[Serusers] SIP call to ISDN subscriber

Jiri Kuthan jiri at iptel.org
Fri Mar 12 11:35:55 CET 2004


most likely caused by a misrouted ACK. -jiri

At 10:36 AM 3/12/2004, Manuel Goertz wrote:
>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 the SER server. 
>All numeric userparts of the SIP URI are rewritten and 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+
>
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--
Jiri Kuthan            http://iptel.org/~jiri/ 




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