Hi! > Issue solved. > Looks like all I was missing was one parameter: > "fromuser="
That's interesting - could be related to this: http://lists.digium.com/pipermail/asterisk-dev/2006-November/024842.html You were probably caught be the fact that you are using extension numbers also as SIP user names for your phones (here: 3666). This is not a good thing to do, better use an alphanumeric username or the phone's MAC address etc. As for your IAX sound quality issue: I have seen that before as well, and switched to SIP (as others did). My guess is that it will probably go away if you use Asterisk 1.4 on both sides, though. SIP DEBUG on the receiving Asterisk gives you a hint which peer was found if matching is done on the IP address, the text is somethint like "Found peer ..." or "Found no matching peer or user for w.x.y.z" Two quotes from the Wiki to explain things better: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer "When Asterisk receives an incoming SIP call, the SIP Channel Module 1. first tries to find a [user] section matching the caller name (From: username), 2. then tries to find a [peer] section matching the caller's IP address. 3. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf." "As of Asterisk 1.2, there is no reason to actually use 'user' entries any more at all; you can use 'type=peer' for everything and the behavior will be much more consistent. All configuration options supported under 'type=user' are also supported under 'type=peer'. The difference between friend and peer is the same as defining _both_ a user and peer, since that is what 'type=friend' does internally. The only benefit of type=user is when you _want_ to match on username regardless of IP the calls originate from. If the peer is registering to you, you don't need it. If they are on a fixed IP, you don't need it. 'type=peer' is _never_ matched on username for incoming calls, only matched on IP address/port number (unless you use insecure=port or higher)." Philipp -- _____________________________________________________________________ -- 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
