[asterisk-gui] asterisk 1.4.2 and outgoing sip providers
Justin Hamade
jhamade at shorebridge.ca
Fri Mar 30 05:28:38 MST 2007
This belongs in the asterisk-users list.
Seems like a NAT issue, make sure nat=yes in your sip.conf and take a
look at voip-info.org
__
Justin
________________________________
From: asterisk-gui-bounces at lists.digium.com
[mailto:asterisk-gui-bounces at lists.digium.com] On Behalf Of Eric Swanson
Sent: Friday, March 30, 2007 5:25 AM
To: asterisk-gui at lists.digium.com
Subject: [asterisk-gui] asterisk 1.4.2 and outgoing sip providers
I am a newbie to asterisk.
I have an asterisk server that is behind a cisco pix firewall. I have
opened up ports udp / tcp 5060 and rtp 10000 - 20000 ports.
When I make a phone call out through the sip provider, I can not hear
the other person, but the other person can hear me. Do I need to setup a
stun entry for the asterisk server since it is behind the firewall?
How can I fix this issue?
Thanks
Eric
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