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

[Robert Johnson]

If there is a need to manage only a single box, the first thing that 
comes to my mind is Adtran's Total Access 5000 series. The full size 
chassis has 22 slots. Fill those up with FXS cards and you're looking at 
528 FXS ports.


We have never had a use case where each FXS port didn't require it's own 
registration info, so I can't speak to if there can be a shared 
registration for the ports.


I can say, that such a thing is possible with the Adtran TA900 series 
and that the two pieces of equipment are pretty similar in regards to 
SIP/FXS capabilities.


On 3/21/19 8:55 AM, 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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[VoiceOps] Neustar Port PS Alternative?

[Robert Johnson]

Nuestar's Port PS is now a paid service and I have been tasked with 
finding quotes for competing services. Iconectiv was my first choice, 
but I have been informed their tool is not yet available. What other 
options are out there?

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[VoiceOps] Multicast Paging Device

[Robert Johnson]

Does anyone have recommendations on a SIP Multicast paging device that 
would connect to a pre-existing paging amplifier.

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Re: [VoiceOps] flowroute down?

[Robert Johnson]

Traffic is flowing again now. Here's what our monitoring saw:

[05-20-2016 13:45:39] HOST ALERT: flowroute-sip;UP;HARD;3;PING OK - Packet
loss = 61%, RTA = 68.11 ms
[05-20-2016 13:43:57] HOST ALERT: flowroute-sip;DOWN;HARD;3;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:42:28] HOST ALERT: flowroute-sip;DOWN;SOFT;2;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:40:58] HOST ALERT: flowroute-sip;DOWN;SOFT;1;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:38:41] HOST ALERT: flowroute-sip;UP;HARD;3;PING OK - Packet
loss = 66%, RTA = 68.35 ms
[05-20-2016 13:35:27] SERVICE ALERT:
flowroute-sip;SIP-trunk;CRITICAL;HARD;3;SIP timeout: No response from SIP
server after 15 seconds
[05-20-2016 13:34:27] SERVICE ALERT:
flowroute-sip;SIP-trunk;CRITICAL;SOFT;2;SIP timeout: No response from SIP
server after 15 seconds
[05-20-2016 13:33:57] HOST ALERT: flowroute-sip;DOWN;HARD;3;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:33:27] SERVICE ALERT:
flowroute-sip;SIP-trunk;CRITICAL;SOFT;1;SIP timeout: No response from SIP
server after 15 seconds
[05-20-2016 13:33:27] SERVICE ALERT:
flowroute-sip;PING;CRITICAL;HARD;3;PING CRITICAL - Packet loss = 100%
[05-20-2016 13:32:57] HOST ALERT: flowroute-sip;DOWN;SOFT;2;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:32:27] SERVICE ALERT:
flowroute-sip;PING;CRITICAL;SOFT;2;PING CRITICAL - Packet loss = 100%
[05-20-2016 13:31:57] HOST ALERT: flowroute-sip;DOWN;SOFT;1;(Host check
timed out after 30.01 seconds)
[05-20-2016 13:31:27] SERVICE ALERT:
flowroute-sip;PING;CRITICAL;SOFT;1;PING CRITICAL - Packet loss = 100%

On Fri, May 20, 2016 at 1:48 PM, Kraig Beahn  wrote:

> Website is accessible in Florida and Georgia, at least at the moment.
>
> On Fri, May 20, 2016 at 1:45 PM, Joseph Jackson 
> wrote:
>
>> I see the registration failures to their sip.flowroute.com sbc.
>>
>>
>>
>> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *
>> chris
>> *Sent:* Friday, May 20, 2016 12:42 PM
>> *To:* voiceops@voiceops.org
>> *Subject:* [VoiceOps] flowroute down?
>>
>>
>>
>> flowroute appears to be down hard from here on the east coast
>>
>>
>>
>> cant reach their sip proxies, website, cant call support phone numbers
>>
>>
>>
>> chris
>>
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>>
>>
>
>
> --
>
>
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Re: [VoiceOps] outsource bills printing?

[Robert Johnson]

postalmethods.com


On Wed, Mar 23, 2016 at 12:50 AM,  wrote:

> We use these folks and are happy with them:
> http://print.innovsys.com/index.php/2015-05-13-13-54-27/mailing
>
>
>
> Frank
>
>
>
> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *Ryan
> Finnesey
> *Sent:* Tuesday, March 22, 2016 8:51 PM
> *To:* voiceops@voiceops.org
> *Subject:* [VoiceOps] outsource bills printing?
>
>
>
> I am hoping the group might have some recommendation’s.  I am looking for
> a service provider we can use to outsource the mailing and printing of
> bills  Most of our bills are one to two pages
>
>
>
>
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Re: [VoiceOps] VoIP Innovations reliability

[Robert Johnson]

You're right. Was this mentioned anywhere other than buried in that
maintenance notification?

