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
<[EMAIL PROTECTED] To: [EMAIL PROTECTED]
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Sent by: Subject: Re: [Asterisk-Users]
h323 gateway call lost after 74sec always
[EMAIL PROTECTED]
.digium.com
25-07-03 12:33 AM
Please respond to
asterisk-users
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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