virendra bhati
2011-Dec-18 05:26 UTC
[asterisk-users] How to monitor SIP Trunk on production server
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111218/ea2b2353/attachment.htm>
Zohair Raza
2011-Dec-18 06:25 UTC
[asterisk-users] How to monitor SIP Trunk on production server
Hi, http://blog.tmcnet.com/blog/tom-keating/asterisk/using-monit-tool-to-monitor-asterisk.asp Regards, Zohair Raza On Sun, Dec 18, 2011 at 9:26 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 > > > -- > _____________________________________________________________________ > -- 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/20111218/039ed4d7/attachment.htm>
Carlos Rojas
2011-Dec-19 02:36 UTC
[asterisk-users] How to monitor SIP Trunk on production server
Hello, Do you saw this solution? http://linuxnotes.us/ Regards On Sun, Dec 18, 2011 at 12:26 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 > > > -- > _____________________________________________________________________ > -- 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/20111218/17214cad/attachment.htm>
virendra bhati
2011-Dec-19 05:29 UTC
[asterisk-users] How to monitor SIP Trunk on production server
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111219/d360ba1b/attachment.htm>