On Thu, Mar 17, 2016 at 6:45 PM, Nathan Anderson <nath...@fsr.com> wrote:

> Not true re: DR IP.  Details about it are not being broadcast publicly,
> though, to prevent the attackers from getting their hands on it.  You are
> probably thinking of their primary and secondary SBCs within the main data
> center.  For details about the DR SBC you need to contact VI (or read your
> e-mails).
>
>
>
> -- Nathan
>
>
>
> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of *Robert
> Johnson
> *Sent:* Thursday, March 17, 2016 3:43 PM
>
> *To:* voiceops@voiceops.org
> *Subject:* Re: [VoiceOps] VoIP Innovations reliability
>
>
>
> Their primary SBC and DR IP are both in the same IP netblock, so whenever
> the DDOS hits, both IPs are affected. The past few outages have involved
> 80% packet loss or so to both hosts, so some calls do make their way
> through, and plenty of wierdness ensues when an INVITE makes its way
> through but not the OK on the way back.
>
>
>
> Can't wait to get our numbers ported out.
>
>
>
> On Thu, Mar 17, 2016 at 6:27 PM, Nate Burke <n...@blastcomm.com> wrote:
>
> Annnd they're down again.
>
>
>
> On 3/17/2016 5:14 PM, Nate Burke wrote:
>
> 6 calls from 4 different CID numbers.  All within 3 minutes.
>
> On 3/17/2016 4:58 PM, Nathan Anderson wrote:
>
> Yesterday I definitely saw calls coming from the DR IP in my logs, but I
> had not yet added that IP as a peerin our SBC.  I'll have to comb through
> logs today to see if we got any.
>
>
>
> Are you saying that the multiple calls you saw coming to your desk were
> all from the same number?  If I had to guess, their side probably sent
> continuous INVITEs to you when it failed to get back an OK for any of them
> (not that you weren't sending back OK, but that their either didn't reach
> their SBC or did not reach it in a timely manner).
>
>
>
> -- Nathan
>
>
>
> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org
> <voiceops-boun...@voiceops.org>] *On Behalf Of *Jeff Waddell
> *Sent:* Thursday, March 17, 2016 2:53 PM
> *To:* Nate Burke
> *Cc:* voiceops@voiceops.org
> *Subject:* Re: [VoiceOps] VoIP Innovations reliability
>
>
>
> That is the issue a lot of our customers are reporting - where multiple
> calls are sent
>
>
>
> On Thu, Mar 17, 2016 at 5:49 PM, Nate Burke <n...@blastcomm.com> wrote:
>
> I didn't see any traffic increment on the DR IP Address in my firewall
> rules, but this was odd.  During the 15 minute period, I had probably 5 or
> 6 simultaneous calls ring into my desk.  I normally only take a handfull of
> calls a day.
>
>
>
> On 3/17/2016 4:39 PM, Jeff Waddell wrote:
>
> We implemented it too - I haven't checked to see if any traffic was sent
> across it
>
>
>
>
>
>
>
> On Thu, Mar 17, 2016 at 5:20 PM, Nate Burke <n...@blastcomm.com> wrote:
>
> Only 15 Minutes this time though.  I had implemented the Disaster Recover
> Trunk as mentioned previously, but I didn't seem to be getting any calls
> completed through it.
>
>
>
> On 3/17/2016 4:16 PM, Shripal Daphtary wrote:
>
> Down again!
>
> Thanks,
>
>
>
> Shripal
>
>
> On Mar 16, 2016, at 9:50 AM, Nate Burke <n...@blastcomm.com> wrote:
>
> Looks like it just came back up for me.  Just over 30 min.
>
> Nate
>
> On 3/16/2016 8:45 AM, Shripal Daphtary wrote:
>
> We are experiencing an outage as well.
>
> Thanks,
>
>
>
> Shripal
>
>
> On Mar 16, 2016, at 9:36 AM, Nate Burke <n...@blastcomm.com> wrote:
>
> Problems again this morning?  Looks to be acting the same as it has been.
>
> On 3/11/2016 6:00 PM, Alexander Lopez wrote:
>
> I added them to our monitoring platform, stated getting alarms this past
> hour or so.
>
> Up and down.
>
>
>
>  Original message 
> From: Nathan Anderson <nath...@fsr.com> <nath...@fsr.com>
> Date: 3/11/2016 6:31 PM (GMT-05:00)
> To: 'Nate Burke' <n...@blastcomm.com> <n...@blastcomm.com>,
> voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> ...aand we're back.
>
> -- Nathan
>
> -Original Message-
> From: VoiceOps [mailto:voiceops-boun...@voiceops.org
> <voiceops-boun...@voiceops.org>] On Behalf Of Nathan Anderson
> Sent: Friday, March 11, 2016 3:30 PM
> To: 'Nate Burke'; voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> It *feels* like they are under attack again, since I get a response to a
&

