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 WaitForSilence NEVER detects silence

I am trying to use my dialplan to play recordings with WaitForSilence to make sure it wait until the person is done speaking or the message is left on voicemail. However, it doesn't seem to wait for 5 ...
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2answers
334 views

Asterisk new PJSIP driver security option

I have a working asterisk environment, but I get a lot of unwanted traffic, like sip scanners of people who even try to call as a guest. I'm using res_pjsip, the configuration is stored in pjsip.conf....
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1answer
323 views

why do all softphones fail to authenticate after receiving 401 unauthorized from Asterisk?

When I attempt to connect to the Asterisk server from CSipSimple, over a Cisco router, on the same network, Asterisk logs show the attempted connection: <-------------> [Mar 23 20:01:34] ...
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1answer
754 views

FreePBX UK Dial Patterns Route Through Specific Trunk

I've read a ton of links and tutorials and for some reason cannot get my dial patterns to work correctly. I've got Freepbx installed and working properly - no issue there. I want to create a dial ...
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1answer
944 views

Asterisk - very high Load Average with no reason

There is a problem with Asterisk server showing high Load Average I can not explain. Ny setup is: Asterisk 13.1.0 Linux 3.13.0-24 (Ubuntu Server) Dual socket (Xeon E5-2620) server, HT enabled - 24 ...
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1answer
30 views

Attaching Asterisk console to process

I want to open control console on currently running Asterisk instance. When am I supposed to use option -r and when -c in order to connect to Asterisk process? What is a main difference?
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1answer
1k views

WebRTC on standalone asterisk - no audio

After struggling with Asterisk for WebRTC for a few weeks now, I decided to put my problem on this forum. My Problem is as follows: Im not getting audio from WebRTC to WebRTC clients. I work in a LAN ...
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1answer
2k views

Can't dial through SIP trunk: FreePBX/Asterisk

I set up a SIP TRUNK in FreePBX/Asterisk that works perfectly for incoming calls. Here' s the relevant configuration: type=friend host=201.217.31.10 callerid=mynumber username=595XXYYZZZZZZ@prepago....
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508 views

Asterisk Sip Server and “Screen Sharing” function

I'am newbie in asterisk, i followed the tutorials on the asterisk wiki, and installed the latest version (13) of the asterisk server. I was able to setup the voip now i wonder, there is a function im ...
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1answer
869 views

Asterisk: execute shell command on server when call is accepted on SIP extension

I'm trying to configure asterisk to execute shell command for incoming calls - but only when the call is accepted. I've managed to setup extensions.conf so that command is executed when new call comes ...
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1answer
1k 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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1answer
2k 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
209 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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1answer
298 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 client)....
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0answers
494 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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2answers
250 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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1answer
179 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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1answer
713 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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1answer
99 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
210 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
898 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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1answer
242 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
1k 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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1answer
1k 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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2k 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
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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3answers
869 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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2answers
266 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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202 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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1answer
509 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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396 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 10.3.0....
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59 views

Asterisk test ISDN card

I have an AVM ISDN controller C4 plugged into an asterisk machine. It shows up in the PCI device list as: Digital Equipment Corporation StrongARM DC21285 (rev 04) How can I test this card to see ...
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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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1answer
651 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 http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/...
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1answer
2k 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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1answer
3k views

route incoming asterisk sip calls - fake auth rejected

we are trying to handle incoming sip_calls on our asterisk server, but somehow we always end up getting either 403 or 603 (which should be the same?) what should happen if it works: user calls ...
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2answers
2k views

Allow specific SIP connection FreePBX

Hey I am using FreePBX with firmware: 5.211.65-14 and service pack 1.0.0.0 On this asterisk server I have everything up and running, but inbound phone calls might be rejected: WARNING[56522][C-...
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397 views

Dial international number over dahdi trunk

I am trying to perform the following outgoing call: exten => _49.,1,Log(NOTICE,Dialing German number: ${EXTEN}) same => n,Set(route=DAHDI/g1/00${EXTEN}) same => n,Dial(${route}) exten =>...
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1answer
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 ...
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1answer
234 views

sip unreachable after changing routes using linux router

SIP server was unreachable after changing routes. I have 2 ISP ,ISP 1 is my default and ISP 2 is my back-up. I have a dialer behind linux router linux. Linux router has 3 ethernet cards. Router ...
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1answer
452 views

SecAst blocking IP's that fail2ban is not

I'm using SecAst on my Asterisk server as well as fail2ban. (Setup as an option as per the SecAst installation guide). I did this to test if SecAst is working right. I planned to remove fail2ban if ...
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1answer
48 views

Preventing SecAst access from across the network

I have SecAst up and running great, and I realize that any workstation on the network can access the SecAst telnet interface. Is there a way to limit this to only my laptop? I don't want one of the ...
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2answers
3k 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
39 views

secast init file already exists

I'm installing SecAst on a new computer and I'm on step 2.1.7 (copying init files). I copied the first initd file, but when I copy the second one it says file already exists (and I'm overwriting the ...
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71 views

SecAst: Potential documentation issue, Ubutunu 12.04.4 LTS and the command: cp -v

To Generation D: I am writing you about what might just be a documentation issue which hopefully doesn't have significant repercussions. This is something I encountered immediately at the start on ...
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67 views

Asterisk PBX Manager not saving changes

We are implementing Asterisk 11.4 and using the native PostgreSQL database to store call data. So far, we have not been able to change the admin password through the PBX manager. The password gets ...
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2answers
746 views

Installing SecAst on AsteriskNOW with CentOS

Having some issues installing SecAst for IPS, Followed the directions up to 2.1.6 and found a way (on this forum) to install qt5-qtbase (thanks) but when I run ldd /usr/local/secast/secast the return ...
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2answers
190 views

SecAst: Failing to Ban IP ''

To Generation D: This was the setup at time this issue was observed: secast-1.0.1.0-x86_64-ub12 on Ubuntu 12.04.4 Server LTS with Asterisk 11.10.2. The following events were captured and observed in ...
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1answer
53 views

SecAst: Premature Exit in Daemon Mode

To Generation D: This is a new problem with build secast-1.0.4.0-x86_64-ub12 that didn't happen with the previous build, secast-1.0.1.0-x86_64-ub12. When secast is run as a service (service secast ...
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1answer
3k views

Unable to connect to Asterisk via Telnet

I'm trying to connect to asterisk via telnet. In order for PHPAGI to use certain commands, it has to be able to do the same. So I'm cutting out the middle man and testing it manually. I've ...