[asterisk-users] IAX problem through intermediate asterisk box
Brandon B.
brandon at brellsystems.com
Fri Mar 27 00:39:57 CDT 2009
Here's my troubleshooting help -- since the problem sounds like a timing
issue and part of the call is being trunked, then fix your timing problem,
or remove the trunking from A and B then see if the problem goes away.
On Thu, Mar 26, 2009 at 10:50 PM, Andrew Hakman <andrew.hakman at gmail.com>wrote:
> So no one else has a problem routing IAX traffic through an
> intermediate Asterisk server? Does anyone else use Asterisk in such a
> configuration?
>
> On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman <andrew.hakman at gmail.com>
> wrote:
> > I'm having a problem with IAX running through an intermediate asterisk
> > box. Perhaps a small diagram will explain the situation better:
> >
> > *A ------- [cloud (public internet)] ------- *B --------[cloud
> > (private network)]----------- *C
> >
> > Asterisk server's A, B, and C, are all connected together with IAX
> > All asterisk servers are 1.6.0.6
> > Server A and B are geographically close, but connected over the public
> internet.
> > Server B and C are geographically far, but connected over a private
> network.
> > (the latency between A and B, and B and C are roughly equal)
> >
> > Each server has at least 1 phone hanging off of it, with A and C
> > having most of the phones (B only has a couple).
> > A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
> >
> > Phoning from A to B (or vice versa) works well, as does phoning from B
> > to C (and vice versa). Calls can be placed for an indefinite amount of
> > time and everything works great.
> >
> > The problem arises when phoning from A through B to C (or vice versa).
> > For the first small amount of time (which can vary on a call to call
> > basis, and lasts from 0 seconds to 3 minutes or so) everything is
> > fine. After this, the audio in both directions gets garbled, and
> > starts arriving in spurts. Once this happens, it continues forever.
> > The audio never returns to normal no matter how long you wait.
> >
> > A to B uses IAX with trunking. B to C is not using trunking
> > (dahdi_dummy is not working well on C for some reason - the module
> > loads, but no /dev/dahdi is ever created). The same behavior happens
> > when A to B is not using trunking either.
> >
> > Usually only 1 call is being placed at a time. An interesting thing
> > happens when 2 testcalls are in progress at the same time though. If
> > there's a call from A to B, and a call from A to C is made, once the
> > call from A to C becomes garbled, so does the A to B call. When the A
> > to C call is ended, the A to B call clears up. Ending the A to B call
> > first does not improve the A to C call.
> >
> > The dialplans are setup so each server passes all non-local extensions
> > to it's neighbor.
> >
> > Hence, for A, the relevant part of the dialplan is
> >
> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _2XXX,n,Hangup()
> >
> > exten => _3XXX,1,Verbose(1|Extension 3xxx)
> > exten => _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _3xxx,n,Hangup()
> >
> > For B:
> >
> > exten => _1XXX,1,NoOp()
> > exten => _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
> > exten => _1XXX,n,Hangup()
> >
> > exten => _3xxx,1,NoOp()
> > exten => _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
> > exten => _3xxx,n,Hangup()
> >
> >
> > For C:
> > exten => _2XXX,1,Verbose(1|Extension 2xxx)
> > exten => _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _2XXX,n,Hangup()
> >
> > exten => _1XXX,1,Verbose(1|Extension 1xxx)
> > exten => _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
> > exten => _1XXX,n,Hangup()
> >
> > Is this the proper way to set such a configuration up? Is there a
> > better way to call from A through B to C that would work better?
> > Anyone else experience total audio breakup after a while with a
> > similar arrangement? Why does it work initially for up to about 3
> > minutes, then completely fall apart?
> >
> > Thanks,
> > Andrew
> >
>
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