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 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
513 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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277 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
151 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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233 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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115 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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181 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
186 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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115 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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264 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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62 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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905 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
758 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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785 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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115 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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699 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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137 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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532 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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111 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
3k 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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291 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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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
108 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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852 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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165 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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294 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 ...
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2answers
1k views

High Load, low CPU, Memory and Disk IO - Highend Server

This issue has been bugging me for the past several days having spent over 40 hours investigating this issue intensively. Effectively we run asterisk 1.4.42 which I understand is old, however is the ...
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1k views

Asterisk and websocket: “rejecting secure audio stream without encription details”

I'm trying this library using the test service: http://tryit.jssip.net/ When I register, using WS, not WSS, peer is registered succesfully When I make a call i got "rejecting secure audio stream ...
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1answer
81 views

Asterisk console output 'no button pressed' on no DTMF input

I've got an outbound survey program, here is an example of what the survey does from the dialplan. same => n(xtn_age),Read(AGE,rbp_age,2) same => n,Set(CDR(age)=${AGE}) same => ...
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3k views

How to install Asterisk with SRTP support?

I'm rather new to Asterisk, and I need my server to support WebRTC. As far as I know, Asterisk version in Asterisk Now is compiled without SRTP support, which is necessary for WebRTC. So, I try to ...
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272 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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3answers
498 views

Asterisk and SIP trunking, is this configuration possible?

I am new to asterisk and before I dive in, I just want to make sure that what I plan to do is possible/correct. My office will run an asterisk server and have both local and remote extensions. We ...
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2answers
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Interpreting and using the Asterisk “timing test” command

Timing is very important for certain kinds of applications in Asterisk. If DAHDI is the timing source, the dahdi_test command can be used to check the timing provided by the DAHDI kernel module. If ...
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1answer
270 views

Asterisk: Forcing a sip peer to connect via ipv6?

I've got an asterisk server that connects to an upstream provider over a WAN. The upstream provider supports both IPv4 and IPv6 connectivity, and the asterisk server is behind a NAT. When asterisk ...
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2k views

Get details about “Remote UNIX connection disconnected”

I am getting lots of: -- Remote UNIX connection disconnected I would like to get more details about it. How do I get information about who is connecting, ip address, and more details?
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302 views

Monit detecting error, but not restarting asterisk

I am running Ubuntu 13.04 with the corresponding versions of monit (5.5-6) and asterisk (1.8.13.1). I have monit set up to watch my asterisk log file for disconnection to my SIP provider and restart ...
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383 views

can't seem to install dahdi on my server

I am trying to install the dahdi module on my server but it keeps failing when i run the make all command. See below: root@server:~/dahdi-linux-complete-2.6.2+2.6.2# make all make -C linux all ...
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699 views

Asterisk trunk SIP

I have a problem with trunk SIP when I connect my Asterisk to my provider and the Internet connection is lost, the trunk SIP will be unreachable, the problem consist is all local extension are ...
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1answer
2k views

asterisk sip port on ubuntu server

I am working on Ubuntu server 12.04 I have asterisk working. I set the port 5060 as this: iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT iptables -A INPUT -p udp ...
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222 views

SIP (G.711) packets getting out of order

I have 2 asterisk servers on 2 different datacenters. Call is going from one to another. Suddenly audio quality decreased. I have made SIP traces on both servers. Here's what I've found: Server A ...
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397 views

Routing Asterisk traffic evading a VPN

I have two offices set in two distant locations, connected through a IPSec VPN (using pfSense). On one office, I have an Asterisk PBX where all the extensions connect, and the Asterisk PBX is ...
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1answer
368 views

Sending faxes from Asterisk (email to fax) — is any feedback possible?

I am about to make some email to fax gateway --- some python program will make Asterisk call files and put them into some queue (files or db based). Then another program should take "call tasks" from ...
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3answers
2k views

Registering SIP phone (X-Lite) to asterisk server (asteriskNow)

We currently have a school-project going in which we need to set up an PBX with Asterisk. Thus we have installed asteriskNow in a virtual environment (with virtualBox) on ubuntu 12.10. We tried the ...
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What is the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX for?

I am looking for the canonical definition of the “Allow Anonymous Inbound SIP Calls” option under “Asterisk SIP Settings” in FreePBX. Also, how does it relate to "Allow SIP Guests"?
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337 views

When to limit concurrent calls on Asterisk Server?

I'm interested in building an android app that calls another android app via the Asterisk server. It will be strictly SIP to SIP without connection to a PSTN. I've read you can have thousands of ...
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346 views

Asterisk server warning “Unexpected control subclass”

I am not sure why, but our asterisk server keeps having loads of agents (15) on status ready and calls waiting for them, it seems to be griving us this error in our event log. WARNING[7159] file.c: ...
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1answer
495 views

Asterisk can't make nor receive calls through a T1 PRI interface to a Cisco 2430 router

I have an Asterisk 1.8 phone switch with a Digium T1 card. It runs using 5ESS PRI through our present phone provider without a problem. However, we're contemplating switching to Time Warner's fiber ...
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4k views

No voice while making external calls - NAT configuration - Asterisk 1.8.13

As we were trying to setup our Asterisk server, we went on a huge problem: The inability to make call to or from external devices connected to the server. In fact, I can dial and answer the call on ...
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341 views

Asterisk / Voip phone redirects to last called number

I experience a very strange problem with our PBX (Asterisk 1.8, FreePBX, Grandstream GXP1200 phones) If I call an internal number (1020 in this case) the asterisk server redirects me to the last ...