[asterisk-users] How to monitor SIP Trunk on production server

[virendra bhati]

Hi Sammy,

Actually we have 2 voip trunk at our server 1 of *Voipon* and 2nd of *
Gradwell*. When our balance goes down then they don't auto-refill it, I
don't know the reason behind it.
Ans some time goes down means Call will not go through from VoIP trunk.

So want to make a script in AMI / AGI  so that I will check the status all
the time of these VoIP trunk. In case if someone or both will go down then
I will send E-mail / SMS / to all the relevant guys. So that they will
check the issue on that case.



On Mon, Dec 19, 2011 at 9:41 AM, Sammy Govind <govoiper at gmail.com> wrote:

> If you can explain a bit more in detail what you mean by ensuring that
> trunk is not down? By monitoring a trunks health I assume you are talking
> about the qualify response time from a trunk.
> I developed a script for Zabbix monitoring that was executed as a command
> by Zabbix with a prameter of peer/trunk name to return its qualify time.
> Once I get a qualify time from asterisk Zabbix plotted the value on its
> graphs.
> You can use AMI or asterisk concole command to do somehting like below:
>
> #asterisk -rx "sip show peer provider-1" | grep qualify
>
> Use awk to extract only the numeric value from output of above.
>
> Or you can use AMI to fetch sip peer details and parse the value you
> require.
>
>
> On Sun, Dec 18, 2011 at 10:26 AM, virendra bhati <virbhati at gmail.com>wrote:
>
>> Hi List,
>>
>> I have asterisk 1.6.2.20 installed at production server, I have 2 SIP
>> voip trunk for making outgoing and DID for incoming to server.
>>
>> My question is how I can ensure that trunk is not down at production
>> server, So how I can monitor it's automatically by making any scripts?
>>
>> Any hint will be appreciated
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-8885268942
>> Software Engineer
>>
>>
>


-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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