[Asterisk-Users] H.323 and cause code 'user busy'

[Storer, Darren]

Hi Tim,

TR> Anyone else got any comments on this issue? I think it is a fairly
TR> major issue...

This is a problem for a customer of ours that uses * for a simple "follow
me" service on PRI; inbound calls trombone through the asterisk server and
then onwards to the called party's PSTN number. This service suffers from
the exact symptoms that you describe with ringing (due to alerting), during
call set-up, heard before the call is released due to subscriber busy etc.

The Q.931 stack is not clean in this regard and I would echo your comments
about the serious nature of the problem.

Regards

Darren
--
Comgate
Telco>Internet<Broadcast

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Tim Robinson
Sent: 02 June 2004 19:25
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] H.323 and cause code 'user busy'


This is a known architectural issue that does not appear to have been
resolved yet.

See bug http://bugs.digium.com/bug_view_page.php?bug_id=0001337

The problem is the mapping of the various internal states of different
Asterisk channels on to the Q.931 states.

Asterisk currently does not wait for 'alerting' to be confirmed on the
outgoing channel before 'ALERTING' is sent out on the ISDN line.  This
means that if the outgoing channel turns out to be  'busy', it is
already too late to reject the incoming call with a 'SUBSCRIBER BUSY'
cause as this is not valid in the ALERTING phase.  Our switches here in
the UK retuern 'ringback tone' to the calling party as soon as
'ALERTING' is received, and if it is then rejected with a 'busy'
clearing cause you get a message 'The other party has hung up'.

I spoke to Markster at length about this but I am not sure I really made
the issue clear.

Anyone else got any comments on this issue? I think it is a fairly major
issue...

Rgds
Tim
Jan Baumann wrote:
>
> Hi all,
>
> I just installed chan_h323 to interface to a H.323/ISDN gateway.
> It works really well after two days learning and testing except one
> thing somebody of you may have an answer to:
>
> If I call in from PSTN to a busy asterisk SIP extension I can see a SIP
> status 486 BUSY, but don't get it passed to the H.323/ISDN side.
>
> Asterisk jumps correctly to EXTEN+101 in the dialplan. I tried different
> Apps there (Hangup, Busy, Congestion)
>
> They deliver different cause codes to the H.323/ISDN side (normal call
> clearing or call rejected) but none of them returns 'user busy' as
> expected.
>
> In Zaptel with Q.931 PRI (euroisdn) you can do
>
> exten => 123,102,SetVar(PRI_CAUSE=17)
> exten => 123,103,Hangup
>
> to explicitely set the RELEASE cause code.
>
> Is something similiar also possible with H.323?
>
> Thank you and regards,
>
> Jan
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