[Asterisk-Users] Re: failover (was Re: voicepulse)

[Chris Albertson]

I'm having the same concerns.  What we REALLY need is the
ability to test the exact nature of the problem.  OK We could
use SER to front end SIP calls but Asterisk should report
the problem and allow the dial plan to test it.  It's a
needed missing feature in *

What about AGI?  I don't know much about AGI yet but it may help
solve this problem.

--- Matt Lawson <matt at 1control.com> wrote:
> > But this is not to say _you_ can't built a reliable VOIP based
> > system.  Get _two_ providers and set up your dial plan in
> > extensions.conf to "fail over" if one service fails to
> > connect to dial via the next one and finally if both fail
> > use pstn. your users will see a system the "just works".
> 
> Now there's an idea.  
> 
> I'm playing with this now, but there's at least 1 case I'm having 
> trouble recognizing:
> 
> The call connects but then drops due to "unauthorized."  It then only
> 
> goes to the "h" extension and I don't get a chance to try again.  Is 
> there anyway to detect this?
> 
> 
> I have to cover all of the following cases:
> 
> 
> 1.  VOIP IP address is not reachable.  Goes to extension n+101 (seems
> to 
> work as expected)
> 
> 2. VOIP service answers but refuses with call with "unauthorized". 
> It 
> just goes to the "h" extension  Is there any watch to catch this 
> failure?  Perhaps put a timer on it and say if the call was less than
> 5 
> seconds or something try the next one?
> 
> Yes I am using a correct username and password and getting this today
> 
> (not from Voicepulse, from another provider).  But there's also a 
> moderate chance that during our systems' setup a name or password
> could 
> be misspelled so I need to cover this case.

If your providers requires a pre-paid account the the account
bvallance runs out then I gues you'd get "unauthorized".
So this could be a real case that will happen

> 
> 3.  VOIP service connects but reports "all busy."  Well this one is
> hard 
> to test.  But I can make the Zap channel busy.  It goes to extension 
> n+101 as expected, so I'll have to assume that a busy VOIP service
> does 
> the same thing.

I get this from the stwo VOIP providers I use about 20%
of the time.  I guess they have only so  much gateway
hardware.  Normally a quick re-dail does it.
> 
> I was trying to determine if the "t" or "h" extension would be useful
> 
> for these but I think not.  The timeout has to be set long enough for
> 
> someone to actually answer (20-60 sec or whatever).  The "h" is
> always 
> visited at the end of the call, whether it was sucessful or not.
> 
> Any other cases, or suggestions how to handle case #2?
> 
> 
> 
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=====
Chris Albertson
  Home:   310-376-1029  chrisalbertson90278 at yahoo.com
  Cell:   310-990-7550
  Office: 310-336-5189  Christopher.J.Albertson at aero.org
  KG6OMK

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