Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Do you know that it is coming back as FALSE, or are you assuming that from 
examining the expression?

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 1:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with FUNC_MATH

 

John,

 

That is correct. I am trying to figure out why Asterisk is executing the set 
part of the execif, if it's coming back as false.

 

 

 

On Thu, Feb 13, 2020 at 2:10 PM John Kiniston  wrote:

My Apologies Dovid, I think I misunderstood your request.

You don't have the time you need to convert in the format of date string, 
Instead you have your users entering via DTMF when they want something to 
happen?

 

On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender  wrote:

John,

 

>From looking at the wiki won't STRFIME just give me what I need based on the 
>unix time that I put in? What I am actually looking to do is convert over from 
>12 hour format to 24 (unless strftime does just that and I don't kow what am I 
>am doing?).

 

 

 

On Thu, Feb 13, 2020 at 12:03 PM John Kiniston  wrote:

Try using the STRFIME function instead of doing this by hand.

https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME

%H 

The hour as a decimal number using a 24-hour clock (range 00 to 23). 

%I 

The hour as a decimal number using a 12-hour clock (range 01 to 12). 

 

On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender  wrote:

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Is it stored as a floating-point number or an integer? Floating-point decimal 
numbers are often not stored precisely. As in my example, below, “12” may be 
stored as 11.999 (simplified) and, although it will be treated  as “12” 
in most cases, it will appear to be less than 12 in a comparison.

 

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 1:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Help with FUNC_MATH

 

HOUR_SELECTED is going to be 1-12

 

 

On Thu, Feb 13, 2020 at 2:05 PM Don Kelly  wrote:

Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need 
to account for that in your comparison.

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 4:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help with FUNC_MATH

 

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Help with FUNC_MATH

[Don Kelly]

Is HOUR_SELECTED a floating-point number (e.g. 11.999)? If so, you need 
to account for that in your comparison.

 

  --Don

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Dovid Bender
Sent: Thursday, February 13, 2020 4:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Help with FUNC_MATH

 

Hi,

 

I have some dialplan code that is trying to convert 12 hour time with AM/PM to 
24 hour format. The code has something like this:
Exten => 
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))

 

Earlier on in the dialplan HOUR_SELECTED is set to 12. When they press option 2 
they are selecting PM. If the time is from 1PM to 11PM then I want to add 12 to 
the number (so if it's 1 make it 13 etc.). When I run the above the logs show 
the result as false yet if the user sets HOUR_SELECTED to 12 then after this 
line of dialplan code it gets switched to 24. What am I doing wrong here?

 

The exact DP code is:

Exten => 2, 1, Noop(BEFORE CHECK HOUR_SELECTED is ${HOUR_SELECTED})
 same =>n, 
ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
 same =>n, Noop(AFTER CHECK HOUR_SELECTED IS ${HOUR_SELECTED})

 

And the output of the logs is:

[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:1] 
NoOp("SIP/204.145.219.31-81c6", "BEFORE CHECK HOUR_SELECTED is 12") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function MATH(12<12) 
result is 'FALSE'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '12'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Function 
MATH(12+12,int) result is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'ExecIf'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:2] 
ExecIf("SIP/204.145.219.31-81c6", "FALSE?Set(HOUR_SELECTED=24)") in new 
stack
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx_variables.c: Result of 
'HOUR_SELECTED' is '24'
[Feb 13 10:46:18] DEBUG[1580][C-7bc6] pbx.c: Launching 'NoOp'
[Feb 13 10:46:18] VERBOSE[1580][C-7bc6] pbx.c: Executing [2@am_pm_select:3] 
NoOp("SIP/204.145.219.31-81c6", "AFTER CHECK HOUR_SELECTED IS 24") in new 
stack

 

 

TIA.

 

Dovid

 

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Re: [asterisk-users] Half Off Topic Questions

[Don Kelly]

On Tuesday 06 March 2018 at 09:05:25, Markus Weiler wrote:

> Hi Group,
> 
> we're just wondering, in German we call the different types of 
> phone-numbers
> (Geographic,mobile,national,VoIP...) Rufnummerngassen (phone number 
> alleys
> ;-) )
> Is there an english word for this?

No.

It's just another example of German choosing to have a single word for
something, where English uses two or more.

In English I'd say "number type", but that's just my opinion - others might
describe this differently.

Just the word "Rufnummer" on its own is another example - "phone number".


Antony.

--

The North American Numbering Plan doesn't distinguish between the "number
types." With portability of numbers between mobile and the decreasing number
of landlines and the mobility of mobile numbers eliminating geographic
identity within the U.S., there's little use for classifying them.

  --Don



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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Don Kelly]

I've assumed that the client is not present when the cleaners arrive.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Nielsen
Sent: Thursday, May 11, 2017 10:19 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Personally, if I was a client, I would rather have the personell answer the
phone than make a outgoing call, if I would choose.
If you think of billing and costs.
So if a client allows outgoing, I don't think they have any problems with
answering a call immediately following either.

But I assume the client will be billed for the time the personell works
there?
And thats why you have this "phone verification system", to avoid discussion
about how long the company has been there and unfair bills?

Then you could have it this way instead:
1: Give the client (not personell) a PIN code.
2: The client calls and enters PIN.
3: The employee gets a SMS/email/push message/paging tone, that he can start
working.
4: When the employee is done, the client calls again, and enter PIN. This
will stop billing.
5: When billing is stopped, the employee gets a SMS/email/push
message/paging tone he can stop working.


This will be rock solid. The employee only needs to check for the SMSes.
The SMSes prevent the client from cheating the system to get cheaper
service, like claiming to start when client do not, or calling for stop
before the employee is finished, because the employee will only work when he
get start signal, and will stop working at stop signal.

Theres no risk that the client will call in and check in/check out when the
employee is not there, because that would cause the client to Be billed for
rendered services.


-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Don Kelly
Skickat: den 11 maj 2017 17:04
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
<asterisk-users@lists.digium.com>
Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

As a client, I don't want service company personnel answering my phone.

As a service company, I don't want my clients thinking that I do not trust
my employees who are at the client facility.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Goldberg
Sent: Thursday, May 11, 2017 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Seems like this is the best idea (challenge-response), a callback.  No
matter the callerid, you don't know where the caller is.  But if you place a
call BACK to the callerid, it's going to go to the destination.  Then you
either need the phone to be answered, or the phone to be answered and and
the challenge entered.


Adam Goldberg
AGP, LLC
+1-202-507-9900

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J Montoya or A
J Stiles
Sent: Thursday, May 11, 2017 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wednesday 10 May 2017, Steve Edwards wrote:
> On Wed, 10 May 2017, J Montoya or A J Stiles wrote:
> > Presumably your staff carry mobile phones.  What about an app that 
> > gets the ID of the cell tower to which it is connected, and passes 
> > it and the SIM number in a HTTP request to a server you control?
> 
> The problem is that they are supposed to use the 'site landline' to 
> confirm presence -- not their cell phone with the spoofed CID.

Yes; but the whole point is that the caller ID from the site landline is no
longer reliable enough as evidence, by itself, that somebody is actually
there.

A custom app could read the ID of the cell tower to which it was connected
-- or even the phone's GPS co-ordinates -- and transmit that back to base
over the Internet.  Preferrably with some sort of precautions to make the
request harder to forge  (i.e., *not* just a plain HTTP GET with the MCC,
MNC, LAC and CID in the query string).  If your app makes its connection via
the site's wi- fi  (which will require the co-operation of the client)  as
opposed to the mobile network, so much the better, as there will be an IP
address against which you can match.


If you insist to use the site landline for your authentication, you could
extend the protocol to a full challenge-and-response as follows:  Play a
series of digits down the line to the caller, return the call as soon as
they hang up, and ask them to dial the same digits they just heard.  All
this can be done in the dialplan  (you might need to record some

Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Don Kelly]

As a client, I don't want service company personnel answering my phone.

As a service company, I don't want my clients thinking that I do not trust
my employees who are at the client facility.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Goldberg
Sent: Thursday, May 11, 2017 8:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Seems like this is the best idea (challenge-response), a callback.  No
matter the callerid, you don't know where the caller is.  But if you place a
call BACK to the callerid, it's going to go to the destination.  Then you
either need the phone to be answered, or the phone to be answered and and
the challenge entered.


Adam Goldberg
AGP, LLC
+1-202-507-9900

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of J Montoya or A
J Stiles
Sent: Thursday, May 11, 2017 7:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wednesday 10 May 2017, Steve Edwards wrote:
> On Wed, 10 May 2017, J Montoya or A J Stiles wrote:
> > Presumably your staff carry mobile phones.  What about an app that 
> > gets the ID of the cell tower to which it is connected, and passes 
> > it and the SIM number in a HTTP request to a server you control?
> 
> The problem is that they are supposed to use the 'site landline' to 
> confirm presence -- not their cell phone with the spoofed CID.

Yes; but the whole point is that the caller ID from the site landline is no
longer reliable enough as evidence, by itself, that somebody is actually
there.

A custom app could read the ID of the cell tower to which it was connected
-- or even the phone's GPS co-ordinates -- and transmit that back to base
over the Internet.  Preferrably with some sort of precautions to make the
request harder to forge  (i.e., *not* just a plain HTTP GET with the MCC,
MNC, LAC and CID in the query string).  If your app makes its connection via
the site's wi- fi  (which will require the co-operation of the client)  as
opposed to the mobile network, so much the better, as there will be an IP
address against which you can match.


If you insist to use the site landline for your authentication, you could
extend the protocol to a full challenge-and-response as follows:  Play a
series of digits down the line to the caller, return the call as soon as
they hang up, and ask them to dial the same digits they just heard.  All
this can be done in the dialplan  (you might need to record some
announcements of your own, such as "Please memorise the following digits"
and "Please dial the digits you heard in the last call").  

Intercepting incoming calls *to* a number is much harder  (usually requiring
the co-operation of telcos, unless the interloper has access to some
equipment through which they know that the call will be routed; that
potentially includes your Asterisk, but any tampering there would be
evident)  than falsifying outgoing calls *from* a number.  


It would be much more fun to mount a "sting" operation to catch the 
perpetrators red-handed   (say, falsely set off a fire alarm while you know
they 
are slacking off down the pub instead of looking after the site like they
are paid for)  .  but maybe I have just been watching too many detective
dramas on TV!

--
JM

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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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Don Kelly]

It's probably not practical to have them answering the client's telephone!
At a lot of sites, incoming calls would be handled by auto attendant,
diverted to answering service, etc.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sebastian
Nielsen
Sent: Wednesday, May 10, 2017 2:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

Use a callback.
So when clocking in/out, they will hear a random 4 digit PIN, like "Enter
four, three, six, eight at the callback".
After they hangup, the phone will ring, and then they will have confirm with
the 4 digit PIN.

