I've never tried softphones on Linux, but my guess is that since you run kphone and asterisk on the same server you get a port conflict. If the client uses port 5060 (default sip port) it would defenitely have problem connecting to an asterisk on the same port.
Maybe you can change the kphone settings to use some other port or something :) /Anders > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Alex Polite > Sent: den 18 december 2004 14:53 > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Setting up asterisk for one user in > private ip NAT. > > Hi. > > I've just bought SIP telephony service from a Swedish telco. > > I've managed to make and receive calls with kphone. > > Now I want to set up asterisk to be able to add fancy > features like voice mail and recording conversations. But > first I have to get the basic setup right. I'm running > asterisk and kphone on the same machine, behind at NAT-router. > > When I make a call (from my regular phone) to the SIP-number > I get a busy signal and I see my regular phone number in the > debug output of Asterisk. I guess that means I'm doing > something right. > > The problem now is that I can't get kphone or linphone to > connect to asterisk. Trying to connect from kphone to > asterisk does not generate any messages in the asterisk debug > output. Non what so ever. > > Which has me thinking that ip might be something with the > hostnames/ip-addresses that's not right? > > What does "bindaddr" do? I've tried changing it to my private > IP but that doesn't make any difference. > > > I know that I'm not being very specific in my questions but I > feel that I need some handholding here. Some tests that I can > run, for example, to find out if my Asterisk setup is kosher. > So, will someone please hold my hand in this scary land of VOIP? > > Alex > > Here are my config files so far. > > sip.conf > ---------------- > [general] > context=default > port=5060 > bindaddr=0.0.0.0 > srvlookup=yes > > externip = <public ip of router> > localnet = 192.168.0.0/255.255.255.0 ; Internal > NETWORK address > allow=ulaw > allow=alaw > allow=gsm > allow=all > nat=yes > > register => xxxxxxxx:[EMAIL PROTECTED]/1000 > > [alex] > type=friend > host=dynamic > username=alex > secret=zzzzzzzz > context=outgoing > > > [rix] > type=peer > username=xxxxxxxxxx > fromuser=xxxxxxxxxx > secret=yyyyyyyy > host=astrofix.rixtele.com > fromdomain=astrofix.rixtele.com > context=sip-in > insecure=very > nat=yes > ---------------------------- > > > > extensions.conf > ---------------------------- > [default] > exten => 1000,1,Dial(SIP/alex||t) > > > [sip-in] > exten => 1000,1,Dial(SIP/alex||t) > > [outgoing] > exten => _0.,1,Dial(SIP/rix/${EXTEN}|20|t) > > > ---------------------------- > > .qt/kphonerc > ---------------------------- > [Registration] > AutoRegister=No > SipServer= > SipUri="Alex Polite" <sip:[EMAIL PROTECTED]> UserName=alex qValue= > > ---------------------------- > > -- > Alex Polite > http://polite.se > _______________________________________________ > 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
