[asterisk-users] IAX problem through intermediate asterisk box

[Steve Totaro]

On Fri, Mar 27, 2009 at 12:50 AM, 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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I have had, seen, or fixed this problem more times than I can count.

Use SIP.

IAX2 has been a common problem that I have fixed many many times for
people over the years.

OR, "The latest version should fix it", which is the Digium tagline on IAX2.

Please report back your results if you do use SIP.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)