Re: [VoiceOps] i-test alternative

[Fred Posner via VoiceOps]

I’ve used spearline to test calls from local regions via pstn. Was very happy 
with them.

Fred Posner
f...@palner.com




> On Nov 17, 2022, at 9:40 AM, Dovid Bender via VoiceOps 
>  wrote:
> 
> Hi All,
> 
> We have been using i-test.net to test calls around the world, mainly to check 
> for FAS and to see if the CLI we send is what's received on the other side. I 
> vaguely remember there being another service out there like them (for the 
> life of mine I can't remember the name). Anyone know of any alternatives to 
> them?
> 
> TIA.
> 
> Dovid
> 
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Re: [VoiceOps] Numbers listed as spam by Hiya

[Fred Posner via VoiceOps]

Doesn’t it make sense to block numbers from carriers that send spam?

Fred Posner
f...@palner.com




> On Aug 24, 2022, at 10:03 AM, Dovid Bender via VoiceOps 
>  wrote:
> 
> Hi,
> 
> We primarily purchase numbers from Telnyx and Peerless. Lately any number 
> that we purchase is automatically listed as spam. I searched through both of 
> their stocks for random numbers in random area codes (50+ numbers) and every 
> time Hiya says it's likely SPAM. I presume they are marking any carrier where 
> it's easy to get numbers spam. Has anyone fought back against this or are 
> people just going with larger carriers like inteliquent/VZ etc?
> 
> Regards,
> 
> Dovid
> 
> 
> 
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Re: [VoiceOps] SentryPeer: A distributed peer to peer list of bad IP addresses and phone numbers collected via a SIP Honeypot

[Fred Posner]

Hi All,

Re APIBAN...

APIBAN has two main ways that it's used... with a simple block of IP
addresses through firewall or iptables being the most used aspect.

Briefly, through honeypots (global) IP addresses sending SIP or non-SIP
(like dns, fuzz, or malformed SIP) are identified. We capture the
commonly used SIP listener ports with UDP, TCP, and TLS.

Most users utilize the apiban client to automatically block these IPs in
iptables. There is also methods to check individual ip's by API as well
as grabbing all active IPs, etc.

We looked into a community submission, but decided against it as it was
too easily poisoned. The main goal here is quality of the data and
making sure that we're not distributing any valid IP as something that
should be blocked.

I like the idea of community submission, but the poisoning was
determined to be too big of a risk for us.

I also like the idea of sharing some data of numbers being called,
etc... but like that for analysis and approaching hardening in a
non-realtime scenario.

With best regards,

Fred Posner | palner.com
Matrix: @fred:matrix.lod.com
o: +1 (212) 937-7844

On 1/3/22 4:34 AM, Gavin Henry wrote:
> On Mon, 3 Jan 2022, 03:22 Jim O'Brien,  <mailto:jimdoesv...@gmail.com>> wrote:
> 
> Hi Gavin,
>   Thanks for sharing.  In many ways your project reminds me of Fred
> Posner’s APIBAN.  I like your approach here with SentryPeer allowing
> an operator to run their own systems and choose to share with and
> receive IPs from others!  These piecs are fantastic!  Once the crush
> of coming back from holidays is over I cannot wait to give this a try.
> 
> Best,
> 
> Jim
> 
> 
> Thanks Jim. APIBAN, for now, doesn't publish B numbers. I just added
> responsive mode (replying to probes so they then try proper INVITEs),
> but haven't committed it yet and the numbers API so you can check
> customer calling attempts. 
> 
> I'm also adding a SIP agent mode too for SIP redirects. The plan is you
> just run in agent mode with replication on (replication coming soon) as
> a mini SIP proxy etc.
> 
> https://www.linkedin.com/posts/surevoip_sip-sip-fraudprevention-activity-6882708550662070272-9HDL
> <https://www.linkedin.com/posts/surevoip_sip-sip-fraudprevention-activity-6882708550662070272-9HDL>
> 
> I've also done an RPM and Dockerfile / Dockerhub container and my first
> ever proper debian package! That was a long time dream of mine as I
> thought debs were so hard compared to an RPM spec. 
> 
> https://github.com/SentryPeer/SentryPeer/releases/tag/v0.0.4
> <https://github.com/SentryPeer/SentryPeer/releases/tag/v0.0.4>
> 
> Just got Debian salsa git repo access this morning too so I can start to
> get it into Debian proper, hopefully. 
> 
> Gavin. 
> 
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Re: [VoiceOps] Ray Baum's act and all-softphone deployments

[Fred Posner via VoiceOps]

I wonder what ios15's new hide my ip feature will do to this.

