Fwd: Re: SIP-to-TDM gateway appliance
I have to comment that I was pretty dissatisfied with AdTran's customer support and unwillingness to patch a software bug we found in the TotalAccess line (It affects bridging). This bad taste in our mouth has caused us to seek out another vendor to meet our needs. On 2013-02-06 16:42, Nathan Anderson 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> 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
-- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services -- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services
Out of curiosity, what was the bug and why would they refuse to fix it? That seems rather odd. -- Nathan -----Original Message----- From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Joe Fratantoni Sent: Tuesday, February 12, 2013 2:52 PM To: voiceops at voiceops.org Subject: [VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance I have to comment that I was pretty dissatisfied with AdTran's customer support and unwillingness to patch a software bug we found in the TotalAccess line (It affects bridging). This bad taste in our mouth has caused us to seek out another vendor to meet our needs. On 2013-02-06 16:42, Nathan Anderson 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> 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
-- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services -- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
My experience has been the opposite, even when I found a difficult bug in their DNS cache implementation. I had to push hard at first to get them to realize it was a bug, but once I had a test case, they were very, very interested, and ultimately put out a release JUST to fix the bug I found. (I think it was A2.04, if I recall - it had to do with the first DNS lookup after a DNS TTL expiry would cause a DNS lookup failure. It manifested itself as a failed inbound call attempt about once every 10 minutes. Adtran finally realized that the reason they didn't see it more often was we run 5 minute TTLs on our DNS so we can change it quickly, and a lot of people run 60 minute or even 6 hour or 24 hour TTLs on their records, so the adtrans would only fail an inbound call once an hour, once every 6 hours, or once a day, and nobody noticed. Anyway, like I was saying, I'm kind of surprised about the bug patch. Did you follow up by requesting a supervisor, or going through your sales team? Maybe you got a bad tech. -Paul On Feb 12, 2013, at 17:51 , Joe Fratantoni <jfratantoni at cygnustel.com> wrote:
I have to comment that I was pretty dissatisfied with AdTran's customer support and unwillingness to patch a software bug we found in the TotalAccess line (It affects bridging). This bad taste in our mouth has caused us to seek out another vendor to meet our needs.
On 2013-02-06 16:42, Nathan Anderson 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> 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
-- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services
-- Joe Fratantoni Cygnus Communications 19635 97th Ave Mokena, IL 60448 815.680.5686 x206 Business Internet & Phone Services _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops
participants (3)
-
jfratantoni＠cygnustel.com -
nathana＠fsr.com -
paul＠timmins.net