Re: [VoiceOps] VoIP passive monitoring appliances or software - any recommendations?

[Matthew Crocker]

Take a look at http://www.voipmonitor.org/   pretty in-expensive and does a 
great job of capturing pcaps and SIP call detail

-Matt

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



On Feb 12, 2014, at 3:15 PM, Brian Knight m...@knight-networks.com wrote:

 $DAY_JOB is at a national ISP/NSP where we resell VoIP services.  We do 
 peering with the VoIP carrier at one of our remote POP's.  We are looking for 
 a better way to be able to monitor the handoff of those calls to our carrier 
 over that peering link.
 
 We have quite a bit of instrumentation within our walled garden to tell us 
 about call quality.  We can monitor our QOS policies to ensure packets aren't 
 being dropped by intermediate routers.  If the customer uses our routers to 
 terminate their SIP session, we can pull call quality stats from those 
 routers as well.  We can also use our own office telephones to make and 
 receive test telephone calls, and we can of course run Wireshark captures 
 from the switches to which those phones are connected.
 
 However, we can't say for certain that the customer's RTP traffic actually 
 made it on the wire connecting us to the VoIP provider, nor can we say that 
 the traffic is being transmitted and received properly.  The peering link is 
 connected to a Cisco 12k router on our side, so there is no way (afaik) to 
 mirror the port, as on a switch.
 
 For the moment, I am envisioning that we'll need to deploy a server running 
 Wireshark to the remote POP.  It will need two network interfaces; one 
 connected to a management network, the other a capture interface.  The 
 capture interface will connect to a network tap, and the network tap 
 connected in-line between our router and the patch panel.
 
 Wireshark is probably adequate for what we need.  But I'm wondering if there 
 is any software or an appliance that would do the job better.  Given the 
 usual details - calling number, called number, date and time - we want to be 
 able to quickly inspect traffic and dig into the details of the stream.  Do 
 we see any missing packets from the media stream?  What is the MOS score of a 
 particular call?  Do we see any missing packets coming from us?  Any missing 
 packets from the provider?
 
 Alerting on bad call quality would be a nice-to-have addition.
 
 Any recommendation would be appreciated.  Thanks in advance.
 
 -Brian Knight
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Re: [VoiceOps] Looking for whitelabel IPTV provider

[Matthew Crocker]

Colton,

 Yes I meant to send it to the VoiceOps list.  I know many are members are 
CLECs and some have TV solutions already.   Not sure if NANOG is the right list 
either.   Skitter TV looks interesting by the channel line-up is horrible.
I’m looking for something that can stand in for regular TV.   Channel surfing 
and all that.  Roku  AppleTV is a clear winner for the long term.Many 
older residents don’t have the tech savvy to configure a roku and figure it 
out.   They want to turn on the TV and hit channel up/down

Thanks

-Matt

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



On Sep 5, 2014, at 2:02 PM, Colton Conor colton.co...@gmail.com wrote:

 Matt,
 
 Did you mean to send this to the VoiceOps list? This list is mainly for VoIP, 
 not IPTV. You will probably have better luck on NANOG list. 
 
 I have heard Skitter TV has an option. 
 
 
 On Thu, Sep 4, 2014 at 12:42 PM, Matthew Crocker matt...@corp.crocker.com 
 wrote:
 
 Hello,
 
  I have an opportunity to deliver FTTH to about 800 homes in local small 
 town.  I’m looking for someone to provide TV service to the residents (RF 
 overlay or IPTV).   Does anyone know of a white label provider that can 
 deliver service in Massachusetts?I can pick it up in Boston (1 Summer, 
 230 Congress or 300 Bent).800 homes is not enough for me to justify 
 building my own head-end,  hoping I can wholesale off someone else’s head end.
 
 Thanks
 
 -Matt
 
 --
 Matthew S. Crocker
 President
 Crocker Communications, Inc.
 PO BOX 710
 Greenfield, MA 01302-0710
 
 E: matt...@crocker.com
 P: (413) 746-2760
 F: (413) 746-3704
 W: http://www.crocker.com
 
 
 
 
 
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Re: [VoiceOps] T1 Crossover Cable

[Matthew Crocker]

Cross over cable won’t have anything to do with the D channel.  If the T1 comes 
up then the cable is fine.  I.e.  If both ends are getting timing  framing 
then the cable is good.   For the D-channel down make sure you are setting both 
ends to the correct ISDN mode, D channel on 23 and all that.


--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



 On Oct 31, 2014, at 9:17 AM, David Wessell da...@ringfree.biz wrote:
 
 So two different answers :)
 
 It's a 904 connecting on the t1 0/2 DSX interface.
 
 The main issue is that we keep getting D channel is DOWN.
 
 I've run through every step in the Adtran troubleshooting guide 
 (https://supportforums.adtran.com/servlet/JiveServlet/previewBody/4521-102-2-4862/PRI%20Troubleshooting.pdf
  
 https://supportforums.adtran.com/servlet/JiveServlet/previewBody/4521-102-2-4862/PRI%20Troubleshooting.pdf)
  and still get the message. We've matched all setings on the Shoretel side 
 with those in the Adtran..
 
 And I do have a ticket in with Adtran. But they take forever to call back :)
 
 It does appear that when they connect the the Adtran 904 for the Shoretel via 
 a crossover cable the PRI interface itself stays down. A straight  ethernet 
 and the connection does come up. But the D Channel remains down.
 
 Thanks
 David
 
 On Fri, Oct 31, 2014 at 9:08 AM, Fred Posner f...@palner.com 
 mailto:f...@palner.com wrote:
 If you are connecting it directly, then generally yes... a crossover. If 
 there's anything in-between the two, then generally the answer is no.
 
 Fred Posner
 The Palner Group, Inc.
 http://www.palner.com http://www.palner.com/ (web)
 +1-503-914-0999 tel:%2B1-503-914-0999 (direct)
 +1-954-472-2896 tel:%2B1-954-472-2896 (fax)
 
 On 10/31/2014 09:04 AM, David Wessell wrote:
 I'm connecting a Adtran 904 to a Shoretel system for a voice only PRI.
 
 Am I correct that a cross-over cable is needed and not a straight thru
 cable? The IT vendor and I are having a small disagreement.
 
 Thanks
 David
 
 --
 
 Ringfree Communications, Inc http://ringfree.biz/ http://ringfree.biz/
 
 David Wessell / President
 828-575-0030 x101 tel:828-575-0030%20x101/ da...@ringfree.biz 
 mailto:da...@ringfree.biz mailto:da...@ringfree.biz 
 mailto:da...@ringfree.biz
 
 Ringfree Communications, Inc Office: 828-575-0030 tel:828-575-0030 / Fax: 
 888-243-7830 tel:888-243-7830
 PO BOX 1994 Hendersonville, NC 28793
 http://ringfree.biz http://ringfree.biz/ http://ringfree.biz/ 
 http://ringfree.biz/
 
 This e-mail message may contain confidential or legally privileged
 information and is intended only for the use of the intended
 recipient(s). Any unauthorized disclosure, dissemination, distribution,
 copying or the taking of any action in reliance on the information
 herein is prohibited. E-mails are not secure and cannot be guaranteed to
 be error free as they can be intercepted, amended, or contain viruses.
 Anyone who communicates with us by e-mail is deemed to have accepted
 these risks. Company Name is not responsible for errors or omissions in
 this message and denies any responsibility for any damage arising from
 the use of e-mail. Any opinion and other statement contained in this
 message and any attachment are solely those of the author and do not
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  http://ringfree.biz/
 David Wessell / President
 828-575-0030 x101/ da...@ringfree.biz mailto:da...@ringfree.biz
 Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830 
 PO BOX 1994 Hendersonville, NC 28793
 http://ringfree.biz http://ringfree.biz/
 This e-mail message may contain confidential or legally privileged 
 information and is intended only for the use of the intended recipient(s). 
 Any unauthorized disclosure, dissemination, distribution, copying or the 
 taking of any action in reliance on the information herein is prohibited. 
 E-mails are not secure and cannot be guaranteed to be error free as they can 
 be intercepted, amended, or contain viruses. Anyone who communicates with us 
 by e-mail is deemed to have accepted these risks. Company Name is not 
 responsible for errors or omissions in this message and denies any 
 responsibility for any damage arising from the use of e-mail. Any opinion and 
 other statement contained in this message and any attachment are solely those 
 of the author and do not necessarily represent those of the company.
 

Re: [VoiceOps] T1 Crossover Cable

[Matthew Crocker]

Straight through should be fine, that is all I’ve ever used coming out of an 
Adtran.


remember, Cross-over Ethernet != Cross-over DS-1 cable

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



 On Oct 31, 2014, at 9:04 AM, David Wessell da...@ringfree.biz wrote:
 
 I'm connecting a Adtran 904 to a Shoretel system for a voice only PRI.
 
 Am I correct that a cross-over cable is needed and not a straight thru cable? 
 The IT vendor and I are having a small disagreement.
 
 Thanks
 David
 
 -- 
  http://ringfree.biz/
 David Wessell / President
 828-575-0030 x101/ da...@ringfree.biz mailto:da...@ringfree.biz
 Ringfree Communications, Inc Office: 828-575-0030 / Fax: 888-243-7830 
 PO BOX 1994 Hendersonville, NC 28793
 http://ringfree.biz http://ringfree.biz/
 This e-mail message may contain confidential or legally privileged 
 information and is intended only for the use of the intended recipient(s). 
 Any unauthorized disclosure, dissemination, distribution, copying or the 
 taking of any action in reliance on the information herein is prohibited. 
 E-mails are not secure and cannot be guaranteed to be error free as they can 
 be intercepted, amended, or contain viruses. Anyone who communicates with us 
 by e-mail is deemed to have accepted these risks. Company Name is not 
 responsible for errors or omissions in this message and denies any 
 responsibility for any damage arising from the use of e-mail. Any opinion and 
 other statement contained in this message and any attachment are solely those 
 of the author and do not necessarily represent those of the company.
 
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Re: [VoiceOps] CABS processing / billing

[Matthew Crocker]

Yes,  DUF files FTPed from Verizon.

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



 On Feb 10, 2015, at 6:17 PM, Aaron Seelye aseelye-li...@eltopia.com wrote:
 
 When you say CDRs, are we talking DUF files from your tandem operator?
 
 On 2/10/2015 1:54 PM, Matthew Crocker wrote:
 
 I’m looking for a company that can help process my CABS.  Eat up Verizon and 
 switch CDRs then generate invoices to carriers.I was using Mid American 
 Computing Corp to handle my CABS and customer invoicing.   I just switched 
 all customer invoicing over to H2O and MACC doesn’t make sense for just CABS 
 invoicing.
 
 We are pretty small,  $2k/month in CABS.
 
 Any ideas on who is out there?
 
 Thanks
 
 -Matt
 
 --
 Matthew S. Crocker
 President
 Crocker Communications, Inc.
 PO BOX 710
 Greenfield, MA 01302-0710
 
 E: matt...@crocker.com
 P: (413) 746-2760
 F: (413) 746-3704
 W: http://www.crocker.com
 
 
 
 
 
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 This email has been checked for viruses by Avast antivirus software.
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Re: [VoiceOps] SIP packet capture with index

[Matthew Crocker]

http://www.voipmonitor.org/ http://www.voipmonitor.org/

It works,  it is awesome, it is inexpensive.

Our techs live in voipmon to debug issues,  awesome product.

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710

E: matt...@crocker.com
P: (413) 746-2760
F: (413) 746-3704
W: http://www.crocker.com



 On Mar 24, 2015, at 10:47 AM, Nelson Hicks nels...@socket.net wrote:
 
 I'm looking for options to capture SIP/RTP traffic, index it by call, and 
 make it easy to download the capture for a specific call based on 
 calling/called and time. I want the capture to remain ongoing (rotating 
 capture) with, say, a 96 hour window of calls available. I'm open to hardware 
 and software options.
 
 Right now, I have a server that uses tshark running rotating 1-minute 
 captures, but finding and extracting an individual call out of each of the 
 packet segments and merging them together is a slower and more manual process 
 than I'd like, and I'd like to get our techs direct access to these captures 
 as well.
 
 Thanks,
 
 -- 
 Nelson Hicks
 Network Operations
 SOCKET
 (573) 817- ext. 210
 nels...@socket.net 
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Re: [VoiceOps] How to assure LNP accuracy as a small ITSP

[Matthew Crocker]

Get the phone bill from the end-user and pull a CSR to validate the information

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


 On Jun 23, 2015, at 4:11 PM, Carlos Alvarez caalva...@gmail.com wrote:
 
 Without any access to directly verify phone numbers, we run into quite a lot 
 of issues and rejections because of erroneous customer info that doesn't 
 match what the carrier has.  What do some of you do to assure the highest 
 possible accuracy and success?
 
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Re: [VoiceOps] Broadsoft SMS

[Matthew Crocker]

SMS is supported through their BroadCloud service.  I believe if you have 
BroadWorks you can integrate this portion of BroadCloud as well.

-Matt

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


> On Nov 9, 2015, at 9:14 AM, Colton Conor <colton.co...@gmail.com> wrote:
> 
> What options does Broadsoft offer Broadsoft providers for enabling SMS 
> messages? Anything for MMS messages? I know the have the UC-ONE apps, but I 
> am not sure if they support SMS and MMS. 
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[VoiceOps] VocalData/Tekelec/Genband T6000/M6 hardware available

[Matthew Crocker]

Hello everyone.

 I just issued a shutdown on my M6 call agents pulling the switch out of 
production.   I have 4 Intel servers (2 SBCs, 2 DTMF/Media) and a conference 
bridge available to anyone that is still operating the platform.   Free to a 
good home, you only need to pay for shipping & handling.   If you want to drive 
to Springfield Mass you can pick them up and get them for $0.Front door 
warrantee (warrantee ends when you exit my building…)  They have been up and 
running for years without issue.

First come first serve, if you want them send me an email.

Thanks

-Matt

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com



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Re: [VoiceOps] VocalData/Tekelec/Genband T6000/M6 hardware available

[Matthew Crocker]

Wow, never thought the M6 would still be in use by so many people.  I’ve 
received a bunch of requests for the hardware.  So, basically they are gone 
already, if things fall through I’ll post again to the list.

