On 11/22/05, Bharath Khambadkone <[EMAIL PROTECTED]> wrote: > Hello All, > I'm fairly new to asterisk. I have read about the problems about NAT, But > can't seem to find a solution. > My Asterisk is on a public domain, there is no NAT or firewall in front of
If no nat then why do you have nat=1 in sip.conf? > the asteris box. I have sucessfully connected iax2 softphones & was able to > recieve & make calls. In the same locations where I have the iax2 extensions > working I have set up a a SIP softphone & a SIP ATA (Sipura2002). Both teh > sip phones are able to register. I can also make & recieve calls but cannot > hear anything after the call is answered at both ends. I'm not sure what is > causing this problem. By the way I'm using SME server 7(centos 4.2) with > [EMAIL PROTECTED] installed. > > my Sip.conf : > [2008] ;(Sipura2002) > username=2008 > type=friend > secret=2008 > record_out=Adhoc > record_in=Adhoc > qualify=no > port=5060 > nat=1 > [EMAIL PROTECTED] > host=dynamic > dtmfmode=rfc2833 > context=from-internal > canreinvite=no > callerid=device <2008> > > > [2009] ;X-Lite Soft Phone > username=2009 > type=friend > secret=2009 > record_out=Adhoc > record_in=Adhoc > qualify=no > port=5060 > nat=1 > [EMAIL PROTECTED] > host=dynamic > dtmfmode=rfc2833 > context=from-internal > canreinvite=no > callerid=device <2009> > > Thanks in advance.. > > > > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > 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 > > _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
