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

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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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530 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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540 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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1k 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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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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466 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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1answer
572 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
685 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
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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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6k views

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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521 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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556 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
664 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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2answers
7k 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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1answer
497 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 ...
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1answer
237 views

Cheap PSTN Card with caller Id for Asterisk Linux server [closed]

For my house I just wanted to setup a caller Id system that would auto-reply on certain callers to avoid harements. Any sugestion for this? I'm thinking on a Linux + Asterisk + PSTN Card with caller ...
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38 views

ASA 5505 base license and Voice vlan [duplicate]

My current setup contains a cisco asa 5505 with two cisco sf300-24p switches, the data and voice vlans have been setup on the switches and I've started to setup the needed vlans on the ASA, but it ...
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5answers
306 views

VOIP Power outage strategy

In the past 2 years, we have had 4 instances where we have been without power for a period of 4-6 hours - due to construction crews cutting power lines, car accidents involving downed power lines, ...
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2answers
107 views

Trans-atlantic VOIP

We're a Canadian business on the east coast and will soon be opening up a new call centre in Australia. This new call centre will be handling our graveyard shift and will be overlapping with the ...
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1answer
307 views

Asterisk 911 dialing rule

Our company is looking to replace our outdated PBX with VoIP. One function we really like about our current PBX is that it alerts the secretary when a line dials 911 and it shows the extension where ...
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1answer
628 views

AsteriskNOW server routes outgoing calls as if they are incoming

I am very new to Asterisk so this is undoubtedly a misconfiguration on my end. Right now my test setup behaves as follows: When I make an incoming call to an FXO port on my analog card, Asterisk ...
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4k views

How can I find a log of calls message a specific extension in Asterisk?

Question says it all. Trying to view a log of calls to/from an extension using Asterisk. Thanks in advance.
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1answer
200 views

Polycom IP 650 Custom display

I've been googling all over the place and haven't found a good way to do this. I'm hoping that someone out there has done something like this. I have Polycom 650's and 331 deployed to my users via a ...
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1answer
238 views

Variable to store Asterisk extension dialed by user

I am using Asterisk for voice calling. I am using below context If user call back- [from-pstn] exten => _X.,1,Playback(demo-thanks) suppose my number is 74900 on which user can call back. ...
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0answers
472 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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1answer
607 views

Why does asterisk (and other systems) feel the need to regenerate in-call DTMF?

I am using Asterisk to interact with analog telephony devices that can be programmed and tested with DTMF interaction. Some of these guys speak rather quickly. Too quickly, you could convincingly ...
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1answer
406 views

Troubleshooting a failing SIP connection

I have a linux Asterisk server that is behind a firewall. The firewall has three interfaces : eth0 is the LAN, where Asterisk resides, eth2 is the default gateway to the internet (via an ADSL modem ...
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1answer
585 views

Networking issue with asterisk server

I replaced an IVR machine for incoming call after going down. It is running asterisk 1.4.23 on ubunutu 10.04 I decided to put the server behind iptables because my server was under brute force attack. ...
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2answers
1k views

PBXact Behind ASA 5505 nat udp range 10000-20000

I'm installing a new pbxact phone server onsite and I've ran into an issue using a 3rd party sip, I've got the acl's for the traffic created and the traffic is allowed, I'm assuming i need to create a ...
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1answer
1k views

Asterisk IVR - input variable and go to an outbound URL

I have an Asterisk outbound IVR system set up, and I'd like it to talk to my LAMP stack on a different server. I've set up the PHP end to receive data from the IVR server and record it in MySQL, but I ...
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1answer
750 views

Asterisk Connection not working

I have installed Asterisk on VirtualBox by following the steps from here. Everything went ok until I got to navigate to an IP to configure Asterisk using FreePBX: 10.0.2.15 (Shouldn't be something ...
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Inbound SIP calls through Cisco 881 NAT hang up after a few seconds

I've recently moved to a Cisco 881 router for my WAN link. I was previously using a Cisco Linksys WAG320N as my modem/router/WiFi AP/NAT firewall. The WAG320N is now running in bridged mode, so it's ...
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1answer
2k views

asterisk peer with SIP provider through proxy

I'm trying to make a asterisk server connect to a SIP provider (which offers PSTN origination and termination). I've been reading the great Definitive Guide, but it doesn't seem to cover the case ...
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2answers
2k views

