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

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511 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
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2answers
7k views

Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's ...
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1answer
998 views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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1answer
1k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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1answer
235 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...
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2answers
123 views

Determine which user initiated call in Asterisk

I had the following code in my extensions.conf file: [local] exten => _NXXNXXXXXX,1,Set(CALLERID(name)=${OUTGOING_NAME}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${OUTGOING_NUMBER}) Now I ...
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1answer
104 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 ...
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1answer
461 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 ...
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1answer
413 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
231 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
490 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 ...
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1answer
541 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 ...
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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
1k 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
574 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
760 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 ...
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1answer
569 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
824 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
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1answer
164 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 ...
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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
126 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
634 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
88 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
3k 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
922 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
930 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 : ...
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1answer
994 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 ...
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0answers
296 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
579 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
790 views

cli commands not working asterisk on ubuntu

I installed VMware and then Ubuntu. I am running asterisk on top of it. I started the asterisk server successfully by following a tutorial. After making few changes into the sip.conf, I want to ...
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2answers
1k 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. ...
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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 ...
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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
127 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
485 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
343 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
667 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
votes
1answer
833 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
11k 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
6k 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
910 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
66 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
677 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
3k 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 ...
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1answer
798 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
476 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. ...