[asterisk-dev] Early Media on H.323 problem

[Kyle Bresin]

We're having an issue with H.323 on asterisk, passing calls to a Cisco 
5350/5400 Gatekeeper.

The calls are initiated on a PC client, the codec is g723 which asterisk 
passes on the gatekeeper.

The issue we're having is that you never hear any ringing.  And using a 
sniffer, it's confirmed that none of that "Early Media" is getting back 
to the client.

Once the call picks up, audio is fine, no problems.

So we just want ringing to work.

I get the following two errors in the asterisk console:
"Unable to find a codec translation path from g723 to slin"
and
"playtones_alloc: Unable to set 'OH323/XXXXX at XX.XX.XX.XX-2b01a33c' to 
signed linear format"

I've tried the following setups to try to make this go away:
codec_h323 on asterisk 1.2.4
codec_h323 on asterisk 1.0.10
codec_oh323 0.7.3 (against Mimas_patch2 libs) on asterisk 1.2.4
codec_oh323 0.7.3 (against pwlib 1.9.1, openh323 1.17.2.) on asterisk 1.2.4

I get the same error on all of them.  Audio works, no ringing.

Or h323 config is default, we enable only g723.1, set the gatekeeper IP, 
register an extension with it, and set the default context.

The default context in extensions.conf that all calls go through is just 
one dial command:
exten => _99998.,1,Dial(H323/11111#${EXTEN:6},60)

Just enough to strip off the "fake" extension, and pass it onto the 
gatekeeper with a "real" extension.  Again, nothing fancy.

I'm still perplexed why the codec "slin" is even being mentioned.  AFAIK 
nothing involved is using it...

Any help, or advice would be appreciated.

Thanks for reading,

kyle.