asterisk users - May 2013 - Monitoring SIP trunk status on call by call basis

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>
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552

i think DIALSTATUS is not suitable for failover if trunk is down you get
dialstatus after time out in dial string.
it is too late for failover, you can use some script to check if
destination host is up.
if you want to do failover when destination host is up then dialstatus are
good.


On Tue, May 14, 2013 at 5:30 PM, Ishfaq Malik <ish at pack-net.co.uk>
wrote:
> 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>
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: ish at pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
> NORTH, MANCHESTER
> SCIENCE PARK, MANCHESTER, M156SE
> COMPANY REG NO. 04920552
>
>
> --
> _____________________________________________________________________
> -- 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/20130514/e2d9c48c/attachment.htm>

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