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Fwd: [Sheflug] Zero-day rootkit?
by ghenry＠suretec.co.uk Feb. 22, 2013

Feb. 22, 2013

Hi all, Anyone seeing this on any of your softswitches/SBCs/SIP Proxies that are built on Linux kernels? Thanks. ---------- Forwarded message ---------- From: Chris J <cej at nightwolf.org.uk> Date: 21 February 2013 19:25 Subject: [Sheflug] Zero-day rootkit? To: sheflug at sheflug.org.uk Just a heads-up in case it's not been seen. The last couple of days I've seen blogs and forums light up with news of an active zero-day attack - the actual attack vector is currently not known, which makes this more worrying than most. Some folk are placing the blame on SSH, others on cPanel, but really, no-one currently knows. Typically it's been Redhat or CentOS machines affected, although I've seen (unconfirmed) anecdotes on forums that Debian has also been affected. You'll know to be suspicious if you have a file, libkeyutils.so.1.9, on your box, most likely under /lib (but could be elsewhere). The latest "good" version of this file is 1.3... It's also curious that most of the talk is on forums. I haven't seen anything from the distributions about this. Relevent links and more info: http://blog.solidshellsecurity.com/2013/02/18/0day-linuxcentos-sshd-spam-ex… http://blog.configserver.com/index.php?itemid=716 http://www.webhostingtalk.com/showthread.php?t=1235797 A google for libkeyutils.so.1.9 brings back other various forums, etc... Don't know if anyone's got more solid information on this? Cheers, Chris -- Chris Johnson :: cej at nightwolf.org.uk :: PGP 0xBC618B81 :: http://cej.nightwolf.org.uk/ _______________________________________________ Sheffield Linux User's Group http://sheflug.org.uk/mailman/listinfo/sheflug_sheflug.org.uk FAQ at: http://www.sheflug.org.uk/mailfaq.html GNU - The Choice of a Complete Generation -- Kind Regards, Gavin Henry. Managing Director. T +44 (0) 1224 279484 M +44 (0) 7930 323266 F +44 (0) 1224 824887 E ghenry at suretec.co.uk Open Source. Open Solutions(tm). http://www.suretecsystems.com/ Suretec Systems is a limited company registered in Scotland. Registered number: SC258005. Registered office: 24 Cormack Park, Rothienorman, Inverurie, Aberdeenshire, AB51 8GL. Subject to disclaimer at http://www.suretecgroup.com/disclaimer.html Do you know we have our own VoIP provider called SureVoIP? See http://www.surevoip.co.uk Did you see our API? http://www.surevoip.co.uk/api

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PROBLEM: Yate-Status CICs-Yate.Mx is CRITICAL on host ELP-SOT-DC1.Yate.Mx
by mmg＠transtelco.net Feb. 21, 2013

Feb. 21, 2013

Gamaliel Independientemente de las rutas ambas deben estar limpias. Por favor usen equipo de prueba par que validen ambs rutas y no fecten los circuitos hasta que baje el trafico Necesitamos asegurarnos de que ambas rutas esten operando de manera correcta pues hay circuitos de clientes que pasan por donde mismo On Thu, Feb 21, 2013 at 4:29 PM, Gamaliel Bedolla <gbf at transtelco.net>wrote: > NOC, > Se estan viendo algunos errores en 2 de los 3 E1s que van desde Barrancas > a Ezero. Se cree que estan utilizando la ruta larga SNCP > Barrancas-Torres-Ezero. Se van a regresar a la ruta corta. > Aqui le doy seguimiento a esto. > > ---------- Forwarded message ---------- > From: <nagios at opsview.transtelco.net> > Date: 2013/2/21 > Subject: PROBLEM: Yate-Status CICs-Yate.Mx is CRITICAL on host > ELP-SOT-DC1.Yate.Mx > To: gbf at transtelco.net > > > > PROBLEM: Yate-Status CICs-Yate.Mx is CRITICAL on host ELP-SOT-DC1.Yate.Mx > > Service: Yate-Status CICs-Yate.Mx > Host: ELP-SOT-DC1.Yate.Mx > Alias: ELP-SOT.DC1.Yate.Mx > Address: 10.100.0.66 > Host Group Hierarchy: Opsview > Core - Telephony > Core - Telephony > Gateways > State: CRITICAL > Date & Time: Thu Feb 21 16:09:18 MST 2013 > > > Additional Information: > > > > > Advertencia: CICs Bloqueados > \n189=e1_6 > > Service URL: > http://10.11.0.127/cgi-bin/extinfo.cgi?type=2&host=ELP-SOT-DC1.Yate.Mx&serv… > > -- Manuel Mar?n Transtelco Inc. 1.9152172232

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looking for referral for a friend
by eric.fort＠gmail.com Feb. 20, 2013

Feb. 20, 2013

I know some here offer hosted PBX services and or work with those who do so I'm seeking references n referals for an associate who posted the following to another list I'm part of: Anyone using a hosted pbx voip provider in the USA? Any good prices? I have 30 voip phones I need: a system that is compatible with aastra phones includes a iphone/android voip app includes computer based softphone at least 4 incoming/outgoing lines and has all the usual pbx features Anyone here have a good referal or care to pick up some business? If so drop me a note and I'll see what can be done to match up interested parties here. Eric Fort FortConsulting

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Session Border Controllers
by tallen＠acmepacket.com Feb. 20, 2013

Feb. 20, 2013

Acme Packet now has a virtualised SBC offering as well that may suit your requirements. More info from http://acmepacket.com/products-services/enterprise-products/platforms/softw…

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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. 19, 2013

Feb. 19, 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. 14, 2013

Feb. 14, 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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