[HELP] sip sessions are not terminated


I’ve a problem with Asterisk 1.0.9 + bristuff 0.2.0-RC8n

We use Asterisk in the following configuration:

Telecom Carrier <= PRI => Asterisk <= PRI => Siemens HiCom

Most of our users are on the HiCom, but the number on Asterisk is growing.
From SIP phones (eyeBeam or Cisco 7960 V7.5) we have now problems with sip sessions that are not teminating.

Sometimes after a call from a sip phone to a number of the Carrier or the HiCom the sip session is still there.
With “show channels” i get a warning message like this:
Sep 5 15:52:03 WARNING[9460]: channel.c:536 ast_channel_walk_locked: Avoided deadlock for ‘Zap/86-1’, 10 retries!

and with “sip show channels” the sip channel is still up.

any idea?