Since Asterisk isn't converting from one codec to another you should not need a G.729 license.
On Wed, 2003-10-08 at 12:05, Nicolas Gudino wrote: > Hi List, > > I'm new to asterisk. I think it's great! I'm interested in terminating calls > via a SIP provider. I want to know if I need to license G729 on asterisk in > these scenarios: > > CISCO ATA186 - Asterisk - SIP Provider - PSTN > > or this one: > > CISCO ATA186 - Asterisk - CISCO ATA > > To my understanding, in the second case, if one of the ATA is behind NAT, I > should set canreinvite=no, so the RTP channels would go through *, so I > would have to license G729 in order to use this codec with the ATAs. Is this > right? > > But if boths ATA have public IPs, and * issues a reinvite, can the ATAs > negotiate G729 themselves, without needing it on * ? > > And in the first scenario, if the SIP provider supports G729 and the ATA has > a public IP, do I need to license the codec in *? > > Thanks in advance, > > Nicolas Gudino > Buenos Aires - Argentina > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
