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 udp 5060 recv q stuck

I saw a lot of discussion here over about this issue but I still didn't got my answer. I run asterisk 1.4.36 CentOS. Every now and then, the receive q is getting stuck on asterisk process or even ...
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15 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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11 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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9 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
17 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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13 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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21 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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6 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
9 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, ...
2
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1answer
101 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
57 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
45 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
55 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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154 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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40 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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71 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
301 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
112 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
52 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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49 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
62 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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1answer
91 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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25 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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48 views

Snom sip phone strange behavior

Didn't know where to ask this question, if someone can point me out the correct place, i'll be grateful I will be grateful, if someone will be able to help me with the following, I have around 200 ...
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372 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
331 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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1answer
438 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
600 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
585 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: ...
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159 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
494 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
113 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
125 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 ...
2
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1answer
30 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
875 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
23 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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2answers
34 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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0answers
39 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
169 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
89 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
30 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
628 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 ...
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1answer
41 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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2answers
1k views

How do I 'activate' voicemail on an extension on asterisk-Freepbx

I have asterisk-Freepbx (Version 12) hosted on a debian 7 server. I have created an extension (Cisco IP phone SPA 504G). I am able to dial in and out. The extension is configured to go to voicemail ...
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2answers
260 views

Asterisk inbound trunk unauthorised without allowguest=yes

I am trying to configure an Asterisk (Elastix) box to receive SIP calls from a provider without requiring allowguest=yes to be enabled in sip.conf. Basically the SIP trunk provider uses multiple IPs ...
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75 views

Switch from E1 to Analog PSTN line on IP-PBX failure

I have two Asterisk servers, E1(PRI) Gateway Asterisk 1.4 with E1(PRI) card IP-PBX Asterisk 1.8 on VM Server When my e1 gateway is failed (ex. shutdown, no power...), PSTN automatically switch to ...
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1answer
197 views

SIP INVITE packet has WAN address rather than call manager LAN IP

I am using SIP between two subnets (192.168.3.0/24 and 192.168.30.0/24) each connected via VPN. I have a call server on 192.168.3.100, and two phones 192.168.30.118 (Ext. 3128) and 192.168.30.119 ...
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1answer
129 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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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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44 views

Starting SecAstQFSFileEngine::open: No file name specified - error when starting secast

I'm receiving the below error when starting secast service secast start Starting SecAstQFSFileEngine::open: No file name specified [FAILED] Has anyone run into this issue before? I'm running CentOS ...