Problem with incoming traffic

Hello,

I have succeeded to create my Switchboard the way i want. But there is one issue i don’t seem to find a solution for. When i call other users internally there is no sound. According to the statics its because its not getting any incoming packages.

This is how my user looks like in users.conf

[6001]
fullname=User
registersip=no
host=dynamic
callgroup=1
mailbox=6001
call-limit=100
type=peer
username=6001
transfer=yes
callcounter=yes
context=DLPN_DialPlan1
cid_number=6001
hasvoicemail=no
vmsecret=
email=
threewaycalling=no
hasdirectory=yes
callwaiting=no
hasmanager=yes
hasagent=yes
hassip=yes
hasiax=yes
secret=secretpassword
nat=yes
canreinvite=no
dtmfmode=rfc2833
insecure=port,invite
fromdomain=sip.uvtc.net
pickupgroup=1
macaddress=6001
autoprov=yes
label=6001
linenumber=1
LINEKEYS=1
managerread=system,call,log,verbose,command,agent,user,config,originate
managerwrite=system,call,log,verbose,command,agent,user,config,originate
disallow=all
allow=ulaw,gsm

[general]
fullname=New User
userbase=6000
hasvoicemail=yes
vmsecret=1234
hassip=yes
hasiax=yes
hasmanager=no
callwaiting=yes
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
callgroup=1
pickupgroup=1
port=5060

I would be very thankful if anyone could help me resolve this. I really want it to work.

// Ivan

users.conf is deprecated.

Typically you would have a NAT or firewall issue, e.g. you may have forgotten to NAT/pass the RTP port range.

I can’t tell whether you need special NAT handling without more information on your network connectivity and I can’t tell whether you have told Asterisk how to find its external address and possibly override the RTP address without sip.conf.