eric at fnords.org
Guest
|
 Posted: Tue May 13, 2008 11:50 am Post subject: [asterisk-users] More one way audio... |
 |
|
I have never seen a SIP aware firewall work with localnet and
externip/externhost. You should try either disabling the SIP fixup on
your firewall or remove the localnet/externip from sip.conf.
Carlos Chavez wrote:
| Quote: | I am a bit desperate trying to solve this problem. Sorry if I am
abusing the list a bit with the same king of question.
The problem I am having is very specific which is why it is very
difficult to diagnose and fix. Basically an Asterisk server is
connected via E1 PRI to an Avaya PBX. The Asterisk server has 45 PAP2T
and 45 SPA-3102 devices connected via the Internet. The Asterisk server
is behind a Fortinet firewall and has all necessary ports redirected to
it.
By itself, everything is working. I can make and receive calls to all
SIP devices, check voicemail and any other service I configure on the
Asterisk server. I have the relevant parts of NAT configured like
"externip", localnet, nat=yes and canreinvite=no. The problem only
presents itself when a SIP device is trying to call an extension
connected to the Avaya. Since "localnet=192.168.2.0/255.255.255.0" is
defined and the Fortinet firewall rewrites the source IP as its own
"192.168.2.1", I think this may be the cause of my problems but why only
when calling the Avaya and not other SIP extensions or Asterisk
services?
Since the SPA3102 has Symmetric RTP it works fine. The PAP2T on the
other hand gives one way audio when you call any extension on the Avaya.
The only way I can get the PAP2T to work is to change the localnet to
something else then it works properly but the SPA does not. Any call I
make from the SPA hangs up after a minute or so and any call I make
rings the SPA but I do not get any audio.
What is the proper NAT setup for something like this? Is it even
possible to work with this type of NAT? Any comment would be truly
appreciated.
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
|
Search
