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

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How can I limit Asterisk's Chanspy feature to certain channels?

I'm using a standard config of Asterisk 1.4 with FreePBX using the chanspy feature code. I want to restrict what channels it is possible to spy on, ideally doing this in a .conf file. How would I do ...
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131 views

Is there a minimum CPU requirement for a single call using G729?

I am just trying this out on an AWS micro VM, which gives out roughly the equivalent of a 600Mhz processor. Currently call quality is poor, but I am trying to ascertain if this is due to a lack of ...
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409 views

TrixBox Message Waiting Indicator in Device and User Mode

My trixbox installation supports hot-desking, so that users can take any desk and log into the phone. To do this, I placed FreePBX into Device-and-User mode, which allows the devices (phones) and ...
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1k views

Asterisk failing at startup after upgrading to asterisk18

I was using asterisk16 and asterisk16-skypeforasterisk, which was working fine. I have recently upgraded to asterisk18 and asterisk18-skypeforasterisk, after that I am receiving the following error ...
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757 views

Asterisk doesn't start properly at system startup. DNS lookup fails

When I start my Ubuntu system it attempts two DNS lookups. One to find out what my internet-routers external ip is. And one to find the IP of my PSTN-SIP-provider. Both fails. [Apr 7 22:14:54] ...
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47 views

How to setup Asteriesk in WINDOWS machine and how to get started?

How to setup Asteriesk in WINDOWS machine and how to get started? any tutorials would be helpful I googled it but it didn't turn into helpful things
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4k views

How do I configure Asterisk to use G729 on a trunk with FreePBX

I've successfully got the channel from phone->PBX working on G279. I'm just trying to get the PBX->trunk to also be G279. I did this by using FreePBX and putting in allow=g729,ulaw into each ...
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739 views

Setting environment variables for a service without a login shell on Debian

I have created a linux service that runs as a deamon (and gets started from /etc/init.d/X). I need to set some environment variables that can be accessed by the application. Here's the scenario. The ...
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581 views

Can we manage incoming call number landing on asterisk?

I am using Asterisk 1.4.2 in two different machine. I have configured some extensions in asterisk. When any caller, dial my extension number with local number then if I see asterisk console (asterisk ...
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2k views

Asterisk, IAXModem & Hylafax how-tos?

I'm trying to set up Asterisk and IAXModem to send faxes via T38 (Yes, I know I'm swatting a fly with a Buick...) However, since I'm trying to do something so small with a product so large, I'm having ...
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FreePBX: Asterisk in the Cloud (EC2) Audio Problems

Please pardon the newbie question, but I can't seem to figure this out. I followed Voxilla's tutorial to the tee. But in making calls, my softphones connect, yet no audio (in either direction). I ...
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Asterisk terminating outbound call when picked up, sends 'BYE' message

I'm running Asterisk 1.6.1.10 / FreePBX 2.5.2.2 and I've got an outbound trunk setup. Everything use to work fine until recently (perhaps due to upgrade to FC12 or other things I'm not sure). Anyway ...
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862 views

Asterisk/TrixBox and Predictive Dialing/Outbound Call Center

Is anyone out there running predictive dialers or Call Center type products on Asterisk/Trixbox. I'm looking for vicidial type functionality. The best way to characterize it would be political ...
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12 views

Asterisk write cdr now

I'm writing CDR to mssql table with cdr_adaptive_odbc. This works fine in the sense that the record is written after the call. But I'd like to write a cdr_record when the call comes in, before it is ...
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12 views

Asterisk redirect incoming call to last number that called it

I would like to know how to redirect an incoming call to the extension that called it first in Asterisk. I have not found a way to do it, and I haven't found any Asterisk variable that contain a value ...
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24 views

Issue with Asterisk on Google compute engine (GCE)

I installed and configured Asterisk in Google compute engine (GCE). I am facing the below issues: 1) Unable to login to Asterisk CLI by using asterisk -rvvvvv command (But, I am able to login using ...
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166 views

Asterisk dialog destruction on attended transfer timeout

Steps to replicate: Alice receives incoming call from Eve (either from external line or another extension, doesn't matter). Alice makes attended transfer to Bob and hangs up before Bob answers. Bob ...
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91 views

How to know what number was dialled on incoming call

The scenario In the UK. I have one (BT openreach) phone line coming into a Cisco SPA232D. That phone line is ringable on a number, let's say 01235AAAAAA. I have a virtual number, 01235BBBBBB, which ...
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51 views

Asterisk one way audio detection

I have some calls have one way audio. I tried to write ${CHANNEL(rtpqos,audio,all) to CDR and think what if themssrc=0 than no audio, but it's not true. Is any way to detect one way audio? ...
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303 views

OpenLDAP & Asterisk - no global superior knowledge

I'm struggling to setup OpenLDAP to allow Asterisk to store information about users. I've tried to follow several howto's, but unfortunately I'm not able to set it up correctly. It seems like most ...
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114 views

Asterisk logging to Graylog2

Does anyone have any experience with logging Asterisk servers to Graylog2? I'd like to be able to send certain verbosity levels to Graylog2, but I don't have a clue how to go about it. The goal is ...
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116 views

