SIP-to-TDM gateway appliance
(remember to "Reply All"! :-)) Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't). -- Nathan -----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options. Thanks David David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote: I know this has been a topic of conversation in the past, but things might have changed since the last discussion and I'm wondering what the market is currently like for such devices. We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes. Special requirements: 1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN. 2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI". 3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI). 4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this? 5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that. 6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone. ...and, of course, it would be nice if we could find such a device < $1,000. :-P I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible. There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements? Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though. If only somebody made a reasonably-priced single-board-computer that ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!: http://switchvoice.com/index.php?page=shop.product_details&flypage=flypage-a... http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.htm Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities. Would love to hear y'all's thoughts on this subject. Thanks, -- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
Also Netxusa (Our main wholesale supplier) has a similar price. I'd always suggest buying from them over Newegg due to the support. (I'm a huge fan of Netxusa). dw David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz<mailto:david at ringfree.biz> w: ringfree.biz On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com<mailto:nathana at fsr.com>> wrote: (remember to "Reply All"! :-)) Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't). -- Nathan -----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options. Thanks David David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com<mailto:nathana at fsr.com>> wrote: I know this has been a topic of conversation in the past, but things might have changed since the last discussion and I'm wondering what the market is currently like for such devices. We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes. Special requirements: 1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN. 2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI". 3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI). 4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this? 5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that. 6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone. ...and, of course, it would be nice if we could find such a device < $1,000. :-P I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible. There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements? Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though. If only somebody made a reasonably-priced single-board-computer that ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!: http://switchvoice.com/index.php?page=shop.product_details&flypage=flypage-a... http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.htm Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities. Would love to hear y'all's thoughts on this subject. Thanks, -- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com> wrote:
(remember to "Reply All"! :-))
Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't).
The 10 year warranty doesn't suck either ;) I love the Adtran TA-9xx. It is a swiss-army knife of VoIP. The only issue is they don't handle 208v very well (i.e at all). we released the magic blue smoke in our lab. The warranty covered the repair though :)
-- Nathan
-----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options.
Thanks David
David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz
On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote:
I know this has been a topic of conversation in the past, but things might have changed since the last discussion and I'm wondering what the market is currently like for such devices.
We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes.
Special requirements:
1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN.
2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI".
3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI).
4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this?
5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that.
6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone.
...and, of course, it would be nice if we could find such a device < $1,000. :-P
I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible.
There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements?
Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though.
If only somebody made a reasonably-priced single-board-computer that ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!:
http://switchvoice.com/index.php?page=shop.product_details&flypage=flypage-a...
http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.htm
Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities.
Would love to hear y'all's thoughts on this subject.
Thanks,
-- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I'll second (or third . . .) the Adtran TA900 series. We use them for PRI, T1-CAS, analog, pretty much anything you would want to do with them they can handle. They support PAI, you can set the number of digits transferred or you can perform extensive manipulation of DNIS/ANI, pretty much rock solid on t.38, great devices. Good support and (knocking on wood) have never had one actually "fail". On 2/6/13 5:47 PM, "Matthew Crocker" <matthew at corp.crocker.com> wrote:
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com> wrote:
(remember to "Reply All"! :-))
Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't).
The 10 year warranty doesn't suck either ;)
I love the Adtran TA-9xx. It is a swiss-army knife of VoIP.
The only issue is they don't handle 208v very well (i.e at all). we released the magic blue smoke in our lab. The warranty covered the repair though :)
-- Nathan
-----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options.
Thanks David
David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz
On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote:
I know this has been a topic of conversation in the past, but things might have changed since the last discussion and I'm wondering what the market is currently like for such devices.
We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes.
Special requirements:
1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN.
2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI".
3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI).
4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this?
5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that.
6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone.
...and, of course, it would be nice if we could find such a device < $1,000. :-P
I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible.
There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements?
Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though.
If only somebody made a reasonably-priced single-board-computer that ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!:
http://switchvoice.com/index.php?page=shop.product_details&flypage=flypa ge-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30
http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.htm
Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities.
Would love to hear y'all's thoughts on this subject.
