Can you tell me what this error is?

Server name, phone number and IP address edited for privacy :slight_smile:

Can you tell me what this error means? I’ve got a h323 call coming in and I need it to go back out SIP to another server. Works fine on an existing asterisk we have built but this new one is not acting right. Any ideas why I’m getting this error? It’s not running freepbx…just a simple asterisk server.

– Executing [[email protected]:1] Set(“H323/ip$X.X.X.X:17907/16586”, “VOLUME(RX)=2”) in new stack
– Executing [[email protected]:2] Set(“H323/ip$X.X.X.X:17907/16586”, “VOLUME(TX)=1”) in new stack
– Executing [[email protected]:3] Dial(“H323/ip$X.X.X.X:17907/16586”, “SIP/[email protected],30,tr”) in new stack
== Using SIP RTP CoS mark 5
– Called [email protected]

(and here’s the error - line below)
[Sep 4 13:50:46] WARNING[3370]: translate.c:168 framein: no samples for ulawtolin

== Spawn extension (default, xxxxx, 3) exited non-zero on ‘H323/ip$X.X.X.X:17907/16586’

What version Asterisk? What is the target and destination CODEC?

Sorry for the delay. It’s version 1.6.1.5. Codec is G.711