Fred Posner | palner.com
Matrix: @fred:matrix.lod.com
o: +1 (212) 937-7844

On 9/27/21 3:35 PM, Alex Balashov via VoiceOps wrote:
> I wonder if doing GeoIP via a commercial database like MaxMind—and we all 
> know how perfect a location mechanism that is—could be construed as a 
> sufficient “best effort” in this case.
> 
>> On Sep 27, 2021, at 3:28 PM, Carlos Alvarez  wrote:
>>
>> Ours doesn't, but that's a good idea to share with the vendor.  But in most 
>> cases, people are using the softphone on a laptop and there's no native 
>> phone app, nor a GPS, nor any other location info other than proximate wi-fi 
>> networks.
>>
>> I can see no feasible way to meet the letter of the law, and hate to be 
>> depending on the good graces of a bureaucrat to know that it's an impossible 
>> task.
> 
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Re: [VoiceOps] AT Verizon to block text messaging

[Fred Posner]

On 3/4/21 12:52 PM, Oren Yehezkely wrote:
> Overall it is a move by a giant to hurt small carriers and customers of
> other companies by reducing functionality and raising their cost of
> doing business.

That's your take...

Or it's like Carlos and others believe (and me as well) that the spam
generated, perhaps from your customers, is making SMS simply as useless
as email. If this makes it so we get fewer unsolicited sms messages, great.

Nothing I see impacts sms for individiual <-> individual or SMB <->
individual at low volume. In fact, it's specifically stated those do not
need to register.

Your language here is very "sky is falling" and one of two things need
to happen...

Either (a), you're legitimately sending large volumes of sms and need to
register... or (b) you're not sending legitimate sms and should stop.

The registration seems like not a big deal to be honest; as well as
decent time provided.

I like this model a lot better than stir/shaken.

--fred
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Re: [VoiceOps] False 911 calls and old abandoned DID

[Fred Posner]

My understanding was that it was chosen cause it was easy to remember 
(especially for kids), simple to dial, and didn’t conflict with any service or 
area code with no consideration for pulse shortness. 

-- Fred
direct/sms +1 (336) 439-3733


> On Jan 21, 2021, at 6:59 PM, Carlos Alvarez  wrote:
> 
> 
> Island time is chill, yo.  New Yorkers and Southern Californians got to do 
> things like NOW!
> 
> Two pulses, one pulse, three pulses would be Los Angeles.
> 
> I mean, I can test dial something right here at home if you want, I have the 
> means.  Yes, it works.
> https://imgur.com/5m2Fe5b
> 
> 
>> On Thu, Jan 21, 2021 at 4:52 PM Jeff Shultz  wrote:
>> That would be odd  one of the reasons that big cities in the US got area 
>> codes like 212 was because it was shorter to dial with fewer pulses needed. 
>> Hawaii got stuck with 808 
>> 
>>> On Thu, Jan 21, 2021 at 3:43 PM Pete Mundy  wrote:
>>> 
>>> So close! But on the PoTS equipment I'm familiar with (others can chime in 
>>> and correct me if I'm wrong on a larger scale) the numbers are reversed
>>> 
>>> (well for all except 5 :)
>>> 
>>> Basically it counts down from 10. 1 pulse is digit '9', 4 pulses is digit 
>>> '6', 10 pulses is digit '0'.
>>> 
>>> So do dial 911 you need:
>>> 
>>> 1 brief loop interruption
>>> pause
>>> 9 brief loop interruptions
>>> pause
>>> 9 brief loop interruptions
>>> 
>>> 
>>> 
>>> Pete
>>> 
>>> 
>>> > On 22/01/2021, at 12:07 PM, Mike Johnston  wrote:
>>> > 
>>> > One interrupt represents the digit "1" being dialed.  Two rapid 
>>> > interrupts represents the digit "2", and so on.  You need to have at 
>>> > least a brief timeout 
>>> > 
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>> 
>> 
>> -- 
>> Jeff Shultz
>> 
>> 
>> Like us on Social Media for News, Promotions, and other information!!
>> 
>>  
>> 
>> 
>> 
>> 
>> 
>> 
>> 
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Re: [VoiceOps] Automatic 911 location updates

[Fred Posner]

On 10/23/18 7:36 PM, Andrew Haninger wrote:

(This time to the group. Sorry for the dupe, Carlos.)

