
We are running X-Lite on Wyse V90 terminals. They have Windows XP
Embedded, 800mhz CPU and 512 MB Ram. We use USB headsets because the
onboard audio chip is very poor on voice quality. I guess X-Lite has
Windows CE version. Check on www.counterpath.com.
Idris
-Original
Check features.conf. If not uncomment the atxfer line and assign a key
combination (Default is *2). Then use t and T switches in Dial command.
Finally restart asterisk service.
-Original Message-
From: Julian Lyndon-Smith [mailto:[EMAIL PROTECTED]
Sent: Tuesday, June 20, 2006 1:58 PM
Hi Steve,
Thank you for your answers. First of all span 3 is not a satellite link
and no echo occurs when I connect this line to another pbx with HW EC
feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I
have to do something to enable EC for this card ?
Idris
-Original
I have two MP-108 boxes working fine in
both ways. You have to check these;
-
Assign
endpoint phone numbers to FXO ports (Note these numbers you will use in
extensions.conf later)
-
Route
calls to asterisk in Tel to IP Routing table.
-
Add endpoint
phone numbers to asterisks
If you want to send calls using SIP you have to define in dial-peer.
Add session protocol sipv2 to dial-peer voice 2 voip interface.
-Original Message-
From: Wasif [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 8:19 PM
To: asterisk-users@lists.digium.com
Subject:
I forgot to mention one thing. I don't know if it changes anything.
Internal users are connected to another PBX which connects to asterisk
over SIP. Echo is always at internal user side. External user never
hears echo.
External User -- PSTN -- Asterisk -- SIP -- CIC -- Internal User
Hello,
There are 3 PRIs connected to the card each from
different operators. Especially echo occured on span 3 is really annoying.
Configuration files are as follows. Is there something wrong in conf ?
Zapata.conf --
[channels]
context=default
We developed a commercial SIP phone for windows platform. What do you
want to know ?
-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Thursday, June 15, 2006 4:31 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] open
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