[Asterisk-Users] (no subject)

[T. Chan]

Dear All, I am going to deploy a VOIP network here in Canada with nodes all over town. This is for long distance services and hence would need a good reliable solution. I have looked into * and am very interested in it with all the value-added features as well as its capability to do H323 and SIP. I understand that a good portion of VOIP operators in the industy is converting to SIP but there are still more H323 VOIP operators out there. That is one reason why I am interested in this solution as it can do both. Most of my customers and carriers are still on H323 and hence I would need to make sure that the * is able to talk with most H323 gateways out there in the market, such as cisco and quintum. There are two things I would like your comments on to make sure that the system will serve my purpose. I need my carrier customers to be able to send calls to my * via H323 VOIP from most H323 gateways and the * to pass through and route the calls to appropriate carriers with cisco or other gateways via VOIP, the process will be H323 mostly and some SIP, working with G723 and / or G729 which are what most VOIP operators are using. In this situation, the inbound and outbound are both via VOIP, either with H323 or SIP with both G723 and / or G729 codec. Is the application, is * compatible with most gateways out there? Another situation is carrier customers will send calls to me to be terminated on the TDM circuits OR I originate calls from the TDM side to be terminated on any gateways on the carrier side. In such, there will be a encoding / decoding process. Would I be able to send calls to most other gateways from my TDM circuits via VOIP H323 or SIP on g729 / G723 codec? Likewise for customers calling into my *, would the * be able to decode G729 and G723 to pass calls to the TDM side? I would love to get some feedbacks and advices (which is greatly appreciated) from you all who have had experiences in doing this before, thanks a million. Tommy Chan