[Asterisk-Users] Asterisk and H.323 Gatekeeper

[kido noagbodji]

Hi Paul,

Are you using the h323 or the oh323 channel. Please, what is the status of
the bug that you are talking about?

Thanks
----- Original Message ----- 
From: "Paul Davidson" <planac at gmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Monday, November 22, 2004 3:11 PM
Subject: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper


> > Message: 4
> > Date: Sun, 21 Nov 2004 17:56:10 -0800
> > From: "Paul Mahler" <pmahler at signate.com>
> > Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> >         <asterisk-users at lists.digium.com>
> > Message-ID: <20041121205619.GA56563 at mail26f.sbc-webhosting.com>
> > Content-Type: text/plain;       charset="us-ascii"
> >
> > Are you using oh323 ?
> >
> > Paul Mahler
> > pmahler at signate.com
> > Signate, LLC
> > 665 Third Street
> > Suite 100
> > San Francisco, CA
> >  94107-1901
> >
> >  Asterisk Services and Training
> >
> > > -----Original Message-----
> > > From: asterisk-users-bounces at lists.digium.com
> > > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> > > Jorge Alayon
> > > Sent: Friday, November 19, 2004 4:33 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper
> > >
> > > Hello,
> > >
> > > I am new to this list and to asterisk and going through the
> > > archive file I did not find an answer to my problem.
> > >
> > > I have a VoIP network working fine with multiple gateways
> > > registered to a Cisco H.323 Gatekeeper. I have successfully
> > > registered Asterisk as a GW in that network and also
> > > successfully registered two X-Lite SIP Client to asterisk
> > > that call to each other.
> > >
> > > I want to connect to the H.323 network but call does not
> > > progress from the SIP to the H.323 network.
> > >
> > >   This is the ASterisk console output.
> > >
> > >     -- Registered SIP '1154538511' at 192.168.11.46 port 5060
> > > expires 1800
> > >     -- Executing Wait("SIP/1154538511-ed8a", "2") in new stack
> > >     -- Executing Dial("SIP/1154538511-ed8a",
> > > "h323/01145568423") in new stack
> > >     -- Called 01145568423
> > >   == No one is available to answer at this time
> > >     -- Timeout on SIP/1154538511-ed8a
> > >   == CDR updated on SIP/1154538511-ed8a
> > >     -- Executing Goto("SIP/1154538511-ed8a", "#|1") in new stack
> > >     -- Goto (default,#,1)
> > >     -- Executing Playback("SIP/1154538511-ed8a",
> > > "demo-thanks") in new stack
> > >     -- Playing 'demo-thanks' (language 'en')
> > >     -- Executing Hangup("SIP/1154538511-ed8a", "") in new stack
> > >   == Spawn extension (default, #, 2) exited non-zero on
> > > 'SIP/1154538511-ed8a'
> > >
> > > If I dial from an ATA, An AS5300, or an Audiocodes GW the
> > > number 01145568423 through the Gatekeeper, it works.
> > >
> > > Any ideas ?
> > >
> > > Regards,
> > >
> > > Jorge A.
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >
>
> I have been working with this precise same issue, under bug number
> 0002659.  I've seen this problem all the way up to CVS-HEAD-11/21.  In
> my case, I'm using the gnuGK gatekeeper, and connecting to cisco
> callmanager 3.3.3.  While callmanager can call in to Asterisk via the
> gateway, calls do not proceed in the other direction- the only
> difference between this setup and my own (aside from a different
> gatekeeper) is that mine is 100% H.323 with IAX softphones used to
> attempt the call.
>
> I've been bouncing stuff back and forth with JerJer on this isse- one
> thing that might help you (it didn't help me) is to use CVS-HEAD,
> which will require an update to OpenH323 and PWLIB (that was a long
> evening).
>
> Not much help- but at least know you're not alone.
>
> -pbd
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