Re: [asterisk-users] Question on resources

[dk]

Doesn’t that mean, effectively that you are using the equivalent of 100% of 2.7 
CPUs?

 

  --Don

 

 

From: asterisk-users  On Behalf Of 
Jerry Geis
Sent: Thursday, August 4, 2022 7:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Question on resources

 

I am running Asterisk 13.30.0

40 core CPU (VM) VMware.

CentOS 7

32 G ram

10G vmx network

 

Should be plenty of room for anything...

 

Yes asterisk is running 270% CPU...

Is it not taking advantage of the 40 cores ? 

I am bring around 300 SIP endpoints in a muted audio conference (so one way) 
and this spikes up the CPU to 270%.

 

Is there something I dont have set right to take advantage to the resourses?

Thanks

 

Jerry

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[asterisk-users] Just a test

[dk]

Just a test

 

  --Don

 

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Re: [asterisk-users] Between a dumb client and a capable server...

[dk]

Sorry about the top-posting...but that's what my app does!

I think it's doable, but would take me six months to work it out! Let's see who 
jumps in.

  --Don

-Original Message-
From: asterisk-users  On Behalf Of 
Antony Stone
Sent: Wednesday, August 18, 2021 10:07 AM
To: Asterisk Users' Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Between a dumb client and a capable server...

On Wednesday 18 August 2021 at 16:47:35, d...@donkelly.biz wrote:

> I think I would start by finding an open source SIP client that can manage
> calls like you want,

I can certainly find those.

> then figure out how to divide the control and audio responsibilities between
> these two SIP clients.

Do you believe it is possible for one SIP client to place a call, and for 
another one then to contact the server which is handling it and send commands 
to manage that call in progress?

I'm puzzled about how the authentication would work for identifying the call 
to the server in such a way that it thinks the request is valid, and acts upon 
it.

I can put the same SIP credentials (username & password) into two clients, but 
they'd be placing quite independent calls through the server - how could I get 
a second client to manipulate a call placed by the first one?

> Curious about why you can't just use the more capable SIP client.

It's built into a bigger application and can't just be swapped out.


Antony.

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9 of the remaining 10% are routing problems in the other direction.
The remaining 1% might be something else, but check the routing anyway.

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Re: [asterisk-users] Between a dumb client and a capable server...

[dk]

I think I would start by finding an open source SIP client that can manage 
calls like you want, then figure out how to divide the control and audio 
responsibilities between these two SIP clients. Curious about why you can't 
just use the more capable SIP client.

  --Don


-Original Message-
From: asterisk-users  On Behalf Of 
Antony Stone
Sent: Wednesday, August 18, 2021 4:33 AM
To: Asterisk Users' Mailing List - Non-Commercial Discussion 

Subject: [asterisk-users] Between a dumb client and a capable server...

Hi.

I wonder if anyone has some helpful advice or suggestions for me?

I have a very basic SIP client application, which can make and receive phone 
calls, and that's about it.  Regard it as a pretty dumb softphone.  
Unfortunately I cannot change it for a smarter one.

This client is talking to a completely capable SIP server (PBX) which can do 
all the standard PBX stuff like putting calls on hold, transferring them, 
conferencing, etc.

The problem is that the simple SIP client cannot itself tell the server to do 
any of these things - it can send an INVITE to place a call, and it can 
REGISTER and then accept an INVITE to receive a call, but it doesn't know how 
to send any other commands to the server to "manage" calls once they're in 
progress.

I'm looking for something which I can place in the network path between the 
client and the server, which can send these call control commands on to the 
server, so that it can then put calls on hold, transfer them, etc.

I'm assuming this "thing" needs to sit in the network path, so that it sees 
the INVITEs and OKs and is then aware of the Call-IDs and sequence numbers, 
etc, and can therefore present the correct call reference to the SIP server 
when it wants to say "please put this one on hold".  I have full access to the 
SIP credentials used to authenticate the client to the server.

