Re: [cisco-voip] ATA 186 question

2022-02-12 Thread Lisa Notarianni
Hello All-

Just following up on my question and your responses (which I greatly 
appreciated so many).

We decided to bite the bullet and purchase ATA-192s. However, we found that as 
we investigate building by building, many endpoint/devices are no longer used 
so the quantity of ATAs needed is almost half of what we anticipated.

Basically someone removed the fax machine or no longer uses a cordless phone or 
whatever it was at the end of that connection and they never informed us over 
the years, as staff have come and gone, that the use of these devices greatly 
changed.

Thanks again,

Lisa

On Jan 27, 2022, at 7:07 PM, Lelio Fulgenzi  wrote:

 Please let us know what your final solution is. Whether one switch with one 
ATA, or 100/100.  It’s good to know.

Lelio

Sent from my iPhone

On Jan 27, 2022, at 5:46 PM, Lisa Notarianni  
wrote:



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All good ideas. Thanks again!!

On Jan 27, 2022, at 5:13 PM, James Buchanan  wrote:


Hello,

There is also the option of keeping a few of the switches that were refreshed 
to use for the ATAs.

Thanks,

James

On Thu, Jan 27, 2022 at 10:10 PM Lelio Fulgenzi 
mailto:le...@uoguelph.ca>> wrote:

Something to try is to get a cheap DLINK or Linksys switch and insert it 
between the new 9300uxm and the ata.  It might give you some options to get 
them working until you can replace them.

Sent from my iPhone

On Jan 27, 2022, at 4:41 PM, Lisa Notarianni 
mailto:lisa.notaria...@scranton.edu>> wrote:



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content is safe. If in doubt, forward suspicious emails to 
ith...@uoguelph.ca

Thank you all for your quick responses.  We will go beg for funding and replace 
the ATAs.  We cannot set speed to 10 since it is not an option according to our 
network engineer.

Lisa
From: Lelio Fulgenzi mailto:le...@uoguelph.ca>>
Sent: Thursday, January 27, 2022 4:26 PM
To: Lisa Notarianni 
mailto:lisa.notaria...@scranton.edu>>
Cc: cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] ATA 186 question


You might want to double check that. From this, 17.3, it shows 10 is a valid 
option.

https://www.cisco.com/c/en/us/td/docs/switches/lan/catalyst9300/software/release/17-3/configuration_guide/int_hw/b_173_int_and_hw_9300_cg/configuring_interface_characteristics.html#task_uw1_3wc_g1b
Sent from my iPhone


On Jan 27, 2022, at 3:45 PM, Lisa Notarianni 
mailto:lisa.notaria...@scranton.edu>> wrote:

CAUTION: This email originated from outside of the University of Guelph. Do not 
click links or open attachments unless you recognize the sender and know the 
content is safe. If in doubt, forward suspicious emails to 
ith...@uoguelph.ca

We just began Cisco switch upgrade to 9300s and ran into an unexpected issue.

We have many ATA 186s out there (about 100) and did not budget for replacement 
as we were gratefully approved for the massive overhaul of the switches.  This 
ended up being an unforeseen glitch.

We cannot get the ATA 186s to work on this switch with this IOS version:
   1 65 C9300-48UXM 17.03.04 CAT9K_IOSXE INSTALL
However, the 186s are working on this switch with this IOS version: 
 1 64 C9300-48P 16.9.5 CAT9K_IOSXE INSTALL

I know it is not an apples to apples comparison but I was wondering if anyone 
had a workaround for the newer model/IOS.

We believe the problem is that the ATA186s are hard coded at 10Mbps/half 
duplex.  Even if we change the switch port parameter from 0x to 
0x00ff  the ATA will not register.  The switch port is also configured at 
duplex auto but the slowest we can make the port is 100Mbps.

