[Asterisk-Users] hungup problems with SIP and x100P card

[Floris Roset]

Hello all,



I am using Asterisk as a PSTN gateway. I can make a call with a sip client to the PSTN and it works fine.  When I hangup on the sip client, the call is disconnected.  Unfortunately, when I hangup on the PSTN telephone, asterisk does not disconnect the call.  The zap channel is kept occupied and no BYE message is sent. 

On the other hand, when I dial from console that extension, the call is disconnected when I hangup on the PSTN phone. Did someone experienced a similar problem? Any ideas? I can send some debug information.

Thanks in advance.

Floris

 ========

Here is the setup.

 PSTN ----- ASTERISK ---- SER – SIP Phone

1 ) Asterisk registers with ser as floris, and extension 667516956 is mapped to that registration in sip.conf. X100P used for PSTN connection

2) SIP Phone registers with SER as florisc.

3) florisc calls floris

4) Hangup from florisc disconnects the call. 

5) Hangup from PSTN 667516956 does not disconnect the call. When dialling from console the extension 667516956, PSTN hangup is detected and call is disconnected.





---------------------------------
  Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040114/ace8acc9/attachment.htm