Depends what you want to do! Do you want to do VoIP over that T1 to a provider or IP telephones? Do you want to hook up to the PSTN through that T1 as 24 voice channels, through a T1 card on your asterisk? If you want to use the T1 as 24 voice channels, the Telco is going to have to re-provision the T1 as a voice T1, because currently, presumably it is one big channel of data. You could have the telco do any combination of 24 channels, some voice and some data, if your DSU or router allows drop and insert of channels. It would then split the T1 into a "voice side" and a "data side", each with part of the channels available. Once you have a channelized voice T1, it can plug into a voice T1 card in your Asterisk, but typically can't do data anymore, so if that's not what you intend, then please explain further.. -----Original Message----- From: Warren [mailto:warren-lists@icruise.com] Sent: Monday, June 19, 2006 10:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] How to use a data T-1? I have a data T-1 available to me to do some testing of a new asterisk systemthat I am putting together. Do I just leave this T routed through my cisco router and plug in the asterisk system through a network card or do I need to get a T-1 card and use that? I looked on the voip-info wiki and it did not seem to answer this for me. TIA, Warren
Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren Steve Jones wrote:>Depends what you want to do! > >Do you want to do VoIP over that T1 to a provider or IP telephones? >Do you want to hook up to the PSTN through that T1 as 24 voice channels, >through a T1 card on your asterisk? > >If you want to use the T1 as 24 voice channels, the Telco is going to >have to re-provision the T1 as a voice T1, because currently, presumably >it is one big channel of data. You could have the telco do any >combination of 24 channels, some voice and some data, if your DSU or >router allows drop and insert of channels. It would then split the T1 >into a "voice side" and a "data side", each with part of the channels >available. > >Once you have a channelized voice T1, it can plug into a voice T1 card >in your Asterisk, but typically can't do data anymore, so if that's not >what you intend, then please explain further.. > >-----Original Message----- >From: Warren [mailto:warren-lists@icruise.com] >Sent: Monday, June 19, 2006 10:16 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] How to use a data T-1? > >I have a data T-1 available to me to do some testing of a new asterisk >systemthat I am putting together. Do I just leave this T routed through >my cisco router and plug in the asterisk system through a network card >or do I need to get a T-1 card and use that? I looked on the voip-info >wiki and it did not seem to answer this for me. > >TIA, >Warren > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
If you get it figured out, please post details on the wiki. I tried about a year ago. I think I was close but I didn't have enough time to pursue it. Looks to be trivial with Sangoma though I haven't tried that either. Thanks, Steve Totaro> -----Original Message----- > From: Michael Welter [mailto:mike@telecommatters.net] > Sent: Monday, June 19, 2006 11:15 AM > To: john@millican.us; Asterisk Users Mailing List - Non-Commercial > Discussion > Subject: Re: [Asterisk-Users] How to use a data T-1? > > Is anyone using the HDLC facility in Zaptel to bring a data T1 into an > Asterisk system? I know this was available in kernel 2.4.19--isanyone> using it in kernel 2.6.x? > > -- > Michael Welter > Telecom Matters Corp. > Denver, Colorado US > +1.303.414.4980 > mike@TelecomMatters.net > www.TelecomMatters.net > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren <warren-lists@icruise.com> wrote:> > Steve, > > I want to end up with a system that will let me send and receive voice > calls. I guess what I want to do depends on the best way to do that. > Can I do more than 23 (decent sounding) voice calls on a data T-1 with > someone else handling the final part of the call to the copper for me? > If so than that is my likely final destination. > > I have a channelized voice T-1 currently plugged into my meridian > system, but I would like (if realistically possible) to do as much of > this over IP as possible for maximum flexibility. Is that a pipe dream > or just silly given the current state of technology? > > I am lucky enough to work for a company that is letting me take my time > with this, test the various options and come up with the proper > solution. I am assuming (I know: dumb to assume) at this point that > VoIP over a T-1 to a provider that can then route it to hard phones for > me would be the way to go. Similarly, I would point my 800 number to a > DiD hosted by a VoIP provider that would then route the call back to > me. If that is an incorrect assumption, please let me know. > > Regards, > Warren-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/505e1c98/attachment.htm
Remember to add the RTP, UDP and IP overheads. And then just do the math. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of William Piper Sent: 19 June 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren < warren-lists@icruise.com> wrote: Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/66bf739b/attachment.htm
After all the overhead, for uLaw you would need about 90kbps (give or take) and
for G.729, you would need about 32kbps (give or take). Therefore, you would
have the following:
uLaw= about 17 calls
g729= about 48 calls
I am trying to start a voip service in my local area and sometimes seeing these
numbers make me wonder how using VoIP for larger companies could possibly be
profitable if you require a $500+ data T1 just have a decent connect (unless you
use g729....?)
- Gabe
Depends on the codec. If you are using ulaw, you will only be able to have
about 23 calls. If you use g729 you can have as many as 187 simultanious calls
on a data T1.
Remember, you have 1544Kbs of bandwidth.
g279=8Kbs per call
uLaw=64Kbs per call
Just do the math.
