Hi all I have badly NATed Clients proble with one way Voice After reading some documents people ask me to use STUN Server But still i have some problem with one way Voice I have setup like below iam trying with 2 extensions 1 extention in the same LAN where the * installed 2 extension in different network, NATed IP , 3. both the side iam use SIPURA 4. i have 2 DID from provider 5. i have route them to appropriate extensions Iam able to make calls in and out but the problem where iam setting up server have limited bandwidth So i have installed G729 codec So i want to make RTP so i made setup caninvite=yes since my provider support that option then my NAT Clients have One way Voice problem So after Reading some DOCS SER, should be able to do this Job so SER can be integrated with *, if yes can any one point me to some URL thanks ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/cf50ac94/attachment.htm
Andrei Sotirov
2006-Mar-16 03:24 UTC
[Asterisk-Users] SER & Asterisk with DID incoming and out going
ram wrote:> Hi all > > I have badly NATed Clients proble with one way Voice > > After reading some documents people ask me to use STUN Server > But still i have some problem with one way Voiceuse stun on dinamic ip :)> > I have setup like below > > iam trying with 2 extensions > > 1 extention in the same LAN where the * installed > 2 extension in different network, NATed IP , > 3. both the side iam use SIPURA > 4. i have 2 DID from provider > 5. i have route them to appropriate extensions > > Iam able to make calls in and out > > but the problem where iam setting up server have limited bandwidth > So i have installed G729 codec > > So i want to make RTP > > so i made setup caninvite=yes >canreinvite=no nat=yes> since my provider support that option > > then my NAT Clients have One way Voice problem > > So after Reading some DOCS SER, should be able to do this Job > > so SER can be integrated with *, if yes > can any one point me to some URL > > thanks > > ram > > ------------------------------------------------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- ????????, ?????? ???????
Hi thanks for the reply ya the default is NAT=YES only if i keep reinvite=no, the my server b/w consuming lot since i have bottleneck of server bandwidth ram On 3/16/06, Andrei Sotirov <andrei@unixsol.org> wrote:> > ram wrote: > > Hi all > > > > I have badly NATed Clients proble with one way Voice > > > > After reading some documents people ask me to use STUN Server > > But still i have some problem with one way Voice > use stun on dinamic ip :) > > > > I have setup like below > > > > iam trying with 2 extensions > > > > 1 extention in the same LAN where the * installed > > 2 extension in different network, NATed IP , > > 3. both the side iam use SIPURA > > 4. i have 2 DID from provider > > 5. i have route them to appropriate extensions > > > > Iam able to make calls in and out > > > > but the problem where iam setting up server have limited bandwidth > > So i have installed G729 codec > > > > So i want to make RTP > > > > so i made setup caninvite=yes > > > canreinvite=no > nat=yes > > since my provider support that option > > > > then my NAT Clients have One way Voice problem > > > > So after Reading some DOCS SER, should be able to do this Job > > > > so SER can be integrated with *, if yes > > can any one point me to some URL > > > > thanks > > > > ram > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > ????????, > ?????? ??????? > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060316/ed2a4e2d/attachment.htm