[Asterisk-Users] Asterisk and H.323 Gatekeeper

[Jorge Alayon]

Thank you, I will see into it.

Regards,

Jorge A.


-----Mensaje original-----
De: Paul Davidson [mailto:planac at gmail.com]
Enviado el: Monday, November 22, 2004 12:12 PM
Para: asterisk-users at lists.digium.com
Asunto: 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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