I had thought that Kamailio might be what I was looking for, but I've asked on 
their mailing list and people are telling me that it isn't, and that I need 
something like Asterisk to do this.  I'm trying to get some specifics from them 
about *how* I would get Asterisk to do this (because I personally can't see 
how Asterisk could sit between a SIP client and a SIP server, and generate 
commands to manipulate the RTP stream and send them to the server, which is 
what the Kamailio people are saying I should do), but I thought it was worth 
asking here just in case what they're telling me is in fact quite easy when 
you only know enough about Asterisk.

So, if someone here thinks this is possible using Asterisk, please could you 
point me at some documentation showing what commands I would use or the basics 
of how I should go about it?

If anyone thinks there is another, perhaps better, way of achieving this, then 
I'm quite open to alternative solutions (as I say, I was initially thinking 
that Kamailio might be the way forward), so anything that shows me *how* such 
a thing might be achieved, with any tool at all, would be very welcome.

Just to summarise: I have a SIP client talking to a SIP server, and I need 
something which can send commands to that server to put calls, which were 
created by the existing client, on hold (that's the simplest scenario).  I do 
not want to build a SIP server / PBX myself which can itself perform call hold 
& transfer etc (I know how to do that with Asterisk) - I need those functions 
to be performed by the existing server.

Any constructive ideas are most welcome :)


Thanks,


Antony.

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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] Stir Shaken

[dk]

 

 

From: asterisk-users  On Behalf Of 
Saint Michael
Sent: Tuesday, July 14, 2020 2:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Stir Shaken

 

I need to point out the this is factually misleading and materially false:

"I think this, being the basis of your whole argument, is the fallacy. 

S/S is forcing people to take responsibility, for sure, but carriers
won't just let their customers leave because they don't want to sign
calls.  It will force them to make sure they know who their customers
are, and make it impossible for those customers to escape consequences if they 
misbehave."

 

There is Law of The Land that is about to take effect. Use google and search 
"stir shaken" Whoever thinks I am exaggerated is dreaming. Also: it is true 
that my service is the only one for asterisk --worldwide. The model proposed by 
Transexus (302 redirect with a new header) can't be used by Asterisk. 

But don't take my word for it:

https://issues.asterisk.org/jira/browse/ASTERISK-28924 

 



 

 

I need to point out again that this is not the forum for your business 
proposition. Please take it to the business list.

 

  --Don

 

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Re: [asterisk-users] Stir Shaken

[dk]

-Original Message-
From: asterisk-users  On Behalf Of 
Matthew Fredrickson
Sent: Monday, July 13, 2020 2:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Stir Shaken

On Mon, Jul 13, 2020 at 2:34 PM Saint Michael  wrote:
>>
>> There is a big confusion here about Stir Shaken. It is NOT a provider issue. 
>> Un fact, all providers are whasing their hands and modifying their swihtches 
>> to pass-through the Signature. They cannot sign the call because then the 
>> become the responsible party for the call before the FCC, and liable for any 
>> illegal call. Every owner of a PBX that sends calls to the network, except 
>> if you use a trunk for the likes of Vonage, needs to sign their calls. So if 
>> you send calls with any kind of dialer and use DIDs, real or "borrowed", you 
>> need to get the signature service urgently or your business will stop 
>> terminating calls. You cannot self-sign, you cannot get around it, the calls 
>> will either go to straight to voicemail or fail. Even worse, the carries wil 
>> play a fake voicemail and charge you a fee, something that some already a 
>> are doing when they detect robocallig.
>
> Don't even think about Transnexus, because they use 302 Redirect with a  
> header, and no version of Asterisk supports it.  I am the only game in the 
> world for Stir-Shaken and Asterisk. I know it sounds arrogant but it is 
> literally true. If you need to sign your calls to get through, with Asterisk, 
> you need to connect to my service. I am an approved Service Provider from the 
> FCC. If you keep thinking this is not happening, it is, and your business 
> will disappear overnight.
> The issue is that Vicidial, for example, does not provide res_odbc and 
> func_odbc, so you need to solve that first with Vicidial. Then you can apply 
> the code I provided earlier and your calls with have a legal, binding 
> signature. The carriers verify each signature and discard the ones that fail 
> the cryptography test.

