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

[Chris Bagnall]

On 14/5/13 4:30 pm, Ishfaq Malik wrote:
> 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.

Well, the obvious start point might be ChanIsAvail() - that'll at least 
weed out an upstream SIP peer that's unavailable (assuming you're using 
qualify) before you even get as far as Dial().

However, one of the problems you might encounter when sending calls to a 
provider is an inability to distinguish between Congestion and Busy. 
Ideally, of course, you want to route the call to upstream2 if you get 
Congestion from upstream1, but not if the dialled number is Busy. 
There's not always a good way around that.

As others have said, the only real way around it is to send calls 
periodically to verify end to end operation - at least this way you're 
testing both your upstream's SIP connectivity and also their PSTN 
termination.

Kind regards,

Chris
-- 
This email is made from 100% recycled electrons