Hi My English is bad SORRY for that

i setup Elastix server
- Change ip to 192.168.1.53 static, setup hostname exemple.domain.com, DNS & gateway setup 192.168.1.1
- Change in sip_nat.conf
nat=yes
externhost=exemple.domain.com
localnet=192.168.1.53/255.255.255.0

extentions are online but i cant hear

what is wrong

i have also tested localnet=192.168.1.53/255.255.255.0 to remove

then work internel extentions working ok but externel extentions are online is server but no sound

i have Linksys router Ports are forwed to server 5060 tcp/udp & 10000-20000 udp

Plz help me out friends

localnet=192.168.1.53/255.255.255.0


Does not describe a legitimate network:

192.168.1.53 is a host in a network, if the netmask is 255.255.255.0 (a /24 network)
then it is in the network described by

localnet=192.168.1.0/255.255.255.0 (or 192.168.1.0/24)

Oups.
I was used the bad post.
It was funny a post in french into an English post, no?

hi

i change sip_nat_conf to

nat=yes
externhost=exemple.domain.com
localnet=192.168.1.0/255.255.255.0

stil no sound

1 extention is in internel network
1 extention is out (externel network)

dicko wrote:
localnet=192.168.1.53/255.255.255.0


Does not describe a legitimate network:

192.168.1.53 is a host in a network, if the netmask is 255.255.255.0 (a /24 network)
then it is in the network described by

localnet=192.168.1.0/255.255.255.0 (or 192.168.1.0/24)

Exactly Dicko

Hi Dicko , how are you?

Which is the internal and which the external Atifa or Azhar and is the internal extension a softphone on the notebook or 7000? Did you you reload SIP (or restart asterisk or reboot) after the change

(please edit your previous post to obfuscate IPs or names that are resolvable, please check your security settings in FreePBX as your server is way too open.)

extention 7000 is Local in my huis
extention 8000 is Externel user

i restart always after changing

Then recheck the NAT boxes involved both ends for correct PNAT translation of UDP ports 10-20000, (tcpdump is a good diagnostic tool here) also use the standard debugging tools:

sip set debug (to verify the SIP headers are rewritten properly)

and

rtp debug (to check bi-directional rtp (audio) streams)

p.s. this is usually caused by PNAT boxes that are too clever for themselves if the sip_nat is right (and yours now looks correct)

sorry it is to dificcult english

can u help me out with easy english

i have been checked my router again 10-20000 UDP is open

but stil no sound if u like i like to add u in msn

then wee can have a small chat

In your scenario there are two routers involved, one between Elastix and the internet and one between the remote extension and the internet, BOTH need to be set up to correctly forward sip and audio.

I noticed that you have no SIP trunks yet, and I suggest you set one up (there are free ones available) to eliminate any possible remote extension router problem. When you can make and receive calls from a SIP carrier to your Elastix box, then tackle the remote extension problem.

(Sorry I won't take personal calls, it's a privacy thing, and it defeats the Forum concept)

regards

dicko