[Asterisk-Dev] Re: [Asterisk-Users] SIP change...

[Dave Packham]

Interesting.  I am working on getting CID to work from * to my Cisco routers.  I have a tac case open and they are giving me debug IOS's to work with but this is what they have come up with.   Dont know if this will help


quoted from my talks with Cisco TAC

Hi Dave -

A few more questions from our SIP guys.  

Normally the caller-id is taken from "remote-party-id" in the SIP
INVITE.  We don't see that field poplated in this INVITE.  What is the
originating gateway?  What device is sending the call to the 827?  We
should be seeing "remote-party-id" in the INVITE.

Thanks,

Clay

end quote

Thanks
Dave


>>> markster at digium.com 8/23/2003 1:09:58 AM >>>
I've made a subtle but important SIP change as part of bug #155.
According to Mediatrix, the URI in "Contact" should be used as the URI in
the top part of the SIP request when sending follow up messages, and
they're allowed to use the same IP for from and to.  I made the change to
support that but I want to be sure we didn't break anything else, so if it
did, let me know, on list, off-list, or on IRC (or via the bug tracker).

Mark

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