Sounds like you're trying to sell/direct people towards a service that
you've created.  Feel free to do so on the -biz list but the -users
list isn't the right place for that sort of thing.

Matthew Fredirckson



He has been told before that this is not the right list. Can't someone delete 
him from the list?

  --Don



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Re: [asterisk-users] weather.agi

[dk]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, December 16, 2015 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] weather.agi

Here is a funny story.  We mostly do hotels in the Caribbean, and one of our
first clients (going on ten years now) used the sample "weather.agi" 
that used to be shipped with... asterisk@home? Trixbox?  Can't even recall
where we originally got it from.

This perl script uses festival to speak a brief weather forecast to the
caller.  We told our hotels this was a feature for guests, and assigned a
function code to execute it.  They have been using it for years.  In fact
the front desk used to just dial it in speaker mode for guests checking in.

Here is the funny part - the text for the weather forecast was coming from
an anonymous FTP site at tgftp.nws.noaa.gov.  Sometime in 2013 they stopped
updating the forecast!  So ever since then the caller would get the forecast
for June something 2013.  But because the weather in St Thomas is nearly the
same year round, NO ONE NOTICED!  Well, until just recently.

After we stopped laughing about it, they now want me to fix it.  But I can't
seem to find a new source of textual summary weather data.

So... has anyone else run into this and fixed it by chance?  Or can point me
to weather data?

Cheers,

j


Does something here work for you?
http://search.usa.gov/search?v%3Aproject=firstgov=web+service
e=nws.noaa.gov

  --Don

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Re: [asterisk-users] PTT push to talk solution

[dk]

If I had this request from a customer, I'd give them a box that sits on the
conference table with a  big red light that lights when the ptt switch is
on. Would work with a regular Plantronics, etc., headset. Check with these
folks to make it work: www.sandman.com

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bertrand
LUPART - Linkeo.com
Sent: Friday, August 7, 2015 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PTT push to talk solution

 

I understand you'd like something like this:

http://www.headsetzone.com/ptt.html
 
But with a headset, not a handset.
 
Right?
 
That is correct... 

 

You should be able to achieve this using any Quick Disconnect capable
Plantronics headset and a PTT switch like Plantronics SSP1051 :

 

http://headsetplus.com/product1081/product_info.html

 

I guess other headset vendor may have similar solutions.

 

-- 

Bertrand LUPART

 

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Re: [asterisk-users] showing sip number insted of pri number

[dk]

It is possible.

Are your SIP numbers from the same carrier as your PRI trunks? Some carriers
will not let you present a calling number that's not on your account (to
prevent spoofing).

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi
Sent: Friday, July 31, 2015 11:11 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] showing sip number insted of pri number

Hi,


I have asterisk installed on centos with phpagi. Also I have PRI card
connect to it. it's possible to show the sip number when calling from sip
number to external number thru the PRI, instead of showing the PRI number
show the sip number ?

 
Regards

-Hadi.Salem



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Re: [asterisk-users] Forward loop protection...

[dk]

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles
Sent: Wednesday, June 3, 2015 3:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Forward loop protection...

On Tuesday 02 Jun 2015, Carlos Chavez wrote:
  Ia had a server overload today because someone did a call forward 
 to their own extension.  To do a call forward I write a key called 
 CFWD with the extensión number and number to dial .  The main script 
 tests if the key/value exists and dials the number stored in the 
 database.  What is an easy way to prevent dumb people from creating a loop?

There currently is no easy way to prevent an infinite forwarding loop.  If you 
come up with one, then you might well earn yourself a Nobel Prize for solving 
the Halting Problem .

The obvious bodge is to set a hard limit on depth of recursion; if an actual 
real, live person is not reached within, say, five hops then the call should go 
to  (the originally-called party's)  voicemail.

--
AJS


Deciding on the mailbox to use is problematic! The dialed-party may be away for 
an extended period and wants voice mail handled by the forwarded-to party.

  --Don




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Re: [asterisk-users] Forward loop protection...

[dk]

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Tuesday, June 2, 2015 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Forward loop protection...

