[asterisk-users] SIP Transit problem

[Gary Mensenares]

Hi!

Hope someone can help me. I'm trying to pass SIP traffic from one asterisk
to another through a third server. Here is the desired scenario:

ServerA -- SIP --> ServerB -- SIP --> ServerC

When a call is placed on a ServerA local, I can see that ServerB receives
the call and dials ServerC. But ServerC says:

Jun  8 09:38:32 NOTICE[3269] chan_sip.c: Failed to authenticate user
"asterisk" <sip:ServerB-user at 123.456.789.012>;tag=as15c8b5e0

However, when I change the configuration between ServerA and ServerB such
that:

ServerA -- IAX/2 --> ServerB -- SIP --> ServerC

This works just fine.

If I understand correctly, ServerA only needs to authenticate to ServerB.
The fact that ServerB dials ServerC when both legs are SIP seems to indicate
that there is no AUTH problem between A and B. And with the 2nd scenario, it
proves that there is no auth issue between B and C.

Am I missing something? Has anybody got a recipe for this?

I'd appreciate any info. Thanks

Jug Mensenares