Thanks again Ming, I see what you mean, what kind of connection do you think would be optimal to handle the outgoing route of the calls using say g.711 codec?
On Nov 24, 2007 1:13 AM, Ming Yong <[EMAIL PROTECTED]> wrote: > Igor, > Honestly, I would do 23 channels incoming DID with pure SIP VOIP trunking > via internet WAN to my terminating VOIP provider. > I do not quite understand why you need another 11 channel for outgoing voip > traffic when simple Ethernet SIP outgoing connections will do. > Ming > > > > On Nov 24, 2007 2:10 PM, Igor H <[EMAIL PROTECTED]> wrote: > > 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] > > > > > > -- > > > 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
