[Asterisk-Users] SIP re-INVITES problem

[Glenn Dalgliesh]

When a call is place to xxx9931211 from the pstn the call proceeds normally
until asterisk issues the Second INVITE, which is MESSAGE 14, and instead of
call being sent with INVITE sip:xxx9931211 at proxy.yyyyy.net SIP/2.0. It gets
sent with INVITE sip:xxx9931211 at yyy.33.165.201:5060 SIP/2.0 and this seems
to cause SNOM proxy to return the packet without a Record-Route entry and
then asterisk starts sending the packets to the UA directly. Not sure if
this is a bug or not but it seems odd to me that the INVITE and re-INVITE
messages have different fields in them.

Also, if I test the same scenario with canreinvite=no since * doesn't issue
a re-INIVTE the call completes properly and all messages go thru the SNOM
proxy to reach the UA.

Any insight would be appreciated.

Thanks

Glenn

pstn-> asterisk ---------> snom ------------> UA
            (xxx.99.77.23)     (xxx.93.91.74)    (yyy.33.165.201)

     SIP MESSAGE 3        xxx.99.77.23:5060(2) -> xxx.93.91.74:5060(4)
     UDP Frame 3        19/Apr/04 18:17:47.9517
TimeFromPreviousSipFrame=0.1666 TimeFromStart=0.1676
INVITE sip:xxx9931211 at proxy.yyyyy.net SIP/2.0

- Re-Invite
     SIP MESSAGE 14       xxx.99.77.23:5060(2) -> xxx.93.91.74:5060(4)
     UDP Frame 14       19/Apr/04 18:17:50.4408
TimeFromPreviousSipFrame=0.0003 TimeFromStart=2.6566
INVITE sip:xxx9931211 at yyy.33.165.201:5060 SIP/2.0