Hi there. i am seriously stuck with an asterisk and sip problem.
the following sip.conf works with respect to some_peer: [general] bindaddr = x.y.z.w nat = no [some_peer] type=peer host=somehost secret=somesecret some other unrelated options here x.y.z.w is the ip address of the interface pointing to the network containing somehost. more precisely its the address of tun0 and route -n prints Destination Gateway Genmask Flags Metric Ref Use Iface [..] x.y.z.0 0.0.0.0 255.255.255.0 U 0 0 0 tun0 [..] here 'it works' implies that i have to change and reload sip.conf after ifup tun0, or anything that forces tun0 to go down, like my dsl provider. also, the bindaddr line is suboptimal for the other peers... the same thing -- without the bindaddr part -- doesnt work. more precisely it almost works. its just incoming sound that doesnt. this must have something to do with how asterisk picks up interface addresses and communicates them to the peer in question. inspecting the packages sent to somehost, gave me the impression that asterisk uses the ip adress of ppp0 (a dsl modem) instead. how am i supposed to tell asterisk to use tun0 as the interface for [some_peer] so i can remove the bindaddr line? i've found many nat-related options in the manual, but there is no nat involved here. also, i couldnt find anything similar to "iface=tun0", although the sip dialogue apparently relies on ip adresses and routing. this is about asterisk on sid, version 1:1.8.13.0~dfsg-1, but of course i'm going to switch to whatever you might suggest. regards and thanks felix -- _____________________________________________________________________ -- 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