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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.

Nagios

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.

SmokePing

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

Clipping issue with SIP over satellite

I’m having a wierd clipping issue with one employee who’s using a phone
over a satellite Internet. He was sold that system specifically for use
with VoIP. Ping times show average round-trip time as around 700 ms with a
range of 560 to 841, so considerable jitter.

Things work fine when he’s talking to another Asterisk phone or to a SIP
trunk provider, but when connecting to a T1, there’s clipping where about
1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds
like an echo canceller conflict, but I’ve set echocancel=no in
chan_dahdi.conf (I have hardware echo cancelling) and it didn’t do
anything. I’m forcing his codec to G729 for bandwidth reasons. The
phone is an Aastra 6757iCT.

Does anybody have any suggestions here?

app_swift beta release

Hi folks,

Just a note to let everyone know I’ve finally finished up the new BETA release of app_swift (now v3.0.1 b1).

This release introduces some pretty major changes to app_swift such as:

– The entire code-base has now been unified and the build system auto detects which Asterisk version you’re using (yay! one branch!)

– Auto-detection and support for both the Cepstral 5.0 and 6.0 engines

– Support for Asterisk 1.4, 1.6, 1.8, 1.8 Certified, and 10

– Asterisk 1.2 support has been dropped.

I have only been able to do some basic testing with all these permutations of Asterisk and the Cepstral engines on a few of my machines here at the house and need some volunteers to help out and be guinea-pigs.

Please email me directly with any feedback you might have.

I’ve updated my github repo with the new app_swift code which can be downloaded using git.

git clone git://github.com/dmsessions/app_swift.git

Thanks,

– D

Music instead of Ring Ring

Hello,

How can I achieve to play a music file instead of typical ring ring
(something like MusicOnHold)…

I have the following dialplan, the time when the user calls the context is
executed and the system calls both the user and I hear a Ringing sound.

[inc-call]
exten => s,1,Dial(DAHDI/i1/USER2&DAHDI/i1/USER1,20,A(sound-file))

Please suggest

Getting unwanted pager email from Asterisk voicemail

My guess is that your email provider is forwarding the message since
Asterisk should send the same content to both places.

From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Duncan
Turnbull
Sent: Thursday, May 31, 2012 6:33 AM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: [asterisk-users] Getting unwanted pager email from Asterisk
voicemail

Hi All

I am not sure why but I am getting a pager email as well as a voicemail
email when a voicemail is left. I am guessing its a setting somewhere but I
can’t find it

The system is Ubuntu with Asterisk 1.8.12 from source. I am using Freepbx
for the configs but freepbx doesn’t do much to voicemail

The mail system is Postfix

My test scenario

[general]

format=wav49|gsm|wav

serveremail=asterisk@questterrace.co.nz

attach=yes

skipms=3000

maxsilence=10

silencethreshold=128

maxlogins=3

emaildateformat=%A, %B %d, %Y at %r

pagerdateformat=%A, %B %d, %Y at %r

sendvoicemail=yes ; Allow the user to compose and send a voicemail while
inside

[default]

121 => 1234,Duncan
testing,duncan@e-simple.co.nz,,attach=yes|saycid=no|envelope=no|delete=no

I get the voicemail with attachment

Subject [PBX]: New message 1 in mailbox 121

Dear Duncan testing:

Just wanted to let you know you were just left a 0:08 long
message (number 1)
in mailbox 121 from 21722545, on Thursday, May 31, 2012 at 08:45:02 PM so
you might
want to check it when you get a chance. Thanks!

Common/Reasonable Assumption on DID/Channel over-subscription

Hello All,

just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX.

I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have any real world numbers they can share? Is 10 to 1
a good ratio to ensure a user practically never gets a “circuits are busy”?

Thanks in advance