Thanks a bunch Ming, your help is really appreciated! Take care,
Igor H. On Nov 24, 2007 1:32 AM, Ming Yong <[EMAIL PROTECTED]> wrote: > For least bandwidth, get quality PRI card eg. Sangoma with echo cancellation > + onboard DSP encode the audio in and send out using g729 (10kbps per > channel) SIP. > > For best quality, take PRI channels as ulaw & alaw out using 23 x 32kbps = > 736kbps or 1Mbps will be good enough > > Ming > > > > On Nov 24, 2007 2:26 PM, Igor H <[EMAIL PROTECTED]> wrote: > > 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] > > > > > > -- > > > 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
