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

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Bringing people into an Asterisk conference call

I'm using Asterisk 1.4 and am trying to work out a way to bring people into a conference call. In the ideal scenario two people would be talking and one of them would push some keys, then a phone ...
3
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1answer
3k views

Asterisk - Trying to use call files to create a conference call between two dynamic numbers

I'm trying to setup an Asterisk system that will allow me to create a conference call between two dynamic numbers. It seems I can use 'call files' to make Asterisk initiate the call without needing ...
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1answer
823 views

Calls cut off in Asterisk 1.6.2

We are offering hosted VoIP. The Asterisk servers are running in our data center and customers connect their Aastra phones remotely to it. However, I have one customers connected to his own PBX and he ...
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2answers
2k views

How to enable CDR on AsteriskNow 1.5

I have upgraded PBX to Asterisk 1.6.2.7 and now CDR files are not created. It looks that such logging is disabled: Connected to Asterisk 1.6.2.7 currently running on pbx2 (pid = 5824) Verbosity is at ...
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1answer
3k views

Asterisk cached an old dial plan? rejected because extension not found in context?

I'm trying to fix my asterisk server which has been quite stable up until recently. Here's my problem: a) if i run a sip reload and/or dialplan reload from asterisk terminal, it sometimes breaks the ...
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1answer
25 views

Asterisks Dialplan Extensions

I'm playing around with a very simple asterisks setup. My asterisks server is 192.168.1.10 (Ubuntu 12.04), my 2 sip clients are 192.168.1.20 and .21 (both using ubuntu 12.04 and Jitsi as a sip ...
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0answers
30 views

Asterisk behind NAT sets wrong Contact Header

I'm using SIP with asterisk 13.1.0 behind a statically configured NAT. The servers private_ip differs from the public_ip, where I can reach it. I've already set these options in the sip.conf file. ...
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18 views

Recursion in phpagi script of asterisk server [closed]

I am using this in extension file which execute the my phpagi script:- exten => s,n,Read(NUMBER,,4) exten => s,n,agi(a.php,${CALLERID(num)},${NUMBER}) And this in my phpagi script:- ...
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2answers
1k views

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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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
47 views

Asterisk - Send voicemail to email as client SMTP address

I'm able to setup Elastix (Asterisk) to send email voicemail attachments without any issue. But I'm trying to email them to a particular external address and it seems to be rejecting the emails as ...
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1answer
1k 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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0answers
328 views

How to forward a call to SIP server

I have a SIP server(Asterisk) I have configured a DID for outbound and inbound. My asterisk server has freepbx installed to manage it. My requirement here is, when anyone calls to this DID call should ...
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1answer
25 views

routing for iax-protocol does not work on pfsense

I have problems with a pfsense-box routing iax-traffic (UDP port 4569). Our company has a central office and eight branch-offices. At the branch-offices we use pfsense as firewall and VPN endpoints ...
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2answers
709 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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29 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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0answers
28 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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1answer
140 views

configuring command prompt in Asterisk CLI

I am trying to set up color prompt in asterisk CLI. In the documentation I have found this: %Cn[;n] Change terminal foreground (and optional background) color to specified. A full list of colors may ...
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5answers
1k views

Skype-SIP gateway

Does anyone know if you can setup a Skype-to-SIP gateway on your network? I tried Uplink, which works wonders, but only on Windows and with Skype on the same machine... It would be awesome if you ...
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0answers
23 views

Finish extension (avoid dialplan to silently continue in the next priority of another extension)

I have run into a strange behaviour: [ctx-test] exten => h,1,NoOp(*** hangup ***) exten => _.,1,NoOp(*** ${EXTEN} ***) same => n,... same => n,... same => n,Hangup() Strangely, ...
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31 views

Asterisk Firewall access Rules

I have a questions regarding the firewall rules for Asterisk VoIP server. I have a restricted access to UDP/TCP port 5060, which seems to be blocking calls. I want to make sure that my VoIP Server ...
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9 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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1answer
17 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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1answer
499 views

Elastix “asterisk -r” command not working “-bash: asterisk: command not found”

I'm having trouble administering the current version of Elastix from the command line on a CentOS 6.3 box. I believe the command I'm look for is asterisk -r however I cannot figure out how to access ...
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2answers
82 views

Asterisk, GSM-FXO gateway example of usage

Can anyone explain where to use GSM-FXO gateway. I understand where to use GSM-FXS - for example to connect analogue telephone to gsm gateway. But do we use GSM-FXO to connect to two telephone ...
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1answer
72 views

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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1answer
107 views

