Hi, I installed H323, however when I make a call from SIP Phone -> Asterisk H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct to internet with a public IP. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/442be54d/attachment.htm
Good luck. Try to switch between channel drivers. Chan_oh323, chan_h323 and ooh323. and remember to install the *exact* lib versions recommended on the readmes.... May the force be with you... Isamar On Sat, 1 Apr 2006, Il Neofita wrote:> Hi, > I installed H323, however when I make a call from SIP Phone -> Asterisk H323 > -> Provider H323 the provider can hear me, but I cannot hear nothing. > The asterisk is 1.2.6 with G729 license, and the asterisk is connect direct > to internet with a public IP. > Any thoughts? >
Il Neofita wrote:> Hi, > I installed H323, however when I make a call from SIP Phone -> > Asterisk H323 -> Provider H323 the provider can hear me, but I cannot > hear nothing. > The asterisk is 1.2.6 with G729 license, and the asterisk is connect > direct to internet with a public IP. > Any thoughts?Set a valid bindaddr Ensure G.729 is actually getting allowed If you expect any more assistance, at all, debug information is required - So for now I am totally guessing. Jeremy McNamara
On Sat, 1 Apr 2006 20:09:35 -0500, "Il Neofita" <asteriskmail@gmail.com> wrote:> Hi, > I installed H323, however when I make a call from SIP Phone -> Asterisk > H323 > -> Provider H323 the provider can hear me, but I cannot hear nothing. > The asterisk is 1.2.6 with G729 license, and the asterisk is connect > direct > to internet with a public IP.Try using G.711 oodec.> Any thoughts? > >
Is the SIP phone behind NAT? That's one of the common reasons for one way audio. You might want to try forwarding some port ranges if you are behind NAT just to eliminate that as a possiblity. The SIP port ranges should be something like: SIP: 5060-5061 RTP: 10000-20000 Kyle On 4/1/06, Il Neofita <asteriskmail@gmail.com> wrote:> > Hi, > I installed H323, however when I make a call from SIP Phone -> Asterisk > H323 -> Provider H323 the provider can hear me, but I cannot hear nothing. > The asterisk is 1.2.6 with G729 license, and the asterisk is connect > direct to internet with a public IP. > Any thoughts? > > _______________________________________________ > --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/20060402/e44fec1f/attachment.htm