- VoiceOps - lists.voiceops.org

Fwd: Re: SIP-to-TDM gateway appliance
by jfratantoni＠cygnustel.com Feb. 14, 2013

Feb. 14, 2013

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-… > > > > 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

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VoIP number can't be verified for Google ad campaigns
by carlos＠televolve.com Feb. 11, 2013

Feb. 11, 2013

We're a small ITSP, and get most of our numbers through Onvoy/Zayo/360networks. We also use a variety of aggregators and other carriers. Today a customer called and said that when he tried to create a Google ad campaign with one of his DIDs, it said the number couldn't be verified. The error message said this happens with prepaid cell phones and "some" VoIP services. Neither of us know what Google is looking for. Is it the CNAM for the number, or a white pages listing, or something else? As a normal practice we don't put CNAM on DIDs if they are not used for CLID on outbound calls, and of course we don't do directory entries for DID numbers. Any ideas? The specific number in question is through Onvoy. -- Carlos Alvarez TelEvolve 602-889-3003

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Specific SIP packets cause some Intel gigabit ethernet controllers to reset
by kris＠kriskinc.com Feb. 8, 2013

Feb. 8, 2013

I'm sure some of you have seen this elsewhere but it seems the more places I share it the more confirmations I receive: http://blog.krisk.org/2013/02/packets-of-death.html Long story short some oddly configured Yealink devices could cause our Intel gigabit controllers to lose link. -- Kristian Kielhofner

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OT: What do you recommend for help desk software
by nick＠flhsi.com Feb. 7, 2013

Feb. 7, 2013

We ended up with SmarterTrack from SmarterTools. (We also use SmarterMail). It's working pretty well for us. We started out with OSTicket Which is free, That also did well for awhile. Nick Olsen Network Operations (855) FLSPEED x106 ---------------------------------------- From: "Faisal Imtiaz" <faisal at snappydsl.net> Sent: Thursday, February 07, 2013 1:53 PM To: voiceops at voiceops.org Subject: Re: [VoiceOps] OT: What do you recommend for help desk software We have been using Kayako, very flexible and works well..... Regards. Faisal Imtiaz Snappy Internet & Telecom 7266 SW 48 Street Miami, Fl 33155 Tel: 305 663 5518 x 232 Helpdesk: 305 663 5518 option 2 Email: Support at Snappydsl.net On 2/7/2013 1:43 PM, Carlos Alvarez wrote: This may be considered slightly off topic here, not sure. I figured some of you would have insight and recommendations as many people here have a business similar to ours. We have several small companies who share workers, clients, and contractors in various ways. We have been using Zendesk for our help desk, but it's tough because they won't allow one person to be in multiple organizations. So we have a contractor who is John-A in company A and John-B for company B, etc. Annoying. We are looking for other options, hosted preferred, but we can fire it up in a VM if we need to. We are small and would have about six help desk agent types, and only a few hundred end users. We don't need highly advanced features like RT would provide; simplicity and customer acceptance is the highest priority. Must accept tickets/replies by e-mail. Most of our tickets are handled in a day, and are fairly simple, while maybe 5% linger for weeks as they are complex, but those still don't need a bunch of details in them. -- Carlos Alvarez TelEvolve 602-889-3003 _______________________________________________ VoiceOps mailing list VoiceOps at voiceops.org https://puck.nether.net/mailman/listinfo/voiceops

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OT: What do you recommend for help desk software
by carlos＠televolve.com Feb. 7, 2013

Feb. 7, 2013

This may be considered slightly off topic here, not sure. I figured some of you would have insight and recommendations as many people here have a business similar to ours. We have several small companies who share workers, clients, and contractors in various ways. We have been using Zendesk for our help desk, but it's tough because they won't allow one person to be in multiple organizations. So we have a contractor who is John-A in company A and John-B for company B, etc. Annoying. We are looking for other options, hosted preferred, but we can fire it up in a VM if we need to. We are small and would have about six help desk agent types, and only a few hundred end users. We don't need highly advanced features like RT would provide; simplicity and customer acceptance is the highest priority. Must accept tickets/replies by e-mail. Most of our tickets are handled in a day, and are fairly simple, while maybe 5% linger for weeks as they are complex, but those still don't need a bunch of details in them. -- Carlos Alvarez TelEvolve 602-889-3003

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NOTICE: To all providers using the Grandstream HT502/HT503
by ryandelgrosso＠gmail.com Feb. 6, 2013

Feb. 6, 2013

All, Over the last few months we have uncovered a vulnerability in the HT502 that allows for theft of credentials from customer devices. I am sending this out since the issue has now been resolved in a new release of firmware BUT Grandstream have NOT sent out any kind of pro-active notifications nor included this fix in their release notes for this build. After conferring with some other sizable providers also using this device at scale, they were able to "connect the dots" on their up-tick in fraud based on our discovery. First some history: We currently have over 50,000 deployed HT502's in active customer service. Beginning in December we saw an immediate and sizable up-tick in fraud by easily an order of magnitude. Statistical analysis of the fraud showed the ONLY linking factor to be the fact that the compromised accounts were ALL using the HT502 device AND had WAN port access enabled to the device, and we as the provider were locked out (admin password changed, no longer provisioning from us on scheduled interval) After some digging and conferring with Grandstream technical gurus it was confirmed there was a buffer overflow vulnerability that would allow a remote attacker to change the admin password WITHOUT rebooting the device or otherwise having any administrative access to it. Once the password was changed the attacker could log in with the new password and complete control. On all versions prior to 1.0.5.10 the SIP credentials could be extracted from the admin website with the "Download config" option. On versions up to 1.0.8.4 the sip credentials were STILL extractable from the telnet interface if the provisioning values were known by the attacker. All of these vulnerabilities are fixed in version 1.0.9.1. I encourage you to test and deploy this version ASAP. I am sending this out in a purely advisory capacity in the hopes that education will prevent further monetary damages. Please feel free to contact me on or off list if you want to know more about this issue. -Ryan

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SIP-to-TDM gateway appliance
by nathana＠fsr.com Feb. 6, 2013

Feb. 6, 2013

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-… 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

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Old Morotola ATAs
by nathana＠fsr.com Feb. 4, 2013

Feb. 4, 2013

Does anybody out there either have/know a good source for old-stock Motorola ATAs (VT1000 series) at a reasonable price, or perhaps have a bunch that they want to get rid of? VT1003 or VT1005, new or used (as long as the used are in relatively good physical condition and in good working order), locked or unlocked, branded to Vonage/Speakeasy/whoever or unbranded...doesn't matter. If so, feel free to hit me off-list. Thanks! -- Nathan Anderson First Step Internet, LLC nathana at fsr.com

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Problems with long connect times / PDD to "New Cingular Wireless PCS"
by d＠d-man.org Feb. 3, 2013

Feb. 3, 2013

Hi folks, We're seeing long long connect times across all our VoIP termination carriers when calling numbers whose LNP is coming back as New Cingular Wireless. Can't figure out a rhyme or reason. Anyone else have any suggestions on what to do here? At first we thought it was our upstream provider but we've been able to recreate this with about six different folks now. Thanks much, Darren

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Broadworks on VMWare/Linux
by peeip989＠gmail.com Jan. 31, 2013

Jan. 31, 2013

Is anyone running production Broadworks on virtual (VMWare/Red Hat Linux) servers? Have you had any issues related to it being a virtual machine? I know it is now supported and we have tested in a lab environment without issue but have not yet clobbered it with any serious volumes of traffic and are curious if Broadworks behaves any differently on physical versus virtual servers. Any experience you can share is welcome. Thanks

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