[asterisk-users] IAX problem through intermediate asterisk box

[Gordon Henderson]

On Thu, 26 Mar 2009, Andrew Hakman 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?

I do. Not had a problem apart from when Digium break the protocol.

1.2 -> Interweb -> 1.2 -> Interweb -> 1.2

Also now have 1.4 in the middle too.

I'm moving to SIP though because the last leg is stuck on 1.2 and carrying 
the traffic is not something I want to keep on doing. (No "reinvite" in 
IAX in 1.2)

Gordon

> 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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