[Asterisk-Users] Asterisk and H.323 Gatekeeper

[Jorge Alayon]

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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