Re: [VoiceOps] VoIP Innovations reliability

[Robert Johnson]

Their primary SBC and DR IP are both in the same IP netblock, so whenever
the DDOS hits, both IPs are affected. The past few outages have involved
80% packet loss or so to both hosts, so some calls do make their way
through, and plenty of wierdness ensues when an INVITE makes its way
through but not the OK on the way back.

Can't wait to get our numbers ported out.

On Thu, Mar 17, 2016 at 6:27 PM, Nate Burke  wrote:

> Annnd they're down again.
>
>
> On 3/17/2016 5:14 PM, Nate Burke wrote:
>
> 6 calls from 4 different CID numbers.  All within 3 minutes.
>
> On 3/17/2016 4:58 PM, Nathan Anderson wrote:
>
> Yesterday I definitely saw calls coming from the DR IP in my logs, but I
> had not yet added that IP as a peerin our SBC.  I'll have to comb through
> logs today to see if we got any.
>
>
>
> Are you saying that the multiple calls you saw coming to your desk were
> all from the same number?  If I had to guess, their side probably sent
> continuous INVITEs to you when it failed to get back an OK for any of them
> (not that you weren't sending back OK, but that their either didn't reach
> their SBC or did not reach it in a timely manner).
>
>
>
> -- Nathan
>
>
>
> *From:* VoiceOps [mailto:voiceops-boun...@voiceops.org
> ] *On Behalf Of *Jeff Waddell
> *Sent:* Thursday, March 17, 2016 2:53 PM
> *To:* Nate Burke
> *Cc:* voiceops@voiceops.org
> *Subject:* Re: [VoiceOps] VoIP Innovations reliability
>
>
>
> That is the issue a lot of our customers are reporting - where multiple
> calls are sent
>
>
>
> On Thu, Mar 17, 2016 at 5:49 PM, Nate Burke  wrote:
>
> I didn't see any traffic increment on the DR IP Address in my firewall
> rules, but this was odd.  During the 15 minute period, I had probably 5 or
> 6 simultaneous calls ring into my desk.  I normally only take a handfull of
> calls a day.
>
>
>
> On 3/17/2016 4:39 PM, Jeff Waddell wrote:
>
> We implemented it too - I haven't checked to see if any traffic was sent
> across it
>
>
>
>
>
>
>
> On Thu, Mar 17, 2016 at 5:20 PM, Nate Burke < 
> n...@blastcomm.com> wrote:
>
> Only 15 Minutes this time though.  I had implemented the Disaster Recover
> Trunk as mentioned previously, but I didn't seem to be getting any calls
> completed through it.
>
>
>
> On 3/17/2016 4:16 PM, Shripal Daphtary wrote:
>
> Down again!
>
> Thanks,
>
>
>
> Shripal
>
>
> On Mar 16, 2016, at 9:50 AM, Nate Burke < 
> n...@blastcomm.com> wrote:
>
> Looks like it just came back up for me.  Just over 30 min.
>
> Nate
>
> On 3/16/2016 8:45 AM, Shripal Daphtary wrote:
>
> We are experiencing an outage as well.
>
> Thanks,
>
>
>
> Shripal
>
>
> On Mar 16, 2016, at 9:36 AM, Nate Burke < 
> n...@blastcomm.com> wrote:
>
> Problems again this morning?  Looks to be acting the same as it has been.
>
> On 3/11/2016 6:00 PM, Alexander Lopez wrote:
>
> I added them to our monitoring platform, stated getting alarms this past
> hour or so.
>
> Up and down.
>
>
>
>  Original message 
> From: Nathan Anderson  
> Date: 3/11/2016 6:31 PM (GMT-05:00)
> To: 'Nate Burke' 
> , voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> ...aand we're back.
>
> -- Nathan
>
> -Original Message-
> From: VoiceOps [ 
> mailto:voiceops-boun...@voiceops.org ] On
> Behalf Of Nathan Anderson
> Sent: Friday, March 11, 2016 3:30 PM
> To: 'Nate Burke'; voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> It *feels* like they are under attack again, since I get a response to a
> ping once every 20 or so.
>
> -- Nathan
>
> -Original Message-
> From: VoiceOps [ 
> mailto:voiceops-boun...@voiceops.org ] On
> Behalf Of Nathan Anderson
> Sent: Friday, March 11, 2016 3:28 PM
> To: 'Nate Burke'; voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> Confirmed.
>
> -- Nathan
>
> -Original Message-
> From: VoiceOps [ 
> mailto:voiceops-boun...@voiceops.org ] On
> Behalf Of Nate Burke
> Sent: Friday, March 11, 2016 3:26 PM
> To: voiceops@voiceops.org
> Subject: Re: [VoiceOps] VoIP Innovations reliability
>
> Anyone else show them down again right now?  My traceroutes aren't even
> leaving Chicago.  Dying at a Chicago hop on Level3.
>
> Nate
>
> On 3/6/2016 6:50 PM, Nathan Anderson wrote:
> > Did anybody else just suffer another 45-minute-ish long outage from
> about 4:00p PST to 4:45p PST (ending about 5 minutes ago)?
> >
> > -- Nathan
> >
> > -Original Message-
> > From: 

