[Asterisk-Users] Asterisk and H.323 Gatekeeper

[Paul Davidson]

I'm using the NuFone H323 channel, not the Oh323 channel.  I've had
problems with both in different situations- in this case, I haven't
tried the Oh323 channel in some time.  (but I might, given these
problems).

To date, JerJer (Jeremy McNamara, the NuFone developer who's been
assigned the bug) has not responded to my latest log and description
of the problem.  You can track and monitor the problem yourself
through bugs.digium.com if you like- you'll get the updates as I do in
that situation.

Thanks!
-pbd


On Tue, 23 Nov 2004 19:37:36 -0000, kido noagbodji <kido at cafe.tg> wrote:
> 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.
> > > > _______________________________________________
> > > > Asterisk-Users mailing list
> > > > Asterisk-Users at lists.digium.com
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >    http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > 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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