Asterisk can not send fax via T.38

I am a newer on asterisk. when I tried to send fax, I can not get success. After doing a lot of reseach, I decide to ask my question at here. I hope someone could give me some idea to correct my ...
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1answer
107 views

Call duration limit & least cost routing

We have Elastix version 2.4 setup and we have some little questions which if you could please answer to solve via web based panel of Elastix, as agents are making calls via Elastix there are little ...
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2answers
4k views

Asterisk block incoming call with certain Caller IDs

So I know this is a fairly simple question, but I can't seem to find a good answer to it. I have a DID which is getting hit fairly hard (or at least for me, about 1-3 calls per day) with telemarketers ...
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249 views

Asterisk 11.* TLS Configuration

I have installed asterisk 11.12.0 on CentOS (64-bit). When enabling the TLS support i get the following error WARNING[7620]: tcptls.c:673 handle_tcptls_connection: FILE * open failed! i tried ...
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1answer
644 views

Asterisk Real time does not perform register with SIP TRUNK provider

I am using astersik real time (dynamic). I have entered my SIP trunk details into the sippeers tables. However, the sip trunk does not perform a register with the SIP TRUNK providers servers as it ...
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2answers
989 views

PBXact Behind ASA 5505 nat udp range 10000-20000

I'm installing a new pbxact phone server onsite and I've ran into an issue using a 3rd party sip, I've got the acl's for the traffic created and the traffic is allowed, I'm assuming i need to create a ...
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3answers
170 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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2answers
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exit: 1 [FATAL] Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting

I am struggling with this error after an Elastix upgrade: exit: 1 [FATAL] Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting From what I can see the manager is ...
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1answer
354 views

Installing Asterisk inside a CentOS docker image on Xubuntu

I'm trying to install an Asterisk server inside a CentOS 6.5 docker image running on Xubuntu 14.04. I downloaded Dahdi from wget ...
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49 views

Asterisk GOIP Callerid not working

Have anyone made this work with GOIP or other similar voip gateways with asterisk? I think I have exhausted all possible options regarding setting a caller ID and it still doesn't work. The recipients ...
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1answer
448 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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90 views

Debian DHCP for voip option 66 using tftp

I have configured my tftp and I am able to send configurations from the windows server when it runs dhcp option 66 tftp to my server but when I try to do it from the debian isc-dhcp-server I get ip ...
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1answer
142 views

Polycom IP 650 Custom display

I've been googling all over the place and haven't found a good way to do this. I'm hoping that someone out there has done something like this. I have Polycom 650's and 331 deployed to my users via a ...
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2answers
59 views

Asterisk: Join 2 inbound calls

I have a VoIP box setup with Asterisk and using chan_dongle to provide me with an inbound GSM trunk as well as a couple of DID SIP trunks with local numbers. I would like to be able to have the ...
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1answer
401 views

Replacing Asterisk with MS Communication Server

We are examining replacing our Asterisk PBX server with MS Communications Server. We use Asterisk for the following reasons: VoIP telephony mass calling by streaming messages via SIP to our ...
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2answers
846 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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2answers
4k views

PSTN Trunk TDM400P Install on Asterisk / Trixbox

Hey All, I'm trying to get a TDM400P card with FXO module to connect to our PSTN line. The card is correctly detected by Linux: [trixbox1.localdomain asterisk]# lspci 00:09.0 ...
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65 views

Using IAX2 to call from one Asterisk server to another

I have two asterisk servers serverA and serverB . I am trying to place a call from serverA to use serverB using IAX2. I have defined a user on serverB in iax.conf. When making the call this is what I ...
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2answers
986 views

Asterisk SIP Trunks and routing

I have two asterisk based PBXs, each one of them on a diferent LAN, both LANs connected with a router. First pbx is 10.0.0.1/24 and second one is 10.0.2.1/24. Both pbx have their own extensions and ...
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1answer
75 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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3answers
6k views

How to configure Asterisk to send audio before call is established

Is it possible to configure Asterisk so that it sends RTP packets with audio from the receiver before the remote party actually picks up? This seems to be required for a VoIP compliance test my ...
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1answer
1k views

Sip configuration for Elastix, for Terrasip (Provider) with Multiple DIDs

I have just purchased several DIDs with Terrasip I have configured my Elastix (freepbx) SIP trunk according to their suggestion: Please follow this template configuration. (valid for outbound/inbound ...
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1answer
108 views

Asterisk relay between multiple subnets

I wonder what's the best way to go when you have phones on multiple networks which are not directly reachable. I have 3 networks 10.3.x.x 10.6.x.x 10.17.x.x My asterisk server resides on the ...
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2answers
1k views

Interpreting and using the Asterisk “timing test” command

Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If ...