[Asterisk-Users] Asterisk and H.323 Gatekeeper
Paul Mahler
pmahler at signate.com
Sun Nov 21 18:56:10 MST 2004
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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