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 <[email protected]>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 <[email protected]> > 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 > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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