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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
