No path to translate from H323

I have a small problem with my quintum gateway. I have installed asterisk and am able to make calls using h323 and sip protocol over lan using softphones like xlite(sip) and sjphone(h323). Problem arises when i connect my quintum a400 gateway to the lan. Quintum is configured to throw calls to asterisk server, when ever it recieves any. 1 digital phone is connected to PBX port of quintum. i dial quintum extension from digital phone, quintum throws the request to asterisk server, asterisk uses an ‘s’ extension to dial a sip or h323 phone—> problem is, asterisk does call the softphone but call doesnt get connected, some times it rings once and hangup some times it doesnt even rings the called phone and shows an error which is:

Executing Answer(“H323/ip$”, “”) in new stack
– Executing Dial(“H323/ip$”, “H323/|20”) in new stack
– Called
Jun 29 15:01:37 WARNING[2753]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/ to H323/ip$
== Spawn extension (default, s, 2) exited non-zero on ‘H323/ip$’ is the host ip address for softphone, and is the ip address for quintum gateway

Anybody who has any idea about this problem, plz help

no clues?? so sad :exclamation:

i bet you didn’t even bother to google for this did you ? as this is some research project, i think you need to widen your search area … not just asking people !!

the “unable to translate” bit would probably indicate a problem with the codecs that the endpoints are using. it would appear that Asterisk is unable to translate one to another. so, what codecs are in use ?

hahaha…i think u like the movie title ‘Pride and Prejudice’…anyways, i do know its a codec problem…ihave already tried gsm,ulaw, alaw, g729, g726, g711, slin. i just thought as im new to this technology, i should ask somebody about it.

quintum and softphone both use h323 protocol, so there should be no problem, but there is.

protocol != codec

ofcorse :wink: .

problem not resolved yet…anybody…any indea?

i have the same problem! :frowning:
You found a solution?

I use asterisk 1.2.14 with openh323, if a call come in with h323 and a call foreward is defined to go out with h323 i become the following error:

Jan 17 12:21:51 WARNING[32294]: channel.c:2752 ast_channel_make_compatible: No path to translate from H323/ip$ to H323/00XXXXXXXXXX-6(-2033650) Jan 17 12:21:51 WARNING[32294]: app_dial.c:1603 dial_exec_full: Had to drop call because I couldn't make H323/ip$ compatible with H323/00XXXXXXXXXX-6

If a call come in with SIP and go out with h323 all is ok.
But the external telephone gateway only speak h323! :frowning:

Any idea?