If they arent in presence: the phone at the site will ring, and the person
at site (that isn't your employee) cannot carelessly just OK it because they
haven't heard the PIN.
If they are in presence: the phone at the site will ring, and the employee
will be able to enter the PIN they just heard. If they fake the callerID or
not at the initial call, does not matter, since you have verified with a
callback.

-Ursprungligt meddelande-
Från: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] För Steve Edwards
Skickat: den 10 maj 2017 19:13
Till: Asterisk Users Mailing List - Non-Commercial Discussion

Ämne: Re: [asterisk-users] How to detect fake CallerID? (8xx?)

On Wed, 10 May 2017, J Montoya or A J Stiles wrote:

> Presumably your staff carry mobile phones.  What about an app that 
> gets the ID of the cell tower to which it is connected, and passes it 
> and the SIM number in a HTTP request to a server you control?

The problem is that they are supposed to use the 'site landline' to confirm
presence -- not their cell phone with the spoofed CID.

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] How to detect fake CallerID? (8xx?)

[Don Kelly]

You have an unusual situation--you suspect caller ID spoofing by a known
person. 

Under the Truth in Caller ID Act, FCC rules prohibit any person or entity
from transmitting misleading or inaccurate caller ID information with the
intent to defraud, cause harm, or wrongly obtain anything of value. Anyone
who is illegally spoofing can face penalties of up to $10,000 for each
violation.

Making it clear to your employees that spoofing will result in termination
might be enough.

Requiring employees to have a phone that you can locate would allow you to
check from time to time.

  --Don





-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Wednesday, May 10, 2017 10:12 AM
To: Asterisk Users Mailing List
Subject: [asterisk-users] How to detect fake CallerID? (8xx?)

I have a 'time and attendance' application. Think janitorial or security
kind of thing where an employee goes from location to location.

They're supposed to 'clock in' when they get to a site using a phone at that
site to prove they're there.

Some employees have discovered 'fake caller ID' services can be used to say
they're on site when they are not.

How can I detect a fake CallerID? The INVITE looks the same to me.

If I have the employees call an 8xx number, can I ask my SIP provider to
include more headers to show the real ANI? What would that service be
called?

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 https://www.linkedin.com/in/steve-edwards-4244281

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Re: [asterisk-users] Metaswitch caller ID passing

[Don Kelly]

Is this your Astribank installation? Xorcom may have an answer.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Friday, August 5, 2016 8:05 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Metaswitch caller ID passing

Hi,

I am dealing with a telco that has recently upgraded from a DMS100 switch to
a "Metaswitch", and our PRI no longer passes foreign caller ID information,
i.e. if I place an outbound call with specific caller ID information not
associated with the PRI, it gets replaced with the PRI's primary phone
number.

This is a bit of a problem for follow-me services, which end up showing the
PRI's primary phone number instead of the original caller's phone number.

I know this isn't a "metaswitch" forum, but can anyone point me in a
direction of some metaswitch documentation or know what the option is in a
metaswitch to allow foreign caller ID information?  The telco engineers are
still struggling with this new switch, and I am not sure they understand or
appreciate my urgency in getting this resolved!

Cheers,

--
j


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Re: [asterisk-users] Asterisk Phone ( Telecom feature )

[Don Kelly]

Although you may have found them rude and unhelpful, I think most of the
responses had a common theme-we need more information about what you have
and what you want to do before we can help at all.

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, October 08, 2014 1:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

This is free software supported by volunteers who are helping you because
somewhere in the past someone helped them or perhaps they were just brought
up properly.

No one really cares about how many clients you may have. No one here will
make any money from that.

Have you read the free books on Asterisk?
Have you at least followed the installation instructions to some point where
you are stuck.

I don't see where you have explained what you want to do with the PBX once
you have it set up. There are a lot of different applications possible -
Small office, enterprise with many branches, call center with thousands of
incoming or outgoing calls, etc.

Have you looked at what phones you want to support - brand, features?

Have you selected a type of trunk that your telephone company offers? Do
they have a document describing how to interface an Asterisk PBX to their
trunks?

What were you expecting a total stranger to do for you when you asked your
question?
If you want to hire a consultant to set up your first few clients and train
you, than you should say so.
Free advice comes in many forms but there is a limit about how much free
time each one of us has and you will get different people helping you
depending on the question that you ask. We are not all experts at
everything. Most of us are people like you or end-users who are supporting
their own company's phone system.

Ron


On 08/10/2014 12:34 AM, Dania Asi wrote:

Dear Mr. Adam,

 

Thank you for you kind words and for judging me.

 

I am a system integrator and I have a whale clients in UAE , I will not
proceed further in dealing with Asterisk because of the lack of support and
because of the rude emails. 

 

I have no idea what is wrong with you people. And I hope you get well soon
from whatever is happening to you.

 

 

Best Wishes,

 

Dania Abu Asi

Sales Executive Engineer





Future Trends Establishment

Abu Dhabi - U.A.E.

Mob : +971 50 4948363

Off : +971 2 6730666

Fax : +971 2 6734888

 

From: Adam Goldberg [mailto:a...@agp-llc.com] 
Sent: Tuesday, October 7, 2014 8:51 PM
To: Dania Asi
Subject: FW: [asterisk-users] Asterisk Phone ( Telecom feature )

 





I suggest that your question amounts to please do my homework for me.
This may be understandable given that you are a recent grad and probably
don't have much experience in business communication and/or Asterisk
complexities. 

 

You cannot expect a mailing list to rush to answer vague, unanswerable
questions -- nor emails that don't show that you've tried to answer the
question first.  

 

Consider, if I asked: 

 

I don't understand how to set up Asterisk.  Can someone
tell me how to do that? 

vs. 

 

I have a Dell R210-II with 32g of memory and two gigabit ethernet
interfaces, I've installed Asterisk from the FreePBX Distro v9.99 and have
an assortment of Polycom and Snom IP phones.  I've configured paging as
described in http://wiki.snom.com/Interoperability/PBX/Asterisk and
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page, and it is working for
the Snom phones but not the Polycom phones.  Can someone point me at what
it's going to take to make the Polycom phones work?

 

I'd expect to get attempts at an answer to the second one, but would expect
snide and rude comments (at best) to the first one.

 

Adam Goldberg

AGP, LLC

+1-202-507-9900

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Tuesday, October 07, 2014 9:38 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

JG confirmed that it is possible, but it has not been defined.

 

Without knowing what kind of instruments you are using, a possible it

would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.

 

  --Don

 

-Original Message-

From:  mailto:asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com

[ mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi

Sent: Tuesday, October 07, 2014 3:41 AM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'

Cc: 'Irene Galera'; 'Maysara Orabi';  mailto:moham...@futuretrendsest.com
moham...@futuretrendsest.com

Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

 

Dear Mr. Mitual,

 

Kindly

Re: [asterisk-users] Asterisk Phone ( Telecom feature )

[Don Kelly]

JG confirmed that it is possible, but it has not been defined.

Without knowing what kind of instruments you are using, a possible it
would be for a party to dial a 4-digit extension number to talk to someone
internally, completing a call without using the PRI trunks.

  --Don

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dania Asi
Sent: Tuesday, October 07, 2014 3:41 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: 'Irene Galera'; 'Maysara Orabi'; moham...@futuretrendsest.com
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

Dear Mr. Mitual,

Kindly check the attached mail where Mr. JG confirmed to me that is possible
and I already informed my client of that.


Dear Mr. Steve,

I am not expecting a mailing list to do any work for me. All I was asking is
for you to guide me because this is the first time we deal with Asterisk
phones.


Best Wishes,

Dania Abu Asi
Sales Executive Engineer

Future Trends Establishment
Abu Dhabi - U.A.E.
Mob : +971 50 4948363
Off : +971 2 6730666
Fax : +971 2 6734888

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Tuesday, October 7, 2014 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Irene Galera; Maysara Orabi; moham...@futuretrendsest.com
Subject: Re: [asterisk-users] Asterisk Phone ( Telecom feature )

On 7 Oct 2014, at 09:24, Dania Asi da...@futuretrendsest.com wrote:
 Kindly note that I asked about the capability of the phones and now I 
 am asking about the way I can do it to my client's phones, because he 
 is asking for a demonstration.

Yet you've not even told us the phones in use. You can't just expect a
mailing list to do your work for you. You need to look at the handsets and
see what they support, and how they support it. You don't even say if the
handsets are SIP or not, the PSTN connectivity is pretty irrelevant.

 Sales Executive Engineer

That explains that one.

Steve
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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Don Kelly]

On 09/23/2014 02:17 PM, Steve Edwards wrote:
 For some applications, storing recorded audio (prompts and caller
 recordings) as a BLOB in MySQL has advantages.
Jeff sez:
How about a named pipe (fifo)?  Of course then you might have issues with
simultaneous calls.  You would have to have a pool of them and somehow
manage locking them...

J

I'm curious about what the advantages are of storing audio in a blob.
Wouldn't it be more efficient to store it in a file and just put the
filename in the database?

  --Don



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Re: [asterisk-users] Playback/background audio from MySQL BLOB

[Don Kelly]

On Tue, 23 Sep 2014, Steve Edwards wrote:

  On 09/23/2014 02:17 PM, Steve Edwards wrote:

   For some applications, storing recorded audio (prompts and caller
   recordings) as a BLOB in MySQL has advantages.

 On Tue, 23 Sep 2014, Don Kelly wrote:

  I'm curious about what the advantages are of storing audio in a blob.
  Wouldn't it be more efficient to store it in a file and just put the  
 filename in the database?

 Multiple web servers, multiple Asterisk servers, multiple DB servers, 
 synchronizing filesystems vs db, etc.

 It appears to eliminate some problems, but Asterisk limiting audio 
 playback to files seems like a tough obstacle.



Mike said:
Maybe make the audio files available to all servers via a single NFS
directory?  Probably not a good solution if the servers aren't co-located.


Maybe someone could write a Linux device file that would return the blob's
content as a file read.

  --Don


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Re: [asterisk-users] PRI timing settings

[Don Kelly]

For my NI2 PRIs I've always used 10 digits for everything and no +1

 

   --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, August 20, 2014 9:41 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PRI timing settings

 

On 08/20/2014 07:58 AM, Scott L. Lykens wrote:

 

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote:





I wrote earlier today about a new PRI installation in the Caribbean, where
all outbound calls are functioning fine *except* calls to Sprint phone
numbers, which get rejected immediately as busy.

 

I don't know what expectations for CLID your carrier might have, or for that
matter the upstream carrier, however, we found through our CLEC here in the
US that while the CLEC was happy to take e.164 formatted numbers from us as
CLID, Global Crossing would reject them further upstream resulting in our
calls to many toll frees being rejected.

 

Switching to 10 digit CLID on all outbound calls through that PRI solved the
problem.

 

I don't know if this is your problem but be sure your CLID is in the most
simple format possible for your region to help rule it out.

 

sl

 


This makes me curious... what *is* the simplest format possible for NANPA
numbers?  I'm sure there must be a spec to conform to.  Can anyone point me
to it?

