[asterisk-users] Beginner Issues
Noah Miller
noahisaacmiller at gmail.com
Wed Jul 16 08:35:50 CDT 2008
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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