Thanks

-Matt

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


> On Jan 4, 2016, at 1:43 PM, Matthew Crocker <matt...@corp.crocker.com> wrote:
> 
> 
> Hello everyone.
> 
> I just issued a shutdown on my M6 call agents pulling the switch out of 
> production.   I have 4 Intel servers (2 SBCs, 2 DTMF/Media) and a conference 
> bridge available to anyone that is still operating the platform.   Free to a 
> good home, you only need to pay for shipping & handling.   If you want to 
> drive to Springfield Mass you can pick them up and get them for $0.Front 
> door warrantee (warrantee ends when you exit my building…)  They have been up 
> and running for years without issue.
> 
> First come first serve, if you want them send me an email.
> 
> Thanks
> 
> -Matt
> 
> —
> 
> Matthew Crocker
> President - Crocker Communications, Inc.
> Managing Partner - Crocker Telecommunications, LLC
> E: matt...@corp.crocker.com
> E: matt...@crocker.com
> 
> 
> 
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Re: [VoiceOps] Preventing random SIP connections to handsets

[Matthew Crocker]

We have a Calix ONT in our lab that is ‘on the internet’ for its voice VLAN.   
It gets rogue INVITES and rings constantly (every 5-10 seconds).   Makes for a 
nice honeypot, source IPs go right into the ACL on the firewall

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


> On Nov 20, 2015, at 3:35 PM, Robert Johnson <rober...@bendtel.com> wrote:
> 
> On 11/20/2015 12:14 PM, Carlos Alvarez wrote:
>> We're starting to see customers who get random arbitrary ringing caused by
>> a random connection attempt from the internet.  Most of our customers have
>> Cisco routers with full-cone NAT, so it's easy to do that.  We don't
>> reinvite handsets, we proxy the media, so we've considered using restricted
>> NAT instead.  If we can figure out how, we can't find any documentation on
>> how to do it, and don't have a response to our Cisco TAC case on it yet.
>> 
>> But I figured I'd ask if others have come up with better solutions.  I know
>> there are a few authentication options in the phones themselves, but they
>> seem to vary greatly by vendor and even by model.  I like to do things as
>> simply and system-wide as possible.  We primarily sell Grandstream, and we
>> support Cisco/Linksys SPA as well as Polycom IP series (not VVX).
>> 
>> We're an Asterisk-based hosted service provider.
>> 
>> 
>> 
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>> 
> 
> This may be dependent upon the Cisco router in question, but when we
> deploy routers we always set the ACL to only allow SIP communications
> from our SBC. - When customers provide their own, we recommend the same
> settings.
> 
> -- 
> Robert Johnson
> BendTel, Inc.
> (541)389-4020
> Central Oregon's Own Telephone and Internet Service Provider
> http://bendtel.com/about/
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Re: [VoiceOps] Preventing random SIP connections to handsets

[Matthew Crocker]

> On Nov 20, 2015, at 3:27 PM, Alex Balashov  wrote:
> 
> On 11/20/2015 03:23 PM, Carlos Alvarez wrote:
> 
>> That's the default for all the handsets, I believe.  There are various
>> options such as "accept only from proxy" or "only from registrar," but
>> like I said it varies so it could be more challenging to employ that.
>> Also in our limited testing it seems like it may not have had the
>> intended effect.  Possibly because NAT hides the original IP, but I
>> don't know that for sure.
> 
> Any properly standards-compliant registrar will send a Request URI on 
> incoming INVITEs that is equivalent to the Contact binding provided by the 
> phone originally. It can choose to send that INVITE to a network and 
> transport-layer destination that is different to the network and 
> transport-reachability in the contact provided by the handset, i.e. for 
> far-end NAT traversal, but the integrity of the RURI should not be 
> compromised.
> 
>> Most phones also have an option to force auth for incoming invites,
>> which we have not tested yet.
> 
> I don't think you want that. SIP servers and registrars will certainly 
> definitely expect the registrant to trust them. You can certainly configure 
> Asterisk per se to answer 401/407 challenges from the phone with digest 
> credentials, but that's not a very simple or interchangeable solution.
> 

Broadworks handles the 401 UNAUTHORIZED with nonce fine from a Polycom.  It 
will resend the INVITE with the authentication credentials



> -- Alex
> 
> -- 
> Alex Balashov | Principal | Evariste Systems LLC
> 303 Perimeter Center North, Suite 300
> Atlanta, GA 30346
> United States
> 
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] Virtualized SBC

[Matthew Crocker]

NFV is all about containers and micro services.   It is unix all over again but 
in the cloud.  Small containerized functions that do a specific task.  Spun up 
in the cloud and linked together by an orchestration overlay.  Personally I 
think it is a good thing



—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


> On Apr 7, 2016, at 10:42 AM, Alex Balashov <abalas...@evaristesys.com> wrote:
> 
> Management and licencing are rather ancillary to any discussion of NFs.
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
> 
> Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
> 
> Sent from my BlackBerry.
> From: Pete Eisengrein
> Sent: Thursday, April 7, 2016 08:26
> To: Alex Balashov
> Cc: voiceops@voiceops.org
> Subject: Re: [VoiceOps] Virtualized SBC
> 
> If it requires a bunch of configuration, I agree it is just V. But if there's 
> an orchestration layer that informs all other systems of its presence, 
> including management and licensing, without having to configure all that, 
> then it is NFV. Now, whether the aforementioned vendors are actually doing 
> this, that can be debated (and I'd be curious to hear people's options on 
> it). They may just be doing V with a long-term vision of NFV, I don't know.
> 
> On Thu, Apr 7, 2016 at 7:52 AM, Alex Balashov <abalas...@evaristesys.com 
> <mailto:abalas...@evaristesys.com>> wrote:
> No. That's not NFV. That's just... V.
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
> 
> Tel: +1-800-250-5920 <tel:%2B1-800-250-5920> (toll-free) / +1-678-954-0671 
> <tel:%2B1-678-954-0671> (direct)
> Web: http://www.evaristesys.com/ <http://www.evaristesys.com/>, 
> http://www.csrpswitch.com/ <http://www.csrpswitch.com/>
> 
> Sent from my BlackBerry.
> From: Pete Eisengrein
> Sent: Thursday, April 7, 2016 07:26
> Cc: voiceops@voiceops.org <mailto:voiceops@voiceops.org>
> Subject: Re: [VoiceOps] Virtualized SBC
> 
> > my point was that they, of all
> 
> > things, are a poor standard
> 
> > bearer for the NFV marketing-
> 
> > gasm.
> 
>  
> With centralized licensing and the ability to scale resources when and where 
> you need them, I would argue they are the perfect device for the NFV 
> marketing-gasm (LOL).
> 
> 
> On Wed, Apr 6, 2016 at 11:08 PM, Alex Balashov <abalas...@evaristesys.com 
> <mailto:abalas...@evaristesys.com>> wrote:
> On 04/06/2016 10:54 PM, Ryan Delgrosso wrote:
> 
> an SBC ... is a demarcation and control point between network
> segments where you can inject interworking and business logic.
> 
> Absolutely, at >= Layer 5.
> 
> If it's news to anyone here that you can virtualise applications, they've got 
> some catching up to do.
> 
> Anyway, my argument wasn't that SBCs serve no valid purposes. You've done a 
> good job of outlining them. Instead, my point was that they, of all things, 
> are a poor standard bearer for the NFV marketing-gasm.
> 
> -- Alex
> 
> -- 
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
> 
> Tel: +1-800-250-5920 <tel:%2B1-800-250-5920> (toll-free) / +1-678-954-0671 
> <tel:%2B1-678-954-0671> (direct)
> Web: http://www.evaristesys.com/ <http://www.evaristesys.com/>, 
> http://www.csrpswitch.com/ <http://www.csrpswitch.com/>
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Re: [VoiceOps] media bypass question

[Matthew Crocker]

If the customer truly has an SBC I don’t see why you couldn’t put the Carrier 
information in the SDP going out to the customer.  Same thing going back,  
Customer SDP on 200 OK back to the carrier.   Most customers in Hosted PBX 
don’t have an SBC however, they have a dumb firewall.   In the case of a dumb 
firewall (or worse a broken SIP ALG) you are risking one way audio problems if 
the Carrier SBC doesn’t handle RTP latching.

—

Matthew Crocker
President - Crocker Communications, Inc.
Managing Partner - Crocker Telecommunications, LLC
E: matt...@corp.crocker.com
E: matt...@crocker.com


> On Apr 18, 2016, at 9:39 AM, Kumudu Suriyaarachchi <kumudu.v...@gmail.com> 
> wrote:
> 
> Hello,
> 
>  
> Is there a mechanism to achieve  media bypass from the customer premise SBC 
> to peering/carrier SBC  where in a typical NAT traversal hosted PBX 
> deployment?
> 
> I am talking about multi-vendor SBCs on the access and the peering sides.
> 
>  
> Thanks,
> 
> Kumudu
> 
> 
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[VoiceOps] CNAM lookup options for Broadowks (SIP or SOAP)

[Matthew Crocker]

Hello,

  I’m looking into getting CNAM lookups working in Broadworks instead of having 
my Taqua perform SS7 dips.   Broadworks supports lookups via SIP or SOAP.

  I’m testing ezcnam.com via SIP.Broadworks sends the SUBSCRIBE and I get a 
NOTIFY response.  Broadworks then fails on decoding the Calling-Name from ezcnam

  Does anyone on this list use a CNAM service with Broadworks (R20sp1)?  If so, 
can you send me some contacts for ones that work?

Thanks

-Matt

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President
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Re: [VoiceOps] FCC Numbering Authorization Application?

[Matthew Crocker]

Does anyone have a list of companies that provide the TDM-SIP gateway 
interconnection?   Verizon-MA only does TDM interconnection, I have a Taqua 
T7000 handling that but would prefer moving everything to SIP.

-Matt

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Crocker Communications, Inc.
President

From: VoiceOps <voiceops-boun...@voiceops.org> on behalf of Nicholas Sten 
<nicks...@gmail.com>
Date: Sunday, May 7, 2017 at 12:32 PM
To: Ryan Finnesey <r...@finnesey.com>
Cc: "voiceops@voiceops.org" <voiceops@voiceops.org>
Subject: Re: [VoiceOps] FCC Numbering Authorization Application?

My bad.  Does this help?

For further information regarding the Commission’s electronic filing process, 
please contact ECFS Help at ecfsh...@fcc.gov<mailto:ecfsh...@fcc.gov> or 
202-418-0193; for further information regarding other aspects of the 
Commission’s Numbering Authorization Application process, please contact 
Margoux Brown, Competition Policy Division, Wireline Competition Bureau, at 
margoux.br...@fcc.gov<mailto:margoux.br...@fcc.gov> or (202) 418- 1584.

https://www.nationalnanpa.com/tools/trainGuides/getting-started-for-interconnected-voip-providers.pdf


On Sun, May 7, 2017 at 10:43 AM, Ryan Finnesey 
<r...@finnesey.com<mailto:r...@finnesey.com>> wrote:
I did find that page via Google but I don’t see the Application or an 
instructions on how to file the application.  I just see notices of other 
companies that have filed or been granted an application.

Cheers
Ryan


From: Nicholas Sten [mailto:nicks...@gmail.com<mailto:nicks...@gmail.com>]
Sent: Friday, May 5, 2017 8:28 PM
To: Ryan Finnesey <r...@finnesey.com<mailto:r...@finnesey.com>>
Cc: voiceops@voiceops.org<mailto:voiceops@voiceops.org>
Subject: Re: [VoiceOps] FCC Numbering Authorization Application?

https://www.fcc.gov/wireline-competition/competition-policy-division/numbering-resources/general/voip-numbering#block-menu-block-4


On May 5, 2017, at 5:08 PM, Ryan Finnesey 
<r...@finnesey.com<mailto:r...@finnesey.com>> wrote:
It seems my Google skills have failed  me today ☹  Where can I find the FCC 
Numbering Authorization Application so I can request numbers as a VoIP Provider?
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Re: [VoiceOps] How do you update/manage your notification contacts?

[Matthew Crocker]

Set up a status page, configure RSS and let your customers subscribe to the RSS 
feed.   Their responsibility to maintain notifications and they will drop off 
when they are no longer interested.

Or, you could setup a couple twitter handles for notifications and have 
customers follow them.   No need to maintain a notification list.

-Matt

--
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Crocker Communications, Inc.
President

From: VoiceOps <voiceops-boun...@voiceops.org> on behalf of Carlos Alvarez 
<caalva...@gmail.com>
Date: Thursday, June 1, 2017 at 4:40 PM
To: "voiceops@voiceops.org" <voiceops@voiceops.org>
Subject: Re: [VoiceOps] How do you update/manage your notification contacts?

Sending notifications is the easy part, lots of services for that.  It's the 
maintenance of who the right contacts is which I find challenging.  Since we 
see support tickets arrive from unexpected/new contacts, we know there must be 
people who need to know, but we don't know who they are.


On Thu, Jun 1, 2017 at 1:37 PM, Ryan Delgrosso 
<ryandelgro...@gmail.com<mailto:ryandelgro...@gmail.com>> wrote:

Weve recently starting testing a statuspage internally http://staytus.co/ and 
like it enough we will probably go live with it customer facing. this allows us 
to have pre-formatted emails for different event classifications, and the users 
opt in/out on their own. Its fairly extensible being written in ruby, and has 
an API to take machine-generated events.

Atlassian has their own as well if you want to pay atlassian money

https://www.atlassian.com/software/statuspage

On 5/31/2017 2:58 PM, Carlos Alvarez wrote:
We're at the point where we really need to clean up and update our 
notifications contact list (people to notify of outages, changes, etc).  I'm 
curious what people here use.  We use Freshdesk for our support tickets, and 
that was a good list to start with, but as employees change it wouldn't get 
updated necessarily.

Something we can simply pay for and outsource is ideal.  We're an open source 
company, but time is a precious resource right now.




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Re: [VoiceOps] Local calling tariff database

[Matthew Crocker]

The NPANXX of the CallingParty sets the rate center.  The rate center sets the 
list of all local NPANXX for the CalledParty.  If the CalledParty NPANXX is in 
the list then it is a local call.

LRN has nothing to do with it

On Nov 14, 2017, at 6:54 PM, Alex Balashov 
> wrote:

On Tue, Nov 14, 2017 at 01:48:20PM -1000, Erik wrote:

You can port out of a rate center what u can't do is port out of a
LATA so if you're a clec the LRN does truly matter for cost depending
how you route your tandem traffic vs VoIP traffic.

You can't port out of a rate centre.

--
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] Local calling tariff database

[Matthew Crocker]

Telcordia LCA provides this data

http://www.trainfo.com/products_services/tra/downloads/lca.pdf

Combined with the LERG you can get everything you need for NPANXX & LCA

I have a script that eats it up and injects it into my Broadworks NS for local 
call routing.  I use it for proper rating (billing trusts the Broadworks 
rating) and 7 digit dials when available.