Cheers,

j

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Re: [asterisk-users] PRI timing settings

[Don Kelly]

It's possible that Sprint is burping on the name. Try first dropping the
1.  Then try dropping the name also, if necessary.

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, August 20, 2014 10:03 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PRI timing settings

 


What about the text portion?  Should that never be sent?  I was indeed
sending the '1', and I will remove that to see if it solves my problem, but
I also have the company name in there.  I feel like a newb asking such
questions, but I've never had this issue before :)

Company 1NXXNXX

Cheers,

j

On 08/20/2014 09:46 AM, Eric Wieling wrote:

NXXNXX is the correct format of CallerID numbers in NANPA.   The leading
1 is not part of any NANPA phone number.   Toll free area codes are also
not valid for CallerID.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Wednesday, August 20, 2014 2:41 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] PRI timing settings

 

On 08/20/2014 07:58 AM, Scott L. Lykens wrote:

 

On Aug 19, 2014, at 5:56 PM, Jeff LaCoursiere j...@jeff.net wrote:






I wrote earlier today about a new PRI installation in the Caribbean, where
all outbound calls are functioning fine *except* calls to Sprint phone
numbers, which get rejected immediately as busy.

 

I don't know what expectations for CLID your carrier might have, or for that
matter the upstream carrier, however, we found through our CLEC here in the
US that while the CLEC was happy to take e.164 formatted numbers from us as
CLID, Global Crossing would reject them further upstream resulting in our
calls to many toll frees being rejected.

 

Switching to 10 digit CLID on all outbound calls through that PRI solved the
problem.

 

I don't know if this is your problem but be sure your CLID is in the most
simple format possible for your region to help rule it out.

 

sl

 


This makes me curious... what *is* the simplest format possible for NANPA
numbers?  I'm sure there must be a spec to conform to.  Can anyone point me
to it?

Cheers,

j





 

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Re: [asterisk-users] PRI timing settings

[Don Kelly]

Doubtful that T309 or T316 are causing the problem, but you can always change 
them to correspond with their defaults.

 

http://www.nmscommunications.com/manuals/6272-16/appendxe.htm

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, August 19, 2014 4:56 PM
To: asterisk-users@lists.digium.com  Asterisk Users Mailing List - 
Non-Commercial Discussion
Subject: [asterisk-users] PRI timing settings

 

Hello,

I wrote earlier today about a new PRI installation in the Caribbean, where all 
outbound calls are functioning fine *except* calls to Sprint phone numbers, 
which get rejected immediately as busy.

The telco has been working with their switch manufacturer and took the output 
of pri show span 1 from me and came back with this:

quote---

Please check your timers below.  How did you determine your settings?

*   Timer and counter settings:
  N200: 3
  N202: 3
  K: 7
  T200: 1000
  T201: 1000
  T202: 2000
  T203: 1
  T303: 4000
  T305: 3
  T308: 4000
  T309: 6000Our switch: Telcordia National ISDN 2: Range 10 - 90 seconds, 
default 90 seconds. 

*   
  T312: 6000
  T313: 4000
  T316: -1Our switch: Telcordia National ISDN 2: Range 10 - 120 
seconds, default 30 seconds. 


  N316: 2
  T-HOLD: 4000
  T-RETRIEVE: 4000
  T-RESPONSE: 4000

---end quote---


Now I have no idea what T309 or T316 represent, but it seems odd that timers 
and counters would cause such an odd result... and the failure is immediate, 
not after some amount of timeout.  Can anyone shed light on these settings and 
tell me if they are configurable?

Cheers,

j

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Re: [asterisk-users] Loud Ringers and paging systems...

[Don Kelly]

The basic concept is that the original call will run a script that creates a
call file to call the paging system and play a specific audio file. It also
passes into the paging call its channel name. In the call to the paging
system, I use the SHARED function to write back to the original calls'
channel the channel name of the paging call. Then when the original call is
answered, it runs a subroutine that redirects the paging call to a priority
that hangs that call up. 

Will your approach handle ringing more than one of the three extensions
simultaneously?

 

  --Don

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Re: [asterisk-users] Asterisk call forward for T1 incoming calls

[Don Kelly]

On Fri, Apr 25, 2014 at 9:19 AM, Al lists asteris...@gmail.com wrote:
 Is there a way to divert incoming calls on DAHDI T1 channels so telco 
 gets the diversion and send the call to new number and releasing the
channel?

Rusty Newton:
I'm no PRI expert, but I do remember from my time working with the T1
interface cards that two B-channel transfers did something like this.

Digium has documentation on that here:
http://kb.digium.com/articles/Configuration/Two-B-Channel-Transfers

If that doesn't help, and you have a Digium card; you might call Digium tech
support to ask about it.

Don Kelly adds:
TBCT doesn't actually divert, but it will transfer a call. The
difference, which may not matter to you, is that one of the channels must be
answered by Asterisk (unless something has changed). 

A call comes in, you decide you want to divert it--you place an outbound
call on another channel. You must either answer the incoming call or wait
for the outgoing call to answer before completing the TBCT.

Another quirk to be aware of: You may wish to present the caller's number to
the called party as caller ID. Some carriers will not permit this.

Also, the op referred to T1 channels. TBCT works on PRI, not simple T1.

  --Don



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Re: [asterisk-users] Asterisk 1.6

[Don Kelly]

Shouldn't the secast discussion be on the commercial list?

 

Note that their free version works for five simultaneous calls-then the
price goes 'way up.

 

  --Don

 

(Top posting 'cause that's what's already being done.)

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz
Sent: Friday, April 04, 2014 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.6

 

that sounds feasible, Thanks Michelle, 

 

 

 

On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote:

If you know your users are all from with your country, or state, or even
city, you could restrict geographic access in your secast.conf file like
this:

 

ruledefault=deny

ruleexceptions=NA:CA:Ontario:|NA:US:Michigan:Detroit|::Ohio:|NA

 

The above would:

- By default deny all source IP's anywhere in the world

- Let in only source IP's from:

1. North America (continent), Canada (country), Ontario (region)

2. North America (continent), USA (country), Michigan (region), Detroit
(city)

3. Any region called 'Ohio' anywhere in the world (not sure why you would do
that but fun example)

4. Anywhere in North America

 

So you can open up your system based solely on where you know your real
users are located.

 

-=Michelle=-

 

  _  

From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of motty cruz
motty.c...@gmail.com
Sent: Friday, April 4, 2014 11:15 AM


To: Asterisk Users List
Subject: Re: [asterisk-users] Asterisk 1.6

 

Hello Ishfaq, outside users usually travel around the country and connect
from different network, so it won't be possible to lock it down to specific
IP.  

 

Thanks for your support. 

 

On Fri, Apr 4, 2014 at 8:03 AM, Ishfaq Malik i...@pack-net.co.uk wrote:

 

 

On 4 April 2014 15:22, motty cruz motty.c...@gmail.com wrote:

thank you all for your support. I am using Linux, I only have about 7 users
outside our home network. I will learn fail2ban and will use it accordingly.


 

again Thanks for your support. 

 

 

Do the 7 users outside of your home network always connect from the same IP
addresses? If so, you can just lock down your SIP port to those 7 IPs
explicitly in your IPTables configuration.

 

Another option would be to change which port you're running SIP on. 




 

-- 

Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994 tel:%2B44%20%280%29845%20004%204994 
f: +44 (0)161 660 9825 tel:%2B44%20%280%29161%20660%209825 
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk http://www.pack-net.co.uk/ 
 
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street 
Manchester, M1 2JW
COMPANY REG NO. 04920552


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Re: [asterisk-users] Replying to Posts

[Don Kelly]

 On 13/3/14 6:27 pm, Eric Wieling wrote:
  This is an example of why I top post.   Who wrote what?
 
 Of course, if you use a mail client that's capable of quoting correctly, 
 it all works beautifully.
 

Kevin Larson sez:

Outlook can quote correctly, but it is an all or nothing setting it would
appear. Lotus Notes actually handles it better as there is a Reply option
for normal email and a Reply With Internet-Style History that I use for this
list. I don't have any problems following the rules of the list, but I am
fully on the side of the Replies should go at the top group and would vote
for a change in the rules. 

 

I'll vote again for top posting, and expect my vote to be recognized
internationally about as much as the Crimean referendum.

 

  --Don

 

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Re: [asterisk-users] Call transfer problem.

[Don Kelly]

Does he complete the call as a supervised transfer--waits for the called
party to answer before completing the transfer?

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, February 24, 2014 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call transfer problem.

Hi all,

I have a user who is having trouble transferring calls, using a Grandstream
GXP2xxx.

Here's the use case that I've seen:

I call the user from phone A and he answers on phone B.

Then, he hits the transfer button on his phone and dials an extension that
is reachable by him, but not by me, based on administrative policy.

However, the Asterisk logs indicate that the new call is being initiated by
phone A, not phone B!  Thus the call transfer fails.

I have other users, with other phones, that are able to transfer just fine.
What could be different with this particular user?

Any ideas?

Mike.

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Re: [asterisk-users] Reading DTMF sent by callee during a SIP call

[Don Kelly]

Isn't it easier just to use a SIP door phone?

 

  --Don

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Friday, December 20, 2013 10:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Reading DTMF sent by callee during a SIP call

 

 

 

On 20 December 2013 16:13, Alex ralie...@gmail.com wrote:

Hi everyone,

I am looking for advice about the design of a SIP-based intercom. I
count on your help, as my current attempts are not fruitful (yet).

This will be a pretty long message, so here's my fundamental question:

Is there a way to interpret DTMF tones sent by the calee
(not the caller) while a voice call is in progress?






Here's the desired scenario:

- there is a box with speakers and a mic
- Asterisk is running on a computer inside that box
- the box is embedded in a door
- There are two user accounts, UserA and userB
- UserA is a client that runs on the server*
- UserA calls UserB and they are having a voice conversation


Throughout the call, Asterisk must react to DTMF tones sent by userB;
such that an action is executed when a specific key is pressed.

The idea is to build an intercom that would enable me to open a door
remotely, by relying entirely on SIP, so there would be no need to
have some additional communication channel to send the open door
signal.




I have previously implemented IVRs using `Background` and jumped to
specific extensions, when a button was pressed. But in that case, the
extensions are dialed by the caller; whereas now the input must from
the person who answered the call.

If I use `Dial` and `Read` - the latter is only executed after `Dial`
terminates - so this is not suitable.


`Background` behaves like I need - but it plays back a predefined
file, so it is not suitable for an interactive conversation.



* Having a SIP client on the same machine as the Asterisk server
itself is not possible, because both won't be able to bind to port
5060. My guess is that the solution is to originate a call from the
CLI; but I haven't gotten to that part yet.




Thank you for your patience, I am looking forward to your feedback,
Alex



 

You could create your own feature in features.conf that executes a
Macro/Gosub defined in sip.conf...

