3 X Cisco 2821 configure voice gateway for 3 different sites. Each voice gateway have a voip dial peer to another sites and vice verse.
Let say all 3 sites 3rd party PBX and support E1 truk to Cisco voice gateway. Only use Cisco voice gateway route call to different site.
How can we do the transcoding, MTP and conference call based on this setup? Or this kind of setup only use G711codec?
A voice gateway in the traditional sense will only route calls. Every call will require DSP resources, because DSPs are needed to translate non-RTP voice on analog circuit to packetized RTP voice. Some codecs require larger amounts of DSPs, because for a single DSP you may be able to place 16 G.711 calls but only 8 G.729 calls. In this function, all calls are independent of one another, and there is no controlling manager.
For things like MTP, conferencing, transfers, and transcoding, it requires for multiple streams to interact. For interaction, it requires a CME or CUCM/CCM.
Once you have a CUCM/CME, you can have something where there are 8 gateways, but only 5 of them have the DSPs for conferencing and 2 do transcoding, etc.
Voicegateway does not have the inteligence to do conference.Can be achieved only with Callmanager functionality.
You can use any supported codec on the dialpeer.
What Paolo means is that you can configure the codec preference,then apply it under dial-peer.