Is this not possible?  I am hoping you Cisco geniuses have a solution 😊

Lisa Notarianni
University of Scranton
Telecommunications Engineer
Infrastructure Services
800 Linden St.
Scranton PA 18510
570.941.4325

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Re: [cisco-voip] SIP to iTSP best practices

2022-02-12 Thread Matthew Huff
We have  a few fax phone numbers that have been used for 20+ years. They are in 
corporate documents and regulatory filings. Since there is a just a few, I 
bought a couple of ATA and can play around with them and move them over time. 
For personal faxing, we already use e-faxing.

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | www.ox.com
...

From: Myron Young 
Sent: Friday, February 11, 2022 5:31 PM
To: Matthew Huff 
Cc: Kent Roberts ; cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] SIP to iTSP best practices

If you can just go with E-faxing, do that because it will save you lots of 
headaches as well.


On Feb 11, 2022, at 12:43 PM, Matthew Huff mailto:mh...@ox.com>> 
wrote:

Thanks.

Our new SIP voice gateway is separate and not in production so I have plenty of 
freedom to play.

We have copper based FAX lines, not going over our PRI currently. This is 
something we are looking into though after this conversion is done.

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | www.ox.com
...

From: Kent Roberts mailto:dvx...@gmail.com>>
Sent: Friday, February 11, 2022 12:14 PM
To: Matthew Huff mailto:mh...@ox.com>>; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] SIP to iTSP best practices


Oh yeah.. one more thing...

Test faxing  a fax test is a min of 10 pages, inbound call and out 
don't just do a page and say your good.  Check T38 if your using it... if you 
have to fail back because of T38 non-compliant, is G711 working?  Does your 
faxing software do/support switchback to 711 if T38 doesn't setup.

If you have a fax machine on a ATA or whater, test to it as well.



Isn't fax dead yet? :)   good luck with your go live.


On 2/11/22 8:52 AM, Matthew Huff wrote:
Thanks for the recommendations. I have a lot to dig into. Question about the 
video disable. We have no video hardware, so  think it would be good to disable 
it before we go live. What’s the best way to disable it globally?

Is it

Voice service voip
  Sip
 Audio forced

?

Matthew Huff | Director of Technical Operations | OTA Management LLC

Office: 914-460-4039
mh...@ox.com | www.ox.com
...

From: Kent Roberts 
Sent: Thursday, February 10, 2022 6:14 PM
To: Matthew Huff ; 
cisco-voip@puck.nether.net
Subject: Re: [cisco-voip] SIP to iTSP best practices






I was part of the team that starting a large scale sip migration almost 10 
years ago.  Have moved thousand's of DID since then.  Run multiple gig circuits 
into the cube.

Recommendations:

on the link to your provider, use address outside of the route able block for 
your company.  (say you use 10.x.x.x  then use 172.16 or 192.168)  If you 
can, don't route the itsp connections on your company network, go direct to the 
routers supporting those links.  (BGP peers I would guess depending on 
carrier/build)   If you can use a dedicated router, unless is a small site  
This is important if you wind up doing any kind of call recording, or if you 
have to enable debugs during the day.

Use dedicated dial peers setup exactly for each itsp SBC link  for in and one 
for out.

Use something like the "voice class uri trunk(x) sip"  or equivalent to bind to 
the dial peers for each SBC.

This will help if you have to add additional carriers, or say acquire a 
company, or need to do special routing...

use full E164 to and from the carrier, they may only want to do 10 digit 
in/out, but that is easy enough.  (uri trunkx will help here, as the inbound 
number will be at the cube, then you can route to cucm with outbound dial peer)
From your CUCM still send the 9 or 8 or whatever for outbound, then strip on 
match in the dialpeer to Itsp.   This will keep call looping etc.

define your voice class codecs on the dialpeers... don't just assume it will 
take the default, or work as you want without it.

if the cube will never see VIDEO, disable the options.  The cube software likes 
to release bugs that cause the cube to go south with video errors.

Depending on your carrier, you may need to force G729 or G711 first, even if 
its not your preferred codec, have seen were the SBC will not negotiate a call, 
if the codecs aren't in the order the carriers SBC wants.

do not assume the carriers network will normalize the cal