[asterisk-users] Beginner Issues

John Koenig koenigjm at acalledshot.net
Wed Jul 16 11:34:21 CDT 2008


Thanks! Opening the ports did the trick!

John


Noah Miller wrote:
> Hi John -
>
>   
>> That could be...I only have ports 5060 and 8088 open on the firewall.
>>  Should another port be open?
>>     
>
> If asterisk is inside a firewall/nat and the phone devices are on the
> other side, you need to also open port for the rtp audio stream.  By
> default, this is UDP 10000 - 20000, but this range can be modified in
> rtp.conf
>
>
>   
>> 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.
>>     
>
> Aha.  You're using the GUI.  In that case, the useful info will be in
> users.conf.
>
>
> - Noah
>
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