Thanks,
-- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I'll 4th this as well. I have had a couple of TA900s die from various causes but I am not convinced these were Adtran problems. In every case we open a ticket with Adtran and they issue an RMA without a hassle. Their support has been great and they don't charge you for support and updates. We use the TA900s on the majority all of our PRI/CAS hand offs or when we need to do T38. We make limited use of the Cisco 2431-8fxs and 2431-16fxs for analog POTS type hand offs. They are cheap on the secondary market (you can't find TA900s secondary now) but support is very limited since no one really knows much about them. Richey -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Robert Dawson Sent: Tuesday, February 12, 2013 12:45 PM To: Matthew Crocker; Nathan Anderson Cc: 'voiceops at voiceops.org' Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance I'll second (or third . . .) the Adtran TA900 series. We use them for PRI, T1-CAS, analog, pretty much anything you would want to do with them they can handle. They support PAI, you can set the number of digits transferred or you can perform extensive manipulation of DNIS/ANI, pretty much rock solid on t.38, great devices. Good support and (knocking on wood) have never had one actually "fail". On 2/6/13 5:47 PM, "Matthew Crocker" <matthew at corp.crocker.com> wrote:
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com> wrote:
(remember to "Reply All"! :-))
Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't).
The 10 year warranty doesn't suck either ;)
I love the Adtran TA-9xx. It is a swiss-army knife of VoIP.
The only issue is they don't handle 208v very well (i.e at all). we released the magic blue smoke in our lab. The warranty covered the repair though :)
-- Nathan
-----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options.
Thanks David
David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz
On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote:
I know this has been a topic of conversation in the past, but things
might have changed since the last discussion and I'm wondering what the market is currently like for such devices.
We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes.
Special requirements:
1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN.
2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI".
3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI).
4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this?
5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that.
6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone.
...and, of course, it would be nice if we could find such a device <
$1,000. :-P
I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible.
There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements?
Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though.
If only somebody made a reasonably-priced single-board-computer that
ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!:
http://switchvoice.com/index.php?page=shop.product_details&flypage=fly pa ge-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30
http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.h tm
Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities.
Would love to hear y'all's thoughts on this subject.
Thanks,
-- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
I will 5th the TA900s. These are the most reliable analog/PRI/CAS devices we have used. Brian
From: mylists at battleop.com To: RDawson at alliedtelecom.net; matthew at corp.crocker.com; nathana at fsr.com Date: Thu, 14 Feb 2013 14:13:35 -0500 CC: voiceops at voiceops.org Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
I'll 4th this as well. I have had a couple of TA900s die from various causes but I am not convinced these were Adtran problems. In every case we open a ticket with Adtran and they issue an RMA without a hassle. Their support has been great and they don't charge you for support and updates. We use the TA900s on the majority all of our PRI/CAS hand offs or when we need to do T38.
We make limited use of the Cisco 2431-8fxs and 2431-16fxs for analog POTS type hand offs. They are cheap on the secondary market (you can't find TA900s secondary now) but support is very limited since no one really knows much about them.
Richey
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Robert Dawson Sent: Tuesday, February 12, 2013 12:45 PM To: Matthew Crocker; Nathan Anderson Cc: 'voiceops at voiceops.org' Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
I'll second (or third . . .) the Adtran TA900 series. We use them for PRI, T1-CAS, analog, pretty much anything you would want to do with them they can handle. They support PAI, you can set the number of digits transferred or you can perform extensive manipulation of DNIS/ANI, pretty much rock solid on t.38, great devices.
Good support and (knocking on wood) have never had one actually "fail".
On 2/6/13 5:47 PM, "Matthew Crocker" <matthew at corp.crocker.com> wrote:
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com> wrote:
(remember to "Reply All"! :-))
Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't).
The 10 year warranty doesn't suck either ;)
I love the Adtran TA-9xx. It is a swiss-army knife of VoIP.
The only issue is they don't handle 208v very well (i.e at all). we released the magic blue smoke in our lab. The warranty covered the repair though :)
-- Nathan
-----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options.
Thanks David
David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz
On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote:
I know this has been a topic of conversation in the past, but things
might have changed since the last discussion and I'm wondering what the market is currently like for such devices.
We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes.
Special requirements:
1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN.
2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI".
3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI).
4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this?
5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that.
6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone.
...and, of course, it would be nice if we could find such a device <
$1,000. :-P
I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible.
There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements?
Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though.
If only somebody made a reasonably-priced single-board-computer that
ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!:
http://switchvoice.com/index.php?page=shop.product_details&flypage=fly pa ge-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30
http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.h tm
Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities.
