On Sat, 2008-12-20 at 15:00 +0530, VG wrote:
> My server ip is 192.168.22.240
> PRI Gtw ip is 192.168.22.11
>
> When I call in my local users then there is no problem in
> calling.
> But when i call to PRI gateway then:
> Sipx is returning me 407 and 403 Sip message and PRI gateway
> does not get any Bye signal from sipx user side.
> Means call still gets connected for PRI gateway.
> "Bye from sipx user will not be treated by sipx...."
There is a bug in your phone (Voip Phone 1.0); it is not handling the
Record-Route value correctly.
In your merged.xml:
The authenticated INVITE from the phone is frame 169
The 200 response is frame 212, and contains (edited):
SIP/2.0 200 OK
[...]
Contact: <sip:[email protected];transport=tcp>
Record-Route:
<sip:192.168.22.240:5060;lr;sipXecs-rs=%2Afrom%7EMjY5NDgyMzM1Ng%60%60.400_authrules%2Aauth%7E%2184951f2587b302966909e4f962552882>
The ACK request (frame 213) and the first BYE (frame 259) both have an
incorrect Route header value:
Route:
<sip:192.168.22.240:5060;lr;sipXecs-rs=%2525252Afrom%2525257EMjY5NDgyMzM1Ng%25252560%25252560.400_authrules%2525252Aauth%2525257E%2525252184951f2587b302966909e4f962552882>
The '%' characters in the Record-Route returned by the proxy have been
escaped in your phones Route headers, so they become '%25' (in fact,
this appears to have been done three times, so it's actually '%252525').
The value should not be modified at all - it should just be copied from
the initial Record-Route to the Route header in each in-dialog message.
This corruption of the Route header value causes the proxy to ignore the
dialog permission values in it, and so challenges your BYE. The fact
that the authenticated copy of the BYE fails is an artifact of how the
authorization rules matching works - the user part of the target doesn't
match the number of digits in the dial plan. If your phone were doing
the Route header handling correctly you'd never have been challenged for
the BYE.
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