Thanks Ming, Another question is would it be possible to handle 23 simultenous calls through the PRI or would I need 11 pri channels for the incoming calls and 11 pri channels for the outgoing voip traffic?
On Nov 24, 2007 1:05 AM, Ming Yong <[EMAIL PROTECTED]> wrote: > This is Ming from Voiceroute. > There is a very simple way to do this. > PRI will send 11 callerIDs into the PRI card with 1 context. This context > can be an inbound route context that will route 11 different DIDs into > auto-attendant or DISA that will route to an outbound dialing pattern to the > trunk you want. > > Let me know if you need more help. > Btw, the above can be done using Druid very easily. Many VOIP providers have > implemented the above using Druid Enterprise communications server ECS. > Features & Benefits > http://www.voiceroute.net/site/druidecs/features > Eg of VOIP provider customers > http://www.voiceroute.net/site/druidecs/customers > > We have free trials for the software. > > Ming > > > > On Nov 24, 2007 1:41 PM, emist <[EMAIL PROTECTED]> wrote: > > > > > > > > Hey guys, > > > > I was hoping someone could clarify this for me since i've been trying to > > find out for a while now to no avail. Im thinking of deploying a call > > routing service through asterisk. Basically I want people to be able to > > call a number through the PSTN and then call whatever extension to be > > routed through a voip termination provider. > > > > Im guessing using a PRI is the best way to do this. However, im confused > > as to how it all works. Say a PRI has 23 usable channels, does that mean > > that I will be able to route 23 calls at the same time or does it mean > > that I would have to split 11 channels for incoming voice traffic(from > > PSTN) and 11 channels for outgoing voip traffic? > > > > Im stomped =\ > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-biz mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-biz > > > > > > -- > Ming Yong > CEO, www.voiceroute.net > DID: +1-650-331-1732 ext 301 > SIP/email: [EMAIL PROTECTED] _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
