0
votes

I'm need to use SIP server.

My choice is 'Asterisk'

That's version is 1.8.0

I configured all of the asterisk.

and... i'm calling two users (0000FFFF0001,0000FFFF0002) using X-lite, Zoiper.

User Calling is not problem. It's fantastic.

But i can't hear nothing.

I just can calling other user, end the call.

My source is below.

sip.conf

[office-phone](!)
type=friend
host=dynamic
nat=yes
secret=pspsps
dtmfmode=auto
disallow=all
allow=alaw
allow=ulaw
qualify=yes
canreinvite=no
context=LocalSets

[0000FFFF0001](office-phone)
defaultip=223.33.184.3

[0000FFFF0002](office-phone)
externip=192.168.194.2
localnet=192.168.0.100/255.255.255.0

extensions.conf

[LocalSets]
exten => 100,1,Dial(SIP/0000FFFF0001)
exten => 101,1,Dial(SIP/0000FFFF0002)

exten => 200,1,Answer()
exten => 200,n,SayNumber(5)
exten => 200,n,Wait(1)
exten => 200,n,SayNumber(5)
exten => 200,n,Hangup()

I expected rtp problem.

I was opened TCP/UDP ports 20000~30000.

and my sharer NAT was configured.

help me. i need advise of you.

1

1 Answers

0
votes

I believe you need to put your externip, localnet, and canreinvite in your general setting, Not in the sip peer itself.

Do you have the ports in rtp.conf natted to your server and allowed through your firewall?

I recommend you make two test extensions, Let's use 201 that answers and calls the MusicOnHold application, If you call that extension from your sip phone do you get audio?

Create Extension 202 that answers and calls the Echo Application, If you speak into your phone do you hear your voice Echo'd back to you?