[Asterisk-Users] Asterisk and H.323 Gatekeeper

[Jorge Alayon]

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.