[Asterisk-Users] Asterisk + SER and NAT

[Guy Decarpentrie]

Le mardi 10 Mai 2005 08:01, Laurent Foulonneau a écrit :
> Hi,

Hello,

>
> We are testing a SIP solution * + ser solution for a large implementation.
> All the clients are nated.
> When a client is dialing outside the domain (to a FWD sip account for
> example) all is perfect ! ;-)
> But ,when a call is done to a sip account, the client is ringing, then the
> caller can hear the nated client very well, but the nated client does'nt
> hear anything. RTP issue no ?
> I've follow the SER, Asterisk and Lucent TNT by Michael
> Shuler  (http://www.voip-info.org/wiki-Asterisk+at+large ), but I think
> I've missed something.
> The * and ser are using public ip, no nat for them.
> I've tried different config, with and without rtpproxy, with forward
> instead of t_relay, but same or more problems.
>
> If someone could help me please.
>
> Here are my conf files :
>

Try to add canreinvite=yes in general section of sip.conf, and switch it to 
yes for the ser server.

++

-- 
Guy Decarpentrie - ipnotic - switch to ip
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