Would love to hear y'all's thoughts on this subject.
Thanks,
-- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
And just to be another voice echoing, a 6th.. have 100's of TA's deployed for CAS or PRI translation at the edge.. One thing to note, 30 dsp channels on the base box. Pretty sure it won't run out on a full T, 23 or 24 for CAS channels doing G711/G729/RFC2833. TA's aren't dirt cheap, but roughly equivalent in virtually every capability to something like a Cisco 2431 IAD, for at least 50% of the price, if not less. As others have said, complimentary 10 years hardware and firmware update support out of the box. Their support groups are fairly small, they do a solid job, and I've had times when I was a little frustrated for a minute, but generally they've done a pretty bang up job. I love dealing with them, compared to other vendors.. Things to watch out for: Netflow cache sample rate, you want it to almost anytime you've got ip flow export destination ip.ad.re.ss configured you'll want to. Turn on ip ffe anywhere you can show proc queue on a lightly loaded T1 doing a bunch of short calls and see why (hint, PacketRouting goes about 87, you drop packets). debug isdn l2-formatted is your best friends, but you'd best be in via ssh and ready to kick your connection if you get overloaded. "no events" after logging in will get rid of the scroll messages. Most stuff configures almost exactly like IOS on AOS, but the voice stuff makes a lot more sense. You can virtually go through the options and see how it's done pretty quick. Configs on these devices unless you're going to do massive amounts of DNS or 18 SIP trunks feeding the PRI, are a couple to a few K. Some other cool stuff you can do when looking to control the routing when there are multiple PRI's or CAS hitting the customer equipment are things like accept and reject statements on the CAS/PRI trunks. ACL's are pretty straightforward as is SNMP, Syslog, Traps.. not much to complain about.. a couple of things here an there that you can't poll that you should be able like, IMO having to do with like active calls, call processing and stuff, but I haven't looked at AOS enterprise MIBs too much since A1.xx - the T1 performance stats are readily available, both current and interval.. Like that stuff when landing T1's from the Adtran to someone's gear 350' away on a flaky cable. Peter On Thu, Feb 14, 2013 at 1:19 PM, Brian R <briansupport at hotmail.com> wrote:
I will 5th the TA900s. These are the most reliable analog/PRI/CAS devices we have used.
Brian
From: mylists at battleop.com To: RDawson at alliedtelecom.net; matthew at corp.crocker.com; nathana at fsr.com Date: Thu, 14 Feb 2013 14:13:35 -0500 CC: voiceops at voiceops.org
Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
I'll 4th this as well. I have had a couple of TA900s die from various causes but I am not convinced these were Adtran problems. In every case we open a ticket with Adtran and they issue an RMA without a hassle. Their support has been great and they don't charge you for support and updates. We use the TA900s on the majority all of our PRI/CAS hand offs or when we need to do T38.
We make limited use of the Cisco 2431-8fxs and 2431-16fxs for analog POTS type hand offs. They are cheap on the secondary market (you can't find TA900s secondary now) but support is very limited since no one really knows much about them.
Richey
-----Original Message----- From: voiceops-bounces at voiceops.org [mailto: voiceops-bounces at voiceops.org] On Behalf Of Robert Dawson Sent: Tuesday, February 12, 2013 12:45 PM To: Matthew Crocker; Nathan Anderson Cc: 'voiceops at voiceops.org' Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
I'll second (or third . . .) the Adtran TA900 series. We use them for PRI, T1-CAS, analog, pretty much anything you would want to do with them they can handle. They support PAI, you can set the number of digits transferred or you can perform extensive manipulation of DNIS/ANI, pretty much rock solid on t.38, great devices.
Good support and (knocking on wood) have never had one actually "fail".
On 2/6/13 5:47 PM, "Matthew Crocker" <matthew at corp.crocker.com> wrote:
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana at fsr.com> wrote:
(remember to "Reply All"! :-))
Holy crap. I don't know how I missed the pricing for AdTran Total Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma go for on average, I must have made an assumption about AdTran pricing. That totally blows Digium's seemingly-aggressive pricing out of the water, especially if it covers all of my use-cases (which I already know the Digium doesn't).
The 10 year warranty doesn't suck either ;)
I love the Adtran TA-9xx. It is a swiss-army knife of VoIP.
