Its just that it seems so unrealistic.. WHAT do you need 1M DID’s for? Give
each stone in your company driveway a own phone number?
1M DID’s = Thats 10% of the population of the country I live in. (sweden)
1M DID’s is also three times more than the amount of customers the phone
operator
1: 1M DID’s? Then I would go straight out and say you are a phone operator,
and then getting your own STIR/SHAKEN certificate shouldn’t be a problem at
all. Thats a massive amount of numbers, unrealistically many numbers for any
company ever except for those that are a phone operator.
2:
Sebastian,
There are many reasons why someone would want the DIDs provided by one
provider and outbound calls to go out via 1,2 3, or more providers.
In one of my installs I have a situation where local calls are placed
via a local telco switch but LD calls go out via a voip provider. The
I reallt don’t understand why people simply use the same operator to terminate
your calls, which also provide DIDs for you.
Then you don’t need to touch this at all, your carrier will do all the
STIR/SHAKEN handling for you, you are just a PBX customer.
And then the operator then simply
The "inbound piece" is "what do I do with the tag information"?
Should I find a way to present the fact that a call has an A rating?
Should I offer to block calls with a C rating?
It would be great to see asterisk be able to unpack this stuff and have
it available as a dialplan variable and
To be honest, that is the logic we ended up with, and are dumping our
LCR. The savings aren't worth the headache. We don't have 1M numbers,
but we have a significant number. We can't quite get down to one
carrier (and don't really want to), but we can keep outbound calls on
the carrier that
You said it in your first post when you said “I reallt don’t understand.” You
don’t understand the business that these people are in. A few people showed you
a few examples of why it’s important to use more than one carrier--and there
are other reasons that stir/shaken is a big deal for some of
On 3/11/2021 2:50 PM, Mike wrote:
Thank you for taking the time. I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call.
I should have mentioned before that the scenario I mentioned was with
IAX2
Hi,
I wanted to add some comments to Sebastian's response:
1- When you have a lot of DIDs, you can't just "port" them over from
company1 to company2. Try to have 1M or so DIDs and ask if you can just
port them. No no, not that simple. There is a process that a lot of times
is not worth the
If you operate a small PBX for a business your approach is fine.
If you operate a large PBX, or just have lots of high toll rate calls, the
price difference between carriers can add up to a lot money every day. These
operators will route their calls to whomever offers the best rate for that
I've been able to pass presentation status between tandems without
needing to do anything explicitly. This seems to be part of the Caller
ID that is transmitted without explicit intervention. Have you tested
without using the u option? I've never used the u option and not had
issues with
Hi Jeff. What exactly do you mean by the 'inbound piece'? I've spent
quite a lot of time with the folks at TILTX understanding the framework;
but I am not exactly sure what you mean by the 'inbound piece.
Greg/Doug, like many folks here, we use LCR. So, the terminating carrier
is not
THANK YOU! Case closed, that was indeed the problem.
Michael
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: March 11, 2021 15:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CallerID presentation - presentation
prohibited but still
On Thu, Mar 11, 2021 at 4:50 PM Mike wrote:
> Thank you for taking the time. I believe you misunderstood my question.
> Callerid presence is passed perfectly already, as shown through Verbose
> commands on both sides of the SIP call. The CALLERID name and numbers
> aren't passed properly ONLY
Thank you for taking the time. I believe you misunderstood my question.
Callerid presence is passed perfectly already, as shown through Verbose
commands on both sides of the SIP call. The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".
As if Asterisk decided that
Hi,
Using Asterisk 13.36.0
I have a bit of a technical issue with hidden caller IDs. My setup, at
the moment, is composed of two Asterisk boxes. In some instance, calls
arrive on Asterisk A, and are then sent to Asterisk B for further
processing. The link between them is SIP (both on the
Hi.
I'm using MariaDB (via ODBC) to store realtime SIP peers with Asterisk 13 and
16.
In general it's all working fine, however there's something that puzzles me:
If I connect to the Asterisk console and use the command "sip show peers" I
get a list of the peers including the last qualify
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