 

Ish

 

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Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
 
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street 
Manchester, M1 2JW
COMPANY REG NO. 04920552
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Re: [asterisk-users] Multiple IAX2 Trunks Load balancing

[Don Kelly]

On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
 Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want 
 to load balance incoming calls over IAX2 trunks. If any trunk goes 
 down the calls traffic will be shared with other available trunks.
 When it gets Up the script is supposed to perform as desired i.e in 
 load balance mode.

 Thanks in advance.
 

Hans said:

Hi,

Generally, incoming traffic (http/mail/sip/vpn/etc) can easily spread with
the random functions with iptables. 
Perhaps also have a look at:
http://www.linuxvirtualserver.org/VS-NAT.html 

But i feel that might not work in this case, if all your trunk-connections
are already up-and-running.

If so, you might contemplate an alternative approach.
It sounds that you want to do the load balancing trick upon the application
layer (eg. within asterisk)

Perhaps it is possible to do the L.B. at the O.S. or network level, and let
all trunks appear to asterisk to one single trunk.

Don asks:

What's the value of load balancing multiple IAX trunks between the same
system pair? What resources are being balanced?

  --Don



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Re: [asterisk-users] Call Queue advise

[Don Kelly]

 

yes but I believe that least recent would ring one agent at a time?  If my
understanding is incorrect please correct it.  We are wanting to keep with
multiple phones ring to ensure coverage.

 

From what I've seen, I don't think this is possible. But maybe ask in the
#asterisk channel on Freenode IRC.

 

Could ring the least-recent a couple times to provide workload balance, then
ring everyone to make sure someone answers.

 

  --Don

 

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[asterisk-users] link to MySQL connection

[Don Kelly]

I'm making changes to an Asterisk IVR designed by someone else. 

The application uses both func_odbc.conf and php agi to access an external
MySQL database.

 

In the php routines, I would like to use the persistent connection that is
established in the dialplan, rather than creating a new connection each time
they run. How can I do this?

 

In res_odbc.conf, the context asterisk is established, successfully
referencing asterisk-connector in odbc.ini.

 

pre-connect = yes resulting in a persistent connection at startup.

 

Opening a connection in the php routines takes about 5 seconds, which is
unacceptable.

 

  --Don

 

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Re: [asterisk-users] link to MySQL connection

[Don Kelly]

 In the php routines, I would like to use the persistent connection 
 that is established in the dialplan, rather than creating a new 
 connection each time they run. How can I do this?

You can't, they are completely separate processes and code.
Joshua Colp

Thanks--that's not the answer I wanted, but it sure was quick. :)
Is there anything that would enable me to use a persistent connection in the
agi?
--Don



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Re: [asterisk-users] link to MySQL connection

[Don Kelly]

 In the php routines, I would like to use the persistent connection 
 that is established in the dialplan, rather than creating a new 
 connection each time they run. How can I do this?

You can't, they are completely separate processes and code.
Joshua Colp

Thanks--that's not the answer I wanted, but it sure was quick. :) Is there
anything that would enable me to use a persistent connection in the agi?
--Don

Yes.  Use func_odbc in your PHP AGI.   In Asterisk dialplan functions are
treated like dialplan variables so you can get and set them just like you
would other dialplan variables.

If it takes 5 seconds to open a PDO DB connection inside PHP you have some
OTHER problem.
Eric Wieling

Thanks, Eric. If I can't figure out why the connection takes so long, I'll
try the func_ODBC approach.

  --Don



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Re: [asterisk-users] link to MySQL connection

[Don Kelly]

Then you should analyze why it takes 5s. Opening and closing a mysql
connection should take at most a fraction of a second on a local net.

BTW, classical web sites (plain PHP and HTML) do not maintain state, so
keeping the mysql connection open may not be at all possible. I forgot
whether open db connections get closed automatically after the web page has
been rendered, but I think so. You could test this.

For data processing you do not want to got through Asterisk. You could
actually write your own db proxy, but I still think it makes more sense to
find out why opening the db connection takes so much time.

jg

I'll look into why the connection takes so long.

I'm looking for a quick fix for the problem :)

Long term, I think a web service would be better for what this routine is
doing. It doesn't do any telephony stuff and only requires one variable from
the dialplan.

  --Don

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Re: [asterisk-users] terminating the call, when transferer hangs up the call during attended transfer

[Don Kelly]

On Mon, Nov 25, 2013 at 08:25:44AM +0100, jg wrote:
 So:
 A calls B
 B answers
 B puts A on hold
 B calls C
 B talks to C
 B ends conversation with C
 B talks to A again, regardless
 
 I this correct? Looks like a simple Hold exercise.

Hello JG,

thanks for your reply..

not exactly, it's rather something like:

A calls B
B answers
B starts attended transfer (thus puts A on hold) B dials C B hangs up before
C answers

now I need call to C to be terminated, so I can ring back to B.

n.

Is B deciding C is unavailable and hanging up? In that case, rather than
hanging up, B wants to abandon the transfer and recover the original call.
Is that right?

  --Don



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Re: [asterisk-users] is g729 codec free? or under license???

[Don Kelly]

In your scenario, all the calls are from endpoints on 181 to endpoints on
183. If the endpoint devices are similar, it seems to me that there should
be no need to transcode-you can use a codec common to the endpoints. 729
would not be required.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of s m
Sent: Wednesday, October 02, 2013 2:34 AM
To: Dominik George
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] is g729 codec free? or under license???

 

thank you Dominik you help me a lot.

 and the last question is how many license key should i buy? i read that
license for g729 is per-channel but i don't understand what channel exactly
means here. this is my scenario :

10endpointspbx181...pbx182...pbx183...10endpoints

pbx181 and pbx183 has 10 endpoints connected to them. the call between these
endpoints are established by pbx182. if i want to buy a license for pbx182,
how many license key do i need? just one because i have just one connection
on it?  or two, because two trunks is defined on it? or as many as endpoints
which are connected to each other via pbx182?

please help me to clarify channel concept in my mind.

thanks in advance

SAM

 

On Tue, Oct 1, 2013 at 11:34 AM, Dominik George n...@naturalnet.de wrote:

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512

Hi,


about g729, you mean if it get free g729 and all my systems (PBXs and
routers) use g729 codec for setting a call, call is set without any
problem?

Yes, if all systems use g729 directly, you are ready to go.

- -nik

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=5rja
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Re: [asterisk-users] Auto dialer scripts and software

[Don Kelly]

I think political calling is less restrictive than you think! This is three
years old, but probably mostly applicable.

 

http://www.ilga.gov/commission/lru/Feb2010FirstRdg.pdf

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of cjwstudios
Sent: Thursday, May 23, 2013 2:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto dialer scripts and software

 

As long as you're dialing a screened registered voter list and don't call
.gov or .edu, you're fine.

 

On Wed, May 22, 2013 at 5:54 AM, Don Kelly d...@donkelly.biz wrote:

Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.

--Don





-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Wednesday, May 22, 2013 6:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Auto dialer scripts and software

On 22/5/13 10:54 am, A J Stiles wrote:
 You do know that sort of thing is against the law -- or at least
 requires a permit from the authorities -- in most civilised countries,
right?

And it's worth adding that even if it is legal in your country, you're
almost guaranteed to offend/annoy your target audience. Recorded calls
always do.

Kind regards,

Chris
--
This email is made from 100% recycled electrons

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Re: [asterisk-users] ILEC Interconnect: Basic MUX: M13 vs DCS: VT1.5 vs DS3

[Don Kelly]

Is there an OC-n to SIP solution that makes sense?

--Don

 

Hi Nick,

Going from DS1 to OC-n is a multi-step process.  Typically requiring a
hardware device to handle each MUX step.  But you can find hardware that
handles multiple MUX steps together.

VT1.5 is just a raw OC-n channel containing a single DS1.
An M13 device converts between DS3 and DS1.

A DACS (DCS or DXC) provides M13 conversion, sometimes even capable of
extracting the raw VT1.5 signal directly to DS1.

The ILEC transport option you choose really depends on the terminating
interface.  Do you want to connect with a DS3 or OC-n?

No matter what hardware you choose, you will need to convert to single
copper pairs (DS1/T1) to connect to your Asterisk boxes.  So an M13 or DCS
will be necessary to reach the DS1 level.

The device you choose depends on budget and growth expectations.  Typically
a DCS is an expensive investment, handling hundreds of DS3's. An M13 device
is typically a small unit that handles one or two DS3's.

The advantage comes when you add the 29th DS1.  With VT1.5 it's just adding
a single channel, DS3 will require another whole DS3 to get an additional
DS1.


Sincerely,
Brian LaVallee



 From: Nick Khamis sym...@gmail.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Wed, 12 Jun 2013 16:19:06 -0400
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Subject: [asterisk-users] ILEC Interconnect
 
 Hello Everyone,
 
 We are looking to interconnect with a local ILEC over an OC-n transport
layer.
 They basically gave us two options in terms of mapping the SONET to the
DS3:
 
 * VT1.5s mapping
 * DS1s mapping
 
 The second option is quite clear. We would MUX the connection, and 
 plug the lines into qaud t1 cads etc... The tech mentioned that with 
 the second option we would also need a DACS to convert back to M13 
 mapping. I was scared to tell him that I could not follow can someone 
 explain that to me kindly :).
 
 I don't know much about VT1.5 mapping. Can someone kindly explain what 
 the benefits or lack of are in choosing that option. Also what type of 
 additional equipment we would need?
 
 In case I have overlooked something, can you gents please tell me what 
 I will need in terms of hardware in both cases (minus routers and 
 switches). What we are looking at is:
 
 CO
 |
 |
 | OC-n
 |
 v
 DS3 MUX
 |
 |
 v
 21 Asterisk boxes with quad T1s
 
 
 Kind Regards,
 
 Nick.
 
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Re: [asterisk-users] announcement to be played for attended transfer call

[Don Kelly]

jg
Sent: Tuesday, June 11, 2013 5:28 AM

Playing an announcement like Your call has been... to A after C has
accepted the call is probably not a good idea, because C has to wait until
the the announcement has finished. In environments where callers are
announced to C, C would typically not want to wait for A---believe me.

**OP asked for an announcement to be played to B, not A.

  --Don



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Re: [asterisk-users] Auto dialer scripts and software

[Don Kelly]

Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be possible to pass
legislation preventing them.

--Don

 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Wednesday, May 22, 2013 6:48 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Auto dialer scripts and software

On 22/5/13 10:54 am, A J Stiles wrote:
 You do know that sort of thing is against the law -- or at least 
 requires a permit from the authorities -- in most civilised countries,
right?

And it's worth adding that even if it is legal in your country, you're
almost guaranteed to offend/annoy your target audience. Recorded calls
always do.

Kind regards,

Chris
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This email is made from 100% recycled electrons

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Re: [asterisk-users] Gateway?

[Don Kelly]

Guys and gals - these are all excellent answers - I am not being clear, I
think.

 

Let me see if I can illustrate it.

 

If you cannot see my diagramme, let me know and I will make a word-type
chart.

