No subject

Tue Sep 5 14:32:44 MST 2006

(SIP Message Extension).  I can't find any reference to this and asterisk.
 Is it supported?  Here is what I get on the Asterisk console when I send
a text message.  Asterisk appears to receive and transmit the message to
the destination but it never actually appears.

Sip read:
MESSAGE sip:9002 at asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:14754
From: "inetsup"
<sip:inetsup at asterisk>;tag=97f9a632-d8a4-4eff-9870-92bd8037b421
To: <sip:9002 at asterisk>
Call-ID: e885f487-1ea7-441f-85da-712d10516000 at 192.168.0.103
CSeq: 1 MESSAGE
Contact: <sip:192.168.0.103:14754>
User-Agent: Windows RTC/1.0
Content-Type: text/plain;
charset=UTF-8;msgr=WAAtAE0ATQBTAC0ASQBNAC0ARgBvAHIAbQBhAHQAOgAgAEYATgA9AE0AUwAlADIAMABTAGgAZQBsAGwAJQAyADAARABsAGcAOwAgAEUARgA9ADsAIABDAE8APQAwADsAIABDAFMAPQAwADsAIABQAEYAPQAwAA0ACgANAAoA
Content-Length: 4

test
10 headers, 1 lines
Receiving message!
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.103:14754
From: "inetsup"
<sip:inetsup at asterisk>;tag=97f9a632-d8a4-4eff-9870-92bd8037b421
To: <sip:9002 at asterisk>;tag=as200391f2
Call-ID: e885f487-1ea7-441f-85da-712d10516000 at 192.168.0.103
CSeq: 1 MESSAGE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0

 to 192.168.0.103:2990
linux*CLI>