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Hi all,<br>
<br>
I have problem with inbound call. Outbound works without any problems.
I always get 400 Bad Request on INVITE message. I already try to change
header lines "To" and "Contact" <br>
from:<br>
<br>
<i>To: <<a class="moz-txt-link-freetext" href="sip:38116789766@">sip:38116789766@</a></i>10.41.100.10<i>><br>
Contact: <<a class="moz-txt-link-freetext" href="sip:015672227@">sip:015672227@</a></i>10.41.100.10<i>:5060></i><br>
<br>
to:<br>
<br>
<i>To: <<a class="moz-txt-link-freetext" href="sip:38116789766@">sip:38116789766@</a></i>10.41.100.10<i>><br>
Contact: <<a class="moz-txt-link-freetext" href="sip:015672227@">sip:015672227@</a></i>10.41.100.10<i>:5060></i><br>
<br>
but nothing change. I also try to change parameter in Record-Route
"lr=on" but I always get 400 Bad Request from Genexis OCQ118.<br>
Have I malformed message INVITE or what could be wrong?<br>
<br>
Call is from PSTN to Genexis<br>
<br>
Kamailio External IP: 85.23.65.2<br>
Kamailio Internal IP: 10.41.100.10<br>
SIP Client IP (Genexis OCQ118): 10.41.10.100<br>
<br>
Trace:<br>
<br>
INVITE <a class="moz-txt-link-freetext" href="sip:38116789766@10.41.10.100:5060">sip:38116789766@10.41.10.100:5060</a> SIP/2.0<br>
Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:10.41.100.10;r2=on;lr;ftag=as741efbef;did=41c.12f18333"><sip:10.41.100.10;r2=on;lr;ftag=as741efbef;did=41c.12f18333></a><br>
Record-Route:
<a class="moz-txt-link-rfc2396E" href="sip:85.23.65.2;r2=on;lr;ftag=as741efbef;did=41c.12f18333"><sip:85.23.65.2;r2=on;lr;ftag=as741efbef;did=41c.12f18333></a><br>
Via: SIP/2.0/UDP 10.41.100.10;branch=z9hG4bK52b8.1b7b9b04.0<br>
Via: SIP/2.0/UDP
85.23.65.2:5070;received=85.23.65.2;branch=z9hG4bK265923dc;rport=5070<br>
Max-Forwards: 69<br>
From: "015672227" <a class="moz-txt-link-rfc2396E" href="sip:015672227@85.23.65.2"><sip:015672227@85.23.65.2></a>;tag=as741efbef<br>
To: <a class="moz-txt-link-rfc2396E" href="sip:38116789766@85.23.65.2"><sip:38116789766@85.23.65.2></a><br>
Contact: <a class="moz-txt-link-rfc2396E" href="sip:015672227@85.23.65.2:5070"><sip:015672227@85.23.65.2:5070></a><br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:06bbd69d6e21230a13aadfdd3c0158c7@85.23.65.2">06bbd69d6e21230a13aadfdd3c0158c7@85.23.65.2</a><br>
CSeq: 102 INVITE<br>
User-Agent: Asterisk PBX 1.6.1.9<br>
Date: Mon, 08 Mar 2010 13:41:38 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces, timer<br>
Content-Type: application/sdp<br>
Content-Length: 303<br>
<br>
v=0<br>
o=root 492361776 492361776 IN IP4 10.41.100.10<br>
s=Asterisk PBX 1.6.1.9<br>
c=IN IP4 10.41.100.10<br>
t=0 0<br>
m=audio 11576 RTP/AVP 8 0 101<br>
a=rtpmap:8 PCMA/8000<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20<br>
a=sendrecv<br>
a=nortpproxy:yes<br>
<br>
<br>
<br>
SIP/2.0 400 Bad Request<br>
Via: SIP/2.0/UDP 10.41.100.10;branch=z9hG4bK52b8.1b7b9b04.0<br>
Via: SIP/2.0/UDP
85.23.65.2:5070;received=85.23.65.2;branch=z9hG4bK265923dc;rport=5070<br>
CSeq: 102 INVITE<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:06bbd69d6e21230a13aadfdd3c0158c7@85.23.65.2">06bbd69d6e21230a13aadfdd3c0158c7@85.23.65.2</a><br>
Content-Length: 0<br>
<br>
<br>
Thanks for help<br>
<br>
BG<br>
<br>
Ernest<br>
<br>
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