Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

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 ”tre” ( https://www.tre.se ) has in sweden, one of sweden’s largest 
phone operators, they are 4th the largest phone operator. (1: Telia, 2: Tele2, 
3: Telenor, 4: Tre)

 

Then you understand why I wonder WTF people are doing… 

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
 För d...@donkelly.biz
Skickat: den 12 mars 2021 03:14
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' 

Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 us.

 

It clearly isn’t a big deal for you, so you probably don’t have much to add to 
the discussion.

 

--Don

 

 

From: asterisk-users mailto:asterisk-users-boun...@lists.digium.com> > On Behalf Of Sebastian 
Nielsen
Sent: Thursday, March 11, 2021 7:21 PM
To: 'Mailing List' mailto:asterisk-users@lists.digium.com> >
Subject: Re: [asterisk-users] STIR/SHAKEN

 

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: For me, its seems like hunting for nano-cents. I checked around when I got 
my DID and call account for my own personal use, and the prices aren’t that 
different. Its really not worth the effort for what you save. Checked with 
several operators and the prices are almost the same per minute, its like one 
operator has like 0.016 per minute and another has 0.014 … not gonna save much 
on that. Might save like 1$-2$ per month on choosing the latter operator.

 

3: Why? Consolidiate all your agreements to 1 single operator that handles 
everything, and everything will be so much simpler. Then you are simply a trunk 
ccustomer to that particular operator, no need to handle all this with signing 
and certificates and everything..

To save a little tiny nano-cent from each minute of call..

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Joel Serrano
Skickat: den 12 mars 2021 01:52
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com> >
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 
cost/risk/etc.

2- What happens if company1 has very good pricing for DIDs, but extremely high 
rates for placing outbound calls, and company2 has super aggressive pricing for 
the destinations you use most, but sells DIDs very expensive? Mix and match? :)

3- What do you do, when instead of having 1 outbound carrier, you have several 
50? 

 

At the end I think you are mistakenly comparing apples to oranges, your DID 
provider has nothing to do with your outbound carrier, can the DID provider 
also give you outbound calling? Most likely, but that doesn't mean that the 
best way to go is to route outbound calls via the carrier that is providing you 
DIDs.

 

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

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 limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com 

Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

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: For me, its seems like hunting for nano-cents. I checked around when I got 
my DID and call account for my own personal use, and the prices aren’t that 
different. Its really not worth the effort for what you save. Checked with 
several operators and the prices are almost the same per minute, its like one 
operator has like 0.016 per minute and another has 0.014 … not gonna save much 
on that. Might save like 1$-2$ per month on choosing the latter operator.

 

3: Why? Consolidiate all your agreements to 1 single operator that handles 
everything, and everything will be so much simpler. Then you are simply a trunk 
ccustomer to that particular operator, no need to handle all this with signing 
and certificates and everything..

To save a little tiny nano-cent from each minute of call..

 

Från: asterisk-users-boun...@lists.digium.com 
 För Joel Serrano
Skickat: den 12 mars 2021 01:52
Till: Asterisk Users Mailing List - Non-Commercial Discussion 

Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 
cost/risk/etc.

2- What happens if company1 has very good pricing for DIDs, but extremely high 
rates for placing outbound calls, and company2 has super aggressive pricing for 
the destinations you use most, but sells DIDs very expensive? Mix and match? :)

3- What do you do, when instead of having 1 outbound carrier, you have several 
50? 

 

At the end I think you are mistakenly comparing apples to oranges, your DID 
provider has nothing to do with your outbound carrier, can the DID provider 
also give you outbound calling? Most likely, but that doesn't mean that the 
best way to go is to route outbound calls via the carrier that is providing you 
DIDs.

 

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

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 limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com  
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   

Re: [asterisk-users] STIR/SHAKEN

[John Millican]

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 
Local telco has the DID but the LD does not so I have to verify the DIDs 
with the Voip provider(s).

Another case may be for least cost routing.
There are other reasons but you can see that it is not always as simple 
as using the same provider for DID and origination.

Thanks,
John

On 3/11/21 3:34 PM, Sebastian Nielsen wrote:


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 limits your account to only present 
your DID as outgoing number.


Seems to be a unneccesary complicated solution just to have your 
numbers at company 1 and have your call termination at company 2.


