Re: [Asterisk-Users] Three (quick?) questions...
An ordinary T1 (non-ISDN) doen't have a separate channel for signalling. See: http://www.voip-info.org/wiki-T1 ISDN T1's have a separate signalling channel. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 10, 2004 6:52 PM Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi Paul, you would know better than I would but I always thought a T1 was 24 channels of voice with the signalling additional like we have in Australia a Pri or E1 is 30 channels voice channels plus signalling. Can anyone else clarify? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 11 July 2004 2:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three (quick?) questions...
Paul is right. For voice, there are two varities of T1's. The straight channelized T1 provides 24 pots lines and no custom services, like caller ID. The other is a PRI ISDN, which is 23 voice and 1 'D' channel. The telco can now provide more services, including caller id over the D channel, backup routing of voice calls when the T1 goes down, and outbound caller id of the extension placing the call for E911 translation. Traditionally splitting a T1 for voice and data used a special CSU/DSU that you can program to provide two T1 streams and divided the channels between the PBX and the router. It's very easy to configure a CSU/DSU for this. The voice stream is usually a T1 data stream that connects to the PBX with the data piece sent via v.35 serial to the data router. Most of the installs I have seen were ISDN type voice, so chan 24 was always reserved for the D chan signalling. I think these CSU/DSU's run between $2,000 $3,000 new(I have not priced them for a while), but are quite easy to configure and install. I am new to this forum, but worked for 23 years for the telco here doing the central office piece of this puzzle and then for 5 years doing new office turnups for a wireless telco where we put in many split T1's using Adtran CSU/DSU's to split the data stream. Lyle - Original Message - From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 10, 2004 11:52 PM Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi Paul, you would know better than I would but I always thought a T1 was 24 channels of voice with the signalling additional like we have in Australia a Pri or E1 is 30 channels voice channels plus signalling. Can anyone else clarify? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 11 July 2004 2:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http
RE: [Asterisk-Users] Three (quick?) questions...
Caller-ID can be sent over Channelized T1's, as well as a few of the other class-features (stutter dial tone, etc...) Setting outbound ANI is something that can't be done on Channelized T1's which may be what you were referring to. The csu's you reference are usually called Drop and Insert CSU's... An Adtran 850T can be a drop and insert unit, and do the de-channelization to FXO's. It has a DSX port on it (and a v.35 port on it) for the remaining channels to pass through the unit and go to a router... You can get a DI CSU on ebay for around $ 500, and an 850T w/ a few 4 port FXO's for arounf $ 1000 - $ 1200 W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Sunday, July 11, 2004 9:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Three (quick?) questions... Paul is right. For voice, there are two varities of T1's. The straight channelized T1 provides 24 pots lines and no custom services, like caller ID. The other is a PRI ISDN, which is 23 voice and 1 'D' channel. The telco can now provide more services, including caller id over the D channel, backup routing of voice calls when the T1 goes down, and outbound caller id of the extension placing the call for E911 translation. Traditionally splitting a T1 for voice and data used a special CSU/DSU that you can program to provide two T1 streams and divided the channels between the PBX and the router. It's very easy to configure a CSU/DSU for this. The voice stream is usually a T1 data stream that connects to the PBX with the data piece sent via v.35 serial to the data router. Most of the installs I have seen were ISDN type voice, so chan 24 was always reserved for the D chan signalling. I think these CSU/DSU's run between $2,000 $3,000 new(I have not priced them for a while), but are quite easy to configure and install. I am new to this forum, but worked for 23 years for the telco here doing the central office piece of this puzzle and then for 5 years doing new office turnups for a wireless telco where we put in many split T1's using Adtran CSU/DSU's to split the data stream. Lyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three (quick?) questions...
Ken D'Ambrosio wrote: [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] X100P/X101P... but people don't like it, so take the little more expensive TDM400P with a single FXO interface. 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? Some people like budgetone's, but the site www.voip-info.org should reveal more information... If you want something cheaper... you can always get a second FXS module for the TDM400P and plug a standard analogue phone in it, maybe use one that supports FSK signalling for CallerID number + name. Even if you won't use this in production, a TDM400P with both FXS and FXO interface is very nice for testing stuff in combination with Asterisk. 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Yep, T100P should do in that case. Call Digium sales for details. Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Three (quick?) questions...
1) A digium FXO card ($100) will do. Works great for me, ymmv. ~$100. My system is running on a Celeron 2.7GGhz with 256 of RAM and the processor never really blips. I'd say a ~1 GHz would be plenty enough for one or two channels. 2) $10 Walmart Special connected to a $100 Sipura SPA-2000. The Sipura gives you 2 FXS ports (=two extensions) bringing the cost of an analog port to $50/each. Any analog phone will do... I have two cordless, two SWB Freedom Phones ($9.95, with caller-id) and a couple of Aastra 390s (refurbed from ebay, $40ish) because of their excellent speakerphone. 3) Dunno, I do POTS and VOIP only. -Original Message- From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED] Sent: Saturday, July 10, 2004 11:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three (quick?) questions...
1) X100P from Digium. We currently have four of them in one machine (started with one, then added a second, then went all the way up to four) and they're working great. Put the X100P into a cheap PC (we're using a 1.7 Ghz Celeron system, I believe), add Linux and Asterisk, and you have a full-featured PBX for cheap. 2) Either put a TDM400P in your box and use up to 4 standard analog phones, or get something like the IAXy from Digium or a Grandstream HandyTone or BudgeTone. The Grandstream phones aren't the best, but they're probably the cheapest for SIP connectivity. A last option would be a software solution using IAX or SIP, but that requires another computer for each extension. 3) No experience with T1 here, but we'll probably need to look at something else when we eventually need to add more lines. -Zak ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Three (quick?) questions...
Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Three (quick?) questions...
Hi Paul, you would know better than I would but I always thought a T1 was 24 channels of voice with the signalling additional like we have in Australia a Pri or E1 is 30 channels voice channels plus signalling. Can anyone else clarify? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 11 July 2004 2:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