-Matt 
 
-- 
Matthew Crocker
Crocker Communications, Inc.
President

On 11/14/17, 2:45 PM, "VoiceOps on behalf of Alex Balashov" 
<voiceops-boun...@voiceops.org on behalf of abalas...@evaristesys.com> wrote:

Not for routing every single call out of a service provider it isn't. :-) 

On November 14, 2017 2:24:27 PM EST, BackUP Telecom Consulting 
<mary...@backuptelecom.com> wrote:
>Localcallingguide.com and it's free!
>
>Mary Lou Carey
>
>BackUP Telecom Consulting
>
>615-791-9969
>
>On 2017-11-14 02:44 AM, Alex Balashov wrote:
>> Where can one get a digest of local access tariffs that is 
>> complementary
>> to machine processing?
>> 
>> Ten years ago, when I last cared about this, the canonical answer was
>> CCMI. Is that still the answer?
>> 
>> Any idea on pricing? Does anyone offer
>> "can-this-NPA-NXX-call-that-NPA-NXX-locally"-aaS, but reliably?


-- Alex

--
Sent via mobile, please forgive typos and brevity. 
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[VoiceOps] Verizon New England routing contact

[Matthew Crocker]

Does anyone have a contact for someone in Verizon New England for call routing? 
  They stopped routing inbound calls to our tandem (Inteliquent) yesterday 
morning.  All calls from Verizon/ Verizon Wireless in LATA 126 to my customers 
are failing with a fast busy.   It is impossible to get anyone on the phone to 
help resolve the issue.

-Matt

--
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Crocker Communications, Inc.
President
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Re: [VoiceOps] Verizon New England routing contact

[Matthew Crocker]

Inteliquent NOC was great, pushing the tickets with Verizon.   It has been 24 
hours now, just received an update that a DS-3 in Springfield, MA is causing 
the issue.  Verizon dispatching a tech

Thanks,

-Matt

--
Matthew Crocker
Crocker Communications, Inc.
President

From: Paul Timmins 
Date: Thursday, June 14, 2018 at 9:25 AM
To: Matthew Crocker 
Cc: "voiceops@voiceops.org" 
Subject: Re: [VoiceOps] Verizon New England routing contact

When we have this with intelliquent they usually have contacts for the other 
tandem operator and help us open tickets. Sometimes you have to push them a bit.

On Jun 14, 2018 07:47, Matthew Crocker  wrote:

Does anyone have a contact for someone in Verizon New England for call routing? 
  They stopped routing inbound calls to our tandem (Inteliquent) yesterday 
morning.  All calls from Verizon/ Verizon Wireless in LATA 126 to my customers 
are failing with a fast busy.   It is impossible to get anyone on the phone to 
help resolve the issue.

-Matt

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Crocker Communications, Inc.
President
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[VoiceOps] Netsapien reviews

[Matthew Crocker]

Hello,

I’m a Broadsoft Broadworks shop but I’m looking at alternatives,  Not a YUGE 
fan of Cisco…

If you have an opinion on NetSapien as a HostedPBX platform (good, bad, ugly) 
please send to me on or off list.
 Thanks


-Matt

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Crocker Communications, Inc.
President
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Re: [VoiceOps] Broadworks call transfer destination lookup

[Matthew Crocker]

Sounds like selective call forwarding which is a premium license feature.   Not 
really a database lookup but works 

You could also setup another sip feature server have bworks route to that, do 
the lookup and return a 302 moved temporarily back to the AS

> On Feb 12, 2018, at 3:16 AM, Julien Lamarche  wrote:
> 
> Hello,
> 
> Is it possible to configure Broadworks to use a third party tool to determine 
> the destination (number / extension / bwUserID ) of a call transfer, such as 
> a database SQL lookup or a bash / perl script? The input of that lookup would 
> be the originating number.
> 
> ie.
> 
> if calling number is X, send to called number A
> 
> if calling number is Y, send to called number B
> 
> if calling number is not found, send to called number C
> 
> 
> 
> Julien
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Re: [VoiceOps] HD Voice / Wideband audio?

[Matthew Crocker]

Most of my traffic runs over Inteliquent and about 14% of that traffic is g722. 
 I’m not seeing any g722 traffic through bandwidth.com or ThinQ


-Matt

--
Matthew Crocker
Crocker Communications, Inc.
President

From: Colton Conor 
Date: Wednesday, August 15, 2018 at 9:45 AM
To: Matthew Crocker 
Cc: Ryan Delgrosso , "voiceops@voiceops.org" 

Subject: Re: [VoiceOps] HD Voice / Wideband audio?

Matt,

Are those G722 calls between users hosted on your switch (internal calls) or 
external calls from other Bandwidth.com carrier customers?

On Wed, Aug 15, 2018 at 8:21 AM Matthew Crocker 
mailto:matt...@corp.crocker.com>> wrote:

I’m 100% SIP in and out via Inteliquent/Neutral Tandem/Bandwidth.com.  I run 
Broadworks which is codec agnostic and lets the end points figure it out.   I 
enable all codecs supported by end points (Polycom).   I’m seeing 15.6% of 
calls establish as g722,  1.5% as g729, 0.1% as g722.1, 0.5% t38 and the rest 
g711u


-Matt

--
Matthew Crocker
Crocker Communications, Inc.
President

From: VoiceOps 
mailto:voiceops-boun...@voiceops.org>> on behalf 
of Ryan Delgrosso mailto:ryandelgro...@gmail.com>>
Date: Wednesday, August 15, 2018 at 12:52 AM
To: "voiceops@voiceops.org<mailto:voiceops@voiceops.org>" 
mailto:voiceops@voiceops.org>>
Subject: Re: [VoiceOps] HD Voice / Wideband audio?


Nope, becoming table stakes. I push opus everywhere I can these days. Smaller 
footprint, HD experience, packet loss concealment. The downside is its 
relatively processor expensive to transcode.

Most of the wireless guys are using AMR which is royalty bearing and not quite 
as advanced.

Look at it like this, change codecs because you want better experience in less 
network footprint, the HD part is a bonus.

-Ryan

On 8/14/2018 2:33 PM, Ryan Finnesey wrote:
I know this topic has come up a few times on the list before but now I can’t 
seem to find the discussion(s)  now.Is it mostly the wireless carriers  
that support HD Voice / Wideband audio?



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Re: [VoiceOps] HD Voice / Wideband audio?

[Matthew Crocker]

I’m 100% SIP in and out via Inteliquent/Neutral Tandem/Bandwidth.com.  I run 
Broadworks which is codec agnostic and lets the end points figure it out.   I 
enable all codecs supported by end points (Polycom).   I’m seeing 15.6% of 
calls establish as g722,  1.5% as g729, 0.1% as g722.1, 0.5% t38 and the rest 
g711u


-Matt

--
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Crocker Communications, Inc.
President

From: VoiceOps  on behalf of Ryan Delgrosso 

Date: Wednesday, August 15, 2018 at 12:52 AM
To: "voiceops@voiceops.org" 
Subject: Re: [VoiceOps] HD Voice / Wideband audio?


Nope, becoming table stakes. I push opus everywhere I can these days. Smaller 
footprint, HD experience, packet loss concealment. The downside is its 
relatively processor expensive to transcode.

Most of the wireless guys are using AMR which is royalty bearing and not quite 
as advanced.

Look at it like this, change codecs because you want better experience in less 
network footprint, the HD part is a bonus.

-Ryan

On 8/14/2018 2:33 PM, Ryan Finnesey wrote:
I know this topic has come up a few times on the list before but now I can’t 
seem to find the discussion(s)  now.Is it mostly the wireless carriers  
that support HD Voice / Wideband audio?




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Re: [VoiceOps] LNP, tandems, etc.

[Matthew Crocker]

Mike

You either need to connect to the other tandem or pay another carrier to get 
you there.  It is the responsibility of the originating carrier to get the call 
to the terminating carriers switch.  Depending on the call volume you should 
probably just route through your LD carrier

On Aug 28, 2018, at 9:15 PM, Mike Hammett 
mailto:voice...@ics-il.net>> wrote:

I drew a picture, hoping it would clear things up a bit.

Through my switch, we're trying to call a number in a block assigned to 
Sycamore, with an LRN in a DeKalb number block, but the common tandem operator 
is telling me that the LRN actually lives on a different tandem.



-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com




From: "Mike Hammett" mailto:voice...@ics-il.net>>
To: voiceops@voiceops.org
Sent: Tuesday, August 28, 2018 4:46:52 PM
Subject: [VoiceOps] LNP, tandems, etc.

I thought you had to be on the same tandem to port a number, but with what our 
tandem operator (Frontier) is telling me, this isn't the case.

Comcast ported a number from us in town A. The LRN they pointed to is based in 
town B (per TelcoData). The tandem generally used by carriers in both towns is 
based in town B. Naturally, we send traffic to that tandem.

The operator of that tandem is telling us that the LRN is actually homed off of 
a different tandem in our LATA (operated by CenturyLink) in town C. 
Unfortunately, I can't corroborate this information with TelcoData the only 
rate center I see off of that tandem in TelcoData is an AT town next door.

Where can I read up authoritatively on the porting requirements that would 
apply to this and related bits of info I should know?

I'm checking on our LERG access as I know that has the authoritative 
information, but I don't have that access at the moment. Maybe we're not 
subscribed to it.

Number NPA-NXX in town A: 
https://www.telcodata.us/search-area-code-exchange-detail?npa=815=991
LRN NPA-NXX in town B: 
https://www.telcodata.us/search-area-code-exchange-detail?npa=815=901
Tandem in town B: 
https://www.telcodata.us/search-switches-by-tandem-clli?cllicode=DKLBILXA50T
Tandem in town C: 
https://www.telcodata.us/search-switches-by-tandem-clli?cllicode=DIXNILXA50T


Thanks.



-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com



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Re: [VoiceOps] Broadworks / Merging calls with Voipmon

[Matthew Crocker]

Well, would you look at that. The 183 Session Progress from Broadworks -> 
Polycom has the header.  VoipMon should be merging them together.  I’ll dig 
deeper but I think it is an issue with the voipmon config at this point.

Thanks for the help ☺


-Matt

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President
From: "Zilk, David" <david.z...@cdk.com>
Date: Monday, January 22, 2018 at 4:21 PM
To: Matthew Crocker <matt...@corp.crocker.com>, Matthew Beckwell 
<matth...@aitech.net>, "voiceops@voiceops.org" <voiceops@voiceops.org>
Subject: RE: [VoiceOps] Broadworks / Merging calls with Voipmon

In my experience the leg gets identified by the correlation ID that broadworks 
adds to the responses associated with the original INVITE. VoIPmonitor will use 
that to complete the correlation.

From: Matthew Crocker [mailto:matt...@corp.crocker.com]
Sent: Monday, January 22, 2018 1:19 PM
To: Zilk, David <david.z...@cdk.com>; Matthew Beckwell <matth...@aitech.net>; 
voiceops@voiceops.org
Subject: Re: [VoiceOps] Broadworks / Merging calls with Voipmon


Thanks David,


The Broadworks Correlation Id is enabled but the original INVITE from the 
end-user (Polycom) doesn’t have it so voipmon isn’t picking it up (which make 
sense).  I was hoping there was a way for Broadworks to pull something out of 
the original invite for tracking.



Selecting both calls in voipmon and merging works, just doesn’t merge 
auto-magically

-Matt

--
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President
From: VoiceOps 
<voiceops-boun...@voiceops.org<mailto:voiceops-boun...@voiceops.org>> on behalf 
of "Zilk, David" <david.z...@cdk.com<mailto:david.z...@cdk.com>>
Date: Monday, January 22, 2018 at 4:11 PM
To: Matthew Beckwell <matth...@aitech.net<mailto:matth...@aitech.net>>, 
"voiceops@voiceops.org<mailto:voiceops@voiceops.org>" 
<voiceops@voiceops.org<mailto:voiceops@voiceops.org>>
Subject: Re: [VoiceOps] Broadworks / Merging calls with Voipmon

If you click on the ‘Merge’ dropdown while displaying the Legs by Header, you 
can select ‘SIP History’ to display the SIP Ladder diagram of all the legs 
together.

David

From: VoiceOps [mailto:voiceops-boun...@voiceops.org] On Behalf Of Matthew 
Beckwell
Sent: Monday, January 22, 2018 1:02 PM
To: voiceops@voiceops.org<mailto:voiceops@voiceops.org>
Subject: Re: [VoiceOps] Broadworks / Merging calls with Voipmon

Hi Matt,
Here's what I've done to get close to what you're looking for...

In the BroadWorks Application Server, set these parameters:

AS_CLI/Interface/SIP>
sendCallCorrelationIDAccess = true
sendCallCorrelationIDNetwork = true


Once you do that, you'll see BroadWorks start to add a header for "related" 
calls. They will have have a common correlation header (but different call-id) 
like this:

X-BroadWorks-Correlation-Info:1126786:1


Then, in voipmonitor.conf you can set this parameter in the sniffer 
configuration to keep track of that header's value in the database:

matchheader = X-BroadWorks-Correlation-Info


VoIPmonitor won't merge them into a single SIP Diagram (at least not that I've 
found)-- but you should start to see the "related" call legs (with different 
Call-ID's but the same X-BroadWorks-Correlation-Info header) show up in the 
"Legs by header" tab when you're looking at a call's details in the VoIPmonitor 
GUI.

~Matthew





On Mon, Jan 22, 2018 at 2:36 PM, Matthew Crocker 
<matt...@corp.crocker.com<mailto:matt...@corp.crocker.com>> wrote:

Hello,


We currently have Broadworks/AcmePacket handling calls to/from customers.  We 
have a couple VoipMon sensors running watching all traffic inside/outside our 
SBC.   Currently calls are presented in VoipMon as two different calls (PSTN -> 
Broadworks & Broadworks -> Polycom) or (Polycom -> Broadworks & Broadworks -> 
PSTN). The Call-id on each call is different,  Broadworks generates a new 
SIP Dialog for the call.   If the customer has a hunt group there could be 
several INVITEs going out to multiple phones, all with different Call-Id. 
Does anyone know of a way to get VoipMon to merge the calls into a single 
CDR/SIP Diagram?  Is there a way to configure Broadworks to embed the original 
Call-Id in the new INVITE (Parent-Call-Id Header)?

I’m running R20sp1

Thanks

-Matt

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Crocker Communications, Inc.
President

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[VoiceOps] Broadworks / Merging calls with Voipmon

[Matthew Crocker]

Hello,


We currently have Broadworks/AcmePacket handling calls to/from customers.  We 
have a couple VoipMon sensors running watching all traffic inside/outside our 
SBC.   Currently calls are presented in VoipMon as two different calls (PSTN -> 
Broadworks & Broadworks -> Polycom) or (Polycom -> Broadworks & Broadworks -> 
PSTN). The Call-id on each call is different,  Broadworks generates a new 
SIP Dialog for the call.   If the customer has a hunt group there could be 
several INVITEs going out to multiple phones, all with different Call-Id. 
Does anyone know of a way to get VoipMon to merge the calls into a single 
CDR/SIP Diagram?  Is there a way to configure Broadworks to embed the original 
Call-Id in the new INVITE (Parent-Call-Id Header)?