 

So, the Ip device at the top is a SIP phone

Asterisk Server 

Gateway /IP

 

*   This gateway is where the SIP Trunk is - so, a provider like Packet
8 or Broadcomm would have this
*   this connects directly to the public telephone system (somehow)
*   a Digium card would not work for me as I am not looking to connect
to a dial tone.
*   Does this make sense?

So, the Gateway/IP based - what the hell is that called? I am sure there is
such an animal as most of us have configured SIP trunks on Asterisk - so,
I'm thinking that this thing that connect to the public phone system is what
we see as a SIP trunk - right?

 

So, how the hell do I do that? Probably not that simple.

 

Thanks!

 

Glen

 

 

No, it doesn't make sense to me J

 

If you don't need a dial tone, you don't need the PSTN.

 

If you are using Broadcomm, etc., you simply use your Asterisk's system's
Ethernet connection.

 

Let's start with your application-what do you want to accomplish?

--Don

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Re: [asterisk-users] multiple provider for incoming

[Don Kelly]

If inbound reliability is important, you may be able to accomplish what you
want with redundant servers, multiple sip providers and toll-free numbers
that can be readily switched between the sip providers.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Tuesday, April 30, 2013 10:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] multiple provider for incoming

 

The process will depend on your provider, of course, but I know some have
an option that if they are unable to reach
your box, then they can auto-forward to another DID, or to a voicemail box,
or to a user-defined function, etc.  

Forwarding to another DID will/should work for us assuming they are going to
be able to do that during a failure on their side. During a recent outage (I
think they had some major issues at one of their switches), they were not
able to send the calls to our box which was online.  

Thanks,
Matt



  _  

Date: Tue, 30 Apr 2013 20:38:19 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] multiple provider for incoming

On Tue, Apr 30, 2013 at 7:50 PM, David Wessell da...@ringfree.biz wrote:

Hi Matt,

 

You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though.

 

Thanks

David

 

 

David,

 

I'm not sure where you got this information, but it's not accurate.  I've
had multiple inbound and outbound SIP providers for years going to a single
box.  You just get a separate DID from each provider.  

Matt,

The process will depend on your provider, of course, but I know some have an
option that if they are unable to reach your box, then they can auto-forward
to another DID, or to a voicemail box, or to a user-defined function, etc.  



-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


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Re: [asterisk-users] multiple provider for incoming

[Don Kelly]

The idea is to get your toll-free numbers from a top-tier carrier, pointed
to ring-to numbers on the sip provider of your choice. You can then quickly
(hopefully automatically) switch the toll-free numbers to other ring-to
numbers on your backup sip provider.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Tuesday, April 30, 2013 11:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] multiple provider for incoming

 

Don,

Inbound reliability is very important. We don't use toll-free numbers, but
we will look into that. I thought porting numbers - not sure about toll-free
though - from one provider to the other took days (not technically, but
paperwork, etc.)

Thanks,
Matt

  _  

From: d...@donkelly.biz
To: asterisk-users@lists.digium.com
Date: Tue, 30 Apr 2013 22:38:44 -0500
Subject: Re: [asterisk-users] multiple provider for incoming

If inbound reliability is important, you may be able to accomplish what you
want with redundant servers, multiple sip providers and toll-free numbers
that can be readily switched between the sip providers.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Hamilton
Sent: Tuesday, April 30, 2013 10:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] multiple provider for incoming

 

The process will depend on your provider, of course, but I know some have
an option that if they are unable to reach
your box, then they can auto-forward to another DID, or to a voicemail box,
or to a user-defined function, etc.  

Forwarding to another DID will/should work for us assuming they are going to
be able to do that during a failure on their side. During a recent outage (I
think they had some major issues at one of their switches), they were not
able to send the calls to our box which was online.  

Thanks,
Matt

  _  

Date: Tue, 30 Apr 2013 20:38:19 -0500
From: wcse...@selbytech.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] multiple provider for incoming

On Tue, Apr 30, 2013 at 7:50 PM, David Wessell da...@ringfree.biz wrote:

Hi Matt,

 

You can't have multiple providers for inbound traffic. You can have multiple
providers for outbound traffic though.

 

Thanks

David

 

 

David,

 

I'm not sure where you got this information, but it's not accurate.  I've
had multiple inbound and outbound SIP providers for years going to a single
box.  You just get a separate DID from each provider.  

Matt,

The process will depend on your provider, of course, but I know some have an
option that if they are unable to reach your box, then they can auto-forward
to another DID, or to a voicemail box, or to a user-defined function, etc.  



-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


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Re: [asterisk-users] Asterisk AMI - Create a daemon (background process)

[Don Kelly]

On Tue, 26 Feb 2013, Eric Wieling wrote:

 For me, PHP with its C-like syntax...

Steve Edward said:
For me, C with it's C-like syntax...

So that brings up the question I have. Shouldn't a daemon be a compiled
process?

  --Don



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Re: [asterisk-users] Can't detect remote answer

[Don Kelly]

Press # after entering the number to see if it's an extension pattern
matching delay

--Don

 

-Original Message-
On Behalf Of Steve Edwards
Sent: Monday, February 11, 2013 4:16 PM
 
Extension pattern matching (waiting for the digit timeout) can also induce
perceived dialing delays.



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Re: [asterisk-users] OT - Chan-mobile -Bluetooth dongle on remote LAN workstation

[Don Kelly]

-Original Message-
snip

 What I had in mind is to use someone's cellphone as a presence detector.
 Let me explain:
 - as the first thing you take along when leaving a room or location, 
 is your own cellphone, why not use chan_mobile and a bluetooth dongle 
 on your on PC (as you're not supposed to be within bluetooth range 
 from an asterisk server ;-)) to advertise you're away from your desk

Being completely ignorant of Bluetooth dongles, and knowing less than I
should about Asterisk, I'm happy to throw in my two-cents worth.

It seems that the Asterisk server need know nothing about the Bluetooth
dongle--it only cares if the user is in their office.

An application can run on a pc in the office that simply decides if a device
it's paired with is nearby or not. Probably want to have something that
makes sure it's lost a few times over a minute or so. Then the application
informs the Asterisk server (a web service?) that the individual is gone. I
would expect that a single pc could keep track of several people.

That said, I was thinking this was a simple Bluetooth presence detection
issue, so I Googled that. One result was this:

http://nerdvittles.com/?p=78

  --Don



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Re: [asterisk-users] Asterisk voicemail minimum length / silence settings

[Don Kelly]

But with max silence at 2 seconds, couldn't someone leave a 30-second
message, pause for a couple seconds to gather their thoughts or dig up a
phone number, and get hung up on?

 

I'd think that the 3-second and 10-second settings are sensible. Saving a
message that is known to be total silence has no value.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen
Sent: Tuesday, January 22, 2013 5:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk voicemail minimum length / silence
settings

 

Depending on exactly what you want, those may be the right settings. What
the warning is telling you is that with those settings, someone could reach
voicemail, say nothing (silence on the line) and after 10 seconds it will
hang up on them and deliver a 10 second silent message to their voicemail
box. If you set maxsilence to something like 2 seconds, then it would hang
up after 2 seconds and no message would be delivered. 

The only risk you run by ignoring the warning is that some users *WILL* get
voicemails delivered that have no real audio, but simply be silent. If you
can live with that, you settings will work fine. 

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 



From:asterisk users ast4...@gmail.com 
To:Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, 
Date:01/22/2013 05:22 PM 
Subject:[asterisk-users] Asterisk voicemail minimum length / silence
settings 
Sent by:asterisk-users-boun...@lists.digium.com 

  _  




What I'm trying to achieve is that a voicemail message should be at
least 3 seconds long for it to be saved, but *after that* a prolonged
silence (e.g. 10 seconds) should terminate the call and recording.

My current settings (Asterisk 10.7.0 and 11.2.1) are:

  ; Minimum length of a voicemail message in seconds for the message to be
kept
   ; The default is no minimum.
   minsecs=3

   ; How many seconds of silence before we end the recording
   maxsilence=10

With these settings, I'm getting the following warning message.

   WARNING[21671] app_voicemail.c: maxsilence should be less than
minsecs or you may get empty messages

What are the right settings for this situation?

Thanks all!

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Re: [asterisk-users] Mail list settings?

[Don Kelly]

I get direct replies when people reply to my posts. I thought that was just
'cause they wanted to make sure I saw their replies!

--Don

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pete Mundy
Sent: Thursday, January 17, 2013 6:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Mail list settings?

On 18/01/2013, at 12:37 PM, Andrew Latham lath...@gmail.com wrote:

 On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com
wrote:
 For some reason the mailing list is sending all messages from the 
 sending party.
 This makes it less than ideal when responding; as selecting reply 
 goes to the person and not the list.
 Can we have it set back to the old way please?
 
 I just checked back over the list emails and Bryant's email appears to 
 be unique in this problem.  I assume it is a simple issue somewhere.
 List admins?

My 2c...

Looking back over recent e-mails, it looks to me like Bryant just has a
reply-to header on his outbound e-mail's with his e-mail in there. The
mailing list is simply allowing his address to remain in the 'reply-to'
header (while adding the list's address too).

I've noticed some others do this too (Chui Kingh Man) is an example, but
there are others.

So is this a case of the mailing list no longer stripping 'reply-to' headers
before adding it's own, or is this simply a case of a few users setting
reply-to when most don't, and those users getting replies directly as well
as to the list (as one would expect)?

Ie, unless I'm mistaken, it all looks to be operating normally.

But I'd be happy to be proven wrong ;)

Pete Mundy


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Re: [asterisk-users] Delay in call asterisk

[Don Kelly]

If you dial 2001# does it complete the call immediately?

 

Your dial plan may be ambiguous about numbers starting with 2, so it waits
a few seconds to see if you're going to dial a longer number.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of upendra
Sent: Thursday, January 17, 2013 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Delay in call asterisk

 

Hi,

 

i am using elastix 2.3 and created some dahdi extensions,now i dialing
between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4
second before it ring the destination. so cany anyone know how fix it so
that after dialing the digits the destination should ring . without any
delay after dialing.

 

 

 

regards

Upendra.

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Re: [asterisk-users] special conference room

[Don Kelly]

Sounds like a conference with all attendees permanently muted  (except the
moderator).

 

The moderator uses whisper to communicate with individuals.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] special conference room

 

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many switches
and options and
can help me a lot... but as you already said... does _almost_ all
features.. unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried it
in deep, because just
from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the mute switch to disable the callers to talk to the
conference.. (so to say
it mutes the callers microphone, not his earphones am I wrong? 
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all feature
you ask for in it

On Jan 16, 2013 5:37 AM, Yves A. yves...@gmx.de wrote:

Hi list,

I am in need of a special asterisk conference room with the following
constraints:

- there is one admin / moderator and several normal callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] DIDForSale spam

[Don Kelly]

Jai,

 

It should not be necessary for me to remove my email address from your list.
It should not be on there to start with-we do not have, and have never had,
a relationship that justified you sending me email.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jai Rangi
Sent: Wednesday, January 09, 2013 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DIDForSale spam

 

Guys, 
Since I am attached to did for sale: 
My apology to every one who received the DIDForSale 2012 Achievement
email and you hated it. 

