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Yes, if you look at the BYE there is NO WAY SER can find out where to
send it. The To header cannot (by RFC) be used for routing and no other
header contains either location destination (as in Route) or aor (as in
request uri).<br>
<br>
To check your record routing, look at the INVITE outgoing from your SER
towards the gw. <br>
<br>
If you do proper record routing, take the RFC snippets below and hit
the vendor hard on the head. (it may seem that the GW is doing strict
routnig, i.e. old-old RFC).<br>
g-)<br>
<br>
The relevant snippet from RFC3261, 12.2.1.1:<br>
<br>
If the route set is empty, the UAC MUST place the remote target URI<br>
into the Request-URI. The UAC MUST NOT add a Route header field to<br>
the request.<br>
<br>
If the route set is not empty, and the first URI in the route set<br>
contains the lr parameter (see Section 19.1.1), the UAC MUST place<br>
the remote target URI into the Request-URI and MUST include a Route<br>
header field containing the route set values in order, including all<br>
parameters.<br>
<br>
If the route set is not empty, and its first URI does not contain the<br>
lr parameter, the UAC MUST place the first URI from the route set<br>
into the Request-URI, stripping any parameters that are not allowed<br>
in a Request-URI. The UAC MUST add a Route header field containing<br>
the remainder of the route set values in order, including all<br>
parameters. The UAC MUST then place the remote target URI into the<br>
Route header field as the last value.<br>
<br>
Kostas Marneris wrote:
<blockquote cite="mid:4649B048.4080409@otenet.gr" type="cite">
<pre wrap="">As far as I know and understand it seems that the
R-URI of the BYE mesg (1st line) which GW sends to SER is wrong.
I expected to see :
BYE sip:user@SIP_Phone_IP_Address SIP/2.0
or the problem is that the 'Route:' header is missing from the BYE mesg
so it's not loose_routed.
Do you Record_Route the original INVITE ?
Check also : rfc3261 / 16.12.1
Kostas
flavio wrote:
</pre>
<blockquote type="cite">
<pre wrap="">2007/5/15, Kostas Marneris <a class="moz-txt-link-rfc2396E" href="mailto:K.Marneris@otenet.gr"><K.Marneris@otenet.gr></a>:
</pre>
<blockquote type="cite">
<pre wrap="">Hello,
The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg
which GW sends to SER.
</pre>
</blockquote>
<pre wrap="">Kostas,
Thank you for suggestion. I've already posted SIP BYE message found
with ngrep :-D. However here is:
U 2007/05/15 14:26:15.072602 10.28.52.105:5060 -> 10.28.19.202:5060
BYE sip:10.28.19.202 SIP/2.0.
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:06605XXXXX@10.28.19.202"><sip:06605XXXXX@10.28.19.202></a>;tag=23d09968cf17c929.
From: "06720XXXXX"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:06720XXXXX@10.28.52.105"><sip:06720XXXXX@10.28.52.105></a>;tag=DXsf22384254031Yaz079220.
Via: SIP/2.0/UDP 10.28.52.105:5060;branch=z9hG4bK2384256106429488807479222.
Contact: <sip:10.28.52.105:5060>.
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:238425403179219@10.28.52.105">238425403179219@10.28.52.105</a>.
Max-Forwards: 70.
User-Agent: Netsynt-Gateway/SIP_AC_SRVlkup_3.1.12_4.
CSeq: 23334 BYE.
Content-Length: 0.
where:
06605XXXXX is the called
06720XXXXX is the callee
10.28.52.105 is IP for GW
10.28.19.202 is IP for SER
Thanks for support.
********************************
* (o< ing. Patria Flavio
* //\ phone 0823451358
* V_/_ mobile 3407873357
*
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</blockquote>
<pre wrap=""><!---->
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