I’m running R20sp1

Thanks

-Matt

--
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Crocker Communications, Inc.
President
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Re: [VoiceOps] Troubles with Comcast

[Matthew Crocker]

You probably need to work with your tandem provider to find out they are 
presenting the all circuits busy error

On Jan 23, 2018, at 5:37 PM, Mike Hammett 
> wrote:

I apologize in advance for I am new to this level of voice.

I'm sending calls to the local tandem and I'm getting all circuits are busy 
when I call a Comcast number (or one ported to Comcast). Calls to those numbers 
work from other places (like my cell phone). Calls to other carriers on that 
tandem work.

How can I reach someone at Comcast to troubleshoot this?


We're using the Metaswitch platform.



-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com


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[VoiceOps] Retiring my Taqua T7000

[Matthew Crocker]

I’m about a month away from removing power from my Taqua T7000 and shutting it 
down.   If anyone is interested in it in whole or in part let me know

I have a ‘blue’ chassis,  Linux based AP.  It is currently in production, in 
Springfield MA

Running 6.2.0pr13

5 TIC1 + cold spare
2 TIC2
2 PIC2
Clocks & commons

I think I have a PIC1 & clock cold spare as well


-Matt

--
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Crocker Communications, Inc.
President
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Re: [VoiceOps] Automatic 911 location updates

[Matthew Crocker]

Location information should be pushed to the phone via DHCP.  The DHCP server 
can get option82 information from the PoE switch port.  The phone should then 
include the information in an INVITE for emergency calls based on its dial 
plan. The carrier can pass the information through to the NG911 PSAP.  

> On Oct 26, 2018, at 8:36 AM, Carlos Alvarez  wrote:
> 
> That’s A-GPS using WiFi and related location databases. A raw, basic, $40 gps 
> module attached to an arduino gets nothing useful in the middle of my 
> standard wood and stucco house. 
> 
> 
> -- 
> Sent from my iPad
> 
>> On Oct 25, 2018, at 10:48 PM, Aaron C. de Bruyn  wrote:
>> 
>> Depends on the construction.
>> I usually get a good GPS fix on my cell phone indoors.
>> 
>> Then again, I don't work in the 4th sub-basement of a skyscraper. ;)
>> 
>> -A
>> 
>>> On Thu, Oct 25, 2018 at 10:36 PM Carlos Alvarez  wrote:
>>> 
>>> They don’t work indoors.
>>> 
>>> 
>>> --
>>> Sent from my iPad
>>> 
 On Oct 25, 2018, at 10:07 PM, Aaron C. de Bruyn via VoiceOps 
  wrote:
 
 I've always wondered why VoIP phones don't have a cheap GPS chip in them.
 
 Sure, it could raise all sorts of problems from a spoofing
 perspective, but for valid 911 calls it could come in handy...
 
 -A
> On Tue, Oct 23, 2018 at 4:41 PM Ryan Delgrosso  
> wrote:
> 
> Automatic, nope. Psuedo-auto im looking at it now.
> 
> West supports sending custom headers to indicate a phones location inside 
> a building to provide additional routing info but nothing using 
> geo-location.
> 
> You could leverage this to have phones send custom headers based on 
> network location.
> 
> I have not seen anything using geo-ip or GPS primarily because most IP 
> handsets dont to those things.
> 
> 
> 
> On 10/23/2018 4:21 PM, Carlos Alvarez wrote:
> 
> Is anyone doing this yet?  The ability to detect the location of a phone 
> and automatically update the 911 address?  Yes, I realize it's a 
> difficult if not impossible thing, yet thought I'd ask since a customer 
> asked me.
> 
> 
> 
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 VoiceOps mailing list
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[VoiceOps] Oracle (Acme Packet) help

[Matthew Crocker]

Hello,


 Does anyone know if I can do some basic call routing on an Acme Session 
Director?   Basically I want to build a dialplan to route calls to various 
session-groups based on the SIP URI.  Example:  911@* goes to SAG:EMERGENCY,  
011*@* goes to SAG:INTERNATIONAL, +1*@* goes to SAG:LONGDISTANCE,  etc.

I’m trying to avoid bouncing the calls through something like a Kamailo or 
freeswitch to process and 302 Moved the call back to the Acme.

Thanks
-Matt

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Re: [VoiceOps] Oracle (Acme Packet) help

[Matthew Crocker]

Thanks everyone,  reading up on LRT now.   Going to read up on ENUM as well, 
certainly sounds easier to manage large datasets with.  Easier to upload a DNS 
server then gzip & upload a LRT XML file

Thanks

From: VoiceOps  on behalf of Ryan Delgrosso 

Date: Monday, November 12, 2018 at 3:07 PM
To: "voiceops@voiceops.org" 
Subject: Re: [VoiceOps] Oracle (Acme Packet) help


You can do this with local policy pretty easily.

LRT is an option but I detest LRT. I would deploy enum before using LRT.



On 11/12/2018 11:25 AM, Matthew Crocker wrote:

Hello,


 Does anyone know if I can do some basic call routing on an Acme Session 
Director?   Basically I want to build a dialplan to route calls to various 
session-groups based on the SIP URI.  Example:  911@* goes to SAG:EMERGENCY,  
011*@* goes to SAG:INTERNATIONAL, +1*@* goes to SAG:LONGDISTANCE,  etc.

I’m trying to avoid bouncing the calls through something like a Kamailo or 
freeswitch to process and 302 Moved the call back to the Acme.

Thanks
-Matt





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Re: [VoiceOps] Oracle (Acme Packet) help

[Matthew Crocker]

Alex,

 Thanks, I actually have 2 projects, LRT will work for my 'call type' routing 
requirements (LD, EMER, TF, INT) should be easily handled with LRT.   I'll also 
need ENUM to handle my inbound with 20k-ish numbers being routed to a couple 
different switches.  I'm in the process of moving customers from one switch to 
another and updated DNS with the move makes sense.   Now I just need to get all 
the pieces flying in formation to work


On 11/12/18, 4:07 PM, "VoiceOps on behalf of Alex Balashov" 
 wrote:

However, if you only have like < 10 rules, use something purely internal
as Ryan suggests. 

On Mon, Nov 12, 2018 at 08:22:28PM +0000, Matthew Crocker wrote:
> 
> Thanks everyone,  reading up on LRT now.   Going to read up on ENUM as 
well, certainly sounds easier to manage large datasets with.  Easier to upload 
a DNS server then gzip & upload a LRT XML file
> 
> Thanks
> 
> From: VoiceOps  on behalf of Ryan 
Delgrosso 
> Date: Monday, November 12, 2018 at 3:07 PM
> To: "voiceops@voiceops.org" 
> Subject: Re: [VoiceOps] Oracle (Acme Packet) help
> 
> 
> You can do this with local policy pretty easily.
> 
> LRT is an option but I detest LRT. I would deploy enum before using LRT.
> 
> 
> 
> On 11/12/2018 11:25 AM, Matthew Crocker wrote:
> 
> Hello,
> 
> 
>  Does anyone know if I can do some basic call routing on an Acme Session 
Director?   Basically I want to build a dialplan to route calls to various 
session-groups based on the SIP URI.  Example:  911@* goes to SAG:EMERGENCY,  
011*@* goes to SAG:INTERNATIONAL, +1*@* goes to SAG:LONGDISTANCE,  etc.
> 
> I’m trying to avoid bouncing the calls through something like a Kamailo 
or freeswitch to process and 302 Moved the call back to the Acme.
> 
> Thanks
> -Matt
> 
> 
> 
> 
> 
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> 
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> 
> https://puck.nether.net/mailman/listinfo/voiceops
> 
> 

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-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) 
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] is there an Answering Service geared towards telecom industry ?

[Matthew Crocker]

Crocker Communications, Inc.
Family owned & operated.  Our operators are in Greenfield, MA.  We can build 
scripts to follow any procedure and push/pull information from your systems 
(SOAP,REST, etc).

Service is billed monthly, base rate + per minute.  If you are interested I can 
get a sales person to give you specifics.

-Matt


From: Izzy Goldstein - TeleGo 
Date: Tuesday, November 27, 2018 at 4:22 PM
To: Matthew Crocker 
Cc: "Voiceops.org" 
Subject: Re: [VoiceOps] is there an Answering Service geared towards telecom 
industry ?

im looking for the answering service to send us the call information via API in 
addition via Email

whats your company information ?

On Tue, Nov 27, 2018 at 4:05 PM Matthew Crocker 
mailto:matt...@corp.crocker.com>> wrote:

Self Promotion:

I own an Answering Service (50+ years) which operates along side our ISP (25 
years) & ITSP.

Your system can access APIs via standard protocols and Tx/Rx information

What type of services are you looking for?


From: VoiceOps 
mailto:voiceops-boun...@voiceops.org>> on behalf 
of Izzy Goldstein - TeleGo mailto:igoldst...@telego.net>>
Date: Tuesday, November 27, 2018 at 4:03 PM
To: "Voiceops.org" mailto:voiceops@voiceops.org>>
Subject: [VoiceOps] is there an Answering Service geared towards telecom 
industry ?

Which answering service do you use for your Business ?

I am looking for one that supports API's (to send the call details to an API)

--

Izzy Goldstein

Chief Technology Officer

Main: (212) 477-1000 x 2085

Direct: (929) 477-2085

Website: www.telego.com<http://www.telego.net/>



Error! Filename not specified.<http://www.telego.com/>



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TeleGo Hosted PBX<https://youtu.be/DaT8YAZ4V0w>


--

Izzy Goldstein

Chief Technology Officer

Main: (212) 477-1000 x 2085

Direct: (929) 477-2085

Website: www.telego.com<http://www.telego.net/>



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Re: [VoiceOps] is there an Answering Service geared towards telecom industry ?

[Matthew Crocker]

Self Promotion:

I own an Answering Service (50+ years) which operates along side our ISP (25 
years) & ITSP.

Your system can access APIs via standard protocols and Tx/Rx information

What type of services are you looking for?


From: VoiceOps  on behalf of Izzy Goldstein - 
TeleGo 
Date: Tuesday, November 27, 2018 at 4:03 PM
To: "Voiceops.org" 
Subject: [VoiceOps] is there an Answering Service geared towards telecom 
industry ?

Which answering service do you use for your Business ?

I am looking for one that supports API's (to send the call details to an API)

--

Izzy Goldstein

Chief Technology Officer

Main: (212) 477-1000 x 2085

Direct: (929) 477-2085

Website: www.telego.com



[Image removed by sender.]



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any other legal right by email communications. Any such communication is 
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Re: [VoiceOps] Hotel Phone System

[Matthew Crocker]

Have you thought about hanging Analog phones off a large ATA (Adtran TA5000) or 
a stack of Adtran TA924s?   Buid a SIP trunk from Broadworks to the ATA and 
assign the TNs to the SIP trunk (Creating trunk users).  You can then assign 
Broadworks services to the users (voicemail, etc).   The SIP trunk user only 
uses up a license when it has an active call,  you can overcommit your licenses 
that way

Check out https://xchange.broadsoft.com/node/1034202

Section 4.4 covers licensing and the overcommit process.


From: VoiceOps  on behalf of Colton Conor 

Date: Saturday, March 30, 2019 at 10:14 AM
To: "voiceops@voiceops.org" 
Subject: [VoiceOps] Hotel Phone System

Anyone have recommendations on brands and models to deploy for a hotel? We use 
Broadsoft as our voip switch, but the though of using standard licenses for a 
100 room hotel would be expensive in monthly license cost alone. Hotel only 
wants 10 phone lines, so we are thinking about providing an onsite PBX with 10 
SIP trunks as the input.




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Re: [VoiceOps] International (Non-US/Canada) Termination Providers

[Matthew Crocker]

I recommend ThinQ for international.  Their LCR is pretty good and they have 
some nice fraud protection.

On Apr 5, 2019, at 10:17 AM, Ivan Kovacevic 
mailto:ivan.kovace...@startelecom.ca>> wrote:


Hello,

We have been struggling to find good international routes. We mostly handle 
contact centre traffic in NA and do not have much traffic ($500ish per month) 
internationally, but it has been exceedingly painful to manage.

We've tried a couple of carriers (Level3 - expensive and bad, TATA - cheaper 
but bad and will randomly block traffic, couple of CLECs who also offer A-Z).

Financially it's not worth the hassle and managing the providers, but our 
contact centre clients do need to be able to make non-contact centre 
international calls on occasion, so we have to support it.

Any suggestions?

[http://www.nuvoxx.ca/images/star-telecom.png]


Ivan Kovacevic

STAR TELECOM

www.startelecom.ca




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Re: [VoiceOps] Hotel Phone

[Matthew Crocker]

You will also want to notify hotel security when a 911 call is placed.The 
911 call still goes to the PSAP but a second connection goes to security so 
they can possibly respond.  It also helps when security knows what is going on 
as the police/ambulance show up at the front door.



From: VoiceOps  on behalf of Jason Kuylen 

Date: Thursday, April 4, 2019 at 12:10 PM
To: "VoiceOps@voiceops.org" 
Subject: Re: [VoiceOps] Hotel Phone

I’ve never heard of any other 911 requirement for hotel or business phones 
outside of not have to dial 9 to dial 911, ie 9911. Kari’s Law.

Can anyone provide a link with more information?

Thanks.

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Re: [VoiceOps] Request for Opinions: High density ATA's

[Matthew Crocker]

Years ago I did something similar (college instead of hospital) with 200+ 
analog lines using a Zhone MALC.   
Originally I MGCPed them back to a Vocaldata/Tekelec T6000/Genband M6 and then 
converted them to SIP on a Broadworks switch.

The Zhone MALC was a bit of a pain to configure but became cookie-cutter once 
we got it working.   
We installed it in a Zhone supplied outdoor cabinet in their very wet basement 
and included DC & batteries in a self contained system.

For cut over we did the following:

Existing Centrex lines from Verizon were terminated on a 'Centrex 66 block'
Existing College lines to the campus phones were terminated on a 'College 66 
block'
Existing cross connects from Centrex block to College Block

We established new 'Crocker 66 blocks' with 25 pair Amphenols to the Zhone 
(disconnected)
We moved the cross connect from the centrex block and connected to the Crocker 
block. One at a time, mapping numbers to Zhone config
We created a temporary cross connect from Crocker block to Centrex Block

On the day of the cut over we pulled all of the temporary cross connects to 
disconnect Verizon from the campus and plugged the Amphenols into the Zhone.
200+ phones cut over in under 10 minutes.

