[asterisk-users] sip.conf and binaddr issue

[Felix Salfelder]

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