Wonder if someone could enlighten me.
My current setup is * behind a nat firewall
Remote extensions behind their own nat firewalls
I can call any extenstion just fine as I’m behind the firewall with *
The remotes can call me fine
The problem is, when the remotes try to call each other. Phone rings (because it’s being passed thru * I presume) but when connected, niether side has any audio. My guess this is rtp related.
Anyone solved something similar?
rgds
What phones are the remote users using? If they are sip phones, can they capable of STUN?
Problem sorted.
I hadn’t specified all of the local networks in sip.conf. All traffic now passes via the VPN.
Works like a charm
rgds
I may be having the same problem you had. Where I work, we have two networks, a DSL connection, and a cable connection, both static IPs. I have my asterisk server on the DSL connection behind a router with ports 5060 and 10001-20001 forwarded to the server. My phones are on the other network, behind a router.
Things are working okay, but for about every 1 out of 7 calls, there is no audio.
here’s my sip.conf:
[general]
context=incoming
recordhistory=yes
port=5060
fromdomain=dsldomainname.com
bindaddr=192.168.6.103 ;server ip address, to which ports are forwarded
srvlookup=yes
videosupport=yes
language=en
nat=yes ;because i’m behind the router?
externip=external.dsl.ip.address
localnet=192.168.6.0/24
[15]
type=friend
context=internal
host=dynamic
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
qualify=yes
nat=yes
and my other extensions are configured the same.
[quote=“mstodd”]Ilocalnet=192.168.6.0/24
[/quote]
And you only have one local net?