[Asterisk-Users] PRI to SIP Problem

[OTR Comm]

Hello all,

I have a problem calling into asterisk on a PRI going out to a SIP phone
(PRI -> SIP).  The calling party does not hear ringing and after about five
seconds gets an *All circuits are busy* recording.  However, the called SIP
phone does ring, and if the called party answers the phone within a few
seconds, the call stays in service.

CLI messages:

...
    -- Accepting call from '9288532045' to '6023432727' on channel 0/23,
span 1
    -- Executing Dial("Zap/23-1", "SIP/102|20|rt") in new stack
    -- Called 102
!! Don't know how to add an IE High-layer Compatibility (125)
!! Unable to add IE 'High-layer Compatibility'
    -- SIP/102-935d is ringing
    -- Channel 0/23, span 1 got hangup request
  == Spawn extension (incoming, 6023432727, 1) exited non-zero on 'Zap/23-1'
    -- Hungup 'Zap/23-1'
...

NOTE: There is no problem calling from SIP phone out (SIP -> PRI).


Any body ever have this problem?

Thanks,
Murrah