On Tue, Oct 23, 2018, 19:22 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.


The Skype for Business system does this where I work. It uses the 
location of the nearest AP or the location recorded for the jack you're 
using if you're wired. If neither are available, the SfB application 
asks the user to enter their location.


The analog phones we're switching from only update when an order is 
processed, but then customers generally have to put in a request for us 
to do a MAC order. The local PSAP will report to us if they get a 911 
call with incorrect name or location info.


It's not fool proof and there is plenty of room for errors, but it's not 
too bad, either.


Andt



The best approach I saw was in a research demo at cluecon:

https://youtu.be/A8i2psbeYT4

We're still not there in most aspects because either (a) you need to 
constantly enter and verify information or (b) information will not be 
up to date.


--fred
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Re: [VoiceOps] Wireless Carrier Lookups: CNAM DIPS

[Fred Posner]

T-Mobile does have a contract service that provides a database of 
supposedly bad actors. That's a known program, quite separate from what 
the smart phone OS might be doing.




Nice

--fred
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Re: [VoiceOps] Wireless Carrier Lookups: CNAM DIPS

[Fred Posner]

On 4/13/18 4:42 PM, Glen Gerhard wrote:

SLIGHTLY OFF TOPIC:

Lately I've been getting calls to my Tmobile cell phone with CNAMs such 
as Scam Likely and Nuisance Call.  Even without the CNAM I know they are 
call centers so don't pick up, but I'm wondering who assigns that CNAM?  
I'd expect Tmobile to dip the CNAM but doubt the official database would 
flag the number this way. If it's coming from the call center app it's 
pretty funny. I don't know anywhere else along the call path it would be 
changed.


~Glen



Android phone? It's from Google, not T-Mobile.

--fred
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Re: [VoiceOps] 411 provider recommendations

[Fred Posner]

Stopped even integrating this a few years ago. With the Internet, 
there's really no application for this anymore.


--fred

On 7/24/17 1:21 PM, Ball, Jared wrote:
My company has had a long-term relationship with a 411 provider for 
about 10 years.  We have never really had any problems.  We received a 
letter about them no longer offering this service and transitioning us 
to another vendor.  Can I get some recommendations on 411 providers? I 
might ask our caller-id-name provider if they have a 411 service.   Any 
recommendations on providers that have had no problems for 5+ years 
would be appreciated.


--Jared Ball



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Re: [VoiceOps] What does you use for Hosted PBX and why?

[Fred Posner]

On 03/16/2017 09:36 AM, Colton Conor wrote:
> Mark/Fred,
> 
> With 2600hz / Kazoo, are you paying them to install, maintain, and
> update their platform, or did you download and install it on your own? I
> talked to 2600hz's long ago about their hosted product, and at the time
> it just was not mature enough. Not to mention they basically wanted
> something like $5 per seat, and support on-top of that which was
> equivalent to the price I was paying for broadsoft seats on a monthly
> basis. No they didn't charge extra for group features like auto
> attedants and hunt groups like broadsoft does, but still I considered
> their product high priced for the limited functionality it had. 
> 

I have used Broadsoft and tried to work that into a hosted model with
great disappointment. The problems I had with a Broadsoft/Acme Packet
solution in handling 10k users was mind blowing... especially, since I
was able to eliminate these problems and see incredible growth when I
switched to openSER/Kamailio.

We jumped to over 60k users and eliminated (completely) problems with
NAT, one way audio, as well as increasing the features we could offer
our clients.

I think some vendors are starting to aggressively attack open source
(Mitel recently did, I responded:
http://www.fredposner.com/1935/mitel-thinks-knowledge-is-a-risk/).

Some people look at open source as a "cheaper" solution, but that's not
the reason I became involved... I needed a better solution, and again,
Kamailio solved the problems I had 15 years ago and it continues to be a
rock solid performer today.

--fred
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Re: [VoiceOps] SIP ladder diagram builder

[Fred Posner]

If there is one that can do this via an API, I haven't found it.
Although you could probably script something with either cloudshark or
homer.

Homer can export to png.

