[Asterisk-Users] ooh323 issues

[Mark Tinka]

Hi all.

Trying to setup H.323 via Asterisk between a PLANET H.323 box and 
my SIP phones.

When calling from the SIP phones, it connects but quickly 
disconnects citing the following error message:

****

---   build_peer
+++   build_peer
+++   reload_config
+++   ooh323_do_reload
    -- Executing Dial("SIP/yyy-2965", "OOH323/203 at xxx") in new 
stack
---   ooh323_request - data 203 at xxx format 0x4 (ulaw)
---   find_peer
+++   find_peer
+++   ooh323_request
---   ooh323_call- 203 at xxx
---   onNewCallCreated ooh323c_o_22
---   find_call
+++   find_call
setting callid number 203
 Outgoing call xxx(ooh323c_o_22) - Codec prefs - (gsm|ulaw|g723)
        Adding capabilities to call(outgoing, ooh323c_o_22)
        Adding gsm capability to call(outgoing, ooh323c_o_22)
        Adding g711 ulaw capability to call(outgoing, 
ooh323c_o_22)
        Adding g7231 capability to call (outgoing, ooh323c_o_22)
---   configure_local_rtp
+++   configure_local_rtp
+++   onNewCallCreated ooh323c_o_22
+++   ooh323_call
    -- Called 203 at xxx
---   onCallEstablished ooh323c_o_22
---   find_call
+++   find_call
+++   onCallEstablished ooh323c_o_22
    -- OOH323/xxx-a6f1 answered SIP/yyy-2965
    -- Attempting native bridge of SIP/yyy-2965 and 
OOH323/xxx-a6f1
---   onCallCleared ooh323c_o_22
---   find_call
+++   find_call
---   ooh323_hangup
    hanging xxx
+++   ooh323_hangup
  == Spawn extension (internal, 00263203, 1) exited non-zero on 
'SIP/yyy-2965'
---   ooh323_destroy
 Destroying xxx
+++   ooh323_destroy

****

When calling from the H.323 box to my Asterisk server, my SIP 
phone rings, and I get a ringing signal from the H.323 server, 
but when the SIP phone is answered, it goes dead with the 
following error message:

****

---   onNewCallCreated ooh323c_10
+++   onNewCallCreated ooh323c_10
---   ooh323_onReceivedSetup ooh323c_10
---   find_user
+++   find_user
        Adding capabilities to call(incoming, ooh323c_10)
        Adding gsm capability to call(incoming, ooh323c_10)
        Adding g711 ulaw capability to call(incoming, ooh323c_10)
        Adding g7231 capability to call (incoming, ooh323c_10)
---   configure_local_rtp
+++   configure_local_rtp
+++   ooh323_onReceivedSetup - Determined context internal, 
extension 203
--- onAlerting ooh323c_10
---   find_call
+++   find_call
+++ onAlerting ooh323c_10
    -- Executing Dial("OOH323/Customer-7849", "SIP/yyy") in new 
stack
    -- Called yyy
    -- SIP/yyy-8a35 is ringing
----- ooh323_indicate 3 on call ooh323c_10
++++  ooh323_indicate 3 on ooh323c_10
    -- SIP/yyy-8a35 answered OOH323/Customer-7849
----- ooh323_indicate -1 on call ooh323c_10
Jun 20 12:00:43 WARNING[18607]: src/chan_h323.c:951 
ooh323_indicate: Don't know how to indicate condition -1 on 
ooh323c_10
++++  ooh323_indicate -1 on ooh323c_10
--- ooh323_answer
+++ ooh323_answer
    -- Attempting native bridge of OOH323/Customer-7849 and 
SIP/yyy-8a35
---   onCallEstablished ooh323c_10
---   find_call
+++   find_call
+++   onCallEstablished ooh323c_10
---   onCallCleared ooh323c_10
---   find_call
+++   find_call
  == Spawn extension (internal, 203, 1) exited non-zero on 
'OOH323/Customer-7849'
---   ooh323_hangup
    hanging Customer
+++   ooh323_hangup
---   ooh323_destroy
 Destroying Customer
+++   ooh323_destroy

****

I've seen a couple of threads about this on the web, pointing 
toward codec mismatches, e.t.c. I've toggled the various codecs 
on the H.323 server and Asterisk, with no luck.

I'm running Asterisk 1.2.9.1 and Add-Ons 1.2.3.

All help appreciated.

Cheers,

Mark.

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