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If your T1 is currently configured for connecting you to the Internet, then your Asterisk just becomes a client on your network, and can terminate calls to Internet based providers by SIP or IAX. No reason for a T1 card or connection to the Asterisk. I don't have enough experience to say who may be the most reliable provider, but you can use any of them for testing. Others have given details of bandwidth requirements for the different codecs, and know more than I about that.. Once you get the basics connected, then any 800# provider should be able to point a number to any existing DID, or you can use a VoIP provider to provide an 800# directly. -Steve -----Original Message----- From: Warren [mailto:warren-lists@icruise.com] Sent: Monday, June 19, 2006 11:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren Steve Jones wrote:>Depends what you want to do! > >Do you want to do VoIP over that T1 to a provider or IP telephones? >Do you want to hook up to the PSTN through that T1 as 24 voicechannels,>through a T1 card on your asterisk? > >If you want to use the T1 as 24 voice channels, the Telco is going to >have to re-provision the T1 as a voice T1, because currently,presumably>it is one big channel of data. You could have the telco do any >combination of 24 channels, some voice and some data, if your DSU or >router allows drop and insert of channels. It would then split the T1 >into a "voice side" and a "data side", each with part of the channels >available. > >Once you have a channelized voice T1, it can plug into a voice T1 card >in your Asterisk, but typically can't do data anymore, so if that's not >what you intend, then please explain further.. > >-----Original Message----- >From: Warren [mailto:warren-lists@icruise.com] >Sent: Monday, June 19, 2006 10:16 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: [Asterisk-Users] How to use a data T-1? > >I have a data T-1 available to me to do some testing of a new asterisk >systemthat I am putting together. Do I just leave this T routedthrough>my cisco router and plug in the asterisk system through a network card >or do I need to get a T-1 card and use that? I looked on the voip-info >wiki and it did not seem to answer this for me. > >TIA, >Warren > >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
If you're going to have to open ports on your firewall for SIP anyway, then why not put the server on the inside? That being said, I don't know if you'd need to punch holes for the phones being trusted and the server on the outside.. Personally I don't like the ideas of having a server outside, but maybe I'm too paranoid?! ________________________________ From: Warren [mailto:warren-lists@icruise.com] Sent: Monday, June 19, 2006 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? So let's assume I am going to use G.729A. I am looking at using Polycom IP601 phones which support G729A directly, so the only licenses I believe I would need are for the calls going to voicemail or in the menu system at once - realistically that number never exceeds 5 simultaneous, since the phones can handle the CODEC and no transcoding is needed, so those do not need licenses according to http://www.voip-info.org/wiki-Asterisk+G.729+Licensing. It looks to me like, for testing, I can get a couple of the polycom phones and have a server using an IP on the unused T1. Assuming that is correct (which I will write up as an article for the Wiki if anyone is interested when this is all done), the next thing I need is a provider of VoIP service. Also, it seems like the server would go on the outside of my firewall with holes punched through for the phones which would be on the ind=side of the firewall. Would that be correct? W Steve Langstaff wrote: Remember to add the RTP, UDP and IP overheads. And then just do the math. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of William Piper Sent: 19 June 2006 17:12 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. bp On 6/19/06, Warren <warren-lists@icruise.com> wrote: Steve, I want to end up with a system that will let me send and receive voice calls. I guess what I want to do depends on the best way to do that. Can I do more than 23 (decent sounding) voice calls on a data T-1 with someone else handling the final part of the call to the copper for me? If so than that is my likely final destination. I have a channelized voice T-1 currently plugged into my meridian system, but I would like (if realistically possible) to do as much of this over IP as possible for maximum flexibility. Is that a pipe dream or just silly given the current state of technology? I am lucky enough to work for a company that is letting me take my time with this, test the various options and come up with the proper solution. I am assuming (I know: dumb to assume) at this point that VoIP over a T-1 to a provider that can then route it to hard phones for me would be the way to go. Similarly, I would point my 800 number to a DiD hosted by a VoIP provider that would then route the call back to me. If that is an incorrect assumption, please let me know. Regards, Warren ________________________________ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/141f3d00/attachment.htm
I would say it's only profitable if you're getting ONE T1 instead of two...?? ________________________________ From: Gabriel Afana [mailto:asterisk@gafana.com] Sent: Monday, June 19, 2006 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How to use a data T-1? After all the overhead, for uLaw you would need about 90kbps (give or take) and for G.729, you would need about 32kbps (give or take). Therefore, you would have the following: uLaw= about 17 calls g729= about 48 calls I am trying to start a voip service in my local area and sometimes seeing these numbers make me wonder how using VoIP for larger companies could possibly be profitable if you require a $500+ data T1 just have a decent connect (unless you use g729....?) - Gabe Depends on the codec. If you are using ulaw, you will only be able to have about 23 calls. If you use g729 you can have as many as 187 simultanious calls on a data T1. Remember, you have 1544Kbs of bandwidth. g279=8Kbs per call uLaw=64Kbs per call Just do the math. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060619/25030642/attachment.htm