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

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89 views

Parse and validate Asterisk dialplan before commiting

I recently made a number of changes to my Asterisk dialplan and would like to validate these changes before I commit. I am thinking more from a "write code" -> "compile" -> "debug" approach. I am ...
1
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1answer
428 views

Asterisk outbound calls hanging up imediatellly on Zap trunks

The inbound calls work fine. Outbound calls fail as follows: Dial("SIP/xxxxxxxxx","ZAP/g0/14442223333|300|") Couldn't call g0/14442223333 Hungup 'Zap/2-1' Everyone is busy/congested at this time ...
1
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1answer
379 views

How to establish SIP connection, when SIP-proxy is required?

I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. ...
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1answer
201 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
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1answer
419 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
2
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1answer
493 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
4
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3answers
2k views

How do I secure my Asterisk server?

Our asterisk server was compromised. some calls were made to Asia countries last weekend. Thought we have improved our network configuration, we still want to determine how the intrusion was done, we ...
2
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1answer
946 views

cant make outbound calls - asterisk

I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: ...
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2answers
1k views

Can anybody explain why to use Asterisk PBX rather than standard PBX from Matrix or Siemens [closed]

Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
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2answers
1k views

Outgoing calls from vtiger to an asterisk extension

I've successfully installed vtiger and asterisk on Ubuntu, and I managed to configure them such that I get an incoming and an outgoing popup in vtiger. The problem is that when I click on an extension ...
2
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1answer
529 views

Elastix and two ISDN lines

Since I am not an expert on communications, I will describe everything from the user perspective. We had a PBX with elastix embedded and two ISDN cards in it (we have two isdn lines). The mobo of the ...
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1answer
654 views

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
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1answer
532 views

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 ...
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2answers
749 views

Timing Source on Asterisk [closed]

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
0
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1answer
156 views

Enable auto dial on Polycomm VoIP phone? [closed]

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 ...
3
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2answers
2k views

How can I make Asterisk keep track of dynamic SIP agent statuses?

I am setting up a new server using Asterisk 1.8.11-certified4. In testing, we're seeing that agents dynamically logged into the queue will receive a second queue call as a call-waiting when ...
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2answers
118 views

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 ...
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1answer
591 views

Inbound Caller gets Fast Busy and Extension gets ringing

Inbound caller (1636) calls a sipstation direct dial to my phone on my desk. Cellphone sounds like fast busy but then desk phone rings. Cant connect the call. then when desk phone hangs up It ...
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2answers
86 views

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 ...
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1answer
2k views

What are the possible reasons for a SIP register failure?

I have setup an Elastix box (asterisk/freepbx based) and added a trunk for an external VoIP provider. When I get to the Asterisk command line interface and type sip show registry I always get the same ...
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3answers
861 views

Elastix and extensions.conf

My VOIP provider required me to add this into /etc/asterisk/extensions.conf: exten => h,1,Hangup exten => _00X.,1, Answer exten => _00X.,n, Set(CALLERID(num)=96354205144239) ; exten => ...
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2answers
869 views

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 : ...
0
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1answer
895 views

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 ...
2
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0answers
279 views

Asterisk ${CALLERID(name)

I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf). I wanted to get caller Name from database which i have stored for example. CLI> ...
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2answers
550 views

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 ...
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2answers
657 views

cli commands not working asterisk on ubuntu

hi guys today my first day on asterisk on ubuntu. I installed vmware and then ubuntu and then on ubuntu i am running asterisk. i started the asterisk server successfully by following a tutorial. ...
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2answers
958 views

asterisk incoming/miss call notification (to skype)

My environment: Centos 5.6, Asterisk with freepbx , skype i.e.sends message with parameter skype.sh skype_user message. Now i wanted to send skype notification so that my asterisk server notification ...
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2answers
3k views

Asterisk does not recognise DTMF tones from mobile phones

We have an Asterisk 1.8.7.0 (the Elastix derivative) switchboard. Every since a month ago, seemingly out of the blue, the switchboard does not recognise DTMF tones any more from mobile phones. ...
4
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2answers
97 views

Using an extension to block a caller

I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this: Once I get a junk call, I ...
2
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3answers
1k views