 

  Ia had a server overload today because someone did a call forward 
 to their own extension.  To do a call forward I write a key called CFWD 
 with the extensión number and number to dial .  The main script tests if 
 the key/value exists and dials the number stored in the database.  What 
 is an easy way to prevent dumb people from creating a loop?

Right after you have read the number to call forward to, compare it to the
number you are call forwarding from. If it matches, play the user an error
message and have them try again. 

And no matter what you do, the dumb people will come up with more creative
ways to tank your phone system. A large amount of my dialplan code is taking
into account the stupid things they have done and handling it properly if
they do it again. I swear, if you could harness their creativity for good
you could solve the world's problems 10 times over.

 

The loop checking is a bit more challenging than that. If Bob forwards to
Fred and Fred forwards to Sue, all is well when Bob and Fred head out for a
beer. A little later, we’re in deep doo-do0 when Sue forwards to Bob. 

 

  --Don

 

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Re: [asterisk-users] Forward loop protection...

[dk]

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Tuesday, June 2, 2015 5:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Forward loop protection...

 

 The loop checking is a bit more challenging than that. If Bob 
 forwards to Fred and Fred forwards to Sue, all is well when Bob and 
 Fred head out for a beer. A little later, we’re in deep doo-do0 when
 Sue forwards to Bob. 

 Could this possibly mean that any person who has CF set should never
 be available as CF Destination. Simple db entry/check can have this done. 

That just goes to show that the problem can get complex pretty quickly. Using 
the original example above, it might be that you want to allow the Bob to Fred 
to Sue forwards, but only stop it if the Sue to Bob link is established, thus 
creating the loop. I wonder if you could do some kind of recursive check where 
you follow each forward and if you ever come back around to a number you have 
already checked you know there is a loop. 

To reuse the example above, on the creation of the Bob to Fred forward, the 
database is checked to see if Fred has any forwards. He doesn't, so is at the 
end of the forwarding chain. Now Fred forwards to Sue. Again, she is at the end 
of the chain, so it is allowed. When Sue goes to forward to Bob, the check 
shows that Bob has a forward. Not a problem, but we create a temporary list 
that has Sue's number in it. Then we check the next stage of forwarding. Bob 
forwards to Fred. Fred's is checked against our temporary list and doesn't 
match, so we are still good. Bob's number is now added to the temporary list 
and we check the forward Fred has in place. Fred forward's to Sue. We check 
Sue's number against the temporary list and it does exist. Thus we have a loop 
detected and the forward can now be denied. 

I am guessing with the recursion involved you might want to do the check 
outside of Asterisk and pass the result back in. I will also state that I have 
not had to do this deep checking in the past, so these are just some initial 
thoughts on how I would start approaching the problem. Of course, this also 
assumes that Bob, Fred, and Sue are all on the same phone system. If you don't 
have a shared database to look at, the problem just got harder indeed. 

 

Classic Centrex forwarding required that the target party answer or that call 
forwarding is forced by repeating the attempt. If calls were provisionally 
forwarded and an attempt was made to complete a call from the forwarding party 
to the target, it wouldn’t be too challenging to see if the forwarded call ends 
up at the initiating extension. This would be effective even if there were 
multiple phone systems involved (if caller ID wasn’t messed with).

 

  --Don

 

 

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Re: [asterisk-users] PRI Callerid Passthrough

[dk]

This depends on what you mean by “not involving the service provider.”

 

If you are literally forwarding calls that come in on the PRI back out on the 
PRI, the most efficient way is with Two B-Channel Transfer (TBCT). Check it out 
in the wiki.

 

You need to make sure your carrier supports the feature.

 

When you want to do a “transfer,” you have an incoming call alerting or 
answered, you initiate an outgoing call (using the originating ANI). You 
initiate the TBCT and the CARRIER completes the transfer, disconnecting both of 
your B channels. The carrier will later notify you when the transferred call is 
done, but I don’t think Asterisk handles this directly. 

 

Note that at least one of the calls must be answered when you initiate the 
transfer. If you are doing “unattended” transfer, you will, typically, leave 
the incoming call alerting until the outbound call answers, then complete the 
transfer. An “attended” transfer would generally answer the incoming call, play 
a message, do some IVR doodling, or chat with an agent then initiate the 
transfer.

