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

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100 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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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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55 views

How to set Mitel 5224 ringer volume in configuration?

We have several Mitel 5224 SIP phones, but can't seem to find out to make the ringer volume setting persist after it reboots. Can the ringer volume can be set somewhere in the XML configuration file?
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280 views

Asterisk ${CALLERID(name)

I am working with asterisk (Asterisk 1.8.11.0), freepbx (2.x) and was building a dialplan (extension_custom.conf). I wanted to get caller Name from database which i have stored for example. CLI> ...
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326 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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8 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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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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134 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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121 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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193 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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181 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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490 views

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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91 views

Change the caller id from my asterisk server

I successfully set up an asterisk server. When people call my asterisk server via PSTN, the server will place another PSTN call to my phone at 33344455555. When I receive the call, my phone shows ...
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112 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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334 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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465 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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321 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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2k views

Configuring Hylafax to work with Asterisk over a single SIP provider

I'm setting up an Asterisk 1.8 based PBX, and as part of the requirements, I need to make provisions for sending and receiving faxes. The 'big picture' goal is installing a fax printer at each Windows ...
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146 views

asterisk Dial-plan?

I want to make a dial plan for asterisk to do this: for incoming-call check the caller-id and if caller id is equal with specific number (for example 666) then hangup that call.(this dial-plan also ...
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41 views

add presence for a remote user to a legacy telephone system?

we have a small call center that uses an old nortel phone system with analog lines. one of our sales people works from home so her calls do not go through the phone system. this creates a problem at ...
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287 views

How to send T.38 from a mac?

I'm trying to set up a fax-server on a macintosh. I have Hylafax, and we're going to use an internet FOIP fax provider (Haven't decided who yet, that may be another question). The problem is how to ...
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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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10 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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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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20 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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153 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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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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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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369 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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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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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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150 views

Iperf UDP listening on used port

I get WARNING[9157] chan_sip.c: Retransmission timeout reached on transmission error on asterisk and trying to troubleshoot the problem. Now asterisk is listening on 5060 port: udp 0 0 ...
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33 views

Adding milliseconds to Asterisk v1.4 log

I'd like to add milliseconds to the timestamp for an Asterisk v1.4 instance. I currently have the following entry in /etc/asterisk/logger.conf: dateformat=%B %d %H:%M:%S I'd like to add ...
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84 views

Asterisk SendText application

I am trying to send text message to the callee using the below config in extensions.conf exten => 5000,1,Answer() exten => 5000,2,SendText('hello world') exten => 5000,n,Hangup() And I am ...
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76 views

Asterisk virtual machine stops responding to network connections when I connect with Phoner Lite

I've installed Asterisk Now with PreePBX from bootable CD to a VirtualBox machine. Then I updated through FreePBX all the modules available for update and noticed a strange behaviour: After I ...
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50 views

check if CCSS callback is available

I've just discovered Call Completion Supplementary Services (CCSS) Is there a way for the dialplan to know if callback is available? If a call terminates with BUSY or NOANSWER, I'd like to offer ...
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72 views

Connect Cisco IP phone on two Asterisk sip servers

I have a cisco ip phone with sip firmware v9.3.1. I want to connect two lines on two seperate Asterisk sip servers. In section i can only define USECALLMANAGER (in node) and not the IP of an ...
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367 views

Soft phone not getting register to my asterisk server

I have config in my local lan asterisk for test. which is working fine. With the same OS version and packages on "AWS Cloud" I am setting as following. But it don't get register "error = sip 408 - ...
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11 views

integrating asterisk with Ravenclaw dialog system

I want to integrate Asterisk with AudioServer in Ravenclaw dialog management system, which is running on Galaxy Architecture. The purpose is to integrate voice calls with the dialog management ...
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246 views

Elastix Tranfser config files

I have installed Asterisk PBX to a new machine. What I want is to transfer configuration files from my running Asterisk PBX to the new one. I do not prefer the solution to backup my running Asterisk ...
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38 views

Asterisk pass-through fax to another Asterisk

I'm having two asterisk machines ("A" with external IP and "B" - local), and one of them (A) is like voip gateway. I have DID (for example 1(234)567-89-00) on asterisk "A" and I want to receive a fax ...
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134 views

Nagios nrpe commands error out with asterisk

as the nrpe user when Command: /scripts/nagisk.pl -c version Outputs: Asterisk 1.8.11-cert10 Using nrpe Command: /usr/lib/nagios/plugins/check_nrpe -H [HOSTNAME] -c check_asterisk_version Outputs: ...
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90 views

Can't listen to port 4445; sqlite3

So I have a dedicated machine running Ubuntu 12.04 server with FreePBX 2.9. Earlier I was attempting to familiarize myself with SQL via sqlite3, but I received this dreaded error: SQLite header and ...
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613 views

Avaya-Asterisk call drops immediatly via SIP (sip:anonymous@anonymous.invalid)

I'm trying to call from Avaya to Asterisk (11.5.0) in SIP. Avaya calls to Asterisk, then Asterisk rings my softphone but when I answer the call (in the softphone), call drops immediately. This is the ...
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212 views

using iptables to secure asterisk

Following the post at http://www.dslreports.com/forum/r26964868- I created some IP tables for my asterisk install. The server is behind a router and the router only had 5060 and the RTP ports ...
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778 views

Not able to register via SIPclient on asterisk server?

I'm unable to register a SIP client on my Asterisk server. What changes do I need to make to my Asterisk server configuration to fix this? Here are my configuration files for Asterisk. sip.conf ...
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10 views

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 ...