Geetings to all. I am having a hell of a time getting incoming SIP connections to work properly, and am hoping that someone can help me. Here is what I am using as a guide (from the wiki):
"Incoming SIP Connections When Asterisk receives an incoming SIP call, the SIP Channel Module first tries to find a [user] section matching the caller name (From: username), then tries to find a [peer] section matching the caller's IP address. If no matching user or peer is found, the call is sent to the context defined in the [general] section of sip.conf." I am mainly concerned with the second point. I want to match an incoming SIP connection to a particular IP address. I have tried just about everything, and the connection always goes to the default context, or the context defined at the top of the sip.conf file. I would like to be able to direct incoming SIP connection to a particular set of extensions. There is no username and password involved as there will be many users coming from this one IP. This is what I have tried recently: [sipin_test] type=peer defaultip=195.27.242.120 context=test_trunk deny=0.0.0.0/0.0.0.0 permit=195.27.242.120/255.255.255.255 dtmfmode=rfc2833 disallow=all allow=ulaw nat=no I have also tried changing what is inside the brackets to the IP address. I have tried many many different combinations of the above, but the IP address never seems to get picked up correctly. I am testing the SIP connection using sipsak. I realize that Asterisk is probably not the best SIP server to use, and plan on migration to SER, but if anyone can offer any suggestions I would really appreciate it. Regards to all, Joe _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
