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/
/[email protected] | //www.ox.com <http://www.ox.com>/
*...........................................................................................................................................*
*From:* Kent Roberts <[email protected]>
*Sent:* Thursday, February 10, 2022 6:14 PM
*To:* Matthew Huff <[email protected]>; [email protected]
*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 calls. For
instance, if the destination is on the same carrier, its a direct
ip route via the SBC. If that end side can't accept say G729
(cheaper sbc) the call will just fail.
NEVER user debug ccsip all
debug CCSIP messages is safer, and unless your cube is peeked,
it won't add to much cpu.
make sure your CPU never exceeds 80% at the max possible peek of
routing.
Check how the calls work with MOH. Inbound and out. make sure 2
way audio remains after the on hold event..
Do you need to force early offer? (yes sounds silly, but have run
into issues where some phones had no audio unless this was set)
Ask your carrier, how they handle TFNs outbound, if you pass the
ANI from a 3rd party. (this is all billing stuff to the TFN owner)
Some may allow calls to process not caring what the number is.
Some may allow you to provide a alternate billing number.
Some will just 603 decline the call if the ANI isn't in your
number poll assigned to you.
with a 603 the cube will try the next dial peer so you can
add a header to re-write this with your number.....
Diversion headers exist, however most carriers pass them through
to the destination, and IVRs or Voice Mail systems on the far side
will try to process that information, and do unexpected things.
(the party your calling doesn't exist for example.)
define the default sip control/media source interface, this will
be your destination from cucm. The URI trucks will define the
sip control/media on the ITSP side.
If you use firewalls any where in your company, that will pass
voip... Set the rtp-port range on the cube match the smaller
range of what your going to use. (say the old days 16384-32767)
don't assume the firewall will pass all the UDP ports by default.
speaking of firewalls, check, double check, and triple check, then
do your own research if you will use them, when it comes to SIP
inspection. Some FW's have options that need to be tweeked and
defined, for the SIP port. (this may control anything from
timeouts, which media ports engage) This is especially true
with expressway in the DMZ. It might be safer to not use sip
inspection and just pass the port. But for some FWs this is not
true.
define the FAX-relay, rats and protocols for T38
ask your carrier how they handle QOS. some don't since the trunk
to them might be dedicated.
use option pings on the dial peers, so if the SBC goes away that
dialpeer disables. The sbc side just has to respond, even if its
an error saying what is this... that will keep the peer up.
Setup the event manager applet. have it email you on syslog
patterns for dialpeer status. Then you will know if the link goes
down.
if you can get a bug scrub on the version of IOS, don't be
determined to use the newest code. newest is not always best.
Hope at least one thing here was helpful.
On 2/10/22 9:09 AM, Matthew Huff wrote:
We are in the process of migrating for a legacy PTSN voice
gateway (PRI) to a new CUBE based SIP connection to a iTSP
connected via a private metro ethernet (not Internet based).
Does anyone have a good source for recipes / dial-plans
recommendations / best practices for this?
*Matthew Huff*| Director of Technical Operations | OTA
Management LLC
/Office: 914-460-4039/
/[email protected] | //www.ox.com <http://www.ox.com>/
*...........................................................................................................................................*
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