I think the MALC is EoL'd but the Zhone MXK is basically the same thing.

You can also do this with Adtran TA5000s. or you can stack a bunch of Adtran 
TA924s


On 3/21/19, 11:55 AM, "VoiceOps on behalf of Ryan Delgrosso" 
 wrote:

I have found myself with a number of hospital opportunities and 
servicing the staff with IP phones is a no-brainer, however there is the 
need for multi-hundred room connectivity for patient room phones and the 
staff mandate is to keep it analog because "ip phones there will grow 
legs".

I am looking for 24+ port density with amphenol connectors, and ideally 
some kind of rudimentary internal routing so i dont need to register all 
24 discreet ports and can route by some header (to or uri) within a 
single registration.

Right now im looking at AudioCodes and the Sangoma Vega series. Obihai 
would be my natural choice here but don't have anything that fits my 
density requirements.

Any opinions on these or others I should consider. Anyone deploy these 
and can speak to the experience?

Thanks in advance

-Ryan

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Re: [VoiceOps] Disclosing Restricted Caller ID to customer

[Matthew Crocker]

If they are an end-user the SIP INVITE should be sanitized by the providing 
carrier.

Carrier <-> Carrier include CallerID with privacy bits set
Carrier -> End-user include Sanitized CallerID (i.e. Anonymous or Private in 
the From header,  no Remote-Party-ID)



From: Faisal Imtiaz 
Date: Monday, September 9, 2019 at 2:02 PM
To: Victor C 
Cc: Matthew Crocker , Nick Olsen 
, "voiceops@voiceops.org" 

Subject: RE: [VoiceOps] Disclosing Restricted Caller ID to customer

Cool,
Looks like it is similar laws here in the USA too..
I stand corrected…

https://www.federalregister.gov/documents/2017/12/01/2017-25917/calling-number-identification-service-caller-id

Though I am curious about a scenario…
Client is running his own pbx, and running Homer ..
Would the Caller ID be visible to them in the homer trace ?

Regards

Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: supp...@snappytelecom.net

From: Victor C 
Sent: Monday, September 9, 2019 1:23 PM
To: Faisal Imtiaz 
Cc: Matthew Crocker ; Nick Olsen 
; voiceops@voiceops.org
Subject: Re: [VoiceOps] Disclosing Restricted Caller ID to customer

Can’t speak for US, but at least in Canada what you described wont fly.

You have obligations as a carrier to honour restricted caller id received from 
PSTN if the caller choose to withhold it. If your paying customer is not happy 
with a private incoming call, they should contact police as someone earlier 
suggested. If deemed necessary, police or court or whatever authority will 
reach to you for the private caller id.

If you just disclose caller id on your customers request as you described, you 
may just as well disregard rpid / whatever privacy flag you have from pstn all 
together. But people dont do that afaik.


On Sep 9, 2019, at 13:13, Faisal Imtiaz 
mailto:fai...@snappytelecom.net>> wrote:
Who is your customer ?  The Caller or the Called Entity ?

Your obligations are to your paying customer….. (which in this case is the 
Called Entity)
You have zero obligations to the caller….
If your client is asking for the information, and you have it, you may choose 
to provide it.
What your client does with it, is not your concern.
(Law enforcement overrides your agreement of keeping your clients information 
confidential )

My two cents !

Faisal Imtiaz
Snappy Internet & Telecom
http://www.snappytelecom.net

Tel: 305 663 5518 x 232

Help-desk: (305)663-5518 Option 2 or Email: 
supp...@snappytelecom.net<mailto:supp...@snappytelecom.net>

From: VoiceOps 
mailto:voiceops-boun...@voiceops.org>> On Behalf 
Of Matthew Crocker
Sent: Monday, September 9, 2019 1:02 PM
To: Nick Olsen 
mailto:n...@floridavirtualsolutions.com>>; 
voiceops@voiceops.org<mailto:voiceops@voiceops.org>
Subject: Re: [VoiceOps] Disclosing Restricted Caller ID to customer


You don’t know if it really is harassment.

Tell the customer you have the call details and will retain the data for 90 
days.  Have them call the police and open a case for harassment.  The police 
can get a subpoena and request the call data.

From: VoiceOps 
mailto:voiceops-boun...@voiceops.org>> on behalf 
of Nick Olsen 
mailto:n...@floridavirtualsolutions.com>>
Date: Monday, September 9, 2019 at 12:50 PM
To: "voiceops@voiceops.org<mailto:voiceops@voiceops.org>" 
mailto:voiceops@voiceops.org>>
Subject: [VoiceOps] Disclosing Restricted Caller ID to customer

Greetings all, Had an interesting case come up today that I wanted some 
feedback on.

Customer called claiming they had been receiving harassing calls to their 
business number, But the calls were caller ID blocked (Caller likely dialed *67 
before the call). I found the CDR's for the call in question, And sure enough 
"Anonymous"  was the displayed Calling number and CNAM.

Out of curiosity, I went and pulled the capture of the same call from Homer. 
And sure enough, The actual calling number is delivered in the Remote-Party-ID 
field, With Privacy=full.

Obviously, The caller asked for... and expected that data to be private. What's 
everyones thoughts on the legality of disclosing that information to my 
customer receiving the call? Would you provide it on request to the end user? 
Or limit that information only if requested by legal request? (Court order or 
request from law enforcement)

Nick Olsen
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
[Image removed by sender.]
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Re: [VoiceOps] Disclosing Restricted Caller ID to customer

[Matthew Crocker]

You don’t know if it really is harassment.

Tell the customer you have the call details and will retain the data for 90 
days.  Have them call the police and open a case for harassment.  The police 
can get a subpoena and request the call data.

From: VoiceOps  on behalf of Nick Olsen 

Date: Monday, September 9, 2019 at 12:50 PM
To: "voiceops@voiceops.org" 
Subject: [VoiceOps] Disclosing Restricted Caller ID to customer

Greetings all, Had an interesting case come up today that I wanted some 
feedback on.

Customer called claiming they had been receiving harassing calls to their 
business number, But the calls were caller ID blocked (Caller likely dialed *67 
before the call). I found the CDR's for the call in question, And sure enough 
"Anonymous"  was the displayed Calling number and CNAM.

Out of curiosity, I went and pulled the capture of the same call from Homer. 
And sure enough, The actual calling number is delivered in the Remote-Party-ID 
field, With Privacy=full.

Obviously, The caller asked for... and expected that data to be private. What's 
everyones thoughts on the legality of disclosing that information to my 
customer receiving the call? Would you provide it on request to the end user? 
Or limit that information only if requested by legal request? (Court order or 
request from law enforcement)

Nick Olsen
Network Engineer
Office: 321-408-5000 x103
Mobile: 321-794-0763
[Image removed by sender.]
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Re: [VoiceOps] Recommendation for End User Billing Service

[Matthew Crocker]

We use Rev.io which has worked well.  It handles our Broadworks provisioning 
and CDRs.  Also integrates with suretax and a bunch of other systems.

On Feb 21, 2020, at 6:58 PM, Jeff Waddell  
wrote:


Are you using Freshbooks Classic or their new platform?

We are on Classic (which I really like) but they are going to be retiring it 
and I'm beginning to come with terms that we may move. - We have been on there 
over 11 years...

On Fri, Feb 21, 2020 at 6:51 PM Gavin Henry 
mailto:ghe...@suretec.co.uk>> wrote:
https://killbill.io/

--
Kind Regards,

Gavin Henry.
Managing Director.

T +44 (0) 330 44 50 000
D +44 (0) 330 44 55 007
M +44 (0) 7930 323266
F +44 (0) 1224 824887
E ghe...@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

OpenPGP (GPG/PGP) Public Key: 0x8CFBA8E6 - Import from 
hkp://pool.subkeys.pgp.net
or http://www.suretecgroup.com/0x8CFBA8E6.gpg
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Re: [VoiceOps] Three Digit Numbers

[Matthew Crocker]

Intelliquent testing is on 933


From: VoiceOps  on behalf of Carlos Alvarez 

Date: Tuesday, March 24, 2020 at 11:21 AM
To: "voiceops@voiceops.org" 
Subject: Re: [VoiceOps] Three Digit Numbers

We're in the US, and yes 811 is the underground utility line, but I don't think 
any of our carriers will pass it.  I don't recall the details on who does what, 
but carriers like Intelliquent, Bandwidth, and thinQ all do 911 testing on 
711/811.

Yeah, I just tried 811 to Intelliquent and it read back my phone number.


On Tue, Mar 24, 2020 at 8:14 AM Mike Hammett 
mailto:voice...@ics-il.net>> wrote:
What country is this? I believe 811 is supposed to be a USA-wide number to call 
for locating utilities for digging projects.


-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com




From: "Carlos Alvarez" mailto:caalva...@gmail.com>>
To: voiceops@voiceops.org
Sent: Tuesday, March 24, 2020 10:10:37 AM
Subject: Re: [VoiceOps] Three Digit Numbers
We don't handle any others in a traditional way.  Well, 611 is actually in 
place, to our support line, but it has never once been used.  811 and 711 are 
used for 911 testing without a real 911 call, as carriers mostly use those for 
automated systems that return your 911 info.

On Tue, Mar 24, 2020 at 7:55 AM Mike Hammett 
mailto:voice...@ics-il.net>> wrote:
What three digit numbers are commonly in use and how are people routing them?

Obviously there's 911 and that has a whole routing ecosystem.
What about 811? 311? X11?


What other special numbers are people handling?


-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
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Re: [VoiceOps] Outsourcing STIR/SHAKEN Setup

[Matthew Crocker]

ClearIP by Transnexus

Get Outlook for iOS

From: VoiceOps  on behalf of Dovid Bender 

Sent: Monday, August 31, 2020 6:37:26 AM
To: Voiceops.org 
Subject: [VoiceOps] Outsourcing STIR/SHAKEN Setup

Hi,

Does anyone have a recommendation for a company that get us everything needed 
for STIR/SHAKEN setup? By setup I mean helping us file to get a cert etc. From 
the small research I have done there is a lot of fragmented information out 
there and it would be easier for us to pay someone else to do this then invest 
our own time to take care of this.

TIA.

Regards,

Dovid

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Re: [VoiceOps] Carrier Robocall Blocking

[Matthew Crocker]

ClearIP from TransNexus

https://transnexus.com/clearip/



From: VoiceOps  on behalf of Calvin Ellison 

Date: Tuesday, May 26, 2020 at 8:53 PM
To: "Voiceops.org" 
Subject: [VoiceOps] Carrier Robocall Blocking

What solutions are people using to stop robocalls at the carrier level? We're 
dealing with aggregator and international gateway traffic that sometimes needs 
aggressive filtering.


Regards,


Calvin Ellison
Senior Voice Operations Engineer
calvin.elli...@voxox.com
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Re: [VoiceOps] USF is 33.4% for 2Q2021

[Matthew Crocker]

I tell my customers to complain to their congress critter if they don’t like 
the 33% tax on the services.

It’s for the children after all….

From: VoiceOps  on behalf of Paul Timmins 

Date: Thursday, June 10, 2021 at 12:38 PM
To: voiceops@voiceops.org 
Subject: Re: [VoiceOps] USF is 33.4% for 2Q2021
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


On 6/10/21 6:23 AM, Alex Balashov wrote:
>
> Yeah, observing it as an outsider who is not a service provider, I'm a
> little shocked to say the least. It's hard to understand where that
> kind of money is supposed to come from with the margins in this business.
>
> -- Alex
>
Passthru fees to the end user, duh. There's nothing us telcos can't cram
on the bottom of the bill.

Customers are gonna be ticked, but what ya gonna do. It's a line item
now, and a line item later.

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Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data

[Matthew Crocker]

The acme/oracle way of doing Hosted NAT Traversal is to set the expire time 
down to 30 seconds and have the phones REGISTER every 30 seconds.   The SBC 
eats the registration so it doesn’t overload the switch.   If the CGN NAT drops 
the entry it gets recreated with the new registration in 30 seconds.

We have had very good results with the acme/oracle approach

From: VoiceOps  on behalf of Pete Mundy 

Date: Thursday, June 10, 2021 at 5:11 PM
To: VoiceOps 
Subject: Re: [VoiceOps] "Timeout" on VoIP call traversing Verizon data
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


Precisely. And those "NAT table entries" eventually time out. On CG-NAT they 
often time out aggressively; <60 seconds. Hence sending OPTIONS over SIP over 
UDP regularly keeps the NAT table entries refreshed and active and therefore 
the UDP 'connection' open. I've come across firewalls with 30 second timeouts, 
so we use 25 second keepalives (OPTIONS).

Pete

>> On 11/06/2021, at 8:24 AM, Alex Balashov  wrote:
>>
>> Not to muddy the waters here with needless pedantry, but:
>>
>> While UDP may be "connectionless", the only way UDP, and in particular, 
>> symmetric SIP signalling, can work through NAT is if a stateful firewall + 
>> NAT gateway has some awareness (that is, state) of UDP "flows", or groups of 
>> packets flowing between ports consistently in some kind of temporary logical 
>> association--one might say, the endpoints have a "connection" of sorts...
>>
>> -- Alex
>>
> On 6/10/21 4:07 PM, Peter Beckman wrote:
> u SIP here is UDP, no?
> There's no connection to close for UDP.
> The source port for UDP doesn't matter. It's not part of the whole
> conversation, unless your switch cares that all communications continue to
> come from the source port. It's connectionless.
> TCP 5060 isn't even listening on our switches.
> So, maybe you're doing SIP over TCP?
> On Thu, 10 Jun 2021, Mark Wiles wrote:
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Re: [VoiceOps] CSR BTN

[Matthew Crocker]

Depends on where the CSR is coming from.   If coming from an RBOC I’ve always 
used it as an inventory of TNs and features.  I’m mostly interested in the TNs 
so I ensure I don’t miss a number when porting.From non-RBOCs the CSR is 
essentially worthless as the information is incomplete or invalid.

From: VoiceOps  on behalf of Mike Hammett 

Date: Wednesday, June 2, 2021 at 9:14 AM
To: VoiceOps 
Subject: Re: [VoiceOps] CSR BTN
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.

I guess to back up to a more basic level...

What's the intent of the CSR?

I thought it was for the gaining carrier to verify customer information with 
the losing carrier that would be required to port a number before actually 
issuing the LSR\port request.


