[Asterisk-Users] Strange T1 Problem

[willy at yponeinc.com]

Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
   Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well.  It will just ring
forever.
Cheers,
Willy

----- Original Message Follows -----
> 
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
> 
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently. 
> 
> This must be a different problem, because neither of those
> solutions worked.
> 
> 
> 
> zapata.conf sends call to fixup context:
> 
> 
> [fixup]
> 
> ; Receive call as *<calling>*<called>
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
> 
> 
> [default]
> 
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
> 
> 
> user1's phone rings, but no ring from PSTN caller. user1
> picks up, both can talk ok.
> 
> 
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Willy Wouters
ypOne Publishing