--fred

On 02/25/2017 05:21 PM, Alex Balashov wrote:
> Does it have an API one can call to get a diagram out automatically? 
> 
> On February 25, 2017 5:20:42 PM EST, Colin Brown  wrote:
>> Alex I'm a huge huge fan of cloudshark.
>>
>>
>>
>> On Sat, Feb 25, 2017 at 5:17 PM Alex Balashov
>> 
>> wrote:
>>
>>> Hello,
>>>
>>> Is there a nice SIP ladder diagram building utility these days into
>> which
>>> one can feed a packet capture and get a ladder diagram out?
>>>
>>> I see the decaying skeletons of several such ventures online, but
>> what's
>>> the new news?
>>>
>>> sngrep is fantastic for live troubleshooting, but unfortunately its
>>> rendering is not export-orientated and doesn't lend itself to
>> automaton.
>>>
>>> Either image or ASCII output is fine. Just needs to be automatic and
>>> exportable.
>>>
>>> Many thanks!
>>>
>>> -- Alex
>>>
>>> -- Alex
>>>
>>> --
>>> Principal, Evariste Systems LLC (www.evaristesys.com)
>>>
>>> Sent from my Google Nexus.
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> 
> 
> -- Alex
> 
> --
> Principal, Evariste Systems LLC (www.evaristesys.com)
> 
> Sent from my Google Nexus.
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Re: [VoiceOps] Opinions on Netsapiens or Kazoo

[Fred Posner]

Never been super impressed with Netsapiens and would recommend looking
at CSRPswitch.com as an alternative; although it depends on what you're
looking to do with the system.

--fred


On 01/28/2017 09:16 PM, Rob Dawson wrote:
> Interested in getting opinions from folks with experience running hosted
> offerings on Netsapiens or Kazoo. Looking to see what types of struggles
> and successes you have had with the platforms, support experience,
> perhaps a little info about your target market and what differentiation
> the platform has helped you to develop – in broad terms.
> 
>  
> 
> Thanks,
> 
> Rob
> 
> 
> 
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Re: [VoiceOps] Faxing over SIP - (88) INCOMPATIBLE_DESTINATION

[Fred Posner]

"This cause indicates that the equipment sending this cause has received 
a request to establish a call which has low layer compatibility, high 
layer compatibility or other compatibility attributes (e.g. data rate) 
which cannot be accommodated."


--fred

On 11/28/16 1:18 PM, Nathan Anderson wrote:

What specifically is displaying "(88) INCOMPATIBLE_DESTINATION"?  That's not a 
normal SIP message.  Is this Zoiper that shows this?  Freeswitch?  Kamailio?  The gateway?

What you really need is a packet capture of the SIP conversation that is 
happening between the entity throwing that error and whatever it is INVITEing.  
So if Zoiper is displaying that message, make a capture of the signalling 
between Zoiper and Freeswitch.  It will certainly supply you with some clues 
about what is going on...

-- Nathan

From: VoiceOps [mailto:voiceops-boun...@voiceops.org] On Behalf Of Tanguy 
Dangelser
Sent: Sunday, November 27, 2016 8:46 PM
To: voiceops@voiceops.org
Subject: [VoiceOps] Faxing over SIP - (88) INCOMPATIBLE_DESTINATION

Hi,

I know that faxing over SIP can be source of a whole lot of issues, but I am 
wondering if someone had the same that I am currently running into.

I have a softphone (Zoiper) sending a fax (TIFF image sent to a physical fax on 
PSTN) through a SIP platform with freeswitch / kamailio which is forwarding the 
fax to my PSTN Gateway (telcobridges TMG800), the fax is detected as T38 and 
everything looks good, except that it ends with (88) INCOMPATIBLE_DESTINATION

Does anyone has an idea what might cause this and what could help resolve the 
issue ? I can send the trace if you email me. Any help appreciated.

Thank you,

Tanguy

--

[http://www.guyacom.net/wp-content/uploads/2015/02/logoSignature-150x124.png]Tanguy
 DANGELSER
Résidence petit Chalet – Châton
56 avenue Louis Pasteur – 97300 CAYENNE
Guyane Française – French Guiana

Tél : +594 (0) 594 25 00 00 - Fax : +594 (0) 594 25 37 85

[http://www.guyacom.net/wp-content/uploads/2015/02/facebook.png]  
[http://www.guyacom.net/wp-content/uploads/2015/02/linkedin.png] 
   
[http://www.guyacom.net/wp-content/uploads/2015/02/email.png]    
[http://www.guyacom.net/wp-content/uploads/2015/02/skype.png]   
[http://www.guyacom.net/wp-content/uploads/2015/02/youtube.png]    
[http://www.guyacom.net/wp-content/uploads/2015/02/www.png] 



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Re: [VoiceOps] Astricon 2016

[Fred Posner]

I'll be there. 

