Hi Michael,

I have just updated to 0.5.4 and the problem is still there.  Are there any
parameters or logs that I should be checking?

When I run SJPhone (h323) direct to the Cisco 2600 fxo gateway - the call
remians up without error.

When I run SJPhone (h323) to Asterisk and then to a SIP extension the call
also remains up without error.

The issue only displays on outbound h323 connections from asterisk.

Thanks for your help.


Regards,

Steve.



                                                                                       
                                                       
                      Michael Manousos                                                 
                                                       
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                      Sent by:                          Subject:  Re: [Asterisk-Users] 
h323 gateway call lost after 74sec always              
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                      25-07-03 12:33 AM                                                
                                                       
                      Please respond to                                                
                                                       
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Steven Thomas wrote:
>
>
>
> Hi,
>
> I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an
FXO
> port.  Asterisk talks to the router via h323 and opens a call and
connects
> with no problem.
>
> At exactly 74 secs (timer on the phone) the call drops, and Asterisks
> displays this message:
>
>     -- H323:29764 answered SIP/6000-9794
>  15:20.606          H225 Caller:80eea08 H225    Received connect PDU.
>
> ********************************
> **** CONTROL PROTOCOL ERROR ****
> *  Roundtrip Delay             *
> ********************************
>     -- Hungup 'H323:29764'
>   == Spawn extension (default, 5500, 1) exited non-zero on
'SIP/6000-9794'
>
>
> Any ideas?  Thanks.

Yes, try the 0.5.4 version of asterisk-oh323. It's been fixed.

>
>
> Steve.
>


Michael.




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