PBX with Fax and Google Voice

Looking to replace/port my home number which I use for mainly faxing for my home business to a PBX server ( Thinking Asterisk or Elastix ). My question is: Does Asterisk/Elastix support Faxing ( ...
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2answers
122 views

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. ...
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0answers
465 views

Lync 2010, Kamailio, & Trixbox 2.6.23 (Asterisk 1.4)

I'm having an issue trying to connect Lync 2010 phone calls with our trixbox PBX. I've gotten to the point where Kamailio seems to be functioning properly and acting as a bridge between TCP traffic ...
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2answers
75 views

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?
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1answer
325 views

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 ...
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1answer
625 views

Asterisk / Elastix Address Book and Call Recording

I am trying to build a small asterisk based PBX using Elastix. It'll be having 4 FXO (2 nos.no ISDN, normal analog POTS, 2 nos. GSM connections using a GSM Terminal) and 4 FXS (2 IP Phones and 2 ...
3
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1answer
774 views

Make call with alternate provider if NOANSWER

I have two voip providers, one free an the other paid. The free provider only allows local calls to certain area codes, so I need to fall back to the the paid provider if a call fails. At the moment, ...
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4answers
10k views

How to stop registration attempts on Asterisk

The main question: My Asterisk logs are littered with messages like these: [2012-05-29 15:53:49] NOTICE[5578] chan_sip.c: Registration from '<sip:912@xx.xx.xx.xx>' failed for '37.75.210.177' - ...
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1answer
5k views

Asterisk SIP/2.0 401 Unauthorized

I'm running into a funny little issue with Asterisk 10.3, but it seems to be applicable to 10.4 as well. The server running Asterisk was relocated from a VPS to dedicated hardware, and now only 1 of ...
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3answers
797 views

Convert a WAV (Asterisk recordings) to MP3

I have a large collection of WAVs. They are: RIFF, little-endian WAVE audio GSM 6.10 Mono 8000Hz effective bitrate of 13.6 I need to convert these to MP3. Current I am forced to send the file ...
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1answer
61 views

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 ...
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3answers
653 views

Is It a Bad Idea to Install the OS GUI on a Server That Will Host Asterisk?

We have a server onto which we plan to install CentOS 6.2. The server will host Asterisk for our office, but is not a webserver. If we install the minimal installation for CentOS - that is, no GUI ...
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2answers
2k views

Asterisk: Authenticated SIP trunk

Let's say I have an Asterisk system with a bunch of connections: there are phones (who register itself with *) and providers (who wish to establish SIP trunks to put a lot of calls over, with ...
0
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1answer
719 views

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 ...
2
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3answers
445 views

How to add more incoming/outgoing lines capability to Asterisk?

We have 4 people call us from around the country at once and it causes our phones to be busy. We want to be able to have at least 8 people call us at once without anyone getting the busy signal. ...
2
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1answer
139 views

Transfer to Voicemail only when call not answered, not on rejecting a call (Asterisk)

I am working with Asterisk now and observed call transfers to voicemail even though i have rejected it. Is any way to transfer call to voicemail only when on not answered or not available ?
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1answer
238 views

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 ...
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2answers
616 views

Server Freeze Up Under Load

I'm having a problem with a debian server that I thought was due to bad RAM, but is persisting. It's a Dell Poweredge 6800 with two dual-core 3.6GHZ Xeon processors and 5GB of DDR2 ECC 333. I've ...
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1answer
158 views

How to add a user who is out of state as if they were just another extension AT our office? [closed]

We want to add a person who is in another state to our phone system as though they were simply in another cubicle here at the office. So, they can dial "302" and get the office assistant's extension ...
0
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1answer
692 views

How to play and record wav in Asterisk

Please, how could I do these things on asterisk: 1) Externally (with some API) make a call from asterisk and play wav and hangup the call 2) How should I configure the extension/route/whatever on the ...
0
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1answer
63 views

How to install app_echo.so in FreePBX?

What is the proper way of installing app_echo module when FreePBX is installed ? I cannot find it in Module admin and manual editing of configuration files when freepbx presented is not recommended, ...