-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com




From: "Mike Hammett" 
To: "VoiceOps" 
Sent: Friday, May 28, 2021 9:51:14 PM
Subject: [VoiceOps] CSR BTN
Does a CSR have to have the specific number being ported or just any number on 
the account?

We're testing a third party network that we'll utilize in the short-term for 
out-of-area usage.
We ported some of our own numbers to our account at the third party.
The CSRs arrived with the main BTN on the account, not the number being ported.
Our porting person rejected the CSRs because they weren't for the number being 
ported. Of course she knew what number was being attempted because it was an 
internal operation, but normally she wouldn't.


It's my thought that the CSR is a customer information verification process. We 
wouldn't know the number actually being ported until the LSR came. Does the 
name, address, account number, TN, etc. match an account you have? If so, great.


Where can I find a good resource on the process and what's normally acceptable 
for rejections?


Of course I said, "Let me verify this before we do anything" and then she 
replied to the CSRs rejecting.  *sigh* I guess we'll really be testing all 
scenarios.


-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com



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Re: [VoiceOps] Bandwidth vs Inteliquent

[Matthew Crocker]

We use clearip.com for our fraud protection along with our Inteliquent trunks.  
ClearIP also handles our STIR/SHAKEN, LCR and CNAM

-Matt


From: VoiceOps  on behalf of Colton Conor 

Date: Wednesday, July 7, 2021 at 4:26 PM
To: Carlos Alvarez 
Cc: VoiceOps 
Subject: Re: [VoiceOps] Bandwidth vs Inteliquent
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.

Do either Bandwidth or Inteliquent have any fraud / overspending limits you can 
put in place? I have looked at something like thinQ in the past, and was 
impressed that their platform could put limits in place, like if international 
calls exceed $1,000 in a day, block all calls. I know this is more a function 
of the SBC / Softswitch, but a nice to have option.

On Wed, Jul 7, 2021 at 12:31 PM Carlos Alvarez 
mailto:caalva...@gmail.com>> wrote:
We moved all of our services off of Inteliquent.  The support was absolute 
garbage, and they constantly raised rates a little at a time.  We took 20% off 
our costs going to Bandwidth, and support is lightning fast.  Coverage is 
identical as far as we can see in the US.  Most of our term and origination are 
with Bandwidth, and a bit with thinQ for a variety of reasons.  I'm considering 
porting all that to Bandwidth.  They say it's dangerous to have all your eggs 
in one basket, but BW has simply been 100% for us in every measure.  We'd 
probably still use thinQ as backup term.


On Wed, Jul 7, 2021 at 6:17 AM Colton Conor 
mailto:colton.co...@gmail.com>> wrote:
If you had to choose only between these two providers for wholesale services, 
which would you choose and why? We are only considering these two as the 
softswitch we are planning on using (Netsapiens) has only built out Group MMS 
support for these two carriers APIs, and no one else.

I have used many services over the years that have utilized both of these 
carriers, but I have never had a direct relationship with them.

Which has a better portal, API, and company overall? What are the true 
differences?

Are there any resellers to consider where we would still have direct access to 
these carriers APIs?
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Re: [VoiceOps] Bandwidth vs Inteliquent

[Matthew Crocker]

I’m a small Broadworks hosted PBX with about 5k devices attached.   I use 
clearIP for all inbound and outbound calls. I’m running about $1100/month.  
Most of that is CNAM dips.   Pricing is based on a per call/function with a 
$500/month minimum

LCR, STIR/SHAKEN, anti-fraud, CNAM just to make a few functions.

Insanely easy to configure and integrate with our oracle SBCs.

On Jul 7, 2021, at 11:36 PM, Alex Balashov  wrote:



CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


No, but highly curated access to the Bandwidth API is common. I cannot confirm 
or deny that we do a lot of that, but I mean, I wouldn’t deny it per se.

—
Sent from mobile, with due apologies for brevity and errors.

On Jul 7, 2021, at 10:18 PM, Ross Tajvar  wrote:


> I don't know of any resellers of Bandwidth that would allow you direct
access to the Bandwidth API. The API is based on Account and you'd have
access to the full reseller account.

I believe Bandwidth allows subaccounts.

On Wed, Jul 7, 2021, 10:08 PM Peter Beckman 
mailto:beck...@angryox.com>> wrote:
I've been happy with Bandwidth. Started with them mid-2015. They handle
most of our inbound and outbound calling and SMS. APIs do what we need for
the most part (there are some limits in viewing their inventory that I'm
not a fan of), SMPP connections never die, tickets get addressed quickly,
at least when the solution is straightforward.

I haven't used Inteliquent directly, so I cannot speak to the true
differences.

I hear that Inteliquent wins for termination, and though I can get coverage
in most places for Bandwidth on-net and Level3 off-net through Bandwidth,
Inteliquent, with their acquisition of Onvoy/Voyant/Vitelity over the past
few years, has coverage in more ratecenters. It just depends if you care
about small rural switches or smaller cities.

I'm sure the APIs are probably similar, though Bandwidth has several SDKs
available, and I have not been able to find mention of Inteliquent SDKs on
the Internet, and definitely not in GitHub.

I don't know of any resellers of Bandwidth that would allow you direct
access to the Bandwidth API. The API is based on Account and you'd have
access to the full reseller account.

https://dev.bandwidth.com/

Beckman

On Wed, 7 Jul 2021, Colton Conor wrote:

> If you had to choose only between these two providers for wholesale
> services, which would you choose and why? We are only considering these two
> as the softswitch we are planning on using (Netsapiens) has only built out
> Group MMS support for these two carriers APIs, and no one else.
>
> I have used many services over the years that have utilized both of these
> carriers, but I have never had a direct relationship with them.
>
> Which has a better portal, API, and company overall? What are the true
> differences?
>
> Are there any resellers to consider where we would still have direct access
> to these carriers APIs?
>

---
Peter Beckman  Internet Guy
beck...@angryox.com 
http://www.angryox.com/
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Re: [VoiceOps] Vendor Pros\Cons

[Matthew Crocker]

We were a VocalData (aka Tekelec T6k, aka Genband M6, aka Broadsoft M6) shop 
that migrated to Broadworks.  We moved from Broadworks to Cisco PaaS 
(Broadworks) 4 years ago.   I just purchased Netsapiens with their fully 
managed solution and will be migrating customers away from Cisco PaaS to  
Netsapiens.

IMHO Cisco has ignored Broadworks since they purchased it, feature set is 
falling behind.   We are 100% SIP in/out with 0 TDM anywhere in the network.

We’ve had customers we have migrated 3 times already,  time to migrate them a 
4th (last??) time.



From: VoiceOps  on behalf of Peter Rad 

Date: Friday, December 17, 2021 at 3:25 PM
To: voiceops@voiceops.org 
Subject: Re: [VoiceOps] Vendor ProsCons
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.



Metaswitch will soon be running on Azure for Operators
Ribbon
Cisco/BSFT or talk to Averistar about running BroadWorks for you
Alianza
netsapiens (now CXDO)
2600Hz (Kazoo)
cloudya by NFON

Biscom
WLC
and a few smaller others


On 12/17/2021 2:15 PM, Mike Hammett wrote:
We've been running Metaswitch for 15 years and are looking to refresh what we 
have.

Who else operates at that tier? I'm talking to Ribbon. I'm hesitant to talk to 
to Broadsoft because they're part of Cisco, but if I should, I should...

Obviously, there are rather limited venues to gain third-party feedback on the 
platforms, so I figured I'd ask here.


Thanks.



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Mike Hammett
Intelligent Computing Solutions
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Re: [VoiceOps] Misrouting 911 Calls?

[Matthew Crocker]

Get a small SIP trunk from another provider.  Order 1 DID from that provider 
for each physical office location.  Register appropriate 911 information for 
the DID.  Configure FreePBX to rewrite the CallerID to the appropriate DID for 
outbound 911 calls.  You may be able to get new DIDs from Comcast for the sole 
purpose of 911 and configure outbound 911 calls to those DIDs.



From: VoiceOps  on behalf of Aaron C. de Bruyn 
via VoiceOps 
Date: Tuesday, January 4, 2022 at 1:41 PM
To: voiceops@voiceops.org 
Subject: [VoiceOps] Misrouting 911 Calls?
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.

One of my clients has a large SIP trunk with Comcast based out of Washington 
State.

They have all their offices across Oregon and Washington hooked into a FreePBX 
phone server that is attached to the Comcast SIP trunk.

911 calls *constantly* get misrouted to the local PSAP where the SIP trunk 
lives.

I must have called Comcast 30 times over the last few years to try and get this 
addressed, but Comcast flat-out refuses to fix the issue.

The short answer is that Comcast refuses to fix it.  In some (but not all) 
cases, our phone numbers are RCF'd numbers, so they don't actually exist on the 
trunk...and Comcast forcibly re-writes them to our 'main' number...and then 
routes the 911 call incorrectly.  In other cases, we have provided Comcast with 
the e911 information, they say it's updated, and then we find out months later 
(when an office dials 911 during an emergency) that it's still not correct.

Not only does this affect 911 calls, but also customers who get the re-written 
caller ID and have no idea which office called them.

The "easy" solution is to ditch Comcast and move to a provider that doesn't 
play the RCF and caller-ID-rewrite games.  Unfortunately my client is locked 
into their Comcast contract for another ~18 months.  Early termination would 
incur a ~$35,000 bill.

Is there a list of PSAP numbers somewhere so I can set up an internal redirect 
to the PSAP 10-digit number?  I know those 10-digit numbers are guarded like 
Fort Knox, so I'm betting this option isn't very realistic.

Maybe a separate service provider that can just handle 911 calls without 
"owning" my client's phone numbers?

Any other thoughts on how I can route around Comcast brain damage?

Thanks,

-A
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Re: [VoiceOps] SS7 Landscape

[Matthew Crocker]

Can you order SS7 links directly to the tandem (f-links?).  Should just be a 
internal cross connect in the building. Then you build ISUP routes for the 
tandem point code over the direct links with backups over your current A- links

On Jan 25, 2022, at 1:15 PM, Mike Hammett  wrote:



CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


Oh, sure. Pros and cons to everything.



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Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
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From: "Mary Lou Carey" 
To: "Mike Hammett" 
Cc: "voiceops" 
Sent: Tuesday, January 25, 2022 11:59:14 AM
Subject: Re: [VoiceOps] SS7 Landscape

There are pros and cons to everything. It sounds like in your case you
may just want to keep your ISUP trunks and SS7 links, but maybe look at
another provider for SS7. Syniverse is also an option but I'm not sure
if their service has gone downhill like TNS' apparently has.

MARY LOU CAREY
BackUP Telecom Consulting
Office: 615-791-9969
Cell: 615-796-

On 2022-01-25 10:38 AM, Mike Hammett wrote:
> Right, but as I said earlier, I'm already in the same building as the
> tandem, so I'm just adding points of failure for moving an existing
> operation somewhere else.
>
> -
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>
> Midwest Internet Exchange
> http://www.midwest-ix.com
>
> -
>
> From: "Mary Lou Carey" 
> To: "Mike Hammett" 
> Cc: "Paul Timmins" , "voiceops"
> 
> Sent: Monday, January 24, 2022 10:42:22 PM
> Subject: Re: [VoiceOps] SS7 Landscape
>
> When you use a PSTN connection provider you route your traffic to them
>
> via SIP and they handle all the SS7 so you can eliminate your direct
> LIS
> trunks with the LEC and your SS7 Links.
>
> MARY LOU CAREY
> BackUP Telecom Consulting
> Office: 615-791-9969
> Cell: 615-796-
>
> On 2022-01-23 05:33 AM, Mike Hammett wrote:
>> Well right, but wouldn't I still need all of the same stuff (perhaps
> a
>> few less trunks to specific switches, only now I have fewer minutes
> to
>> spread the costs over?
>>
>> -
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>> Midwest Internet Exchange
>> http://www.midwest-ix.com
>>
>> -
>>
>> From: "Paul Timmins" 
>> To: "Mike Hammett" 
>> Cc: "voiceops" 
>> Sent: Sunday, January 23, 2022 12:01:01 AM
>> Subject: Re: [VoiceOps] SS7 Landscape
>>
>> Even without IPES you can switch your feature group D to voip
> through
>> these aggregators and get rid of basically everything that isn't the
>> local LEC's legacy network.
>>
>>> On Jan 22, 2022, at 7:03 PM, Mike Hammett 
>>> wrote:
>>>
>>> Sure, but then I just transfer the problem to someone else, until
>>> the traditional PSTN goes away or modernizes. I am working on IPES
>>> for expansion markets, though.
>>>
>>> Also, for my home LATA, I'm needing to be in the same building as
>>> the primary tandem, so it's kind of a selling point for large
>>> customers in that area that I would have fewer opportunities for
>>> failures. I don't have to go to Chicago and back to get to the
>>> operator in the next row of racks over. Though, I suppose my SS7
>>> diversity and availability is the weakness, not the actual call
>>> path.
>>>
>>> -
>>> Mike Hammett
>>> Intelligent Computing Solutions
>>> http://www.ics-il.com [1]
>>>
>>> Midwest Internet Exchange
>>> http://www.midwest-ix.com [2]
>>>
>>> -
>>>
>>> From: "Paul Timmins" 
>>> To: "Mike Hammett" 
>>> Cc: "voiceops" 
>>> Sent: Saturday, January 22, 2022 1:46:46 PM
>>> Subject: Re: [VoiceOps] SS7 Landscape
>>>
>>> Reduce your reliance on it more and more. Switch to IP tandems
>>> (Inteliquent/Peerless/etc) and ip interconnect. The people who make
>>> and support SS7 equipment are in a dying market, the brain drain on
>>> it is immense as people retire, and the longer you rely on it, the
>>> worse it's gonna get.
>>>
>>> TNS is a big dog in the space. But that's not saying much anymore.
>>>
 On Jan 22, 2022, at 10:00 AM, Mike Hammett 
 wrote:

 We currently get our SS7 via TNS. I (maybe incorrectly) understand
 them to be a big dog in that space.

 I've had two major problems with them in six months.

 What are my alternatives? Are they less pleasant?

 -
 Mike Hammett
 Intelligent Computing Solutions
 http://www.ics-il.com [1]

 Midwest Internet Exchange
 http://www.midwest-ix.com [2]

 ___
 VoiceOps mailing list
 VoiceOps@voiceops.org
 https://puck.nether.net/mailman/listinfo/voiceops
>>
>>
>>
>> Links:
>> --
>> [1] http://www.ics-il.com/
>> [2] http://www.midwest-ix.com/
>> ___
>> VoiceOps mailing list
>> 

Re: [VoiceOps] Identity Header Test Tool

[Matthew Crocker]

My call to the Sinch TFN resulted in a valid Identity header.  Calls to other 
TFNs received no identity header.  Calls to Sinch 1-425 ## answer & disconnect 
immediately

Thanks for setting this up  


From: VoiceOps  on behalf of David Frankel 

Date: Sunday, July 3, 2022 at 11:13 AM
To: voiceops@voiceops.org 
Subject: [VoiceOps] Identity Header Test Tool
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


Last week I was forwarded a note from this list regarding tools to test and
debug SHAKEN Identity headers. That prompted us to stitch together some
modules we already had in an attempt to help.

