[Asterisk-Users] Asterisk and H.323 Gatekeeper

[Jorge Alayon]

Thank you,

I will need a SIP client with G723 and/or G.729 then. Do you know any sip
clients that do both ?

Regards,

Jorge A.



-----Mensaje original-----
De: kido noagbodji [mailto:kido at cafe.tg]
Enviado el: Monday, November 22, 2004 8:42 AM
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Asterisk and H.323 Gatekeeper


Hi Jorge,

The oh323 channel and h323 channel by NuFone are different.
As far as your problem, this looks like a codec problem i had. Try to look
that way.

K.
----- Original Message ----- 
From: "Jorge Alayon" <j.alayon at ses.com.ar>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Sent: Monday, November 22, 2004 11:06 AM
Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper


> I compiled the channel on usr/src/asterisk/channels/h323, which I believe
is
> the Nufone Channel.
> Previously I did compile the PWLIB and OH323 packets.
>
> Is that correct ?
>
> Regards,
>
> Jorge A.
>
> -----Mensaje original-----
> De: Paul Mahler [mailto:pmahler at signate.com]
> Enviado el: Sunday, November 21, 2004 10:56 PM
> Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper
>
>
> 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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