will your service provider accept sip OPTIONS? (same as what setting
qualify=yes) in asterisk does for the PEER...
I dont know if astlinux inscludes sipsak but on my large servers i use sipsak
and a script to test both a provider and my asterisk.. (we have had SIP
deadlock issues at one site).. OPTIONS would at least let you know if 2 way
sip traffic is progressing to and from the provider-Christopher
From: Shamus Rask <sha...@srask.ca>
To: astlinux-users@lists.sourceforge.net
Sent: Tuesday, September 15, 2015 10:40 AM
Subject: [Astlinux-users] how to monitor outbound trunks
I’m posting this here as I find the AstLinux users offer the best and most
responsive insights to all things Asterisk.
I’ve started a SaaS-based project that uses Asterisk to make outbound calls.
I’d like to implement some form of rigorous monitoring to ensure that my 2x
outbound SIP trunks are able to dial numbers on the PSTN.
I’ve created a cron job on my Asterisk server that generates a .call file every
10 minutes. This dials a separate DID (hosted by a different SIP provider) on
my personal AstLinux box. By doing an Answer() immediately followed by a
Hangup(), I’ve limited the duration of the call to < 1s. However, I’m still
being charged $0.005 for each call by my SIP provider. At 2x calls every 10m,
this adds up to $1.44/day or $43.20/month.
Given the SIP signalling that is taking place, is there any way to parse the
SIP messages and terminate the call attempt when the Asterisk server sees a
“180 - ringing” message? This would eliminate the daily cost of the tests and I
would only incur the monthly DID charge.
Thank you in advance,
Shamus
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