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<DIV><FONT face=宋体><FONT size=2><SPAN class=359343218-18032005>We encouter a
situation where we need to use SIP info to convey infomation for one end point
to another endpoint. I use asterisk to do the test and find asterisk does not
forward the SIP info to another endpoint, but act as UAS and returns a 4xx error
message. I</SPAN> think asterisk is not right to handle <SPAN
class=359343218-18032005>this </SPAN>SIP info message. </FONT></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=宋体 size=2>In RFC 3261<SPAN class=359343218-18032005> </SPAN>Page
70 "This protocol is designed to be extended. Future extensions may define new
methods and header fields at any time. An element MUST NOT refuse to proxy a
request becasue it contains a method or header field it does not know about". In
this case, asterisk does not understand this INFO message, so it acts as a UAS
instead of proxy.</FONT></DIV>
<DIV><FONT face=宋体 size=2></FONT> </DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体 size=2>How to let asterisk
just forward this request to the other endpoint and instead processing it as a
UAS?</FONT></SPAN></DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体 size=2>Thank
you,</FONT></SPAN></DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体 size=2>Wei</FONT></SPAN></DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=359343218-18032005></SPAN> </DIV>
<DIV> </DIV>
<DIV><FONT face=宋体 size=2>Here is the log from the asterisk server:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=宋体 size=2>Mar 17 12:01:31 WARNING[2804]: chan_sip.c:6134
receive_info: Unable to parse INFO message </FONT></DIV>
<DIV><FONT face=宋体 size=2></FONT> </DIV>
<DIV><FONT face=宋体 size=2></FONT> </DIV>
<DIV><SPAN class=359343218-18032005><FONT face=宋体 size=2>Here is the
trace:</FONT></SPAN></DIV>
<DIV><FONT face=宋体 size=2></FONT> </DIV>
<DIV><FONT face=宋体 size=2></FONT> </DIV>
<DIV><FONT face=宋体 size=2>Frame 96 (808 bytes on wire, 808 bytes
captured)<BR>Session Initiation Protocol<BR> Request-Line:
INFO sip:6002@192.168.10.90
SIP/2.0<BR> Method:
INFO<BR> Resent Packet:
False<BR> Message
Header<BR> Call-ID: <A
href="mailto:60b8596c-4135c-c0a81e68@192.168.10.90">60b8596c-4135c-c0a81e68@192.168.10.90</A><BR>
From:
Demo2<sip:6003@192.168.10.90;user=phone>;tag=221a0-a1cf<BR>
SIP Display info:
Demo2<BR> SIP
from address:
sip:6003@192.168.10.90<BR>
SIP tag: 221a0-a1cf<BR> To:
<sip:6002@192.168.10.90;user=phone>;tag=as6b294484<BR>
SIP to address:
sip:6002@192.168.10.90<BR>
SIP tag: as6b294484<BR> CSeq: 102
INFO<BR> Via: SIP/2.0/UDP
192.168.10.164:5060<BR> Contact:
Demo2<sip:6003@192.168.10.164:5060;user=phone><BR>
Max-Forwards: 70<BR> Supported:
timer<BR> Proxy-Authorization: Digest
username="6003",realm="asterisk",uri="sip:6002@192.168.10.90",response="034d6b15ec1b2fa91f59c55d51c0a8e7",nonce="70c7fe86"<BR>
Content-Type:
application/media_control+xml<BR>
Content-Length: 195<BR> Message
body<BR> <?xml version="1.0"
encoding="utf-8" ?>\n<BR>
<media_control>\n<BR>
<vc_primitive>\n<BR>
<to_encoder>\n<BR>
<picture_fast_update>\n<BR>
</picture_fast_update>\n<BR>
</to_encoder>\n<BR>
</vc_primitive>\n<BR>
</media_control></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=宋体 size=2><BR>Frame 97 (430 bytes on wire, 430 bytes
captured)<BR>Session Initiation Protocol<BR> Status-Line:
SIP/2.0 415 Unsupported media type<BR>
Status-Code: 415<BR> Resent Packet:
False<BR> Message
Header<BR> Via: SIP/2.0/UDP
192.168.10.164:5060<BR> From:
Demo2<sip:6003@192.168.10.90;user=phone>;tag=221a0-a1cf<BR>
SIP Display info:
Demo2<BR> SIP
from address:
sip:6003@192.168.10.90<BR>
SIP tag: 221a0-a1cf<BR> To:
<sip:6002@192.168.10.90;user=phone>;tag=as6b294484<BR>
SIP to address:
sip:6002@192.168.10.90<BR>
SIP tag: as6b294484<BR> Call-ID: <A
href="mailto:60b8596c-4135c-c0a81e68@192.168.10.90">60b8596c-4135c-c0a81e68@192.168.10.90</A><BR>
CSeq: 102 INFO<BR> User-Agent:
Asterisk PBX<BR> Allow: INVITE, ACK,
CANCEL, OPTIONS, BYE, REFER<BR>
Contact:
<sip:6002@192.168.10.90><BR>
Content-Length: 0<BR></FONT></DIV></BODY></HTML>