[asterisk-users] Still true: only first peer matched on incoming call?

[sean darcy]

I've got two 1.6.2 asterisk boxes. I'd like to be able to set up two 
separate sip connections. But when I try that I get:

chan_sip.c:12671 check_auth: username mismatch, have <one-sip-peer>, 
digest has <another-sip-peer>

Looking around I found this in a 2007 bug report on version 1.4.4,
https://issues.asterisk.org/view.php?id=9678:

THis is well known. There is a lot of available documentation out there. 
Basically: We only match the first peer on the incoming call, which is 
the last peer in the sip.conf file. Yes, I know it is awkward, but it is 
the way it works now.

Still the case? Or is there some clever way around this?

sean