 

Have fun

 

   --Don

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rizwan H Qureshi
Sent: Wednesday, March 18, 2015 7:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PRI Callerid Passthrough

 

Thanks AJ and David,

We were actually using GSM gateways by setting busy forward number on the SIMs 
and just giving busy signal on every incoming call, telco took care of the 
forwarding and the line was free within seconds. Now we need to scale up the 
setup but GSM gateways a very very expensive if we want to scale upto a 1000 
DIDs, which means thousand SIMs and a gateway/gateways big enough.

 

 

 

On Wed, Mar 18, 2015 at 3:43 PM, A J Stiles asterisk_l...@earthshod.co.uk 
wrote:

On Wednesday 18 Mar 2015, Rizwan H Qureshi wrote:
 Hi All,
 I have to forward incoming call on PRI back out to PRI but I need the
 original Callerid to passthrough. Is it possible with DAHDI PRI cards
 without involving the service provider?

 Thanks

It depends who your service provider is!

Any PRI card can send the commands down the D-channel to set any caller ID you
like, but it's still up to the telco whether or not they will honour your
request.  I know the hard way that BT will only let you identify with a number
you're entitled to use.

Also, remember if you have a call coming in on a PRI line and going out on
another PRI line, that's eating two of your thirty lines .

--
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .


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-- 

Best Ragards

Rizwan H Qureshi

 

V: +971 (0) 528272154

linkedin.com/in/rhqureshi

 

 

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Re: [asterisk-users] PRI Callerid Passthrough

[dk]

The way it's expected to work:

 

Inbound call to our toll-free number, we pay for the call TO US until
terminated

 

Inbound call to our local number, caller pays for the call TO US until
terminated (if long distance charges apply for caller)

 

In either case, we pay for the outbound call if long distance charges apply

 

As a practical matter, some carriers don't have this figured out very well,
so anything could happen!

 

  --Don

 


On 03/18/2015 10:02 AM, Jeff LaCoursiere wrote:



Got a question for you - with TBCT, who pays for the call once it is
transferred?  Still me as the owner of the trunk?




 

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Re: [asterisk-users] Investigating international calls fraud

[dk]

It's very unlikely that this was an employee calling Mom for 66 hours (I'm
assuming these calls appeared on a single bill). It's also unlikely that
someone inside would benefit financially from making these calls. (Follow
the money!) Don't discount the possibility that you've overlooked something
in the firewall.

Meanwhile, does the client need to do international calling? If not, they
may request that international calls be blocked by the carrier; once
blocked, any international calls are the carrier's responsibility, not the
client's.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Platt
Sent: Thursday, January 29, 2015 12:11 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Investigating international calls fraud

 Hmm the calls are made during the day (and sometimes very early in the 
 morning). Right now it looks like someone actually made these calls. 
 If that is the case it's somewhat comforting to know the system wasn't 
 compromised. However, the $25,000 phone bill still remains. Yikes. 
 $6.25 per minute to Cambodia seems quite steep to me.

Since the Mitel had a default admin password, it seems possible that
somebody accessed its UI over the network, and then accessed and copied its
SIP credentials for your Asterisk server.

If that's the case, the calls might not have been placed through the phone.
The miscreant could have configured the purloined credentials into another
hardphone, or a softphone app on any PC or tablet or cellphone which was
able to access your LAN.
The cloned phone would not have needed to actually register with
Asterisk... it could simply have send an INVITE to place a call, and
Asterisk would have challenged it and then accepted the credentials.

If your CDR log shows IP addresses for each call, you might be able to
compare these with your DHCP (or whatever) IP registration service, and see
if the calls actually came through the phone or not.  If not you might be
able to identify the device which initiated the calls.

The bad news is, I suspect that you're probably on the hook for the cost
of the calls.  In the case of an inside job it's often hard to
legitimately disavow the charges.  You may have to pay the bill and then
(if you can identify whomever placed the unauthorized calls) attempt to
recover the cost from him/her in court.  This sort of misused by an insider
might be theft by conversion.



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