Incoming call no sound asterisk behind Linksys WRT54GP2 NAT


#1

Hi,

I’ve been trying to solve this problem for the last week and I would really appreciate any help with it. I have a linksys WRT54GP2 NAT router, with a asteris server behind the NAT and users from from a larger private network connecting from outside the NAT. My problem is they can hear me but I can’t hear them. I’ve notice that there are no packets being sent back to me accross the network and it uses port 8000 to try and send them. Below is my configuration:

sip.config
[general]
nat=yes
externip=xxx.xxx.xxx.xxx
localnet=xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx
bindport=6060
bindaddr = 0.0.0.0
srvlookup=yes
context=general

rtp.config
rtpstart=10000
rtpend=16000

A users xlite softphone settings would look like this:
Enabled: Yes
Display Name: username
username: username
authorization user: username
password: password
domain/realm:asterisk
sip proxy: xxx.xxx.xxx.xxx:6060
out bound proxy: xxx.xxx.xxx.xxx:6060

where sip/out bound proxy match the externip of sip.config

I have port forward the following ports:
6060
10000-16000

Again I be very grateful if anyone can stir me in the right direction:-)

cheers,

Jonathan


#2

first off, I had come across a thread a couple of months ago where someone was having problems with that Linksys and after a firmware upgrade, was ok - in that case though I don’t think it was the same issue.

Next - it woudl be generally good to try and stick with the standard SIP ports or you may likely be asking for more complications than you need. (At least get it working that way.) So - try forwarding UDP 5060-5080 (to be safe), 5004 for some devices, and 10k-20k rtp and put Asterisk back to 10-20K in rtp.conf. Once you have that working, if you have reasons to go changing to non standard ports, you will have a known working configuration to start from.

p

(another quick and dirty test would be to briefly toss Asterisk on the DMZ and see if you have the same porblem or not, it will help determine if in fact the NAT and/or firewall are creating your problems or something else. (But don’t leave it on the DMZ for any amount of time without properly closing up the firewall on Astersisk…)


#3

Hi,

Thanks for the response… I should have pointed out that my router is an ATA router and I have connected it to an external VoIP provider. My router listens to 5060 and when I had asterisk set to this default port no one can login, this is why I changed it to port 6060. I have read through all the tutorial and posts and I couldn’t find anything relating to the problem I am having. I returned the rtp.config ports to the normal 10000-20000. I have tried making calls between pc’s inside the NAT and this works fine. I put the asterisk server onto DMZ and unfortunately I am still having the same problem. :unamused: