the client that is behind nat is [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
do I have to enable nat on all of them? On Thu, Sep 16, 2010 at 1:36 PM, Sebastian <s...@open-t.co.uk> wrote: > > > 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 >
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