On 09/16/2010 06:58 PM, Thomas Johnson wrote: > I am having a one way audio issue with xlite clients behind NAT. They > can connect to the server and make calls but no audio is heard on the > other end. > > my sip conf > > [general] > context=default > bindport=5060 > bindaddr=0.0.0.0 > srvlookup=yes > canreinvite=no > > [tomfmason] > type=friend > secret=secret > callerid="Thomas Johnson" <XXXX> > host=dynamic > nat=yes > canreinvite=no > disallow=all > allow=gsm > allow=ulaw > allow=alaw > qualify=yes > context=sip > > [1001];Work > type=peer > dtmfmode=rfc2833 > context=sip > insecure=very > host=sip.domain.com <http://sip.domain.com> > nat=no > > [1000];IPKall > type=peer > dtmfmode=rfc2833 > context=sip > insecure=very > host=voiper.ipkall.com <http://voiper.ipkall.com> > nat=no
You seem to be using nat=no shouldn't that be nat=yes? > > > > I pasted the log here -> http://pastie.org/1163238 > > > I have tried connecting both of the clients to another sip > service(sip2sip.info <http://sip2sip.info>) and did not have the same > problems. > > > Any suggestions would be great. > > Thanks, > > Tom > -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
