I'll have to get some VPN's setup, but I will give it a try with SIP. Thanks for the input - you saved me building 2 more asterisk servers for testing this issue locally (rather than across 3 networks).
Andrew On Thu, Mar 26, 2009 at 11:12 PM, Steve Totaro <[email protected]> wrote: > On Fri, Mar 27, 2009 at 12:50 AM, 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 >> > > 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) > > _______________________________________________ > -- 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
