I initially had no trunking anywhere, and had the same behavior. I thought trunking would help, but I can't figure out why the /dev/dahdi device doesn't get created on C. The dahdi tools / modules don't seem to have much error / debugging info available, or if they do, I sure can't find it anywhere obvious.
Andrew On Thu, Mar 26, 2009 at 11:39 PM, Brandon B. <[email protected]> wrote: > 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 > > > _______________________________________________ > -- 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 > _______________________________________________ -- 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
