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 <[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
