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@xxx") in new stack --- ooh323_request - data 203@xxx format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 203@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@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. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 827 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060620/417d7f22/attachment.pgp