I don't receive rtp message

Asterisk Asterisk Support
1

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
defaultip=223.33.184.3

0000FFFF0002
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.

2

This is a misconfiguration of Your sip.conf.
You should consult sip.conf.sample from the asterisk source tree.

3

externhost, etc. must go in the General section. Also, for localnest, (address & (~mask) should equal zero. They don’t, in your case.

Please also read the documentation for type, nat, and canreinvite, as these are commonly misused or deprecated options, where two are definitely poorly chosen and one is suspect.