So fricking unneccessary.

What I know there is requirements of number portability, so as long as 
company 2 can handle DIDs (ergo ”own” DIDs) you should be able to move 
your DIDs from company 1 to company 2 – then company 2 owns your DIDs.


Best regards, Sebastian Nielsen

*Från:* asterisk-users-boun...@lists.digium.com 
 *För *Alexander Perkins

*Skickat:* den 12 mars 2021 01:23
*Till:* asterisk-users@lists.digium.com
*Ämne:* Re: [asterisk-users] STIR/SHAKEN

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 necessarily the one that issued us the telephone 
numbers.  So, they will not sign it or simply cannot sign it.  
Remember that a very limited number of companies can actually sign the 
calls; the rest have to buy it from these 'Service Providers'.


And there is another situation - the company you purchase your numbers 
from and the company you place your calls through may be different and 
both may not be able to sign your calls.  Again, a very limited number 
of service providers that can actually sign your calls. So what do you 
do in that scenario?  You have to find a Service Provider that can:


1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

So, that's that...  I hope this makes sense.

Alex




-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Sebastian Nielsen]

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 limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
 För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex



smime.p7s
Description: S/MIME Cryptographic Signature
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

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 in the CDRs.


Jeff LaCoursiere
StratusTalk, Inc.

On 3/11/21 6:21 PM, Alexander Perkins wrote:
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 necessarily the one that issued us the telephone 
numbers.  So, they will not sign it or simply cannot sign it.  
Remember that a very limited number of companies can actually sign the 
calls; the rest have to buy it from these 'Service Providers'.


And there is another situation - the company you purchase your numbers 
from and the company you place your calls through may be different and 
both may not be able to sign your calls. Again, a very limited number 
of service providers that can actually sign your calls.  So what do 
you do in that scenario?  You have to find a Service Provider that can:


1.  Verify you own that telephone number(s).
2.  Sign your calls.
3.  Provide you with the technical means to do so.

So, that's that...  I hope this makes sense.

Alex



--
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Jeff LaCoursiere]

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 "owns" them, and not worry about this.


Jeff LaCoursiere
StratusTalk, Inc.

On 3/11/21 8:12 PM, d...@donkelly.biz wrote:


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 us.


It clearly isn’t a big deal for you, so you probably don’t have much 
to add to the discussion.


--Don

*From:* asterisk-users  *On 
Behalf Of *Sebastian Nielsen

*Sent:* Thursday, March 11, 2021 7:21 PM
*To:* 'Mailing List' 
*Subject:* Re: [asterisk-users] STIR/SHAKEN

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: For me, its seems like hunting for nano-cents. I checked around 
when I got my DID and call account for my own personal use, and the 
prices aren’t that different. Its really not worth the effort for what 
you save. Checked with several operators and the prices are almost the 
same per minute, its like one operator has like 0.016 per minute and 
another has 0.014 … not gonna save much on that. Might save like 1$-2$ 
per month on choosing the latter operator.


3: Why? Consolidiate all your agreements to 1 single operator that 
handles everything, and everything will be so much simpler. Then you 
are simply a trunk ccustomer to that particular operator, no need to 
handle all this with signing and certificates and everything..


To save a little tiny nano-cent from each minute of call..

*Från:*asterisk-users-boun...@lists.digium.com 
 
> *För *Joel Serrano

*Skickat:* den 12 mars 2021 01:52
*Till:* Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>

*Ämne:* Re: [asterisk-users] STIR/SHAKEN

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 cost/risk/etc.


2- What happens if company1 has very good pricing for DIDs, but 
extremely high rates for placing outbound calls, and company2 has 
super aggressive pricing for the destinations you use most, but sells 
DIDs very expensive? Mix and match? :)


3- What do you do, when instead of having 1 outbound carrier, you have 
several 50?


At the end I think you are mistakenly comparing apples to oranges, 
your DID provider has nothing to do with your outbound carrier, can 
the DID provider also give you outbound calling? Most likely, but that 
doesn't mean that the best way to go is to route outbound calls via 
the carrier that is providing you DIDs.


On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen > wrote:


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 limits your account to only
present your DID as outgoing number.

Seems to be a unneccesary complicated solution just to have your
numbers at company 1 and have your call termination at company 2.