What we have is at http://identity.legalcallsonly.org. You can call one of
the test numbers listed on that page, and if we receive your header, we'll
read you a six-digit code. Disconnect and then plug the code into the box on
the web form, and we'll show you details of that Identity header.

Perhaps most importantly, you'll be able to see if the header we received is
the one you sent. In addition, we parse the header and try to tell you if it
is correctly formatted and valid.

Currently we have a couple of geographic DIDs and three toll-free numbers
(each using different underlying providers). So far we aren't having a lot
of success getting the Identity headers on the TFNs; we're working to
improve that.

Suggestions welcome. We hope the tool provokes more discussion about best
practices regarding making the Authentication Framework as functional and
useful as possible.

Happy 4th of July!

David Frankel
ZipDX LLC
St. George, UT USA


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[VoiceOps] Establishing a new LRN

[Matthew Crocker]

Hello all,

What is the process for establishing a new LRN in a LATA?   We have 3 codes 
assigned to us for LATA 126 (413-200, 413-654, 413-370).   I currently have an 
LRN 413-654-1234 assign to a switch.  I need a new LRN (413-200-1234 ???) 
assigned to a new switch.   All of my interconnect stuff is handled via 
Inteliquent, they can route the calls to my codes to the appropriate switch.  I 
have all of the SIP trunks built for Inteliquent.

I need to migrate customers from one switch to another.  My plan is to 
establish a new LRN on an existing code (413-200-1234 for example).  Have 
Inteliquent route the code to my new switch.  Then I can update the porting 
database for customers to the new LRN as they are migrated.

I believe I need to update the LERG but I don’t know how to do that

Thanks

-Matt

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Re: [VoiceOps] Carriers / TCR requiring NetNumber ID for SMS, even P2P

[Matthew Crocker]

www.campaignregistry.com   The process is 
quick,painless and a campaign costs a small one time fee ($4 I think)


From: VoiceOps  on behalf of Peter Beckman 

Date: Wednesday, May 4, 2022 at 5:09 PM
To: VoiceOps 
Subject: [VoiceOps] Carriers / TCR requiring NetNumber ID for SMS, even P2P
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


I'm hearing from one of my carriers that in order to continue sending P2P
(NOT A2P) SMS from our DIDs, my $dayjob needs to purchase and maintain a
NetNumber ID to use as the SPID for SMS.

I'm also hearing that ALT-SPIDs are being discontinued.

Does anyone else know about this change, who is driving it, and why? Anyone
want to share costs involved in getting and maintaining an NNID for this
purpose?

This feels more like pay-to-play than a solution to reduce unwanted SMS.

Beckman
---
Peter Beckman  Internet Guy
beck...@angryox.comhttps://www.angryox.com/
---
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Re: [VoiceOps] Carriers / TCR requiring NetNumber ID for SMS, even P2P

[Matthew Crocker]

Well, this is all new to me so I apologize for misleading or incorrect 
information.   I’m going through the CSP process now and it has been pretty 
quick/easy.   I’ve registered one brand ($4) and one campaign ($2/month).

I’m using Inteliquent as my interconnect, they charge fees per SMS/MMS.  I 
don’t believe I will be receiving invoices from individual carriers (ala CABS)

From: Peter Beckman 
Date: Wednesday, May 4, 2022 at 7:08 PM
To: Matthew Crocker 
Cc: VoiceOps 
Subject: Re: [VoiceOps] Carriers / TCR requiring NetNumber ID for SMS, even P2P
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


Your statement is misleading, borderline wrong.

It's $200 just to sign up with TCR.

Then it is a variable fee based on how many DIDs you are attempting to
register.

 Low Volume: $2 per MONTH per DID (non-time-sensitive)

Undisclosed how much it costs if NOT Low Volume.

Then carriers can add on their own per-DID and per-campaign fees.

And this is ONLY for 10DLC A2P SMS -- BUSINESS or BULK SMS, not P2P,
person-to-person SMS.

Look at the pricing Commio is putting out regarding T-Mobile:
https://www.thinq.com/blog/t-mobile-10dlc-text-messaging-fee-changes/

 - $50 Campaign Service Activation Fee
 - $50 Campaign Migration Fee
 - $2,000 one-time NNID Registration
 - $10,000 pass-through fee if T-Mobile doesn't like your SMS
 - $1,000 pass-through fee if they think you're evading the rules

Bandwidth increasing per-SMS fees too:
https://support.bandwidth.com/hc/en-us/articles/152422242-10DLC-Overview

Can you imagine your cost basis as a small carrier, to increase your direct
costs of a DID by 10x or more, just to allow your customer to send and
receive SMS messages?



Please note:
 I am discussing P2P -- PERSON-TO-PERSON -- NOT A2P, BUSINESS-TO-PERSON

See the CTIA Best Practices for the difference between the content of the
two.

Beckman

On Wed, 4 May 2022, Matthew Crocker wrote:

>
> www.campaignregistry.com<http://www.campaignregistry.com><http://www.campaignregistry.com%3chttp:/www.campaignregistry.com%3e>
>The process is quick,painless and a campaign costs a small one time fee 
> ($4 I think)
>
>
> From: VoiceOps  on behalf of Peter Beckman 
> 
> Date: Wednesday, May 4, 2022 at 5:09 PM
> To: VoiceOps 
> Subject: [VoiceOps] Carriers / TCR requiring NetNumber ID for SMS, even P2P
> CAUTION: This email originated from outside of Crocker. Do not click links or 
> open attachments unless you recognize the sender and know the content is safe.
>
>
> I'm hearing from one of my carriers that in order to continue sending P2P
> (NOT A2P) SMS from our DIDs, my $dayjob needs to purchase and maintain a
> NetNumber ID to use as the SPID for SMS.
>
> I'm also hearing that ALT-SPIDs are being discontinued.
>
> Does anyone else know about this change, who is driving it, and why? Anyone
> want to share costs involved in getting and maintaining an NNID for this
> purpose?
>
> This feels more like pay-to-play than a solution to reduce unwanted SMS.
>
> Beckman
> ---
> Peter Beckman  Internet Guy
> beck...@angryox.comhttps://www.angryox.com/
> ---
> ___
> VoiceOps mailing list
> VoiceOps@voiceops.org
> https://puck.nether.net/mailman/listinfo/voiceops
>

---
Peter Beckman  Internet Guy
beck...@angryox.comhttps://www.angryox.com/
---
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Re: [VoiceOps] Voice Peering

[Matthew Crocker via VoiceOps]

With STIR/SHAKEN (in theory) all calls will be signed, authenticated so you can 
trace the originating carrier.  In an open peering environment you can use it 
to accept/reject calls

Open SIP proxy handles all of the SIP traffic,  RTP goes directly between 
carriers.
All calls originated must be signed (STIRred)

  *   Call isn’t signed, gets rejected by the SIP peering proxy
Terminating carrier can validate the signed calls (SHAKEN)

  *   Don’t like the signing CA?  reject the call
  *   Don’t like the signing carrier? Reject the call
  *   Carrier sending too many spam calls,  adjust treatment based on customer 
spam settings


Routing is handled between terminating carrier and SIP peering proxy.  
Originating carrier sends all calls to peering proxy first, if proxy doesn’t 
have the route it sends a 4XX error back and originating carrier can continue 
routing on other paths.

So terminating carriers would need to export/upload (hacked BGP?) numbers they 
are willing to receive calls on to the peering proxy.

Proxies can be spun up in various AWS/Azure/GoogleCloud VPS


From: Pinchas Neiman 
Date: Wednesday, October 25, 2023 at 11:18 AM
To: Jawaid Bazyar 
Cc: Matthew Crocker , voiceops 
Subject: Re: [VoiceOps] Voice Peering
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.

By reading the RFCs I was able to grasp 75% of it, it's well written and covers 
your clear constraint, at least on how to verify the SIP header comes from a 
trustworthy authority (If you agree on the root authority)
Practically implementing STIR/SHAKEN has bureaucracy involved.

On Tue, Oct 24, 2023 at 9:38 PM Jawaid Bazyar via VoiceOps 
mailto:voiceops@voiceops.org>> wrote:
Is there a good clear document somewhere describing how STIR/SHAKEN is supposed 
to work?

On Tue, Oct 24, 2023 at 9:33 PM Matthew Crocker via VoiceOps 
mailto:voiceops@voiceops.org>> wrote:


> On Oct 24, 2023, at 9:13 PM, Peter Beckman via VoiceOps 
> mailto:voiceops@voiceops.org>> wrote:
>
> CAUTION: This email originated from outside of Crocker. Do not click links 
> or open attachments unless you recognize the sender and know the content is 
> safe.
>
>
> The challenge is how do you authenticate the end "carrier" or service
> provider?
>

STIR/SHAKEN


> Sure, anyone who leases numbers directly from NANPA can look up the carrier
> of record and exchange traffic directly, but any business who also leases
> numbers INDIRECTLY gets cut out and still needs to pay their upstream
> carrier(s) to place/receive calls, either by channels or per minute, even
> if their upstream is directly peered and not transiting the PSTN at all.
>
> If this would be for the end user, then NANPA would have to delegate to the
> leasee, the leasee delegate to the reseller, the reseller to the end user,
> then the end user could publish their VoIP contact info, and anyone could
> call directly via VoIP, cutting out all of the middle peers.
>
> But, as another person said, this is ripe for abuse, and with no motivation
> by NANPA or the larger carriers to make calls less expensive for the
> reseller or end user, I see this going nowhere. Until there is some value
> in NANPA (plus all the other country telephony organizations) and the
> direct carriers leasing numbers to do so.
>
> Beckman
>
>> On Tue, 24 Oct 2023, Ross Tajvar via VoiceOps wrote:
>>
>> I can think of a few ways that could be adapted into a platform more like
>> an Internet exchange, but as others have said, it just doesn't seem worth
>> it.
>>
>> On Tue, Oct 24, 2023, 5:31 PM Jawaid Bazyar via VoiceOps <
>> voiceops@voiceops.org<mailto:voiceops@voiceops.org>> wrote:
>>
>>> I think schemes like DUNDI (and some of the others mentioned here) suffer
>>> from a trust issue – what’s to prevent operator X from poisoning the
>>> protocol with bogus “stolen” numbers?
>>>
>>>
>>>
>>> On Tue, Oct 24, 2023 at 5:25 PM Jared Smith via VoiceOps <
>>> voiceops@voiceops.org<mailto:voiceops@voiceops.org>> wrote:
>>>
>>>> On Tue, Oct 24, 2023 at 8:49 AM Mike Hammett via VoiceOps <
>>>> voiceops@voiceops.org<mailto:voiceops@voiceops.org>> wrote:
>>>>
>>>>> This was in another thread, but I broke it out into it's own
>>>>> conversation. Someone had asked:
>>>>>
>>>>> ---
>>>>> I am joining this thread late, but, would anyone out there be interested
>>>>> in exchanging traffic with other carriers directly over SIP?
>>>>>
>>>>
>>>> Just another point

Re: [VoiceOps] Voice Peering

[Matthew Crocker via VoiceOps]

I never said STIR/SHAKEN would be used to ‘look up’ for call routing.

Earlier someone mentioned an issue with open peering is spam calls.  
STIR/SHAKEN can solve that issue.

You can certainly use STIR/SHAKEN to reject calls from $COMPANY once you have 
determined you don’t like $COMPANY.  That can easily be done off line by CDR 
analysis.  Sure you let a couple dozen calls in but you can pretty quickly find 
‘$BAD_COMPANY’ and start rejecting their calls.   The system would settle our 
fairly quickly

From: Peter Beckman 
Date: Wednesday, October 25, 2023 at 12:04 PM
To: Matthew Crocker 
Cc: Pinchas Neiman , Jawaid Bazyar 
, voiceops 
Subject: Re: [VoiceOps] Voice Peering
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


STIR/SHAKEN does not delegate any authority to anyone.

It merely allows me to sign a call that I originate, so that someone else
can say "Oh this came from $COMPANY."

Besides, STIR/SHAKEN is done at the time of an origination call, it cannot
be "looked up" to see where to route a call.

The suggestion that STIR/SHAKEN could be used to authoritatively assign a
DID endpoint to someone demonstrates a lack of understanding in how it
works and what it does and does not do.

Beckman

On Wed, 25 Oct 2023, Matthew Crocker via VoiceOps wrote:

>
> With STIR/SHAKEN (in theory) all calls will be signed, authenticated so you 
> can trace the originating carrier.  In an open peering environment you can 
> use it to accept/reject calls
>
> Open SIP proxy handles all of the SIP traffic,  RTP goes directly between 
> carriers.
> All calls originated must be signed (STIRred)
>
>  *   Call isn’t signed, gets rejected by the SIP peering proxy
> Terminating carrier can validate the signed calls (SHAKEN)
>
>  *   Don’t like the signing CA?  reject the call
>  *   Don’t like the signing carrier? Reject the call
>  *   Carrier sending too many spam calls,  adjust treatment based on customer 
> spam settings
>
>
> Routing is handled between terminating carrier and SIP peering proxy.  
> Originating carrier sends all calls to peering proxy first, if proxy doesn’t 
> have the route it sends a 4XX error back and originating carrier can continue 
> routing on other paths.
>
> So terminating carriers would need to export/upload (hacked BGP?) numbers 
> they are willing to receive calls on to the peering proxy.
>
> Proxies can be spun up in various AWS/Azure/GoogleCloud VPS
>
>
> From: Pinchas Neiman 
> Date: Wednesday, October 25, 2023 at 11:18 AM
> To: Jawaid Bazyar 
> Cc: Matthew Crocker , voiceops 
> 
> Subject: Re: [VoiceOps] Voice Peering
> CAUTION: This email originated from outside of Crocker. Do not click links or 
> open attachments unless you recognize the sender and know the content is safe.
>
> By reading the RFCs I was able to grasp 75% of it, it's well written and 
> covers your clear constraint, at least on how to verify the SIP header comes 
> from a trustworthy authority (If you agree on the root authority)
> Practically implementing STIR/SHAKEN has bureaucracy involved.
>
> On Tue, Oct 24, 2023 at 9:38 PM Jawaid Bazyar via VoiceOps 
> mailto:voiceops@voiceops.org>> wrote:
> Is there a good clear document somewhere describing how STIR/SHAKEN is 
> supposed to work?
>
> On Tue, Oct 24, 2023 at 9:33 PM Matthew Crocker via VoiceOps 
> mailto:voiceops@voiceops.org>> wrote:
>
>
>> On Oct 24, 2023, at 9:13 PM, Peter Beckman via VoiceOps 
>> mailto:voiceops@voiceops.org>> wrote:
>>
>> CAUTION: This email originated from outside of Crocker. Do not click links 
>> or open attachments unless you recognize the sender and know the content is 
>> safe.
>>
>>
>> The challenge is how do you authenticate the end "carrier" or service
>> provider?
>>
>
> STIR/SHAKEN
>
>
>> Sure, anyone who leases numbers directly from NANPA can look up the carrier
>> of record and exchange traffic directly, but any business who also leases
>> numbers INDIRECTLY gets cut out and still needs to pay their upstream
>> carrier(s) to place/receive calls, either by channels or per minute, even
>> if their upstream is directly peered and not transiting the PSTN at all.
>>
>> If this would be for the end user, then NANPA would have to delegate to the
>> leasee, the leasee delegate to the reseller, the reseller to the end user,
>> then the end user could publish their VoIP contact info, and anyone could
>> call directly via VoIP, cutting out all of the middle peers.
>>
>> But, as another person said, this is ripe for abuse, and with no motivation
>> by NANPA or the larger

Re: [VoiceOps] Voice Peering

[Matthew Crocker via VoiceOps]

A solution (where to send the call) was offered (open peering) but then 
devolved into ‘how to you stop spam’ and I offered STIR/SHAKEN.   There have 
been plenty of open routing solutions thrown about over the past 30 years, none 
have ever taken hold.