-- Fred
direct/sms +1 (503) 914-0999


> On Sep 16, 2016, at 9:27 PM, Alex Balashov  wrote:
> 
> Aye.
> 
>> On 09/16/2016 09:25 PM, Aryn Nakaoka 808.356.2901 wrote:
>> 
>> Anyone going to Astricon ?
>> 
>> 
>> 
>> 
>> Aryn H. K. Nakaoka
>> anaka...@trinet-hi.com 
>> 
>> Direct: 808.356.2901
>> Fax : 808.356.2919
>> 
>> Tri-net Solutions
>> 518 Holokahana Lane Suite #200
>> Honolulu, HI 96817
>> http://www.trinet-hi.com
>> 
>> https://twitter.com/AlohaTone
>> 
>> Aloha Tone PBX  https://www.youtube.com/watch?v=96YWPY9wCeU
>> 
>> Aloha Tone (HA) High Availability http://youtu.be/rJsr4k0RBH8
>> 
>> CONFIDENTIALITY NOTICE:  The information contained in this email and any
>> attachments may be privileged, confidential and protected from
>> disclosure.  Any disclosure, distribution or copying of this email or
>> any attachments by persons or entities other than the intended recipient
>> is prohibited. If you have received this email in error, please notify
>> the sender immediately by replying to the message and deleting this
>> email and any attachments from your system. Thank you for your cooperation.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
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> 
> 
> -- 
> Alex Balashov | Principal | Evariste Systems LLC
> 
> Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [VoiceOps] Virtualized SBC

[Fred Posner]

On 04/07/2016 10:56 AM, Matthew Crocker wrote:
> 
> 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
> 
> 

"in the cloud" is like fingernails on a chalkboard.

Building anything that is only cloud based, for your independent
business, is like building a solution for nest using resolv.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
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Re: [VoiceOps] Cox cable "outages" and a solution

[Fred Posner]

I had a major issue with Cox when they were deploying ipv6. They set a
rate limiter on packets, and any voip call would trigger it. Once the
call ended, the speed would come back.

The way we proved it was to ping from the router to the immediate hop
after the modem... once we could prove it this way to the field tech,
they got "tier 3" to call a network engineer and resolve it.

Hopefully it's similar.

Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-224-334-FRED (3733) direct

On 03/19/2016 12:48 PM, Carlos Alvarez wrote:
> I'm posting this info here because I know quite a few of us have BYOI
> customers and may be affected by this.  Cox users have been reporting a
> lot of outages lately in certain areas (it seems to cluster by whatever
> areas Cox is making changes in).  I found that when there is a supposed
> outage, the only thing that fails is DNS. You can still
> ping/trace/connect with an IP address. I looked at the router's
> DHCP-assigned info and found that it was seeing an IPv6 DNS as its
> primary, with the usual two IPv4 servers as secondary/tertiary. Oddly,
> this didn't seem to happen right away, but would happen after the router
> had been up for a few hours. My best guess at the root problem is that
> Cox isn't properly handling IPv6 yet, even though they are advertising
> it to the router.  Completely disabling IPv6 in the router fixes it.
> 
> 
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Re: [VoiceOps] 2015 Softswitch

[Fred Posner]

> On Oct 29, 2015, at 9:58 PM, Alex Balashov  wrote:
> 
> Have you considered Enswitch by Integrics?
> 
> ‎It's the best of breed of the sort of thing that it is. Moreover, if you'll 
> tolerate BW price points, you'll think it's practically free.
> 
> ‎https://integrics.com/enswitch/
> 
> It's got the API and integration path requirement covered, too. I know about 
> a dozen operators and they're all pretty happy with it.
> 
> If you talk to Alistair Cunningham, their director, be sure to relate that 
> Alex Balashov sends his regards.
> 
> --
> 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/

This is a great suggestion. 

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Re: [VoiceOps] Software SBC solution recommendations

[Fred Posner]

I'd recommend you look at the Canonical SIP Routing Platform (CSRP). 
http://www.csrpswitch.com

-- Fred
http://palner.com

 On Jul 3, 2015, at 10:37 PM, Ryan Delgrosso ryandelgro...@gmail.com wrote:
 
 So I'm working on a bit of a niche project in need of a virtualizable 
 software SBC and while I am aware of many of the open source solutions out 
 there I'm sure there are facets of the various options that those who live 
 with them every day can share that will influence direction. 
 