Re: [VoiceOps] G.729 A/B Experiences

[Robert Johnson]

On 03/11/2016 07:02 PM, Calvin Ellison wrote:
> ​Could you put their voice on wires (POTS/PRI/VoIP), and the rest of
> their data on fixed wireless?
> This doesn't necessarily give you any more calls per Kbps, but at least
> keeps voice and data independent. Wires for dependability, radio waves
> for bandwidth at the cost of some latency & packet loss.

There are a number of situations in which we do just this, but in doing
so, and depending on the customers location, can price us out of the
market. Such as a customer looking for >23 channels.

> 
> One consideration when using G.729 is how you're going to deliver it to
> a mostly non-G.729 world. Are your not-quite-broadband customers
> attached to some PBX that will handle it and send G.711 to your
> carriers? That's going to cost some CPU or dedicated transcoding
> hardware. Will your providers accept G.729 and transcode for you? Is
> there a cost for it? Or will your carriers blindly throw your G.729 at
> their LCR and hope something sticks?

We don't currently connect to any carrier via VoIP, all of the equipment
to transcode G.711/G.729 -> T1 Trunking is already in place.

Even in the event where we do connect to a carrier via VoIP, we'll
transcode to G.711 (or other) before passing the call off.
> 
> 
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-- 
Robert Johnson
BendTel, Inc.
(541)389-4020
Central Oregon's Own Telephone and Internet Service Provider
www.bendtel.com/about/
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Re: [VoiceOps] G.729 A/B Experiences

[Robert Johnson]

On 03/11/2016 03:50 PM, Alex Balashov wrote:
> ‎As far as I can tell, G.729 is still the best intersection of low bandwidth 
> and call quality, although the OPUS fans have their own opinion. It certainly 
> leads to intelligible speech, though it can make for some amusing gibberish 
> when applied to hold music, given the extreme code word contractions it uses 
> to achieve its vicious compression ratio.
> 
> However, it's relatively CPU intensive and frequently requires transcoding 
> from G.711 PSTN table stakes. Moreover, in general things are going in the 
> other direction, e.g. higher bandwidth ‎HD codecs. 
> 
> This leads me to ask: why, as a North American operator, would you want to do 
> this today, in light of the capacity and price of available bandwidth today? 
> Generally speaking, G.729 is something like a niche interest for 
> international haulers and folk operating in developing world markets where 
> bandwidth remains stubbornly expensive.
> 
> --
> Alex Balashov | Principal | Evariste Systems LLC
> 1447 Peachtree Street NE, Suite 700
> Atlanta, GA 30309
> United States
> 

One of our strategies in combating QoS issues when a customer is
"off-network" is to order a dedicated 1.5/1 ADSL connection and bring it
back to our network on the ILEC's ATM network. But we quickly run out of
call capacity using G.711. Alternatively, we may order a T1, depending
on a number of items (cost, distance, others).

I'm also looking to deploy G.722, but that's another conversation.


-- 
Robert Johnson
BendTel, Inc.
(541)389-4020
Central Oregon's Own Telephone and Internet Service Provider
www.bendtel.com/about/
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[VoiceOps] G.729 A/B Experiences

[Robert Johnson]

Hey everyone,

I'm looking to deploy a lower-bandwidth codec, and am wondering what
everyone's experience has been with G.729, primary regarding voice
quality. Historically, we have limited our codec use to G.711.

Some test calls in the lab are showing promising results, I'm just
curious what might happen in the real-world.

