I tried doing a sample test with a softphone behind nat and trying to connect to my asterisk who has ports forwarded, so far, it can connect but as usual, I can hear the prompt but for example, using the echo test, I don't hear myself back.
By doing a sip show peers I see the softphone connected but instead of showing using port 5060, it shows using port 64112 for example. I have nat=yes and canreinvite=no ... Any ideas? I was thinking about stun and ser but what do you guys think? -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Martes, 29 de Marzo de 2005 10:28 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Anton Krall wrote: > Any problems with RTP or voice just on one side? > > So as long as you use some STUN server, the RTP packets have the right IP. > Did you install your own stund or are you using a public one? > > You didn't have to use SER at all right? Setting nat=yes does pretty much the same as a STUN server. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
