<div dir="ltr"><span style="color:rgb(51,51,255)">Hi Sammy,</span><br style="color:rgb(51,51,255)"><br style="color:rgb(51,51,255)"><span style="color:rgb(51,51,255)">Actually we have 2 voip trunk at our server 1 of </span><b style="color:rgb(51,51,255)">Voipon</b><span style="color:rgb(51,51,255)"> and 2nd of </span><b style="color:rgb(51,51,255)">Gradwell</b><span style="color:rgb(51,51,255)">. When our balance goes down then they don't auto-refill it, I don't know the reason behind it.</span><br style="color:rgb(51,51,255)">
<span style="color:rgb(51,51,255)">Ans some time goes down means Call will not go through from VoIP trunk.</span><br style="color:rgb(51,51,255)"><br style="color:rgb(51,51,255)"><span style="color:rgb(51,51,255)">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.</span><br>
<br><br><br><div class="gmail_quote">On Mon, Dec 19, 2011 at 9:41 AM, Sammy Govind <span dir="ltr"><<a href="mailto:govoiper@gmail.com">govoiper@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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. <div>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.</div>
<div>You can use AMI or asterisk concole command to do somehting like below:</div><div><br></div><div>#asterisk -rx "sip show peer provider-1" | grep qualify<br><br>Use awk to extract only the numeric value from output of above.</div>
<div><br></div><div>Or you can use AMI to fetch sip peer details and parse the value you require.</div><div><div></div><div class="h5"><div><br></div><br><div class="gmail_quote">On Sun, Dec 18, 2011 at 10:26 AM, virendra bhati <span dir="ltr"><<a href="mailto:virbhati@gmail.com" target="_blank">virbhati@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi List,<div><br></div><div>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.</div>
<div><br></div><div>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? </div>
<div><br></div><div>Any hint will be appreciated <span><font color="#888888"><br clear="all"><div><br></div>-- <br><div dir="ltr"><br>Thanks and regards<br><br> Virendra Bhati<br><a href="tel:%2B91-8885268942" value="+918885268942" target="_blank">+91-8885268942</a><br>
Software Engineer<br></div><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br><div dir="ltr"><br>Thanks and regards<br><br> Virendra Bhati<br>+91-8885268942<br>Software Engineer<br></div><br>
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