INVITE retransmission by 1.8…

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I have a site that moved to the latest 1.8 revision, and began to
have problems with phones in “far away places” (South America,
and the MidEast).

What I see is that when a Dial() is issued, the sip channel driver
sends out an INVITE to the phone. Very soon thereafter, Asterisk
retransmits the INVITE. The phone sends back a 100 Trying, and
then, usually, a 400 response. I may be misinterpreting what I see,
but I *think* that the response from the phone is delayed just enough
to invoke the retransmission. The phone responds to the second invite
with a 400 code, which Asterisk interprets as an error, and the call
immediately
goes into hangup.

Has anyone else seen this? It doesn’t happen all the time, and only with
certain
phones. It comes and goes, but when it comes, phones become unreachable. It
seems to be in this state the majority of the time for certain phones.
While most
phones seem far away, some are in Florida.

We replaced the 1.8 version of Asterisk with a 1.6.2 version, and the issue
has
gone away. I know, I know, I could give more detail, fill out a bug
report, but
this is the early stages. I may be misinterpreting what I’m seeing.

Anyone else seen this sort of thing? Any words of wisdom?

murf