So fricking unneccessary.

What I know there is requirements of number portability, so as
long as company 2 can handle DIDs (ergo ”own” DIDs) you should be
able to move your DIDs from company 1 to company 2 – then company
2 owns your DIDs.

Best regards, Sebastian Nielsen

*Från:*asterisk-users-boun...@lists.digium.com

mailto:asterisk-users-boun...@lists.digium.com>> *För *Alexander
Perkins
*Skickat:* den 12 mars 2021 01:23
*Till:* asterisk-users@lists.digium.com

*Ämne:* Re: [asterisk-users] STIR/SHAKEN

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 necessarily 

Re: [asterisk-users] STIR/SHAKEN

[dk]

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 us.

 

It clearly isn’t a big deal for you, so you probably don’t have much to add to 
the discussion.

 

--Don

 

 

From: asterisk-users  On Behalf Of 
Sebastian Nielsen
Sent: Thursday, March 11, 2021 7:21 PM
To: 'Mailing List' 
Subject: Re: [asterisk-users] STIR/SHAKEN

 

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: For me, its seems like hunting for nano-cents. I checked around when I got 
my DID and call account for my own personal use, and the prices aren’t that 
different. Its really not worth the effort for what you save. Checked with 
several operators and the prices are almost the same per minute, its like one 
operator has like 0.016 per minute and another has 0.014 … not gonna save much 
on that. Might save like 1$-2$ per month on choosing the latter operator.

 

3: Why? Consolidiate all your agreements to 1 single operator that handles 
everything, and everything will be so much simpler. Then you are simply a trunk 
ccustomer to that particular operator, no need to handle all this with signing 
and certificates and everything..

To save a little tiny nano-cent from each minute of call..

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Joel Serrano
Skickat: den 12 mars 2021 01:52
Till: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com> >
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 
cost/risk/etc.

2- What happens if company1 has very good pricing for DIDs, but extremely high 
rates for placing outbound calls, and company2 has super aggressive pricing for 
the destinations you use most, but sells DIDs very expensive? Mix and match? :)

3- What do you do, when instead of having 1 outbound carrier, you have several 
50? 

 

At the end I think you are mistakenly comparing apples to oranges, your DID 
provider has nothing to do with your outbound carrier, can the DID provider 
also give you outbound calling? Most likely, but that doesn't mean that the 
best way to go is to route outbound calls via the carrier that is providing you 
DIDs.

 

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen mailto:sebast...@sebbe.eu> > wrote:

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 limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com  
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign 

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Naveen Albert]

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 trunking. I don't use SIP for inter-switch trunking but I do for 
terminating lines. Maybe IAX2 and SIP handle Caller ID differently.

The CALLERID name and numbers
aren't passed properly ONLY when presence is "hidden".
Sorry, by this you mean that Asterisk does not see the number or that 
the telephone set does see it? It sounded like the former.

One thing to note is that prohib and unavailable function differently.
I looked at my code, and if a subscriber does not subscriber to Caller 
ID, I do something like the following:
same => 
n,ExecIf($["${GOSUB_RETVAL}"="1"|"${GOSUB_RETVAL}"="2"]?NoOp():Set(CALLERID(pres)=unavailable))


unavailable and prohib options are interpreted differently by different 
SIP clients (e.g. ATAs or softphones). I did some testing and found that 
there was inconsistent behavior with one option vs. the other.

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.
It may not be Asterisk but the telephone itself. I tested with a soft 
phone and it shows Anonymous . However, an actual telephone 
set shows "PRIVATE", regardless of which option I choose. But, this is 
just a hunch, and I'm not certain.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Joel Serrano]

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 cost/risk/etc.
2- What happens if company1 has very good pricing for DIDs, but
extremely high rates for placing outbound calls, and company2 has super
aggressive pricing for the destinations you use most, but sells DIDs very
expensive? Mix and match? :)
3- What do you do, when instead of having 1 outbound carrier, you have
several 50?

At the end I think you are mistakenly comparing apples to oranges, your DID
provider has nothing to do with your outbound carrier, can the DID provider
also give you outbound calling? Most likely, but that doesn't mean that the
best way to go is to route outbound calls via the carrier that is providing
you DIDs.

