On 6/10/22 03:51, Brian Turnbow wrote:
On 2022-06-09 14:42, Peter Beckman wrote:
Maybe a better question: When providing SIP Trunk Failover as a
feature, what do customer accounts use more often -- per TN or everything?
Everything. It is rare for the customer to request different failover
Many Adtran NetVanta and Total Access 9xx boxes can convert a PRI to SIP
or SIP to PRI.
On 9/2/20 12:56 PM, Christopher Aloi wrote:
We have a customer who needs a PRI to SIP gateway to handle some legacy
applications and a service provider to handle their calls (1T!1,
~25k/month). It is not
I use data24-7.com as the backend for an internal carrier lookup app.
TELEPHONE NUMBER LOOKUP REPORT
TELEPHONE NUMBER: 1-909-701-
WIRELESS: Y
OCN: 6529
SMS: 909701x...@tmomail.net
MMS: 909701x...@tmomail.net
CARRIER NAME: T-Mobile USA, Inc.
On 7/19/20 4:18 PM,
I've seen similar issues with Polycom phones, the fix was to set
voIpProt.SIP.strictUserValidation to "1". I don't use any other
brands, so I don't know about others.
On 08/08/2018 01:43 PM, Carlos Alvarez wrote:
Do most of you have the phones authenticate incoming calls? We haven't
been,
I use data24-7.com
Request:
https://api.data24-7.com/v/2.0?user===json=C=18504901495=sms_address,mms_address,ocn
Response:
{
"response": {
"results": [
{
"status": "OK",
"number": "18504901495",
"wless": "y",
"carrier_name": "Verizon
A few years ago I hacked together something in PHP which we use
internally, it uses pcap2msc and msgen.
I can't release the code, but the snippet below should give you a good
idea..
system("python /usr/local/bin/pcap2msc $pcap_filename_in sip |
mscgen -T png -o $png_filename_out >
This accomplishes "only allow INVITES from SIP registrar" on Polycom
SoundPoint IP and VVX phones.
In a SIP environment many UA's have a feature "only allow INVITES from
SIP registrar" or words to that effect. Another possibility might be
to challenge incoming INVITEs (and other SIP
Would either of these be sufficient? Both have nationwide footprint.
http://www.level3.com/en/solutions/sip-voice-complete/
https://www22.verizon.com/wholesale/solutions/solution/SIP%2BGateway%2BService.html
*From:* VoiceOps [mailto:voiceops-boun...@voiceops.org] *On Behalf Of
*Ryan Finnesey
We use the http://www.data24-7.com/carrier24-7.php We use the product to
look up carrier names, the returned data includes wireless or landline
information. I'm not affiliated with them.
On 8/18/2015 16:30, Carlos Alvarez wrote:
I have a customer in market research who is legally required
I thought PDD / Post Dial Delay was the time between the caller dialing and the
caller hearing ringback. If that is correct you will *never* get PDD info out
of Asterisk. Asterisk only tracks the time between dialing and answer.
-Original Message-
From: VoiceOps
You are referring to the various SIP timer options in Asterisk?
-Original Message-
From: Jesse Howard [mailto:jhow...@shoretel.com]
Sent: Tuesday, April 21, 2015 11:38 AM
To: Eric Wieling; Richard Jobson; Peter Beckman
Cc: VoiceOps
Subject: RE: [VoiceOps] Easy ways to measure PDD
My
We don’t do unlimited. We include a generous (3,000 I think) block of mins in
the monthly service cost. Customers seldom exceed their allowance and when
they do we don’t have to eat the overage costs.
From: VoiceOps [mailto:voiceops-boun...@voiceops.org] On Behalf Of Shripal
Daphtary
Sent:
I get the impression from the thread below that Polycom has something special
with “BroadSoft® Synergy (used to be Sylantro)”. Stumbled across it a couple
of years ago when I was trying to make Asterisk set the Polycom CF and DND
screen indications. If they have done special support for CF
A well spec'd Asterisk box can handle well 500+ calls if audio is not going
through Asterisk.The drawback to Asterisk is you have to add lots of extra
stuff. The few GUIs availabe for Asterisk are all designed for SMBs, not for a
carrier. I love Asterisk, but it would not come close to
Some people, when confronted with a ringback problem, think I know, I'll use
the 'r' option to Dial. Now they have two problems. -- with apologies to
Jamie Zawinski
I don't think the 'r' option to Dial has solved anything for anyone in many
years. Asterisk will normally generate ringback
Peiople get confused when you put them on hold and they hear their own hold
music/messages instead of yours. If you are worred about saving 64K of
bandwidth then you have far more serious issues than which side plays the hold
music..
-Original Message-
From: VoiceOps
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