On 3/29/2011 2:29 PM, Warren Selby wrote:
It looks like you did to me. Is it just OPTIONS packets that are showing the wrong fromuser field? In other words, when you send call traffic over this peer, does it properly create the SIP packets? For some reason, I'm
correct - when i actually invite a call or do the register, the from uri is correct. it's just the options packet that is broken.
sip development may be able to better tell you. Perhaps open a ticket on the bug tracker?
yep, that was the next step - just wanted to run it by a few more eyes before i bothered the devs.
https://issues.asterisk.org/view.php?id=19036 -- Jeremy Kister http://jeremy.kister.net./ -- _____________________________________________________________________ -- 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