On Thu, Mar 11, 2021 at 4:34 PM Sebastian Nielsen 
wrote:

> 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 limits your account to only present your
> DID as outgoing number.
>
>
>
> Seems to be a unneccesary complicated solution just to have your numbers
> at company 1 and have your call termination at company 2.
>
> So fricking unneccessary.
>
>
>
> What I know there is requirements of number portability, so as long as
> company 2 can handle DIDs (ergo ”own” DIDs) you should be able to move your
> DIDs from company 1 to company 2 – then company 2 owns your DIDs.
>
>
>
> Best regards, Sebastian Nielsen
>
>
>
> *Från:* asterisk-users-boun...@lists.digium.com <
> asterisk-users-boun...@lists.digium.com> *För *Alexander Perkins
> *Skickat:* den 12 mars 2021 01:23
> *Till:* asterisk-users@lists.digium.com
> *Ämne:* Re: [asterisk-users] STIR/SHAKEN
>
>
>
> 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 necessarily the one that issued us the telephone numbers.  So, they
> will not sign it or simply cannot sign it.  Remember that a very limited
> number of companies can actually sign the calls; the rest have to buy it
> from these 'Service Providers'.
>
>
>
> And there is another situation - the company you purchase your numbers
> from and the company you place your calls through may be different and both
> may not be able to sign your calls.  Again, a very limited number of
> service providers that can actually sign your calls.  So what do you do in
> that scenario?  You have to find a Service Provider that can:
>
>
>
> 1.  Verify you own that telephone number(s).
>
> 2.  Sign your calls.
>
> 3.  Provide you with the technical means to do so.
>
>
>
> So, that's that...  I hope this makes sense.
>
>
>
> Alex
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Telium Technical Support]

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 
route.  

 

And that’s the problem being solved.  STIR/SHAKEN makes it tough for spoofers, 
but also tough for businesses doing LCR.  Sadly, the easier it becomes to 
implement STIR/SHAKEN (telling the next hop along the route to trust your 
identity), the easier it will be for spoofers to do the same.  I suspect it 
won’t be long until unscrupulous service providers undermine STIR/SHAKEN 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Sebastian Nielsen
Sent: Thursday, March 11, 2021 7:34 PM
To: 'Mailing List' 
Subject: Re: [asterisk-users] STIR/SHAKEN

 

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 limits your account to only present your DID 
as outgoing number.

 

Seems to be a unneccesary complicated solution just to have your numbers at 
company 1 and have your call termination at company 2.

So fricking unneccessary.

 

What I know there is requirements of number portability, so as long as company 
2 can handle DIDs (ergo ”own” DIDs) you should be able to move your DIDs from 
company 1 to company 2 – then company 2 owns your DIDs.

 

Best regards, Sebastian Nielsen

 

Från: asterisk-users-boun...@lists.digium.com 
  
mailto:asterisk-users-boun...@lists.digium.com> > För Alexander Perkins
Skickat: den 12 mars 2021 01:23
Till: asterisk-users@lists.digium.com  
Ämne: Re: [asterisk-users] STIR/SHAKEN

 

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 necessarily the one that issued us the telephone numbers.  So, they will 
not sign it or simply cannot sign it.  Remember that a very limited number of 
companies can actually sign the calls; the rest have to buy it from these 
'Service Providers'.  

 

And there is another situation - the company you purchase your numbers from and 
the company you place your calls through may be different and both may not be 
able to sign your calls.  Again, a very limited number of service providers 
that can actually sign your calls.  So what do you do in that scenario?  You 
have to find a Service Provider that can:

 

1.  Verify you own that telephone number(s).

2.  Sign your calls.

3.  Provide you with the technical means to do so.

 

So, that's that...  I hope this makes sense.  

 

Alex

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[phreak]

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 presentation transmitting as supposed to. This is with 
manual Set(CALLERID(pres)=something) and then seeing if it gets honored 
on the other end. The remote Asterisk gets the full number, of course, 
but the called telephone does not display it. Perhaps the u option is 
intended for lines, not trunks, and so the number never gets sent?


The only time I've found I (may) need to explicitly account for 
presentation is if I am regenerating the call, and then this needs to be 
accounted for in the call file (ignore that if that makes no sense).


