Call Initiation problem

Hi,

I have installed sipxpbx server on Fedora core 4.
It is working perfectly.

Worked scenario:
Users under the same network.

Ex: I have registered from the application sipxezphone with 192.168.1.171 IP.
My friend registered from application sipxezphone with 192.168.1.95 IP.

In the URL
https://sipserver.com:8443/sipxconfig/Registrations
I got two lines

First user:
URI : <sip:madhu@sipserver.com>;tag=1714636915
Contact ID : <sip:madhu@192.168.1.95;LINEID=de852303177a>
Expiration : 285

Second user:
URI :<sip:rakesh@sipserver.com>;tag=1784687515
Contact ID :<sip:rakesh@192.168.1.171;LINEID=39d0c5551948>
Expiration :305

I(Madhu) initiated the call to sip:rakesh@sipserver.com it worked perfectly.

Not working scenario:

I am using one broadband connection.
My private ip is 172.28.21.51
My public ip is 210.212.113.97

If I surf the net my requests for URL's will go through my public ip i.e 210.212.113.97
Nobody knows about private ip. Nobody can ping to my private ip.

Ex: I have registered from the application sipxezphone.
My friend registered from application sipxezphone.He is using own public ip 207.156.210.112. He can directly surf the net. Anybody can ping to his ip.

In the URL
https://sipserver.com:8443/sipxconfig/Registrations
I got two lines

First user:
URI : <sip:madhu@sipserver.com>;tag=1714636915
Contact ID : <sip:madhu@172.28.21.51;LINEID=de852303177a>
Expiration : 285

Second user:
URI :<sip:rakesh@sipserver.com>;tag=1784687515
Contact ID :<sip:rakesh@207.156.210.112;LINEID=39d0c5551948>
Expiration :305

First problem :
I(Madhu) initiated the call to rakesh(sip:rakesh@sipserver.com).The call connected and rekesh is getting my voice but I am(madhu) not getting his voice.

Second problem :
When rakesh initiates a call to madhu(sip:madhu@sipserver.com) the call is not connecting. I think the problem is with ip, because nobody can ping to my private ip. So its unable to create a socket connection.

Expected solution :
Actually in the registration the contact id for the First user: should be
Contact ID : <sip:madhu@210.212.113.97;LINEID=de852303177a>
I think, then it will work.

What you people will say? Can anybody help? Please .....
Thank u.

Hi Madhu and welcome to VoIP User.

The problem is your NAT device.

Have a look through the "One way audio fault-finder" thread in the FAQ forum for more info.

Your first problem and second problem are completely related.

Dean

Thanks a lot.

We have turned on stun server.
Its working perfectly.

Hi,

Now the situation is with adsl connection.
Whenever I login I will get dynamic ip to my system.

When I send a registration request to sipxpbx server,
it is giving status message REGITRATION TIMED OUT : REGISTRATION FAILED

What could be the problem?
Can you help please?

Thanks.

Again, sounds like ports to me.

You could try port forwarding port 5060 UDP to the softphone PC or hardware phone device.

Dean

Hi,

Thanks a lot.

I am requesting you not to get angry for the question I am going to put forward.

If you don't mind can explain in clear what is meant by port forwarding and how to do it?

I am new to this work. Help me please...