asterisk users - Apr 2013 - Logging SIP connection status for review

Is anyone using something to log SIP results (connected/not, latency) that
they really like?  We do some logging using simple scripts writing the
results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some
better reporting.  For example, graphs of latency in a time range, or a
list of unreachable phones within a range, etc.


-- 
Carlos Alvarez
TelEvolve
602-889-3003
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130410/80114c73/attachment.htm>

On Wed, 10 Apr 2013, Carlos Alvarez wrote:
> Is anyone using something to log SIP results (connected/not, latency) 
> that they really like? ?We do some logging using simple scripts writing 
> the results of sip show peers to a text file if customers report issues, 
> but it would be nice to have a tool that logs all the time and lets us 
> do some better reporting. ?For example, graphs of latency in a time 
> range, or a list of unreachable phones within a range, etc.
dumpcap can capture all of the SIP (and RTP) packets into a series of 
files without a huge performance hit.

A cron job can pbzip2 the files and delete if over x days old.

-- 
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards       sedwards at sedwards.com      Voice: +1-760-468-3867 PST
Newline                                              Fax: +1-760-731-3000

http://www.artifact-software.com/?page_id=1666
Would this help?
Put a JasperReport graph or two in a report step.


Ron

On 10/04/2013 2:02 PM, Steve Edwards wrote:
> On Wed, 10 Apr 2013, Carlos Alvarez wrote:
>
>> Is anyone using something to log SIP results (connected/not, latency) 
>> that they really like?  We do some logging using simple scripts 
>> writing the results of sip show peers to a text file if customers 
>> report issues, but it would be nice to have a tool that logs all the 
>> time and lets us do some better reporting.  For example, graphs of 
>> latency in a time range, or a list of unreachable phones within a 
>> range, etc.
>
> dumpcap can capture all of the SIP (and RTP) packets into a series of 
> files without a huge performance hit.
>
> A cron job can pbzip2 the files and delete if over x days old.
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                 http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>     http://lists.digium.com/mailman/listinfo/asterisk-users

-- 
Ron Wheeler
President
Artifact Software Inc
email: rwheeler at artifact-software.com
skype: ronaldmwheeler
phone: 866-970-2435, ext 102

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130410/963ee3a0/attachment.htm>

On On Wed, 10 Apr 2013, Carlos Alvarez wrote: 
>> 
>>> Is anyone using something to log SIP results (connected/not,
latency) that they really like?  We do some logging using simple scripts writing
the results of sip show peers to a text file if customers report issues, but it
would be nice to have a tool that logs all the time and lets us do some better
reporting.  For example, graphs of latency in a time range, or a list of
unreachable phones within a range, etc.
>> 
How about munin

http://munin-monitoring.org/

With these plugins
http://www.venturevoip.com/news.php?rssid=2322

You could easily adapt one to do registrations. I find the sip peers and calls
in the last hour quite interesting

Cheers Duncan
>> 
>> 
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>> 
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> -- 
> Ron Wheeler
> President
> Artifact Software Inc
> email: rwheeler at artifact-software.com
> skype: ronaldmwheeler
> phone: 866-970-2435, ext 102
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20130411/fc7b12f3/attachment.htm>