[Asterisk-Users] SIP call to ISDN subscriber

[Manuel Goertz]

Hi all,

I have a problem calling from a sipset to a ISDN subscriber over
a CISCO 1760 GW.
The following setup is used.
UA ---> GW ---> ISDN
The UA is a sipset, the GW is the CISCO 1760 with ISDN BRI interface
and a standard ISDN subscriber.
The UA is registered with a registrar/proxy. 
All numeric userparts of the SIP URI are routed to the GW. 
The GW's BRI interface is connected to the PSTN.
The call signaling seems to work as the SIP phone indicates ringing
and the ISDN phone is ringing. After picking up the hook of the ISDN
phone the UA shows "In Call". But after a second the call is
terminated. The log shows that the GW sends to both side the call
termination messages. TX -> DISCONNECT "Normal Call Clearing" to the
ISDN side and a BYE message to the SIP side. 
The signaling in short:

UA           GW      ISDN
INVITE ->     |
 <- 100 Try     
              | TX -> SETUP
              | RX <- CALL_PROC 
              | RX <- ALERTING
 <- 183 Sess  |
              | RX <- CONNECT
              | TX -> CONNECT_ACK
 <- 200 OK    |
 Milliseconds later !
              | TX -> DISCONNECT
              | RX <- RELEASE
              | TX -> RELEASE_COMP
 <- BYE       |
200 OK ->     |
 

Any hints how to solve this problem.

Thanks

   Manuel 








-- 
+KOM----------------------------------------------------------------+
|Manuel Görtz                                        Merckstrasse 25|
|Darmstadt University of Technology         64283 Darmstadt, Germany|
|Multimedia Communications                   Tel: (+49) 6151 16-5175|
|Multimedia Networking & Distribution        Fax: (+49) 6151 16-6152|
+----------------------------------------------------------------KOM+