As a asterisk user my question will be. 
If some xyz company send you a so called spam email, what made you think
that you should spam the mailing lists. I am sure we all get lots if spam
emails every day. If you really got some time and talent, why don't you
write some good tips and tricks about asterisk. 

Long story short We have a link where you can unsubscribe your email for any
further communication. 
http://www.didforsale.com/unsubscribe.php  or Send me your email  address I
will personally take care of that and will remove your email. This will take
less than 5 seconds. 
I am sure there will be lot of arguments on why you should that and all. I
will refrain myself on any further unproductive communication.

Happy new year to you all. 




On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:

+1 here.

On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
wrote:

On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
 On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
 What were the senders IP(s)?

 Will have to look it up when I get home.

 Doug

 --
 Ben Franklin quote:

 Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.

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 I have gotten hit with this twice so far. in March and Today:

 Rohit Dhaka ro...@didforsale.com via mail.bingotelecom.com
 3/8/12

 DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
 1/9/13

 UGH, when I asked in March where he got my email he said:

 Hi Chris,
 We got your contact from the Internet. Let me know the good time to
 talk about this in detail.
 Thank you,
 -Rohit Dhaka


Obviously by harvesting these lists.  I received 2 myself.

Thanks,
Steve T

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Re: [asterisk-users] PRI (Primary-NTT)

[Don Kelly]

Your carrier is apparently using a Japanese switch (at least it's a Japanese
standard).

If you don't get a good answer from the list, you might send an email to
louis.lecl...@denphone.com. Denphone is a company installing Asterisk
systems in Tokyo.

--Don

 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edwin Lam
Sent: Sunday, January 06, 2013 3:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PRI (Primary-NTT)


hi folks.

i recently setup an Asterisk system in Hong Kong. their phone
company told me that their T1 PRI switch type is Primary-NTT.
however in chan_dahdi.conf there's no such option. i have it
set to national. it worked fine for a while, but now suddenly
stop working. in coming call just keep ringing and didn't
even show up on console. out going call hang up immediately
with cause code 27. (as usual, phone co. just said it's
problem with our equipment without giving us any detail).
anybody have any suggestions?

here's our /etc/dahdi/system.conf:
loadzone=hk
defaultzone=hk
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
span=3,0,0,esf,b8zs
bchan=49-71
dchan=72
span=4,0,0,esf,b8zs
bchan=73-95
dchan=96

/etc/asterisk/chan_dahdi.conf:
switchtype=national
pridialplan=unknown
prilocaldialplan=unknown
internationalprefix = 001
nationalprefix =
unknownprefix =
signalling=pri_cpe
usecallerid=yes
usecallingpres=yes
echocancel=no
echocancelwhenbridged=no
group=1
callgroup=1
pickupgroup=1
faxdetect=incoming
context=pri_in
channel = 1-23


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Re: [asterisk-users] Paging for Praying

[Don Kelly]

Doesn’t the OP wish to page all phones? So it’s not an issue of dumping dozens 
of call files all at once.

 

Does paging work? 

http://www.voip-info.org/wiki/view/Asterisk+cmd+Page

 

http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom

 

Overhead paging might also be something to consider, requiring just one call to 
page “everyone.”

--Don

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Wednesday, January 02, 2013 5:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Paging for Praying

 

How many people do you plan to page? because if numbers are high (or variable) 
you may have an easier life by using some sort of dialer if numbers are not 
very high and two lines are enough, our WombatDialer is free to use.

l.

 

 

2012/12/29 bilal ghayyad bilmar...@yahoo.com


2) Praying time need to be obtained from text (or database). So, it is not 
always the same time. What actually is needed to be obtained from the text file 
or the database is the time of the pray for each date (for example, if today is 
28 of December so the query will be for this date and then it is required to 
check if the time is same as the current time to page the wave file on the 
Phones).

 

-- 

Loway - home of QueueMetrics - http://queuemetrics.com

Test-drive WombatDialer beta @ http://wombatdialer.com 

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Re: [asterisk-users] Dialing out and recording

[Don Kelly]

I have the same requirement, but it's important that the caller ID
information from the original caller is presented to the destination and we
announce the call before the transfer is complete. The carrier requires a
diversion header if the ANI is not one of our DIDs. Does someone have
experience with this working?

 

(Top-posting 'cause the last guy did)

--Don

 

Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM



Put the AGI call in a macro context and add M(macro) to your Dial string.

 

Henrik Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM



Hi,

 

I am using asterisk via AGI and want to be able to record a call.

The scenario is:

1.  A call comes in
2.  The call is redirected to a mobile number via a local extension and
ChannelRedirect
3.  The local extension looks like something this:

exten = _X.,1,Dial(SIP/${EXTEN},60,.)

exten = _X.,n,Agi(agi://localhost/aj.agi?action=)

 

I have looked through all arguments of Dial but haven't found any way to
continue having a connected call between the caller and the callee and have
AGI control of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?

 

Thanks for help.

 

Regards,

 

Henrik

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Re: [asterisk-users] Top Posting

[Don Kelly]

  I'm the opposite.  I'm likely not to scroll down 10 pages to see the 
  comments at the end.
 
 Wouldn't need to if people trimmed their posts properly.

Precisely (e.g., see above)! Indeed, my sense is that top-posting
*discourages* properly trimming email and that's my main reason against it.
If things were properly trimmed, the email would be short enough that it
really doesn't matter that much if the new material is on the top or bottom,
but people who top-post and don't trim create really hard-to-follow emails.
 

Good point.  I've found myself having to edit and trim replies to  poorly
constructed conversations in the past because we got to the N'th iteration
using either or both formats.


In this properly trimmed example, there's no record of who said what. I
like top posting--with trimming that removes no value and a resultant
message that has the entire history so I can delete the older messages. As
Outlook doesn't support the usenet approach to mail lists, what do people
recommend for a good way of managing this type of list? 

  --Don



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Re: [asterisk-users] Top Posting

[Don Kelly]

It is not hard to follow the rules .  If the nice folks at Digium took the
time to post rules we should at least TRY to follow them. If you do not like
the rules you can always petition Digium to change them but, taking up
bandwidth on the list in this all to frequent pissing match is a futile
waste of time.

Grow up, follow the rules, have a good day.
JohnM

I'm 69, not too likely to do much more growing up, and I do follow the
rules, unless the thread is already top-posted.

I'm young enough, though, that I don't have a problem discussing change, and
I thought I had started a new thread with the Top Posting subject so you
wouldn't need to waste your time looking at it.

If there were change, I'd think it would be better to come from the list
users rather than from Digium.

If you'd like to add real value to this discussion, you might respond to my
request for information on what product/procedure/whatever would enable me
to follow and participate in bottom-posted discussions as it doesn't appear
that Outlook or gmail are very effective.

  --Don



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Re: [asterisk-users] Top Posting

[Don Kelly]

On 1/2/2013  Don Kelly wrote:
 ... what product/procedure/whatever would 
 enable me to follow and participate in bottom-posted discussions as it 
 doesn't appear that Outlook or gmail are very effective.


Umm, what about positioning the cursor below the previous post before
writing your reply in outlook, I used to do it all the time when forced to
use outlook by company policy or such. Click on scroll bar drag - to bottom
of reply - click in message body, about a half seconds time, maybe a full
second if you choose to move slowly. Admittedly though it has been a few
versions since I have been forced to use Outlook, I currently use
Thunderbird for mail and can set it to start my reply on top or at the
bottom.

JohnM

I don't have any problem getting my reply to the bottom of the email, but
Outlook doesn't do any indenting or  or anything (Makes in-line comments
really hard to work with). When people following the rules trim everything,
I end up seeing Works for me Me too with no way of following the thread
to see what they're talking about (especially if the subject is Merry
Christmas and they're talking about Razberry Pi).

I don't think Outlook does what I'd like, so I'm not limiting my options. I
can use different email to keep track of the Asterisk lists.

  --Don



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Re: [asterisk-users] Top Posting

[Don Kelly]

Gmail has just updated some stuff and I've been fiddling with the gmail ap
on Android (Ice Cream Sandwich).

I can select inline reply, delete superfluous stuff and go to the bottom
for my post.

After a few messages back and forth, the thread is displayed with a Show
quoted text link for each post and the current post at the bottom.

In gmail in my Chrome browser, the message is displayed with the subject at
the top and each of the posts (without quoting--even though it's in the
message) all nicely stacked up below. I haven't found the bottom post button
in the browser.

--Don

 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder
Sent: Wednesday, January 02, 2013 2:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Top Posting

On 01/02/2013 03:35 PM, Jim Lucas wrote:
 On 01/02/2013 12:16 PM, Don Kelly wrote:
 I don't think Outlook does what I'd like, so I'm not limiting my 
 options. I can use different email to keep track of the Asterisk 
 lists.

 Thunderbird (by default) bottom posts.  And it does the nice indenting 
 and allows you to turn off that HTML crap...  :)

 Anybody have any suggestions on a good email client for an Andriod 
 device.  A client that actually lets me set BCC or allows me to edit 
 the original message when I replying?  The built in client sucks!!!

maildroid has a lot of features but kills your battery FAST.

I only start it when I am expecting an important email and task kill it
afterwards or it stays running.



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Re: [asterisk-users] Top Posting

[Don Kelly]

Deleting everything would really confuse me


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Re: [asterisk-users] Top Posting

[Don Kelly]

Tongue firmly in both cheeks? How do you do that?


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steven Howes
Sent: Monday, December 31, 2012 3:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting

We should all top *AND* bottom post!

On 31 Dec 2012, at 06:03, isr...@gmail.com wrote:
 Just my pitch in to post
 From a blackberry you can only top post there is no way of bottom 
 posting So if I would have to wait to get to a computer to bottom post 
 I would just never answer

We should all top *AND* bottom post!

(tongue firmly in cheek here..)

S


Tongue firmly in both cheeks? How do you do that?



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[asterisk-users] Top Posting

[Don Kelly]

As I did two years ago, I'm posting a new thread with the Top Posting
subject rather than hijacking the Paging for Praying thread.

 

Two questions:

 

1.   Steve K: What do you mean by /coat?

2.   How do we change rule #5?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

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Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to table ?

[Don Kelly]

This thread confuses me. I've not worked with the Asterisk MySQL CDR, but
have worked with SQL for years.

 

Every table should have a primary key. Is no column identified as a PK?

 

If there is a PK, you will not be able to designate another column as PK.

 

If there is a PK, you don't need to worry about Asterisk duplicating keys in
that column-the database will not permit non-unique PKs.

 

If (and this would be really weird) the table is created with no primary
key, it shouldn't be a problem to add the primary key constraint to a column
IF there are no nulls or duplicates existing.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, December 06, 2012 12:50 PM
To: rwhee...@artifact-software.com; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR - Freepbx - Safe to add primary key to
table ?

