[asterisk-users] Monitoring SIP trunk status on call by call basis

[Ishfaq Malik]

Hi

I'm using asterisk 1.8.7.0 and adding a fail over trunk in case my
primary goes down. I'm wondering what the best method of checking if the
primary being up is. 

Is DIALSTATUS suitable for this or is there any good SIP headers to look
at after the Dial step?

Thanks in Advance

Ish
-- 
Ishfaq Malik <ish at pack-net.co.uk>
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