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 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
47 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 ...
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2answers
734 views

Asterisk Outgoing CDR Logging To Mysql

Trying to utilize the cdr logging (to mysql) using custom fields. The problem I am facing is only when an outbound call is placed, during inbound calls the custom field I am able to log no problem. ...
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2answers
1k views

elastix cdr stop working

CDR was working before 19 march. Unfortunately i dont remember what kind of changes i made to configuration, but this exactly not changes to CDR config. elastix 2.4.0 asterisk 11.7.0 mysql 5.0.95 ...
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251 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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2answers
104 views

Blocking geographic cities from accessing Asterisk using Secast

I am using Secast for intrusion protection on my Asterisk PBX. It’s working great, and I now want to start blocking specific geographic regions. My system is getting hammered from Ramallah ...
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1answer
1k views

Asterisk server running on loopback address instead of local ipv4 address

I installed Asterisk on my local Ubuntu 12.04 machine. After starting the asterisk [asterisk -c] I ran the command netstat -nlpa | grep asterisk and the output shown is this -> tcp 0 ...
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170 views

secAst lib prerequisites too new for Debian/Utuntu 12 LTS?

as requested in FAQ I wrote here the question (there is no question for this product yet on this site?) This Asterisk securing tool looks very nice so I tried to test it out: ...
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117 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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2answers
8k views

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
1k 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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186 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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1answer
1k 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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1answer
116 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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1answer
250 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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197 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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673 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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1answer
8k 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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1answer
12k 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
128 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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1answer
503 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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287 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
373 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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1answer
3k 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
677 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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347 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
185 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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280 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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2answers
134 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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216 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
242 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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145 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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1answer
431 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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79 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?
2
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1answer
1k 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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2answers
1k 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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4answers
2k 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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1k 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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155 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
886 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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153 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
633 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
120 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
4k 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
336 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
48 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
127 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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1k 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
182 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 ...
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1answer
363 views

Asterisk PBX on linux (CENTOS 6.2) hangs up calls that are being recorded after 33 seconds

i have an issue with my calls being made via ASTERISK PBX running on linux CENTOS 6.2. The use case is that a call is triggered from /var/spool/asterisk/outbound/ The caller is dialed The dialplan ...