The only inconsistency I've encountered has to do with presentation 
mismatches between the name and the number. If, for instance, I want the 
number to display but not the same, setting the presentations as the 
documentation would suggest does not work. The behavior is inconsistent 
between different SIP clients and it didn't work for me in any logical 
way. I didn't bother to a file a bug report about it, as I worked around 
this by simply doing Set(CALLERID(name)=) to empty the name and write 
the original name back into the variable after the call. Your mileage 
may vary.


NA

On 3/11/2021 2:22 PM, Mike wrote:


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 same 
switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI link.


When I receive a hidden number (i.e. “presentation prohibited”) call 
on Asterisk A through PRI, I get the following Caller ID information 
(using 444-555- as example):


“ <444555>”

And

CallerID presence is received as “prohibited_not_screened”.

Which is fine – I know the incoming number BUT I am told not to show 
it to the end user. All good.


The problem is when calls are not processed on Asterisk A, but sent to 
Asterisk B for further processing. The dial command I used on Asterisk 
A to send calls to AsterisB is the following:


exten => s,n,Dial(SIP/AsteriskB/123,,f("" 
<444555>)u(prohib_not_screened))


Again, so far so good. But, on Asterisk B in the appropriate context, 
on extension 123, my first command is a Verbose to show Callerid(all) 
and the received called id is shown as “Anonymous ” with 
CALLERID presence still “prohib_not_screened”. I would like Asterisk B 
to receive the actual callerid (“ <444555>”) along with the 
appropriate CallerID presence value (which is correct already).


Basically I want to “pass forward” both CALLERID and CALLERIDPRES 
exactly as received on AteriskA to AsteriskB so that AsteriskB gets 
the exact same info AsteriskA had in the first place.


How do I accomplish this?

Michael



--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
 https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] STIR/SHAKEN

[Alexander Perkins]

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 necessarily the one that issued us the telephone numbers.  So, they
will not sign it or simply cannot sign it.  Remember that a very limited
number of companies can actually sign the calls; the rest have to buy it
from these 'Service Providers'.

And there is another situation - the company you purchase your numbers from
and the company you place your calls through may be different and both may
not be able to sign your calls.  Again, a very limited number of service
providers that can actually sign your calls.  So what do you do in that
scenario?  You have to find a Service Provider that can:

1.  Verify you own that telephone number(s).
2.  Sign your calls.
3.  Provide you with the technical means to do so.

So, that's that...  I hope this makes sense.

Alex
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

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 passing number



On Thu, Mar 11, 2021 at 4:50 PM Mike mailto:mich...@virtutel.ca> > 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 when presence is "hidden".

As if Asterisk decided that since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)



You probably want to set the "trust_id_outbound" option[1] to "yes".



[1] 
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377



-- 

Joshua C. Colp

Asterisk Technical Lead

Sangoma Technologies

Check us out at www.sangoma.com   and 
www.asterisk.org 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Joshua C. Colp]

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 when presence is "hidden".
>
> As if Asterisk decided that since this is a hidden number, to replace the
> number with "Anynomous " as opposed to letting the receiving
> Asterisk process it as desired with whatever logic I choose.
>
> I just tested without any u() or f() or s() functions - same result. No
> improvement or degradation with my issue. (Not sure why I had these
> options)
>

You probably want to set the "trust_id_outbound" option[1] to "yes".

[1]
https://github.com/asterisk/asterisk/blob/master/configs/samples/sip.conf.sample#L377

-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

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 since this is a hidden number, to replace the
number with "Anynomous " as opposed to letting the receiving
Asterisk process it as desired with whatever logic I choose.

I just tested without any u() or f() or s() functions - same result. No
improvement or degradation with my issue. (Not sure why I had these
options)




-Original Message-
From: phr...@phreaknet.org 
Sent: March 11, 2021 15:33
To: Mike ; asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] CallerID presentation - presentation
prohibited but still passing number

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
presentation transmitting as supposed to. This is with manual
Set(CALLERID(pres)=something) and then seeing if it gets honored on the
other end. The remote Asterisk gets the full number, of course, but the
called telephone does not display it. Perhaps the u option is intended for
lines, not trunks, and so the number never gets sent?

