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Questions tagged [asterisk]

Asterisk is software that enables a server to act as an IP PBX system, VoIP gateway, conference server, and more.

see

http://www.asterisk.org/

and

http://en.wikipedia.org/wiki/Asterisk_(PBX)

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asterisk queues priority and penalty

queues.conf shared_lascall=yes strategy=rrmemory wrapuptime=15 A1,A2,A3 are members of 2 queues: queue1(Q1) and queue2(Q2) A3 has penalty 3 in Q1 where min/max penalty are 0/3 and A3 has penalty 0 at Q2 where min/max penalty are 0/3. A3 has just…
MealstroM
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Automatically reconnect to ODBC sources?

I am using Asterisk 1.8.10.1 and a MySQL database connected via ODBC to store CDRs. When my MySQL database isn't available when Asterisk starts or has an outage while Asterisk is running, I would expect Asterisk to retry to connect to the database,…
stefan.at.kotlin
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Timing Source on Asterisk

I have tried to look for simple language of explanation what is timing source for in asterisk? and why it has significant impact on the asterisk Anyone can give me insight will be appreciated
jakarta512
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Enable auto dial on Polycomm VoIP phone?

I was wondering if it was possible to make the polycomm devices automatically dial out without the use of the "Dial" button. We have user's who are mistaking this for the phone being broken when they try to dial out and if it was setup as dial when…
terminaltox1n
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handle actions diferent to Command with Asterisk::AMI

I'm learning Asterisk :: AMI, but all examples deal with the action Command. i've tryed to run the following action (no success) my $action = $astman->action({ Action => "Agents" }); i have the following sub for print response work fine for…
rkmax
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Installing a GUI over a running Asterisk

The server is Ubuntu Server 10.04 have asterisk-1.6.2.2.tar.gz installed. I have read that most of the GUI for Asterisk administration are something more than simple GUIs, and overwrite the previous asterisk configuration at the time of…
rkmax
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Skype Connect and Elastix for incoming and outgoing calls

I ordered Skype Connect, And i want to integrate skype connect with my Elastix server to handle incoming and outgoing calls. I created new SIP Trunk through GUI with the following info : Incoming…
Linux Intel
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Elastix call-center with computer telephony crm integration

The title isn't explanatory enough but I can't explain it better. I want to setup a call-center using elastix which would have the following feature: Suppose bob(client) calls the center and Alice(employee) picks up the phone when that happens I…
ndp
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How do I configure additional phone lines asterisk/trixbox?

I have a 4 port Digium card in there, and have 4 lines running smoothly. Now, we added ANOTHER 4 port card 1AEX410PELF and have 4 more analog lines coming into the Trixbox server. It still runs the 4 fine, but what do I need to do to add the…
Matt
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Can a "4 line" Grandstream GXP 2000 handle 8 lines?

We have a whole bunch of GXP 2000 phones here but now we're growing too big for just 4 incoming lines. We use Trixbox (asterisk) for the phone system with 4 analog lines coming into a Dell server. Is there a way that these Grandstream GXP 2000…
Matt
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Asterisk server in Debian linux

I m tried to download sip.conf files in asterisk server with ssh it replies sip.conf is not a regular file,directory or symlink?
Saravana Karthikeyan
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asterisk 1.8 poor playback quality

i've just installed an asterisk 1.8 IVR using playback() for playing sln files. i've we have some high quality wav files, that i've converted with sox to sln 16KHz files. (Why aren't higher sampling rates supported?) when i call the asterisk IVR…
JMW
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Can I include the same file multiple times in Asterisk?

I'm new to Asterisk, and I, would like to include a file that stores all of the common settings for my SIP phones to avoid duplication. Can I use #include more than once in sip.conf to accomplish this?
Sean W.
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Permission Denied trying to execute a .conf file.

I'm using a virtual machine in Windows 7 and I have installed Linux on it and Asterisk software. I want to configure the sip file to create sip user but when I type ./sip.conf in Linux it replies :permission denied. Please can you tell me how can…
Jaspreet
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Get Asterisk/Freepbx to ignore digits that were unnecessarily dialed (area code for local calls)

This is the second part of this question, worded better. Remove local area code on outbound calls, and remove "dial 1" for long distance in Trixbox/Asterisk I have removed the NEED to dial 933 or 1+933 for long distance or local calls already.…
Matt
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