[asterisk-users] SIP channel problem

[asterisk]

Hi,

We have a small callcenter.
We registered with a provider with SIP with username and pass.
If our customer start a call with an IP phone, he can speek.
BUT
If he hang Up the line, he get  a callback.
with SIP debug I got the next message:

--- (24 headers 9 lines)---
"Using INVITE request as basis request - ....."





Customer's PBX  -> call -> Provider's PBX -> PSTN
If the customor give a Hang Up

Provider's PBX -> call -> Customer's PBX

I think.

I hope this is the situation because I can't find problem in the dial 
plan. :)

Thx for help.
Kind regards
Szolke