 

The reason I add a new column autoincrement is due to the fact I trust more
mysql about uniquness than asterisk. 

Leandro

I am typing from my mobile phone...

Il giorno 06/dic/2012 19:11, Ron Wheeler rwhee...@artifact-software.com
ha scritto:

It seems like a safe thing to do.
You could also ask about the impact of making an existing column a  primary
key, in a MySQL forum.

Leandro's solution seems to be a good one as well and does guarantee
uniqueness.



Ron

On 06/12/2012 12:25 PM, Leandro Dardini wrote:

Yes, go for it. However I have added another autoincrement column and
created the primary key on it. On the other columns I need to search I have
created just an index. 

 

Leandro

2012/12/6 Olivier oza_4...@yahoo.fr

Hello,

I need to develop an application that will query (mostly reading) an
existing MySQL CDR database.
This database (named asteriskcdrdb) was created during Freepbx 2.10 install
on my asterisk 1.8 setup.
This database has a single CDR table which is filled by Asterisk.

The tools I'm planning to use require this table to include a Primary Key.
Is it safe to Alter this table telling it to use UniqueID column as a
Primary Key ?

(Sure, I'll test this on a database copy but I'm not confident my tests will
cover everything)

Regards

--



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Re: [asterisk-users] DTMF digits are coming through twice

[Don Kelly]

Is this happening for all callers, or just iPhone callers?

--Don

 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vik Killa
Sent: Wednesday, October 10, 2012 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF digits are coming through twice

I'm not sure I follow, the packet capture on the asterisk server shows
double digits being entered. Does that mean it's the source?

On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote:
 Hi,

 Not exactly a solution, but I'm sure you must've taken pcap traces of 
 a few such sample calls. See in their RTPs that you are receiving 
 repeatedly same RTPs which will tell you that any DTMF packet is 
 coming in twice by the source or not !
 just one such simple pcap will help you identify at whose end the 
 issue lies. If the source is your vendor sending you RTPs twice for 
 DTMFs then send them the capture and ask them to fix it however they can.

 BR,
 Sammy

 On Wed, Oct 10, 2012 at 8:44 PM, Vik Killa vipki...@gmail.com wrote:

 I've been running an Asterisk server (1.6.2.17.2) for over a year 
 without any major issues. All of a sudden people are unable to login 
 to their voicemail because Asterisk is seeing DTMF twice for each 
 digit the caller pushes. We've noticed the problem only consistently 
 happens to callers from specific locations. All the locations having 
 the issue use te same ITSP and internet provider. The ITSP swears 
 it's not them because they tested outside their phone system, 
 straight from a d-mark. We've tinkered and played with all options in 
 Asterisk related to DTMF with no success. Aside from upgrading 
 Asterisk, I'm at a loss as to how to fix this. This system has worked 
 for over a year without this issue and all of a sudden it appears. My 
 thought is that it's the internet provider (Earthlink) but they say 
 it can not possibly be them. Here is my DTMF settings in sip.conf:
 dtmfmode=rfc2833
 relaxdtmf=yes

 Any input appreciated! Thanks.

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Re: [asterisk-users] VOIP PBX replacement suggestions?

[Don Kelly]

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel
Seagraves
Sent: Wednesday, June 06, 2012 2:40 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] VOIP  PBX replacement suggestions?

The boss wants to move from landline service to VOIP service as a
cost-cutting measure. We have one voice line and one fax line. The telco is
billing over $100 a month for the two. We're using Hylafax for faxing and a
PBX for the voice line.

Our existing PBX is an Intertel Axxess box with the old v5 processor. The
management and voicemail computer died years ago (PSU burned up). I'm
worried that it's going to die before too much longer. We have the IPRC and
several IP Phone+ devices. It's my understanding that the IP Phone+ speaks
only a proprietary Intertel protocol and can never be used with any
non-Intertel equipment. I would like to dump the entire Intertel box and
move to Asterisk instead, but my budget for this project is exactly $0. I
can't afford to buy new devices.

The boss is leaning toward getting digital voice service from the local
cable monopoly. They want to charge us $30 a month per line to start, and we
will have to sign a 3 year contract. The monopoly in question has a
reputation for very poor service, but they are a monopoly so my boss sees
them as the only alternative. My worry is that if we sign that contract, we
are trapped with both the intertel and the cable monopoly, and if I exceed
the capacity of the intertel (or it just dies) I am SOL.

My questions then are as follows:

1) Is there a way I don't know about to get Asterisk to talk to either the
IPRC or the IP Phone+ directly in such a way that gets calls from one to the
other?
2) Are there any reputable VOIP providers that provide business service at a
rate less than $30 per line per month? The boss is adamant that we need
unlimited minutes.


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Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription

[Don Kelly]

The users list probably isn't the best place for this discussion. Send me a
note directly if you like.

--Don

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Sunday, May 27, 2012 1:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel
over-subscription

 

I suspected as much :)

 

Well, it IS a calling card; people call an access number, dial an
international number. 

 

Assuming typical ALOC to be 8 mins which is seen quite often in
International calls esp. in ethnic communities, and since the service hasn't
launched yet, it's hard to tell what the incoming traffic will be like but
in order for us to purchase the channel packs, we do need to figure out the
ratio of over-subscription we can use for the number of channels to buy so
while I understand it's a little vague, just wanted to hear from people
who're running similar services and what is their actual channel usage and
if they have consciously designed it using an assumption for this ratio or
they just buy more channels and/or DIDs looking at historical data (or
customer complaints)

 

 

 

On Sat, May 26, 2012 at 8:46 PM, Don Kelly d...@donkelly.biz wrote:

I don't think it's possible to suggest a ratio without knowing what your
actual application similar to calling card services is.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000 tel:651%20842-1000 
651 842-1001 tel:651%20842-1001  fax

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Saturday, May 26, 2012 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH
Users Help
Subject: [asterisk-users] Common/Reasonable Assumption on DID/Channel
over-subscription

 

Hello All,

 

just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX. 

 

I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have any real world numbers they can share? Is 10 to 1
a good ratio to ensure a user practically never gets a circuits are busy?

 

Thanks in advance


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Re: [asterisk-users] Common/Reasonable Assumption on DID/Channel over-subscription

[Don Kelly]

I don't think it's possible to suggest a ratio without knowing what your
actual application similar to calling card services is.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Saturday, May 26, 2012 5:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH
Users Help
Subject: [asterisk-users] Common/Reasonable Assumption on DID/Channel
over-subscription

 

Hello All,

 

just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX. 

 

I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have any real world numbers they can share? Is 10 to 1
a good ratio to ensure a user practically never gets a circuits are busy?

 

Thanks in advance

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Re: [asterisk-users] Flashphoner

[Don Kelly]

What flavor are flashphoner minties?

--Don

 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Friday, April 27, 2012 12:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Flashphoner

On 04/27/2012 01:24 PM, shayne.al...@gmail.com wrote:

 congratulations @};-

It's a match made in Heaven.  I have spare signature space to sell, and
Pavel wants signature space to rent!  At a low introductory rate of
US$1800/word, he and I are going to make this happen...

--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

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Re: [asterisk-users] Official numbering plan - where to get?

[Don Kelly]

Although I do feel that 100+ Euros/month is more than most of us could
manage, I don't think a one-time list is of much value. I would be
interested in establishing a database if there was interest from enough
users for a modest subscription price.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
 

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Thursday, March 22, 2012 5:50 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Official numbering plan - where to get?

I hope this is not too off-topic. As a kind-of follow up to rate sheet 
normalization I'm still trying to figure out a solution for: throw 10 
ratesheets at a program and get the cheapest codes/providers as output.

For this purpose I believe I need a real, detailed, accurate list of all 
the dialing codes, incl. mobile codes, city codes etc. worldwide as a 
reference for that particular program. There are thousands of A-Z lists 
on the web, and there are thousands of codes in them, but nothing is 
accurate enough or from an official source.

So, I spoke with the ITU today and they, funny enough, too don't have 
such a list. At least they don't have one that is computer parseable, 
like a .csv or .xls or something like that. What they have is: a single 
.doc or .pdf file for EACH country (1 file per country), which is not 
standardized in its content, with lots of text and descriptions, but it 
has all the codes. They don't even have such a list as a paid service. 
Feels like 30 years ago. :)  Anyway, there is numberingplans.com which 
provide exactly what I'm looking for, but they don't support one-time 
purchases, only subscriptions from around 100 to 990 EUR per month, 
which is above my budget (and I don't need a subscription).

Does anyone have an idea where to find such a list for free, or as a 
one-time purchase? If not, I'll probably go through the effort to 
compile my own list based on the ITU data. Let me know in case you want 
a copy then. :)

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Re: [asterisk-users] Rate sheet normalization

[Don Kelly]

If we had reports of every call, we could downgrade status of routes that
had frequency calls not completed that were outside the norm.

--Don

 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus
Sent: Friday, March 16, 2012 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Rate sheet normalization

Am 16.03.2012 04:14, schrieb Ast Coder:
 I would be more interested in a system where quality routes are tested 
 with different providers because rate really doesn't matter if a call 
 can't be placed or if a destination is a fake one. We have seen many 
 fake destinations with top tier providers but they had the best rates 
 so the strategy to pick them first really didn't work.

 So, maybe a subscription service where a dialler system continuously 
 tests routes with a list of 10 providers so that it's established 
 which routes actually work and then allow that data to be downloaded 
 for usage.

Ok, but how would this system handle FAS? Call shows as connected but it's
still ringing to the caller. And probably won't get connected to the
callee ever. I.e. cases where the route is just faulty in some way and it
will ring forever but actually no one will ever get connected. 
How do you differentiate between nobody is home and route is faulty...



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Re: [asterisk-users] Rate sheet normalization

[Don Kelly]

How about a central coop that manages the “normalized” rate sheet and 
distributes it with “unknown” call quality metrics for each route. Coop members 
report call quality for all calls/routes so the call quality metrics can be 
updated in the rate sheet and distributed to members.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of SamyGo
Sent: Thursday, March 15, 2012 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Rate sheet normalization

 

So, maybe a subscription service where a dialler system continuously tests 
routes with a list of 10 providers so that it's established which routes 
actually work and then allow that data to be downloaded for usage. 

 

I think that it may not be humanly possible and also not possible to have a 
separate automated setup to ring destinations and prioritize according to 
Quality or Least-rates. BUT I am sure that real-time call success rate(or ASR) 
via multiple providers and sorting providers accordingly for particular 
destinations is possible or maybe available. Provided a good piece of code is 
written which analyses the call status / quality and then picks favourite 
carrier/provider for any destination. !! 

Not sure if anyone can understand it completely ;)

I am thinking in terms of DynamicRouting or LCR modules from Kamailio or 
OpenSIPS.