From: Peter Beckman 
Date: Wednesday, October 25, 2023 at 5:08 PM
To: Matthew Crocker 
Cc: Pinchas Neiman , Jawaid Bazyar 
, VoiceOps 
Subject: Re: [VoiceOps] Voice Peering
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


This whole conversation, and topic, is "Voice Peering" -- ORIGINATING CALLS
directly to the endpoint rather than me passing to Level3 who passes to IQ
who passes to their customer who passes to their customer who has a VoIP
device that I could reach directly if I only had the ability to do so.

This has nothing to do with rejecting incoming calls signed with
STIR/SHAKEN.

The call cannot start until I know where to send the call. <-- problem that we 
are discussing

Beckman

On Wed, 25 Oct 2023, Matthew Crocker wrote:

>
> I never said STIR/SHAKEN would be used to ‘look up’ for call routing.
>
> Earlier someone mentioned an issue with open peering is spam calls.  
> STIR/SHAKEN can solve that issue.
>
> You can certainly use STIR/SHAKEN to reject calls from $COMPANY once you have 
> determined you don’t like $COMPANY.  That can easily be done off line by CDR 
> analysis.  Sure you let a couple dozen calls in but you can pretty quickly 
> find ‘$BAD_COMPANY’ and start rejecting their calls.   The system would 
> settle our fairly quickly
>
> From: Peter Beckman 
> Date: Wednesday, October 25, 2023 at 12:04 PM
> To: Matthew Crocker 
> Cc: Pinchas Neiman , Jawaid Bazyar 
> , voiceops 
> Subject: Re: [VoiceOps] Voice Peering
> CAUTION: This email originated from outside of Crocker. Do not click links or 
> open attachments unless you recognize the sender and know the content is safe.
>
>
> STIR/SHAKEN does not delegate any authority to anyone.
>
> It merely allows me to sign a call that I originate, so that someone else
> can say "Oh this came from $COMPANY."
>
> Besides, STIR/SHAKEN is done at the time of an origination call, it cannot
> be "looked up" to see where to route a call.
>
> The suggestion that STIR/SHAKEN could be used to authoritatively assign a
> DID endpoint to someone demonstrates a lack of understanding in how it
> works and what it does and does not do.
>
> Beckman
>
> On Wed, 25 Oct 2023, Matthew Crocker via VoiceOps wrote:
>
>>
>> With STIR/SHAKEN (in theory) all calls will be signed, authenticated so you 
>> can trace the originating carrier.  In an open peering environment you can 
>> use it to accept/reject calls
>>
>> Open SIP proxy handles all of the SIP traffic,  RTP goes directly between 
>> carriers.
>> All calls originated must be signed (STIRred)
>>
>>  *   Call isn’t signed, gets rejected by the SIP peering proxy
>> Terminating carrier can validate the signed calls (SHAKEN)
>>
>>  *   Don’t like the signing CA?  reject the call
>>  *   Don’t like the signing carrier? Reject the call
>>  *   Carrier sending too many spam calls,  adjust treatment based on 
>> customer spam settings
>>
>>
>> Routing is handled between terminating carrier and SIP peering proxy.  
>> Originating carrier sends all calls to peering proxy first, if proxy doesn’t 
>> have the route it sends a 4XX error back and originating carrier can 
>> continue routing on other paths.
>>
>> So terminating carriers would need to export/upload (hacked BGP?) numbers 
>> they are willing to receive calls on to the peering proxy.
>>
>> Proxies can be spun up in various AWS/Azure/GoogleCloud VPS
>>
>>
>> From: Pinchas Neiman 
>> Date: Wednesday, October 25, 2023 at 11:18 AM
>> To: Jawaid Bazyar 
>> Cc: Matthew Crocker , voiceops 
>> 
>> Subject: Re: [VoiceOps] Voice Peering
>> CAUTION: This email originated from outside of Crocker. Do not click links 
>> or open attachments unless you recognize the sender and know the content is 
>> safe.
>>
>> By reading the RFCs I was able to grasp 75% of it, it's well written and 
>> covers your clear constraint, at least on how to verify the SIP header comes 
>> from a trustworthy authority (If you agree on the root authority)
>> Practically implementing STIR/SHAKEN has bureaucracy involved.
>>
>> On Tue, Oct 24, 2023 at 9:38 PM Jawaid Bazyar via VoiceOps 
>> mailto:voiceops@voiceops.org>> wrote:
>> Is there a good clear document somewhere describing how STIR/SHAKEN is 
>&

Re: [VoiceOps] Voice Peering

[Matthew Crocker via VoiceOps]

> On Oct 24, 2023, at 9:13 PM, Peter Beckman via VoiceOps 
>  wrote:
> 
> CAUTION: This email originated from outside of Crocker. Do not click links 
> or open attachments unless you recognize the sender and know the content is 
> safe.
> 
> 
> The challenge is how do you authenticate the end "carrier" or service
> provider?
> 

STIR/SHAKEN


> Sure, anyone who leases numbers directly from NANPA can look up the carrier
> of record and exchange traffic directly, but any business who also leases
> numbers INDIRECTLY gets cut out and still needs to pay their upstream
> carrier(s) to place/receive calls, either by channels or per minute, even
> if their upstream is directly peered and not transiting the PSTN at all.
> 
> If this would be for the end user, then NANPA would have to delegate to the
> leasee, the leasee delegate to the reseller, the reseller to the end user,
> then the end user could publish their VoIP contact info, and anyone could
> call directly via VoIP, cutting out all of the middle peers.
> 
> But, as another person said, this is ripe for abuse, and with no motivation
> by NANPA or the larger carriers to make calls less expensive for the
> reseller or end user, I see this going nowhere. Until there is some value
> in NANPA (plus all the other country telephony organizations) and the
> direct carriers leasing numbers to do so.
> 
> Beckman
> 
>> On Tue, 24 Oct 2023, Ross Tajvar via VoiceOps wrote:
>> 
>> I can think of a few ways that could be adapted into a platform more like
>> an Internet exchange, but as others have said, it just doesn't seem worth
>> it.
>> 
>> On Tue, Oct 24, 2023, 5:31 PM Jawaid Bazyar via VoiceOps <
>> voiceops@voiceops.org> wrote:
>> 
>>> I think schemes like DUNDI (and some of the others mentioned here) suffer
>>> from a trust issue – what’s to prevent operator X from poisoning the
>>> protocol with bogus “stolen” numbers?
>>> 
>>> 
>>> 
>>> On Tue, Oct 24, 2023 at 5:25 PM Jared Smith via VoiceOps <
>>> voiceops@voiceops.org> wrote:
>>> 
 On Tue, Oct 24, 2023 at 8:49 AM Mike Hammett via VoiceOps <
 voiceops@voiceops.org> wrote:
 
> This was in another thread, but I broke it out into it's own
> conversation. Someone had asked:
> 
> ---
> I am joining this thread late, but, would anyone out there be interested
> in exchanging traffic with other carriers directly over SIP?
> 
 
 Just another point of VoIP history trivia at this point... but in
 addition to things like ENUM and ITAD, Mark Spencer of Asterisk fame also
 invented Dundi, which was an encrypted peer-to-peer protocol for route
 advertisement and discovery.  As far as I know, very few people besides me
 ever put it in production, but it worked really well at the time. (Of
 course, it's been about 17 or 18 years now since I used it in production.)
 
 -Jared
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 VoiceOps@voiceops.org
 https://puck.nether.net/mailman/listinfo/voiceops
 
>>> ___
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>>> 
>> 
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Re: [VoiceOps] New Hosted PBX Platform?

[Matthew Crocker via VoiceOps]

Netsapiens.   We moved from Broadworks. Customers love it

On Feb 17, 2023, at 7:35 PM, Mike Hammett via VoiceOps  
wrote:



CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.


We have been using 3CX for our hosted PBX platform. It's come time for us to 
find something else. I chose 3CX at the time because it had the best 
user-facing UI.

I've had a lot of platforms suggested to me by other ISPs, but I thought I'd 
come here to see what you all have to say.

3CX didn't scale well, but that was secondary to having a product that 
customers found easy to learn and easy to use.

We currently use Metaswitch and are considering adding their Max UC, but have 
open minds at this time.



-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com



Midwest Internet Exchange
http://www.midwest-ix.com



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Re: [VoiceOps] Outbound Calls being marked as SPAM

[Matthew Crocker via VoiceOps]

I just verified the outbound calls are going out with 11 digit To & From 
headers,  11 digit TN in the STI Auth Token,  Attest ‘A’.

Trying to work through out contacts at Comcast to get this resolved.

Thanks again

-Matt


From: Mark R Lindsey 
Date: Wednesday, April 5, 2023 at 4:56 PM
To: Matthew Yaklin 
Cc: Matthew Crocker , voiceops@voiceops.org 

Subject: Re: [VoiceOps] Outbound Calls being marked as SPAM
CAUTION: This email originated from outside of Crocker. Do not click links or 
open attachments unless you recognize the sender and know the content is safe.

Matt, That's really helpful. The ATIS NNI documents functionally set the 
standard for many of these matters, and number formatting ("canonicalization") 
is one of the big ones. You can find some details on the canonicalization 
algorithm in some of their docs:
https://access.atis.org/apps/group_public/download.php/67436/ATIS-174.v003.pdf
https://access.atis.org/apps/group_public/download.php/63572/IPNNI-2022-9R000.docx

The number formatting is summarized several places like this:
...treat the calling TN as if it were an E.164 number; i.e., canonicalize the 
calling TN to remove any leading “+” sign or visual separators (i.e., “.”, “-”, 
“(”, and “)”)

Thus the "orig" value for a STIR/SHAKEN header should be something like 
"12293160013"  (because the country code is 1), and when the UK is sending 
PASSporT's someday, the orig value would be something like "442078702900" 
(because 44 is the UK country code).


Mark R Lindsey | +1-229-316-0013 | m...@ecg.co | Schedule a 
Meeting<https://ecg.co/lindsey/schedule> | 
Newsletter<https://www.linkedin.com/newsletters/mark-lindsey-voice-7021614437413330944/>


On Apr 5, 2023, at 16:07, Matthew Yaklin via VoiceOps  
wrote:

I was talking with a comcast guy a while back when we had a SBC incorrectly 
set. Here is what he said. By any chance are you sending a 10 digit in the to 
or from? Comcast wants 11 digits for both with no exceptions. We simply logged 
into the Neustar portal and made a small adjustment on that side as it was 
easier then tweaking the acme sbc.

“Hello. My name is XXX XXX; I am an Engineer in Comcast’s Voice Communications 
Engineering organization. I obtained your contact information from the Robocall 
Mitigation Database. Below is an example call and Identity header that fails 
Comcast’s STIR-SHAKEN verification service. You are signing the call using only 
a 10 digit TO and FROM number. All TO and FROM numbers need to be 11 digits. 
Almost all 8M Comcast residential voice customers have Xfinity Voice Spam 
Blocker enabled with the default settings. Failed STIR-SHAKEN calls are sent 
directly to voicemail and do not ring my customer’s phone.”

matt


From: VoiceOps 
mailto:voiceops-boun...@voiceops.org>> On Behalf 
Of Matthew Crocker via VoiceOps
Sent: Wednesday, April 5, 2023 3:59 PM
To: voiceops@voiceops.org<mailto:voiceops@voiceops.org>
Subject: [VoiceOps] Outbound Calls being marked as SPAM


We have a customer whos outbound calls are being marked as SPAM by the 
terminating carrier.   We are sending the calls out fully signed (STIR/SHAKEN) 
with attest ‘A’ and all of the propery identity headers.  The terminating 
carrier is Comcast from what we can tell,  does anyone have any tricks we can 
use or something we may have missed to help get the calls marked correctly?

Thanks

-Matt



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[VoiceOps] Outbound Calls being marked as SPAM

[Matthew Crocker via VoiceOps]

We have a customer whos outbound calls are being marked as SPAM by the 
terminating carrier.   We are sending the calls out fully signed (STIR/SHAKEN) 
with attest ‘A’ and all of the propery identity headers.  The terminating 
carrier is Comcast from what we can tell,  does anyone have any tricks we can 
use or something we may have missed to help get the calls marked correctly?

Thanks

-Matt



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[VoiceOps] Releasing numbers back to original carrier

[Matthew Crocker via VoiceOps]

Hello,

  It is our practice to release numbers back to the original carrier when a 
customer cancels service and doesn’t port the number away.   We have some 
residential customers that want to cancel service over the winter and have us 
retain the number so they can re-use it in the spring.   I’m trying to find 
some FCC or NANPA documentation that says we are/aren’t allowed to do that for 
a customer.   Ultimately we will probably convert the customer from their 
FTTH/ONT voice to a soft phone voice during the off-season, that way they are 
still a customer and still paying for service.   I don’t want to just camp on 
numbers and have to maintain the inventory.

Anyone have any documentation on the correct way to handle disconnected numbers?

-Matt

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