<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=gb2312">
<META content="MSHTML 6.00.2900.3395" name=GENERATOR>
<STYLE>BLOCKQUOTE {
MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px; MARGIN-LEFT: 2em
}
OL {
MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px
}
UL {
MARGIN-TOP: 0px; MARGIN-BOTTOM: 0px
}
</STYLE>
</HEAD>
<BODY style="FONT-SIZE: 10pt; FONT-FAMILY: verdana">
<DIV><FONT face=Verdana size=2></FONT> </DIV>
<DIV><FONT face=Verdana size=2>Hi,everyone<BR> <BR> For voice communication I
use a SIP+RTP proxy together with SER.<BR> For PSTN calls SER routes the INVITE
messages to the RTP proxy.<BR> <BR> Everything works fine there, but when the
call is ended by one client (which is connected to SER) or the PSTN user, the
BYE message arrive to SER but dont forward it. If our RTP proxy does not see the
message coming from SER, call doesn't end.<BR> <BR> That's why I need all of the
BYE messages to go through SER.<BR> How can I do that ?<BR></FONT></DIV>
<DIV><FONT face=Verdana size=2>In the documents they said that the bye message
loose route if there are ";lr" and If the message should be<BR>loose routed
then SER should simply relay the message to the next destination as specified in
the top-most<BR>Record-Route header field .but i have a try
and nothing happened!</FONT></DIV>
<DIV><FONT face=Verdana size=2></FONT> </DIV>
<DIV><FONT face=Verdana size=2>that is my config block:</DIV>
<DIV><BR> # ------------------ module loading
----------------------------------</DIV>
<DIV> </DIV>
<DIV> loadmodule "/usr/local/lib/ser/modules/sl.so"<BR> loadmodule
"/usr/local/lib/ser/modules/tm.so"<BR> loadmodule
"/usr/local/lib/ser/modules/rr.so"<BR><BR> #
------------------------- request routing logic -------------------</DIV>
<DIV> </DIV>
<DIV> # main routing logic</DIV>
<DIV> </DIV>
<DIV> route<BR> { record_route();<BR>
if(message=="BYE")</DIV>
<DIV> loose_route();</DIV>
<DIV> };<BR> </DIV>
<DIV> my ser is 1.2.3.4:5060<BR> this is my sip message:</DIV>
<DIV> BYE sip:0000123@19.68.150.20:31562 SIP/2.0^M<BR> Via:
SIP/2.0/UDP
23.43.53.4:5060;branch=z9hG4bK5a69ae91;rport^M<BR> Record-Route:
<sip:1.2.3.4:5060;ftag=28118cbe-30554463;lr>^M<BR> Record-Route:
<sip:12.45.32.61:5060;r2=on;lr;ftag=28118cbe-30554463>^M<BR> From:
<sip:123456@1.2.3.4:5060;user=phone>;tag=as7232fbd6^M<BR> To:
"beurself212"
<sip:0000123@19.68.150.20:31562>;tag=28118cbe-30554463^M<BR> Call-ID:
<A
href="mailto:76ad6e55-86f19c97-MzA1NTQ0NjM@192.168.50.50^M">76ad6e55-86f19c97-MzA1NTQ0NjM@192.168.50.50^M</A><BR></DIV>
<DIV> </DIV>
<DIV> Thanks,<BR>Jimmy</FONT></DIV>
<DIV align=left><FONT face=Verdana color=#c0c0c0 size=2>2008-09-12
</FONT></DIV><FONT face=Verdana size=2>
<HR style="WIDTH: 122px; HEIGHT: 2px" align=left SIZE=2>
<DIV><FONT face=Verdana color=#c0c0c0 size=2><SPAN>ÇñÀÚ</SPAN>
</FONT></DIV></FONT></BODY></HTML>