- VoiceOps - lists.voiceops.org

Session Border Controllers
by GBaxley＠hostinfinity.com Feb. 20, 2013

Feb. 20, 2013

200 concurrent calls would be a good start with 20 calls a second. If there is a platform that can be virtualized on my current equipment, that would be the preferred method. We have a high available cloud that currently runs many Centos virtual PBX instances. Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704 From: james jones [mailto:james.voip at gmail.com] Sent: Tuesday, February 19, 2013 1:56 PM To: Grant Baxley Cc: voiceops at voiceops.org Subject: Re: [VoiceOps] Session Border Controllers How many concurrent calls? How many calls per second? On Tuesday, February 19, 2013, Grant Baxley wrote: I am looking to implement a cost effective session border controller. Can anyone point me in the right direction? We would need to be able to route calls to and from different IP addresses based on source and destination. Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704

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Business Intelligence Tool for Calling Pattern Analysis
by kleclaire＠mywdt.com Feb. 20, 2013

Feb. 20, 2013

We are looking for a system, either out of the box or custom solution, that would allow us to dig deeper into our customers calling patterns and costs down to the account level. Things like calls per hour, metrics on their destinations, number of calls, length of calls, etc. Just wondering what other people maybe be using to do this sort of analysis and if there are any recommendations out there. Thanks, Keith

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Session Border Controllers
by dave＠3c.co.uk Feb. 19, 2013

Feb. 19, 2013

Hi Grant - That's exactly what we do with it. There's a bunch of GUIs available for it, none of which I've used - details here: http://wiki.freeswitch.org/wiki/Freeswitch_Gui I wrote a web front end/routing/billing engine for it which meets our needs, and it wasn't terribly traumatic. --Dave From: Grant Baxley [mailto:GBaxley at hostinfinity.com] Sent: 19 February 2013 20:15 To: David Knell Subject: RE: [VoiceOps] Session Border Controllers Dave, Does FreeSwitch act as an SBC? I would need to interface with multiple carriers and rout calls accordingly. Also, does it have a user friendly web interface? Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704 From: David Knell [mailto:dave at 3c.co.uk] Sent: Tuesday, February 19, 2013 3:11 PM To: Grant Baxley Cc: voiceops at voiceops.org Subject: RE: [VoiceOps] Session Border Controllers Hi Grant - We use FreeSWITCH running on CentOS, virtualised with OpenVZ. The largest instances run up to 4,000 sessions (=2,000 concurrent calls) and have peaked at over 40 calls/sec; they use less than half of an i7-3770 CPU at this level. --Dave From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Grant Baxley Sent: 19 February 2013 18:58 To: james jones Cc: voiceops at voiceops.org Subject: Re: [VoiceOps] Session Border Controllers 200 concurrent calls would be a good start with 20 calls a second. If there is a platform that can be virtualized on my current equipment, that would be the preferred method. We have a high available cloud that currently runs many Centos virtual PBX instances. Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704 From: james jones [mailto:james.voip at gmail.com] Sent: Tuesday, February 19, 2013 1:56 PM To: Grant Baxley Cc: voiceops at voiceops.org Subject: Re: [VoiceOps] Session Border Controllers How many concurrent calls? How many calls per second? On Tuesday, February 19, 2013, Grant Baxley wrote: I am looking to implement a cost effective session border controller. Can anyone point me in the right direction? We would need to be able to route calls to and from different IP addresses based on source and destination. Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704

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Session Border Controllers
by GBaxley＠hostinfinity.com Feb. 19, 2013

Feb. 19, 2013

I am looking to implement a cost effective session border controller. Can anyone point me in the right direction? We would need to be able to route calls to and from different IP addresses based on source and destination. Thanks, R. Grant Baxley President & CEO Infinity Computer Solutions 813 W Platt St. Tampa, FL 33606 Toll Free: 1.888.287.9198 Local: 813.319.3704

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Problem with ATT (POTS) call forwarding
by orenyny＠gmail.com Feb. 15, 2013

Feb. 15, 2013

Hello, Does anybody happen to know if ATT call forwarding (multi-path) is not working when a call comes in from out of state? We have a client with his ATT line call forwarded and it seems to works fine for all local calls. But a caller from Virginia got a message that "the line is no longer in service.. please check the number and try again". Thanks, Oren

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SIP-to-TDM gateway appliance
by nathana＠fsr.com Feb. 15, 2013

Feb. 15, 2013

(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

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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. 12, 2013

Feb. 12, 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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