The only time I've found I (may) need to explicitly account for
presentation is if I am regenerating the call, and then this needs to be
accounted for in the call file (ignore that if that makes no sense).

The only inconsistency I've encountered has to do with presentation
mismatches between the name and the number. If, for instance, I want the
number to display but not the same, setting the presentations as the
documentation would suggest does not work. The behavior is inconsistent
between different SIP clients and it didn't work for me in any logical
way. I didn't bother to a file a bug report about it, as I worked around
this by simply doing Set(CALLERID(name)=) to empty the name and write the
original name back into the variable after the call. Your mileage may
vary.

NA

On 3/11/2021 2:22 PM, Mike wrote:
>
> 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 same
> switch/LAN). Asterisk A has a Digium PRI card (recent one) and a PRI
link.
>
> When I receive a hidden number (i.e. “presentation prohibited”) call
> on Asterisk A through PRI, I get the following Caller ID information
> (using 444-555- as example):
>
> “ <444555>”
>
> And
>
> CallerID presence is received as “prohibited_not_screened”.
>
> Which is fine – I know the incoming number BUT I am told not to show
> it to the end user. All good.
>
> The problem is when calls are not processed on Asterisk A, but sent to
> Asterisk B for further processing. The dial command I used on Asterisk
> A to send calls to AsterisB is the following:
>
> exten => s,n,Dial(SIP/AsteriskB/123,,f(""
> <444555>)u(prohib_not_screened))
>
> Again, so far so good. But, on Asterisk B in the appropriate context,
> on extension 123, my first command is a Verbose to show Callerid(all)
> and the received called id is shown as “Anonymous ” with
> CALLERID presence still “prohib_not_screened”. I would like Asterisk B
> to receive the actual callerid (“ <444555>”) along with the
> appropriate CallerID presence value (which is correct already).
>
> Basically I want to “pass forward” both CALLERID and CALLERIDPRES
> exactly as received on AteriskA to AsteriskB so that AsteriskB gets
> the exact same info AsteriskA had in the first place.
>
> How do I accomplish this?
>
> Michael
>

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] CallerID presentation - presentation prohibited but still passing number

[Mike]

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 same switch/LAN).
Asterisk A has a Digium PRI card (recent one) and a PRI link.

 

When I receive a hidden number (i.e. "presentation prohibited") call on
Asterisk A through PRI, I get the following Caller ID information (using
444-555- as example):

" <444555>" 

And 

CallerID presence is received as "prohibited_not_screened".

 

Which is fine - I know the incoming number BUT I am told not to show it to
the end user. All good.

 

The problem is when calls are not processed on Asterisk A, but sent to
Asterisk B for further processing.  The dial command I used on Asterisk A
to send calls to AsterisB is the following:

exten => s,n,Dial(SIP/AsteriskB/123,,f(""
<444555>)u(prohib_not_screened))

 

Again, so far so good. But, on Asterisk B in the appropriate context, on
extension 123, my first command is a Verbose to show Callerid(all) and the
received called id is shown as "Anonymous " with CALLERID
presence still "prohib_not_screened". I would like Asterisk B to receive
the actual callerid (" <444555>") along with the appropriate CallerID
presence value (which is correct already). 

 

Basically I want to "pass forward" both CALLERID and CALLERIDPRES exactly
as received on AteriskA to AsteriskB so that AsteriskB gets the exact same
info AsteriskA had in the first place.

 

How do I accomplish this?

 

 

Michael

 

 

 

 

 

 

 

 

 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP Realtime peers

[Antony Stone]

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 time in the "Status" 
column.  The "Realtime" column tells me "Cached RT".

Some of my peers are in the same data centre as the server I'm querying, and 
generally show a qualify time of 1ms; some are in another data centre and have 
qualify times around 15ms.

However, if I go to my database server and ask "select * from sippeers" I get 
the same list of peers but the "lastms" field is always zero.  I had expected 
this to show me the last qualify time in milliseconds for each peer.

Am I just totally misinterpreting what "lastms" means in the table, or do I 
need to do something else to get this value to reflect what Asterisk itself 
will tell me?


Thanks,


Antony.

-- 
"The tofu battle I saw last weekend was quite brutal."

 - Marija Danute Brigita Kuncaitis

   Please reply to the list;
 please *don't* CC me.

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users