Using Asterisk for IVR calls without internet connectivity

I am thinking about using Asterisk as an IVR for a project in Burundi, in a place where we often lose internet connection. I'd like the Asterisk server to deliver IVR calls through, say, an Android ...
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258 views

sip register success on one host but fail on another host in same local network

I acquired antisip account and setup asterisk (lets say PBX-A) to register. It worked fine. However, I attempted to do so with another asterisk (PBX-B). No luck this time. PBX-A and PBX-B are on the ...
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51 views

Log to file in Asterisk: append with newline

(I am testing on Asterisk 11.7.0~dfsg-1ubuntu1) I am using the following format to append to a logfile, according to the documentation: same => n,Set(FILE(/tmp/mylog.txt,,,a)=my-log-message) ...
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120 views

Asterisk Event PauseQueueMember - AMI Calls

We use Asterisk-AMI and we have a little JS server which listens for Events from the AMI to gather & display statistics (Calls, Queues etc). I initialy thought that when a queue member is paused, ...
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Asterisk: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)

Strange issue started with our Asterisk install yesterday. When dialling out via ISDN BRI the call fails and the following error is logged: Unable to create channel of type 'DAHDI' (cause 34 - ...
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448 views

Asterisk ring out even when the client is offline

I need to configure an Asterisk box to go to voicemail but only after X ringouts. exten => 1234,1,Dial(SIP/ivan, 30) exten => 1234,2,VoiceMail(777@mb_tutorial) exten => ...
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Vicidial in Autodial does not connect correctly dongle phone call to agent conference

I want to make work Vicidial with chan_dongle Asterisk module to use Huawei USB modems as GSM gateways (trunks/carriers) for asterisk. In manual mode everything works. Problem appears only if ...
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Asterisk AGI randomly returning 4 and failing to execute PHP AGI

I've got a PHP script that I execute 2-3 times throughout a single calls life time, if it is inbound it will execute once for the queue, once for start of call and a last time for end of call. For ...
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Asterisk IVR setup: disallow direct dial for some extensions?

We've got PBX to configure. It's based on Elastix 2.4.0, freePBX 2.8.1 and Asterisk 1.8.20.0. Currently we have a setup for inbound calls like that: Inbound call > Announcement (30 sec) > IVR ...
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70 views

SecAst: Configuring Email Alerts for Google Apps Account

Requesting assistance with configuring SecAst email alerts to work with Google Apps hosted account (posting here as per request by Generation D)... We have SecAst installed on a system with a new ...
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261 views

Asterisk Penalty for dynamic Agents

Is there a way to use penalty with dynamic agents to order agents call distribution for received calls in a queue? We are using linear ring strategy and this only order calls to dynamic agents by ...
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185 views

Werbrtc2sip crashing intermittently. Aborted (core dumped)

I am having issues with making webrtc2sip actually work. Webrtc2sip is now getting crashed intermittently during a call. I have seen this even when only one call is active in a queue. By the way, my ...
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196 views

Connect Cisco 7975 VOIP phone to two separate Asterisk PBX's

I have a Cisco 7975 registered on an Asterisk SIP server, and I want to register a second line on a second (different IP) Asterisk SIP server. I tried to do this with firmware version 8.5.3 but it ...
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372 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
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346 views

Asterisk congestion reasons

After moving Asterisk from one server to another, I noticed that ASR dropped by 15-20 %. Also I noticed following warning in logs file [Dec 8 15:52:00] WARNING[4771][C-00000031] pbx.c: Maximum ...
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279 views

DTMF Detection in Asterisk PBX

I have an asterisk server as IP PBX. I have created an Auto-Attendant for my company and asking users to enter desired extensions when they call. The problem is that when the message is playing and ...
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Error raised during calls in asterisk server

Today my asterisk could not receive calls for a while. when I checked the asterisk logs, the error below was raising during this period. Purely numeric hostname (1002), and not a peer--rejecting! ...
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153 views

How can I use the FLASH key of my IP phone to transfer a call (atxfer)

My IP phone has a FLASH key. This key sends the DTMF of Flash [16 in decimal] (http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling). How do I set the atxfer parameter in the features.conf?
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How to configure a 2821 ISR to an asterisk PBX on a PRI line?

I'm trying to move my company's phone system from Cisco's CUCM to an asterisk solution. I have setup a PBXinaflash server, and setup an account with www.vitelity.com. I was able to get it working. ...
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570 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
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420 views

UDP Range Nat Translation port forwarding RPT Traffic to inside host PBX Cisco ASA 5505 8.2a

I have an Asterisk based PBX that I'm providing VoIP services with, My gateway is a Cisco ASA 5505 running 8.2 and I only have one public IP Available along with other services I provide with that IP. ...
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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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143 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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566 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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488 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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512 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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890 views

How do I detect call forwarding in Asterisk?

Basically, I want to do the same thing that Google Voice does. I forward my calls to a DID number that rings my Asterisk server via IAX2, which, if it detects the call has been forwarded, sends it to ...
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345 views

Asterisk 1.8, Dedicated SDSL but SIP Provider Inbound Channel heavily delayed?

We have a dedicated Asterisk 1.8 server running as a front end SIP server for authenticated and internal users only, using SDSL 2MB/2MB, in the UK. The provider of our internet also is the provider of ...
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371 views

asterisk call file not deleted

I'm placing outgoing calls using Asterisk call files on Asterisk 1.8. My call files are very simple and have no MaxRetries or RetryTime specified. If a call succeeds, the corresponding callfile is ...