Thank you for your time!!
-- 
Robert Johnson
BendTel, Inc.
(541)389-4020
Central Oregon's Own Telephone and Internet Service Provider
www.bendtel.com/about/
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Re: [VoiceOps] Leveraging a CLEC operating license to secure phone numbers

[Robert Johnson]

I recently jumped though a good portion of the hoops myself; I actually
found that the guys and gals at NANPA and the Pooling Administration
were really helpful. But I'll tell you the first thing they told me.

Read this document along and click through to the links. There's a lot
there and, but I feel it's worth your time to read it.

https://www.nationalnanpa.com/tools/trainGuides/Getting_Started_with_CO_Code_Assignments.pdf

With that said, Mary Lou Carey is one of the recommended contacts for
providing "Administrative Operating Company Number (AOCN)" services, and
I imagine that is a great place to start.

On 12/15/2015 03:01 PM, Kidd Filby wrote:
> I'm not sure how that's possible but... if you really have a CLEC license,
> and don't have any of this other... You have a tough row to hoe with your
> limited experience.  There are several folks on this list that do The Code
> Dance for a living.  I'll let them contact you about that.
> 
> I actually use a company called GVNW (GVNW.Com)  to outsource any of the
> code work that I need done these days.  They do an outstanding job.
> 
> As for ICA's  If you have no Telecom Lawyers... You're better off
> opting into an existing ICA the LEC has available/active on the table.
> You'll need to go through them and decide what is important to you and your
> business plan and choose the one that is most advantageous for your company.
> 
> Good luck;
> Kidd
> 
> On Tue, Dec 15, 2015 at 3:47 PM, Armand De Sio <armand.de...@gmail.com>
> wrote:
> 
>> We've recently been issued a CLEC license in the state of New Jersey and
>> I'd like to leverage that to secure phone numbers for my customers.  Who
>> would I need to contact about Interconnects and how would I go about
>> getting a block of numbers assigned to my company?
>>
>>
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>>
>>
> 
> 
> 
> 
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> 


-- 
Robert Johnson
BendTel, Inc.
(541)389-4020
Central Oregon's Own Telephone and Internet Service Provider
www.bendtel.com/about/
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Re: [VoiceOps] Preventing random SIP connections to handsets

[Robert Johnson]

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
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Re: [VoiceOps] WiFi SIP phones recommendations

[Robert Johnson]

On 09/21/2015 12:32 PM, Aaron Seelye wrote:
> On 9/19/15 8:42 AM, Jay Ashworth wrote:
>> - Original Message -
>>> From: "Alex Balashov" <abalas...@evaristesys.com>
>>
>>> On 09/08/2015 03:35 PM, Carlos Alvarez wrote:
>>>
>>>> Every Wifi phone I've tried will roam just fine between APs assuming
>>>> the APs are properly configured.
>>>
>>> Really? Can this take place seamlessly mid-call? With DHCP? What about
>>> DHCP lease acquisition delay?
>>
>> An access point is an L1 bridge; everyone on the wireless side of every
>> AP are all on the same LAN with the same addressing.
>>
>>> Would such a configuration involve merely bridging all the APs on the
>>> same LAN segment so that the same DHCP server feeds them? If so, where's
>>> the guarantee that the DHCP server will lease out the same IP address
>>> to the client?
>>
>> Access points.  Not routers.
> 
> This implies that a level of intelligence/sophistication is (or isn't)
> in the client unit that when it bounces to a new AP it wouldn't run a
> refresh on the lease.  It would be very reasonable to think that an AP
> with the same name might be on a different subnet (different
> regions/depts of a large building, or whatever), which would then render
> the call dead.
> 
> What you're suggesting is intelligent handoffs similar to a cellular
> network.
> 
> -Aaron
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Correct me if I'm wrong, but doesn't Ubiquiti's UniFi WiFi line do just
that? Is there something I'm missing when a client moves closer to AP
"B" and the network dynamically hands off the connection to AP "B"?

-- 
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] CenturyLink Toll Free Outage ?

[Robert Johnson]

On 09/03/2015 02:17 PM, Mauricio Lizano wrote:
> Anybody else seeing isses with CL Toll free?
> 
> I cant even call their repair line from an ATT landline nor from a
> TMobile cell phone
> 
> Thanks,
> Mauricio Lizano
> 
> 
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I'm unable to reach 8002237881 / Legacy Qwest / from our switch in Bend, OR.

We're also unable to reach it from a Verizon Cell or a google voice account

-- 
Robert Johnson
BendTel, Inc.
(541)389-4020
Central Oregon's Own Telephone and Internet Service Provider
http://bendtel.com/about/
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