Weird issue with "originate"?

Hello! I’m experiencing some weird behavior when generating calls with the Asterisk Management Interface using the “originate” command and dialing through a SIP provider. I originally noticed the issue when empty entries ended up in my CDR logs. Here’s an example of a single call placed but two CDR entries show up (one being empty):

“UNAVAILABLE”,“UNAVAILABLE”,“14232960783”,“message”,“SIP/206.252.200.196-dda9”,
"",“Dial”,“SIP/14232960783@206.252.200.196”,“2006-07-13 08:43:57”,"",“2006-07-13 08:43:57”,
“0”,“0”,“NO ANSWER”,“DOCUMENTATION”,"",“1152794576.1”,""

“”,"",“s”,“default”,“Unknown”,"","","",“2006-07-13 08:43:57”,"",“2006-07-13 08:43:57”, “0”,“0”,“FAILED”,“DOCUMENTATION”,"",“1152794637.2”,""

I’ve been able to recreate the issue when dialing a number that reports a SIT tone, such as the one above. I’m unsure if there are other situations where this might be happening. Here a list of things I’ve done to debug the issue:

[ul]* When dialing the number that reports a SIT tone with a Grandstream GXP-2000 phone through the same Asterisk instance using the Dial() command instead of “originate” and the same SIP provider I’m able to hear the audio with no issues.[/ul]

[ul]* Changing SIP providers doesn’t appear to make a difference.[/ul]

[ul]* Asterisk is receiving the RTP ulaw audio stream but Asterisk isn’t launching my extension application. I’ve proven this by reviewing Asterisk debug logs and sniffing the audio stream. Ultimately, I’d like to process the SIT tone but the audio never makes it to my extension application.[/ul]

Here are links to all the debugging I’ve collected so far. Any insight on what’s going on here would be greatly appreciated. Hopefully I’m just overlooking something stupid…

-----DEBUGGING INFO-----

Asterisk Version: 1.2.7.1 (from ports/source)
OS Version…: FreeBSD 6.0

Sample Asterisk Management Interface Session:

Action: Originate
Channel: SIP/14232960783@206.252.200.196
Context: message
Exten: 14232960783
Priority: 1
CallerID:
Timeout: 60000
Async: 1


Asterisk sip.conf:
http://mike.wadeTN.org/asterisk-problem/sip.conf.txt

Asterisk extensions.conf:
http://mike.wadeTN.org/asterisk-problem/extensions.conf.txt

Successful Call Asterisk Debug Log:
http://mike.wadeTN.org/asterisk-problem/successful-call.txt

Failed (SIT) Call Asterisk Debug Log:
http://mike.wadeTN.org/asterisk-problem/failed-call.txt

Failed (SIT) Call Asterisk Audio Sniff:
http://mike.wadeTN.org/asterisk-problem/failed-call.wav

Successful and Failed Call Asterisk CDR Log:
http://mike.wadeTN.org/asterisk-problem/cdr.txt