 I need this soluton to do: registration caching (but pass reg back to 
 softswitch), nat traversal/media proxy, basic topology hiding, blacklisting, 
 some kind of dynamic routing (ENUM is preferred). No need for fancy multi 
 realm setup, simple inside/outside is fine. 
 
 I am presently looking at: 
 Kamailio + some kind of media proxy (does much of the signaling but is not a 
 B2BUA)
 Freeswitch (doesnt seem to do the reg cache / pass back)
 SEMS (reg caching only introduced last month.
 Various combinations of the above (though most of these feel kludgy)
 Im not precluding commercial solutions or hybrid solutions but the usual 
 suspects (Acme, Meta, Sonus, etc) are off the table here due to either the 
 cost or the software/virtualization requirements.
 
 Thanks in advance,
 -Ryan
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Re: [VoiceOps] Which Softswitch?

[Fred Posner]

You may want to check out the Canonical SIP Routing Platform by Evariste 
Systems. 

http://www.csrpswitch.com

-- Fred

 On Jun 20, 2015, at 2:39 AM, Greg Lipschitz g...@thesummitgroup.com.au 
 wrote:
 
 Hi All,
 
 We are in the process of going down the path of deploying our own Softswitch 
 infrastructure to transition away from purchasing from SIP Trunk providers 
 and moving to getting CTS direct from VoIP Carriers.
 
 There seems to be a lot of solutions out on the market and weeding out the 
 Built on Asterisk / Built on Freeswitch with little or no support vs 
 Commercial Supported solutions is proving to be difficult.
 
 There are the leaders (or strong marketing companies) such as Broadsoft  
 Metaswitch - but what I am trying to uncover is what else is great out there 
 that may not have the exposure but is worth looking at.
 
 What we are looking for...
 
 1. Class 4 Softswitch
 2. Geographical Redundancy (Active - Active)
 3. Preferably not built on Freeswitch or Asterisk (Not saying they're bad 
 just tired of OpenSource for Core products)
 4. Must have commercial support available (preferably during Australian hours)
 5. Windows or Linux doesn't phase us - we have skill set on both platforms.
 
 Happy to hear the good, the bad, the ugly, the war stories so that we can get 
 some real world feedback on what's out there, what works, what doesn't and 
 hopefully not make the same mistakes that others have already made.
 
 Offlist replies are perfectly fine if you don't wish to discuss on-list.
 
 Thanks in advance and enjoy the rest of your weekend! :)
 
 Cheers,
 
 Greg
 
 
 Greg Lipschitz | Director | The Summit Group
 E: g...@thesummitgroup.com.au  W: www.thesummitgroup.com.au
 The Summit Group (Australia) Pty Ltd | P: 1300 049 749 | Level 1, 39 Railway 
 Road, Blackburn  VIC  3130
 The Summit Group (USA) LLC | P: 321 216 3844 | Suite 561, 40E Main Street, 
 Newmark  DE 19711
 Postal:  P.O. Box 3225, Doncaster East  VIC  3109
 
 
 
 
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Re: [VoiceOps] SIP HD Peering with Wireless Carriers

[Fred Posner]

This would be a great reason to get a true North American enum. 

---Fred 


 On Feb 15, 2015, at 11:01 AM, Colton Conor colton.co...@gmail.com wrote:
 
 So does Intelliquent have connections to Sprint, T-Mobile, ATT, and Verizon 
 Wireless? My question here is specifically if SIP providers can buy mintues 
 or peer with these wireless providers using G722 for HD cellular to HD SIP 
 calling. 
 
 On Fri, Feb 13, 2015 at 1:03 PM, Paul Timmins p...@timmins.net wrote:
 If this is happening I want in. I know Intelliquent allows G722 
 interconnect, is anyone actually using it and want to play with it with us?
 
 On 02/13/2015 09:11 AM, Colton Conor wrote:
 As you know most all the wireless cellular carriers are deploying HD voice 
 with VoLTE. Most of these carriers, like Verizon Wireless, have a wholesale 
 wireline division that sells wholesale SIP minutes. Are these carriers 
 going to enable HD G722 SIP peering to their wireless counterparts networks?
 
 That way an office worker, talking on a HD capable handset, like a Polycom, 
 could talk in HD to a wireless cell phone user with a HD capable VoLTE 
 smartphone. What is needed to make this happen? Are any of the big 4 
 carriers already doing this? 
 
 
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Re: [VoiceOps] T1 Crossover Cable

[Fred Posner]

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 (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (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/

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
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