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<DIV><FONT face=Arial size=2>encountered bad request for calls from asterisk to
ser. i can't figure out why?</FONT></DIV>
<DIV><FONT face=Arial size=2>does ser expect a "user" or port in To:
field? failed codec negotiation?</FONT></DIV>
<DIV><FONT face=Arial size=2>here are the sip messages</FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Reliably Transmitting:<BR>INVITE
sip:kelvin@10.17.0.100 SIP/2.0<BR>Via: SIP/2.0/UDP
10.17.0.100:5065;branch=z9hG4bK580fdaaf<BR>From: "2010"
<sip:2010@10.17.0.100:5065>;tag=as0dd619ef<BR>To:
<sip:kelvin@10.17.0.100><BR>Contact:
<sip:2010@10.17.0.100:5065><BR>Call-ID: <A
href="mailto:20be6f95408969da5117a6b82036d540@10.17.0.100">20be6f95408969da5117a6b82036d540@10.17.0.100</A><BR>CSeq:
102 INVITE<BR>User-Agent: Asterisk PBX<BR>Content-Type:
application/sdp<BR>Content-Length: 234</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>v=0<BR>o=root 20301 20301 IN IP4
10.17.0.100<BR>s=session<BR>c=IN IP4 10.17.0.100<BR>t=0 0<BR>m=audio 13602
RTP/AVP 3 0 8 101<BR>a=rtpmap:3 GSM/8000<BR>a=rtpmap:0 PCMU/8000<BR>a=rtpmap:8
PCMA/8000<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-16<BR> (no
NAT) to 10.17.0.100:5060<BR>Sip read: <BR>SIP/2.0 100 trying -- your call is
important to us<BR>Via: SIP/2.0/UDP
10.17.0.100:5065;branch=z9hG4bK580fdaaf<BR>From: "2010"
<sip:2010@10.17.0.100:5065>;tag=as0dd619ef<BR>To:
<sip:kelvin@10.17.0.100><BR>Call-ID: <A
href="mailto:20be6f95408969da5117a6b82036d540@10.17.0.100">20be6f95408969da5117a6b82036d540@10.17.0.100</A><BR>CSeq:
102 INVITE<BR>Server: Sip EXpress router (0.8.11
(i386/linux))<BR>Content-Length: 0<BR>Warning: 392 10.17.0.100:5060 "Noisy
feedback tells: pid=20118 req_src_ip=10.17.0.100 req_src_port=5065
in_uri=sip:kelvin@10.17.0.100 out_uri=sip:10.17.0.30:12532;transport=tcp
via_cnt==1"</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV><FONT face=Arial size=2>
<DIV><BR>9 headers, 0 lines<BR>Sip read: <BR>SIP/2.0 400 Bad Request<BR>Via:
SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf<BR>From: "2010"
<sip:2010@10.17.0.100:5065>;tag=as0dd619ef<BR>To:
<sip:kelvin@10.17.0.100>;tag=49dd952c-6360-4555-be16-601328c6805a<BR>Call-ID:
<A
href="mailto:20be6f95408969da5117a6b82036d540@10.17.0.100">20be6f95408969da5117a6b82036d540@10.17.0.100</A><BR>CSeq:
102 INVITE<BR>Record-Route:
<sip:kelvin@10.17.0.100;transport=tcp;r2=on;ftag=as0dd619ef;lr><BR>Record-Route:
<sip:kelvin@10.17.0.100;r2=on;ftag=as0dd619ef;lr><BR>User-Agent: Windows
RTC/1.0<BR>Content-Length: 0</DIV>
<DIV> </DIV>
<DIV><BR>10 headers, 0 lines<BR>Transmitting:<BR>ACK sip:kelvin@10.17.0.100
SIP/2.0<BR>Via: SIP/2.0/UDP 10.17.0.100:5065;branch=z9hG4bK580fdaaf<BR>From:
"2010" <sip:2010@10.17.0.100:5065>;tag=as0dd619ef<BR>To:
<sip:kelvin@10.17.0.100>;tag=49dd952c-6360-4555-be16-601328c6805a<BR>Contact:
<sip:2010@10.17.0.100:5065><BR>Call-ID: <A
href="mailto:20be6f95408969da5117a6b82036d540@10.17.0.100">20be6f95408969da5117a6b82036d540@10.17.0.100</A><BR>CSeq:
102 ACK<BR>User-Agent: Asterisk PBX<BR>Content-Length:
0<BR></FONT></DIV></BODY></HTML>