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

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Asterisk and FXO issue over DDNS

I've been experimenting with asteriskNow (freePBX/asterisk 11) as a sip server with DDNS. Everything seems to be working fine using externhost and such; calls to remote extensions to and from asterisk ...
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491 views

How to know if graceful restart / stop was executed?

Is there a way to know if restart gracefully or stop gracefully has been executed on asterisk while still running and waiting for calls to end then, restart/stop? When this mode is executed, Asterisk ...
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2k views

Asterisk issue with ConfBridge

I'm receiving an error when I try to call an extension that leads to ConfBridge() app in my dialplan Here is the extensions.conf exten => 601,1,Noop() same => n,ConfBridge(room-one,...
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151 views

Installing FreePBX or Asiterisk-GUI on working server

We have server with Ububuntu and asterisk 1.8 on it without any web-interfaces. Our managers ask me to make them web-interface for looking over config, users, calls and so on. I've installed ...
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422 views

Asterisk ignoring sip_domain part of URI

I am using Asterisk version 11.5. My problem is the following: I literally cannot make any outbound call. That's because, whatever I dial in any client, the sip_domain part gets ignored, it is ...
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470 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...
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541 views

Asterisk: Forcing a sip peer to connect via ipv6?

I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk ...
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577 views

Routing Asterisk traffic evading a VPN

I have two offices set in two distant locations, connected through a IPSec VPN (using pfSense). On one office, I have an Asterisk PBX where all the extensions connect, and the Asterisk PBX is ...
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409 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem /...
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2k 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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1k 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 ...
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2k views

Remove local area code on outbound calls, and remove “dial 1” for long distance in Trixbox/Asterisk

We have Trixbox CE 2.8 and Asterisk 1.6. I want to remove the 933 area code so you can just dial 555-5555 instead of 933-555-5555 for local calls. I also want to remove the +1 for long distance so ...
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1k views

FreePBX How to play an announcement for misdialled calls

Our Asterisk-based PBX (FreePBX "Tango" distro) does not require any prefix for outgoing calls. The system is able to recognise external numbers and route them appropriately. Some of our users who ...
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1k views

GUI for Asterisk 1.8.8 on Debian 6 [closed]

Can someone point me towards a tutorial for installing either FreePBX, or a Webmin GUI for Asterisk 1.8.8 on Debian 6? Asterisk Thirdlane Multi Tenant PBX Manager and Asterisk Thirdlane PBX Manager ...
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519 views

Asterisk and sip station codecs

we have a small instance running on ec2 of voxilla/FreePBX-Asterisk-1.6.0.5-i386.manifest which has been updated as far as i can see We signed up for sipstation (because it removes half the ...
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6k views

Asterisk - inbound calls from SIP DID trunk “rejected because extension not found”

I have a DID with DIDforSale that's pointed at my Asterisk server. When I call it from my landline phone I get the AT&T disconnected line recording. The Asterisk CLI shows the error message: [Oct ...
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2k views

Asterisk DID incoming rules =s

So I have a 10 way SIP trunk from a UK provider. Calls in and out work fine, but as I have multiple numbers for different purposes I set up the inbound rules for each number but they never work, only ...
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513 views

SIP ( Asterisk ) CoS solutions with no VLAN's?

I have a Asterisk box actually two running pure asterisk. No GUI i.e. FreePBX, TL etc. The asterisk box is connected to a an EX4200 VC Stack, the EX4200's then connect to our headquarters via a Qwest ...
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2k views

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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133 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 CPU,...
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467 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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886 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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5k 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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966 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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672 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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1k views

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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2k views

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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877 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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40 views

Asterisk: How to enable the HTTPS mini server?

I am attempting to enable the HTTPS mini server in Asterisk with little luck. Here is my http.conf: [general] enabled=yes bindaddr=192.168.254.26 bindport=8088 tlsenable=yes tlsbindaddr=192.168.254....
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79 views

send fax with Elastix and Hylafax

I need help getting around "Unknown problem" while I am trying to send faxes using Hylafax and Elastix. I have an elastix 4.0 inside a VM, calling outside and inside the box is working great, using ...
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Having trouble with one way adio on an asterisk sip phone

I have an asterisk server (Elastix on CentOS 7) that I am currently running in Amazon Web Services. The server works great and clients can connect without issue using a stun server. However I have ...
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25 views

Voicemail location with two Asterisk servers connected by Follow-me

Here's our current setup: We have two Asterisk servers in different buildings setup and connected by IAX. Call them ServerA and ServerB. We have several DID trunks coming into Server A, where our ...
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46 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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80 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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572 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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238 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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96 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? ssrc=...
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765 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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307 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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207 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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483 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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94 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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209 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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3k views

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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841 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 => 1234,3,PlayBack(vm-...
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2k views

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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2k views

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 ...