Recently I have purchased the Asus RT-N66U router.
Here is the setup:
[Network 1]
Public IP: 100.100.100.10
Local Network: 192.168.1.0/255.255.255.0
Router: Asus RT-N66U
Server: Asterisk 1.4.17
[Network 2]
Public IP: 200.200.200.20
Local Network: 192.168.0.0/255.255.255.0
Router: Linksys with DD-WRT
Sip clients – Asterisk 1.4.17 --Asus RT-N66U–[Network 1]–Internet–[Network2]–Linksys DD-WRT-- Sip clients
Sip clients on the Network 1 can receive incoming calls and make outgoing calls.
Sip clients on the Network 2 can register with the server, can make outgoing calls and receive incoming calls but without audio on both ends.
I understand that Asterisk is behind NAT and enabled port forwarding on ASUS RT-N66U under Virtual Server/Port Forwarding:
SIP_PORT 5060:5082 192.168.1.x BOTH (TCP & UDP)
RTP_PORTS 8000:25000 192.168.1.x BOTH (TCP & UDP)
On the Asterisk:
[sip.conf]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allowexternaldomains=yes
allowguest=yes
allowsubscribe=yes
allowtransfer=yes
alwaysauthreject=yes
autodomain=no
callevents=no
canreinvite=no
checkmwi=10
compactheaders=yes
defaultexpiry=120
domain=
dtmfmode=auto
dumphistory=no
externrefresh=10
fromdomain=
g726nonstandard=yes
jbenable=no
jbforce=no
jbimpl=
jblog=no
jbmaxsize=
jbresyncthreshold=
language=
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
mohinterpret=default
mohsuggest=
mwi_from=
nat=yes
qualify=yes
notifyringing=yes
pedantic=no
progressinband=never
promiscredir=no
realm=asterisk
recordhistory=no
registerattempts=0
registertimeout=20
relaxdtmf=yes
rfc2833compensate=yes
rtpholdtimeout=300
rtptimeout=60
sendrpid=no
sipdebug=no
t1min=100
t38pt_udptl=no
tos_audio=none
tos_sip=none
tos_video=none
trustrpid=no
useragent=Asterisk PBX
usereqphone=yes
videosupport=yes
insecure=very
externip=100.100.100.10
localnet=192.168.1.0/255.255.255.0
subscribecontext=device-hints
disallow=all
allow=ulaw,alaw,gsm,g726,ilbc,speex,adpcm,lpc10,g723
[rtp.conf]
rtpstart=8000
rtpend=25000
Sip clients use Softphone X-Lite and Nortel 1535
I’m not sure whether it is something to do with the ASUS router, Asterisk configurations or sip clients configurations. It seems that sip clients can register and send sip requests but audio packets are blocked or unreachable.
Any help will be much appreciated.