[asterisk-users] Beginner Issues

[John Koenig]

That could be...I only have ports 5060 and 8088 open on the firewall.  
Should another port be open?

The phone I am using are pstn phones connected to a 2 port Linksys PAP2. 
I made sure that I checked the NAT option under the user account and 
enabled NAT Keep Alive under the PAP2 management interface.  I am using 
the G726-16 codec for transmission.

Attached is my sip.conf.

John


Gerard A. Matthew wrote:
> Are your phones behind NAT?
>
> This should be an issue with rtp port communication. 
>
> Gerard.
>
> ------Original Message------
> From: John Koenig
> Sender: asterisk-users-bounces at lists.digium.com
> To: asterisk-users at lists.digium.com
> ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
> Sent: Jul 15, 2008 6:47 PM
> Subject: [asterisk-users] Beginner Issues
>
> I am new to asterisk, and I am having some troubles.
>
> I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and 
> asterisk-gui installed on centos (I built everything using ./configure, 
> make, make install, make samples).  I connected to the GUI interface and 
> created two new users.   I used the two users accounts to connect up a 
> couple of IP phones for testing.  The phones connect to the server just 
> fine, and I can even place a phone call to the other phone.  However, I 
> cannot hear anything on the dialed phone.  The only thing I am able to 
> hear is my own voice looping back to the phone I place the call from. 
>
> Any ideas as to what I am missing?
>
> John Koenig
>
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