* You are viewing the archive for July 13th, 2012

How To Set SIP To Auto Answer In The Dial Plan .


I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone.

- If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup.

Regards Upendra.

Recommended VOIP Monitoring Tools

As system administrator, monitoring the continuity of services is vital. Today I would like to highlight some tools that could come in handy for VoIP monitoring.


For those of you who didn’t know it, Nagios can be configured to monitor pretty much anything you want, including Asterisk servers. Actually, with Nagios the (much) harder part is deciding what’s relevant to monitor, and what your alarm thresholds should be set at.

Some VoIP community members have reported that they used Nagios to monitor ~4,000 hosts and about 8,000 to 10,000 services before they started running into scaling problems on a single box.

For this purpose you might want to take a look at Nagios’ NSCA (Nagios Service Check Acceptor), is a Linux/Unix daemon that allows you to integrate passive alerts and checks from remote machines and applications with Nagios. It’s useful for processing security alerts, as well as redundant and distributed Nagios setups.


This is a free and OpenSource Software written in Perl written by Tobi Oetiker, (the creator of MRTG and RRDtool) that keeps track of your network latency and can provide SIP Ping Probe capabilities. Between the functionality list it provides, we have:

  • Outstanding latency visualisation.
  • Interactive graph explorer.
  • Wide range of latency measurment plugins.
  • Master/Slave System for distributed measurement.
  • Highly configurable alerting system.
  • Live Latency Charts with the most ‘interesting’ graphs.

Monitor PBX (or Монитор АТС in russian, it’s original language)

The homepage of this tool (which is completely in Russian) says that it is an open source utility that allows real-time visualization of key performance indicators Asterisk based telephony server. And that it is designed to evaluate the performance of the system under test, and in commercial operation.

(I think they should really consider adding an English version of the website)


Sure there is a big span of very useful and extraordinary tools around. So far, taking ideas from the Asterisk Community I could recommend this two. If you know about any other tool that could be use for this purpose, don’t hesitate to contact me and I’ll get sure to mention it here.

Last Update: 01-Oct-2012

Trouble With Asterisk Behind Router

Hi guys!

I have a some non standard problem when I register my asterisk into “My SIP Provider” . The trouble is: my asterisk stay behind router with port forwarding, who have Public IP ( – for example), asterisk have a private IP

From “My SIP Provider” cabinet I see:
, but I need from (example):

From wich trukes in linux or asterisk technology I need?
Can you help?

Best regards, Nikolay G. Petrov!