[asterisk-users] More one way audio...

[Eric Wieling]

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:
> 	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.