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