try adding:
progress_ind setup enable 3
progress_ind alert enable 8
progress_ind connect enable 8
to the dial-peer on the Cisco GW...
----- Original Message -----
From: "Manuel Goertz" <mgoertz@KOM.tu-darmstadt.de>
To: <asterisk-users@lists.digium.com>
Sent: Friday, March 12, 2004 2:26 AM
Subject: [Asterisk-Users] SIP call to ISDN subscriber
>
> 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 G?rtz 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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