The only issue is they don't handle 208v very well (i.e at all). we released the magic blue smoke in our lab. The warranty covered the repair though :)
-- Nathan
-----Original Message----- From: David Wessell [mailto:david at ringfree.biz] Sent: Wednesday, February 06, 2013 2:15 PM To: Nathan Anderson Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Seconded. This is a killer topic. We've just closed our first deal for this type of situation. I had planned on going with a Adtran 904 ($725 on NewEgg) but am very interested to hear other options.
Thanks David
David Wessell Chief Packet Slinger Ringfree Communications, LLC t: 828-575-0030 e:david at ringfree.biz <mailto:david at ringfree.biz> w: ringfree.biz
On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana at fsr.com> wrote:
I know this has been a topic of conversation in the past, but things
might have changed since the last discussion and I'm wondering what the market is currently like for such devices.
We deliver voice strictly via SIP/RTP, but naturally there are some potential customers out there that still have an older, non-IP-aware PBX that they're not ready to throw out yet. What are the best and most cost-effective gateway options out there at this time? We are specifically looking for one that has a single T1 interface that can operate in either CAS or PRI modes.
Special requirements:
1) We need to be able to do DID manipulation between T1 and SIP; I presume this is a rather standard feature in most gateways given that most SIP trunk providers will send at least 10-digit DNIS (in the INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 digits of the TN.
2) There may be certain situation where we want to leave the PBX configuration as untouched/unchanged as possible (drop-in replacement service), and where there is no correllation between target DID and the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). We'd like a gateway where static mappings like that for DID manipulation are possible, rather than just a general rule that says "strip the first 6 digits off before sending to the PRI".
3) For outgoing calls, the device needs to put the calling DID (the desired Caller-ID/ANI) in the PAI header, and also needs to be able to be configured to override "From" with a static alphanumeric value (so "From" and PAI should not match; "From" will not contain the desired ANI).
4) In T1 CAS singalling modes such as E&M Wink where it is possible to transmit CLID and target DID information via DTMF to the PBX, different PBXes potentially have different formats that they want to see this information in; for example, a Nortel Norstar would expect to see *CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is 212-555-0001 and the destination is 212-555-1212). Are there any gateways that support this?
5) It needs to have a T.38 gateway mode that can recognize a fax call, either send or accept a re-INVITE with a T.38 SDP as appropriate, and perform the "transcoding" from/to T.38 between the T1 channel and the RTP session. Just resorting to G.711 for fax passthrough is not desireable...any gateway can do that.
6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to place an outbound call, the gateway should generate an audible dialtone.
...and, of course, it would be nice if we could find such a device <
$1,000. :-P
I know I could build one myself with a mini PC and a single-span T1 card that was running Asterisk 10 and easily hit that price point, but I'd rather find a supported, off-the-shelf solution to sell to our customers, if possible.
There are the "usual suspects", of course: AdTran, MediaTrix, AudioCodes, and so forth. AdTran seems to get talked about a lot here. Let's say price was no object for a second. Does anyone know if there is a model amongst any of the ones these manufacturers produce that fulfills the above list of requirements?
Does anybody have any experience with Digium's relatively new line of gateways (G100/G200)? I think it would support some of these scenarios (#1 and #3) but I'm not sure about the remaining ones. Unfortunately, although it most certainly runs on an Asterisk core, that core is only exposed to you through a clever but still-limited GUI; with direct access to the dialing plan (extensions.conf) I could accomplish all of these things myself. The price is certainly right, though.
If only somebody made a reasonably-priced single-board-computer that
ran raw, embedded Asterisk and had a single-span T1 interface on it. Oh wait, somebody does!:
http://switchvoice.com/index.php?page=shop.product_details&flypage=fly pa ge-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30
http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.h tm
Only problem is that the first company doesn't have a U.S. distributor, and the second doesn't have a distributor that sells in single-unit quantities.
Would love to hear y'all's thoughts on this subject.
Thanks,
-- Nathan Anderson First Step Internet, LLC nathana at fsr.com _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
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-- Peter Serwe http://truthlightway.blogspot.com/
participants (7)
-
briansupport＠hotmail.com -
david＠ringfree.biz -
matthew＠corp.crocker.com -
mylists＠battleop.com -
nathana＠fsr.com -
peter.serwe＠gmail.com -
RDawson＠alliedtelecom.net