Accept calls on external cell phones - FreePBX Distro

We installed FreePBX distro last week and configured it so that when someone calls in, it goes to all the phones so someone here can answer the call. After trying all the internal extensions, it ...
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1answer
321 views

can't register a soft phone to asterisk11

I have a VM (on oracle vbox) running Fedora17. I've installed asterisk 11 on it from sources. I've followed the wiki for installation (https://wiki.asterisk.org/wiki/display/AST/Creating+SIP+Accounts) ...
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Determine which user initiated call in Asterisk

I had the following code in my extensions.conf file: [local] exten => _NXXNXXXXXX,1,Set(CALLERID(name)=${OUTGOING_NAME}) exten => _NXXNXXXXXX,n,Set(CALLERID(num)=${OUTGOING_NUMBER}) Now I ...
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1answer
202 views

Parse and validate Asterisk dialplan before commiting

I recently made a number of changes to my Asterisk dialplan and would like to validate these changes before I commit. I am thinking more from a "write code" -> "compile" -> "debug" approach. I am ...
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1answer
644 views

Asterisk outbound calls hanging up imediatellly on Zap trunks

The inbound calls work fine. Outbound calls fail as follows: Dial("SIP/xxxxxxxxx","ZAP/g0/14442223333|300|") Couldn't call g0/14442223333 Hungup 'Zap/2-1' Everyone is busy/congested at this time ...
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1answer
578 views

How to establish SIP connection, when SIP-proxy is required?

I have Asterisk/1.8.13.1 Asterisk GUI-version : SVN--r Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16. ...
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1answer
413 views

Provisioning SIP Phones over the internet

I have a few SIP Phones that are located of site and connect to my PBX over the internet to make calls. For some reason one of these phones has become unprovisioned. In my office phones get ...
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1answer
943 views

Asterisk dialplan context and label clarifications

I have been learning Asterisk dial plan for the past week.I have written down a simple IVR system with two levels of menu and an exit option.I have used concepts from different tutorials on the ...
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1answer
776 views

Configuration of a Cisco IP Phone 7916 and 7975g

Hi everyone, I would appreciate some help configuring my Cisco IP Phone 7975. In fact the phone works well with the SIP firmware (version SIP75.8-4-1ES2) and an Asterisk Server. Now I would like to ...
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3answers
3k views

How do I secure my Asterisk server?

Our asterisk server was compromised. some calls were made to Asia countries last weekend. Thought we have improved our network configuration, we still want to determine how the intrusion was done, we ...
2
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1answer
2k views

cant make outbound calls - asterisk

I have a basic Atcom IP01 with the following config Registered Voip (SIP) Trunk Registered Voip Phone - ext Dial Plan Outbound Call rule I made use of this manual that the manufacturer supplies: ...
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2answers
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Can anybody explain why to use Asterisk PBX rather than standard PBX from Matrix or Siemens [closed]

Can anybody explain why open-source Asterisk is favourable in comparison with a traditional PBX system from a vendor such as NEC/Syntel/Matrix or Siemens.
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2answers
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Outgoing calls from vtiger to an asterisk extension

I've successfully installed vtiger and asterisk on Ubuntu, and I managed to configure them such that I get an incoming and an outgoing popup in vtiger. The problem is that when I click on an extension ...
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1answer
859 views

Elastix and two ISDN lines

Since I am not an expert on communications, I will describe everything from the user perspective. We had a PBX with elastix embedded and two ISDN cards in it (we have two isdn lines). The mobo of the ...
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2k views

asterisk queues priority and penalty

queues.conf shared_lascall=yes strategy=rrmemory wrapuptime=15 A1,A2,A3 are members of 2 queues: queue1(Q1) and queue2(Q2) A3 has penalty 3 in Q1 where min/max penalty are 0/3 and A3 has penalty ...
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998 views

Automatically reconnect to ODBC sources?

I am using Asterisk 1.8.10.1 and a MySQL database connected via ODBC to store CDRs. When my MySQL database isn't available when Asterisk starts or has an outage while Asterisk is running, I would ...
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2answers
1k views

Timing Source on Asterisk [closed]

I have tried to look for simple language of explanation what is timing source for in asterisk? and why it has significant impact on the asterisk Anyone can give me insight will be appreciated