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

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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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17 views

CentOS 6.5 Lampp (Xampp) and MYSQL CDR

I was wondering if it was possible and if so, how to use Lampp's MYSQL to store the call detail records in. I've tried installing the mysql-devel package and i've tried changing the modules.conf, ...
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13 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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28 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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1answer
29 views

Cannot access Asterisk sip from remote location

I've used this tutorial to configure my iptables so I can access asterisk sip from outside of my server. My iptables -L looks like this Chain INPUT (policy ACCEPT) target prot opt source ...
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39 views

Asterisk - Trigger when an extension becomes reachable/unreachable

We have people in our office that use softphones to connect to our Asterisk system. The softphones are online when they are in the office, and offline when they are not. So the Reachable/UNREACHABLE ...
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44 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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51 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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88 views

Asterisk and SIP behind NAT

I am trying to Setup an Asterisk-Server to accept calls from a client in an other Network. The Server and the client are behind an NAT. I have already activated STUN on the client, but I am still ...
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113 views

Asterisk 401 Unauthorized when trying to register sip clients

Trying to register a sip client to my asterisk server often (just about 90% of the times, not always, weirdly) results in 401 Unauthorized errors. This is the config for one of the extensions: [11] ...
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1answer
34 views

Asterisk showing busy

I am trying to pass call via sip phone and I was success when I tried for the first time but after that I can't connect the call showing the message below. Can any one give me a solution why this ...
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6 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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1answer
41 views

Asterisk, How can i play an audio file [closed]

Here is the dial plan [testInComingCalls] exten => s,1,Answer exten => 30953025,1,Dial(SIP/20000,20) I would like to play an audio file as soon as somebody answered the call (Welcome Message)..
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57 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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143 views

Asterisk: Retransmitting again and again

I have following sip settings Name : external Description : Realtime peer: No Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : public ...
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1answer
42 views

Empty Asterisk 12 CDRs in SQLite3 database

Using: CentOS 6 Asterisk 12 cdr_sqlite3_custom.so cdr_sqlite3_custom.conf (deliberately simplified): [master] table=>cdr columns=>calldate, clid, test ...
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1answer
47 views

Security, hardening and NAT for RTP and SIP

I decided to create a VOIP server and it has gotten more complicated then I thought it would. So in order to make my server more secure, I keep it behind a pfSense appliance. I use IP filtering to ...
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11 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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1answer
190 views

Asterisk AMI - get detailed extension status

I am using AMI from Python. I guess the same can be achieved from the Asterisk CLI. I want to be able to get details about every extension in the PBX. More specifically, for every extension I want to ...
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2answers
40 views

asterisk - use a different “SIP CoS mark”

On my asterisk server, when i do a sip reload, i get the message "Using SIP CoS mark 4", followed by a registration time out. I need to make that statement say "Using SIP CoS mark 5". How do I ...
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62 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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3answers
89 views

What is the use of echo canceler in ISDN boards if all my endpoints have echo canceling?

Most digital telephony boards (BRI/PRI) have optional hardware echo canceler presented as alternative to the cpu intensive software echo canceling available in Asterisk and FreeSWITCH. I'm ...
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62 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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1answer
108 views

Asterisk realtime table for confbridge app

I try to configure asterisk with realtime config (configuration stored in mysql database), I configured my IPBX but I don't find a solution for conference application, I found only the table structure ...
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28 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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58 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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1answer
72 views

An extension in use receive more than one call with leastrecent strategy

I have a queue with some members. The strategy is leastrecent. Example: Members: 101, 102, 103, 104 A call goes to 102, and 102 answer the call, and then, the ext. is in use. Another call go ...
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40 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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46 views

How to display incoming trunk name at called device end with Asterisk?

We have an Asterisk with FreePBX, and using Grandstream and Panasonic VoIP phones. Now when an incoming call rings on the phones only displays "Line 1" (as it is configured at account 1 on phone) ...
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61 views

Asterisk Register username with special character like “@”

I am using a SIP provider that has provided me with a username like: +1122233344@aaa.bb.com (Note this is only the username part) And has a numerical password. My Register string looks something ...
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16 views

Alarm when a queue is unattended

Is there a way to make a trigger/alarm when a queue is left with no one to attend it? My setup is freepbx and CDR-stats, but it could be a plain Asterisk solution as well.
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40 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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50 views

FreePBX / Asterisk queue leastrecent Breaks on transfer

I am running into some weird issue's with queue's, It almsot seem's as if when you transfer, or go on hold it does not reset the lastcall time Scenario – John is on a call with a customer. Mary is ...
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90 views

DRBD show StandAlone on Primary node after secondary node attach with it?

I am using Centos5.4 & DRBD8.2.6 heartbeat2.1.3 & one java program with Mysql & Asterisk1.6.2.6. On DRBD disk i put 2 service Asterisk & Mysql. SO if DRBD Disk not up both services ...
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1answer
137 views

Limit simultaneous call per user in Asterisk

I want to limit simultaneous calls per extensions in Asterisk for security reasons. For example when a user is on the call no body else would be able to make a call by that extension. How can I ...
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1answer
205 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 => ...
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102 views

Asterisk Call pickup group picking up wrong call

While executing call pickup within a call pickup group via the soft button on our phones the wrong phone call is picked up. Example: phone A calls phone C and phone B tries to pick up the call. Phone ...
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3answers
424 views

CentOS 6.4 running Asterisk 11.5 with high load, low CPU/memory/disk IO

I have an Asterisk server doing automated calls and I'm noticing an unexplained high load in it. The server is only running Asterisk. Database and other support applications run in a different ...
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137 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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42 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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1answer
171 views

Can't receive phone call on my FreePbx extension

I'm running out of hair to pull here. I have my freepbx installed and configured ok at the beginning. I was testing it with Google Voice account. I can call out and receive call. I'm sure I must did ...
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69 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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1answer
167 views

Asterisk behind a firewall

we have a bit a sticky problem on our Asterisk server, which we are struggling to resolve, I'm hoping someone with more knowledge than me can help. We are running Asterisk 1.8.23.0 on Centos 6.4 and ...
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1answer
50 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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1answer
600 views

How to configure a 2821 ISR to an asterisk PBX on a PRI line?

I'm trying to move my company's phone system from Cisco's CUCM to an asterisk solution. I have setup a PBXinaflash server, and setup an account with www.vitelity.com. I was able to get it working. ...
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1answer
112 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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1answer
40 views

Advice respect an architecture PBX [closed]

i'm new on the world of PBX and network infrastructure. I worked in a model and wanted to know how feasible it is to implement it in a production environment. The requirement is to create a solution ...
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1answer
58 views

How do I set the current working directory of Asterisk?

I need to be able to collect core dumps, but since Asterisk's CWD is / (so claims procfs), it'll never be able to write them. I've confirmed my suspicions by allowing world write to / and SIGABRT the ...
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298 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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3answers
112 views

make web portal visible on two different vlans on same interface

We have recently set up a new Aterisk pbx server based on our previous pbx (also Asterisk) which was installed by an external company. We've kept the configuration of the server the same, all we did ...