Nasir Javaid
2010-Jul-28 16:30 UTC
[asterisk-users] what is rinstance parameter in sip header
hello i was wondering what is the use of "rinstance" in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio when dialing a registered user by his IP:port. I this case "rinstance" parameter is missing. when i dial "SIP/username" audio is fine but when i dial SIP/x.x.x.x:port there is one way audion. Also please tell me what can go wrong by dialing by ip:port.?? Best regards, Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100728/11584028/attachment.htm