 

Regards,

Sammy

 

On Fri, Mar 16, 2012 at 8:14 AM, Ast Coder asteriskcod...@gmail.com wrote:

I would be more interested in a system where quality routes are tested with 
different providers because rate really doesn't matter if a call can't be 
placed or if a destination is a fake one. We have seen many fake destinations 
with top tier providers but they had the best rates so the strategy to pick 
them first really didn't work.

 

So, maybe a subscription service where a dialler system continuously tests 
routes with a list of 10 providers so that it's established which routes 
actually work and then allow that data to be downloaded for usage.

 

 

On Thu, Mar 15, 2012 at 8:42 PM, Markus unive...@truemetal.org wrote:

Am 15.03.2012 17:20, schrieb Raj Mathur (राज माथुर):

 

On Thursday 15 Mar 2012, Markus wrote:

With like 10 different ratesheets from 10 different providers, of
which many change their rates every few days, manually doing it in
Excel is too time consuming...


Is it possible to get samples?  I'd be interested in looking into
developing a script that can handle this problem generically, and
presumably you're available to alpha- and beta-test in any case :)

 

Most definitely! I'll get in touch off-list. :)





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Re: [asterisk-users] Connecting to an Old Phone System

[Don Kelly]

I've done this with US ISDN PRI. Both with a Digium card (PCI) and an
Astribank (USB). I'd expect it's doable with the several products that
support Euro ISDN.

You can set up a simple VoIP gateway, but you can also do all sorts of magic
things in the Asterisk system for selected calls before or in lieu of
passing them along to the legacy system.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
 


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Sunday, January 08, 2012 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Connecting to an Old Phone System

What type of phone system?
And what type of connectivity are you trying to give the old pbx?

On 1/6/12, Dan Journo d...@keshercommunications.com wrote:
 Hi,

 This is not strictly an asterisk questions, but... ive got a client 
 with an old digital pbx phone systems connected to an isdn30e line.

 I've been shown a sip gateway that can connect to asterisk on one 
 side, and also has an ISDN30e socket that the old phone system can 
 connect to. But it's a bit pricey.

 Is there such a thing as an ISDN30e PCI card which can be used with a 
 copy of Asterisk, that can act like a voip gateway between the old 
 phone system, and our asterisk box?
 Or can anyone recommend a gateway that isn't too expensive?
 They use 8 channels of the isdn30e

 Many thanks
 Dan

 Dan Journo
 Kesher Communications (UK)
 Business Phone Systemshttp://www.keshercommunications.com/ | Hosted 
 PBXhttp://www.keshercommunications.com/hostedpbx.html



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Re: [asterisk-users] DID from Direct from Telco

[Don Kelly]

It might be a good idea for you to describe your application and ask for
suggestions.

How many concurrent calls do you need to handle? Do you need a few (or many)
DIDs (actual phone numbers)? Are the DIDs in a single geographic area, or
scattered all over the country(ies)? Is your application inbound-only, or
will you be making outbound calls? Or will you be redirecting calls to
outside agents? What is there about the SIP providers that you find
unsatisfactory?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Friday, November 04, 2011 7:47 AM
To: isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] DID from Direct from Telco

Thank you guys for your response,

 One FXS port can only handle one call. A PRI T1 gateway can handle 23 
 call channels. A single T1 Data line with SIP can  handle about 18 
 call channels running G711, 37 channels running g729

I just want to make sure that a T1 Gateway (capable of 23 call channels),
plugged into an FXS port (capable of one call), is not a bottleneck. I.e.,
even though our network can handle upto 23 channels, we can only support 1
concurrent call becuase of the single FXS? What I am trying to figure out is
what would I need to have the same capabilities as a company offering DIDs.
Which mediant, and maybe a nice illustration?

Thanks in Advance,

Nick.

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Re: [asterisk-users] Fax from FXS to PRI

[Don Kelly]

September 22, 2011 11:20 AM  Kevin P. Fleming wrote:

For many people, with modern CPUs, current versions of DAHDI and Asterisk,
and appropriate configuration (using the faxbuffers option in
chan_dahdi.conf, for example), such a system can be setup to work very, very
close to 100% of the time.


Thanks for the explanation. I think that most of us can be happy with very,
very close to 100%.

  --Don



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Re: [asterisk-users] Console Stereo - One call per ear

[Don Kelly]

(Top posting 'cause that's what others did--and I like it that way, anyway.)

Not so obvious that a single mic can't be shared--if one call is muted, it
would work great. 

This split-ear feature would be handy when, while on hold/in queue on call
A, you want to answer call B. You want to talk to caller B without risking
missing callee A finally picking up after 15 minutes on hold.

(OP said 2 separate calls on each ear. That's 4 calls--would really drive
Kyle nuts.)

--Don



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gohar Ahmed
Sent: Wednesday, September 21, 2011 12:50 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Console Stereo - One call per ear

Couldn't help LOL on Kyle's remarks. But it could be two users listening to
two different streams/calls. Obviously both can't share single mic on their
call(if they ever need it).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kyle Sexton
Sent: Wednesday, September 21, 2011 8:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Console Stereo - One call per ear

I have no solution, but my head hurts thinking about listening to separate
calls simultaneously in each ear.


On Sep 9, 2011, at 9:22 AM, fhirschberg wrote:

 Hi list!
 
 I'm using the latest Asterisk 1.8.6.0 cross compiled for an i.MX27 
 board and it works really good.
 But I need a feature and don't know how to do this. 
 What I need is the ability to have 2 separate calls on each ear on the 
 console channel.
 Is there a way to get this working? It should be possible to have one 
 call on both ears or, if another call is made, to hear this on one 
 (selectable
 L/R) ear, while the other call stays on the other ear.
 Do I need a new console driver? I'm currently using chan_alsa and I
already
 have Alsa devices for left, right and left + right output. 
 It would be great if anybody can help with informations or tips where 
 to start with my problem.
 
 Greetings
 Florian
 
 
 
 
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Re: [asterisk-users] Fax from FXS to PRI

[Don Kelly]

On Tue, Sep 20, 2011 at 4:43 PM, Adam Moffett adamli...@plexicomm.net
wrote:

If I have a 4 port Digium FXS card and a single port PRI card on the same
asterisk box, is it expected that I'd be able to plug a fax machine into the
analog FXS port and have no problems sending or receiving faxes?  Our
connection to the Telco is on the PRI obviously.

snip


Nobody can say for sure.  It is not a supported configuration.  I can tell
you that I have had great success and wasted days messing around with this
configuration.  

snip

 

Again, it has never been a supported configuration by Digium, and everyone
that has dealt with faxing in Asterisk especially on different systems will
tell you that you won't know until you try.  And even then, is it worth days
of your time trying to get it as close to a POTS line as possible?

snip


Thanks,
Steve T

This is a scary answer-you're saying that what should be simple TDM FXS to
PRI does not work?

 

Are you suggesting this is an Asterisk problem or a Digium hardware problem?

 

Is this really everyone's experience?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)



 

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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Don Kelly]

snip

I'll try to explain myself better. The PBX has only one FXO card, connected
to the PSTN. There is no other phones connected to the PBX nor SIP
extensions. 
There are analog phones connected to the same PSTN.

What I try to do is that, when there is an incoming call from the ouside, if
someone answers on a phone, then the PBX won't answer.



If you want to be certain that the Asterisk system won't interfere with an
active call, you can install an exclusion device between the PSTN and the
FXO card.

Google telephone exclusion device.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax


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Re: [asterisk-users] Answering machine answers after pickup a phone.

[Don Kelly]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jorge Barreiro
Sent: Friday, August 05, 2011 12:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Answering machine answers after pickup a
phone.

O Venres, 5 de Agosto de 2011 17:42:28 Shaun Ruffell escribiu:
 On Fri, Aug 05, 2011 at 01:14:58PM +0200, Jorge Barreiro wrote:
  Hi,
  
  thanks for your time!
  
  O Venres, 5 de Agosto de 2011 12:35:05 escribiches:
   Completely normal operation.
   You need to read and understand more basic telephony and analog 
   lines to understand why that won't work.
  
  I definitely have a lot to learn yet.
  
   Asterisk needs to be in control, and once someone answers a phone 
   not under Asterisk control, or the call is abandoned there is 
   little you can do.
  
  What I pretend is that asterisk detects that it's not under control 
  and gets out of the way. The same way it detects a remote hangup and 
  stops the dialplan, it could detect that someone else answered (the 
  line is not ringing anymore) and discard it the same way it does 
  when the remote part hangup.
  
  I've read comments in forums and tutorials that seem to imply that 
  this happens, but I couldn't find any confirmation (and indeed, it's 
  not happening to me).
 
 When I first installed Asterisk in my home I used it in the way that 
 you
 described: as a glorified answering machine to email to me any voice mail.
 
 I think what you want is the WaitForRing()[1] dial plan application.  
 This function will wait x number of seconds, then look for *another* 
 ring to come in. If someone answered the phone before the timeout to 
 that function Asterisk would stop processing the dial plan.
 
 [1] https://wiki.asterisk.org/wiki/display/AST/Application_WaitForRing
 
 I ran into a couple of issues with WaitForRing(). The first being if 
 someone answered the phone and then quickly hung up *and* a new phone 
 call came in within the timeout period, Asterisk wouldn't know that 
 the line was ringing due to a new call. The second problem was I never 
 got the dial tone detection working so that if I tried to *place* a 
 call from Asterisk while someone was on the house line I would aggravate
my wife.
 
 Since coming to work for Digium I've seen in the data sheets for the 
 FXO interfaces that there is a capability to detect when a parallel 
 device on a line goes off hook. This would allow Asterisk to have a 
 better sense of the state of the line (like it currently can detect 
 when a port is unplugged and there is not battery by generating a red 
 alarm.) but I haven't looked into getting that information off the
hardware and up into Asterisk.
 
 Hope this helps,
 Shaun


That application looks like a good solution. I can't test it until Monday,
but I'll try it and let you know. The drawbacks you mention doesn't seem too
inconvenient in my case.

Anyway, I started with this cause I thought it was an easy first step, if it
gets so complicated I think I'll go forward and put all phones under the
control of the PBX.

Thank you everybody for your help.


I don't think this is a solution to the problem you described. No matter how
long Asterisk 'waits for ring,' if the call has already been answered when
Asterisk picks up, things won't work out well. The solution I described
earlier, adding a simple exclusion device, will preclude Asterisk 'stepping
on' a call in progress. This is the approach that Shaun suggests: ...a
capability to detect when a parallel device on a line goes off hook.  As it
has not been implemented in Asterisk, it can be handled by an inexpensive
device. This will enable you to do as you planned--test your implementation
step-by-step, starting with the answering machine.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax


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Re: [asterisk-users] Connect asterisk to normal telephone PBX

[Don Kelly]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Thursday, July 28, 2011 1:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Connect asterisk to normal telephone PBX

 

Hello All,

I don't even know the relevancy of my question. Please answer me if my
question have some sense. 

I have recently implemented an asterisk server with freepbx. I have created
100 extentions and i can make successful calls between extensions from
anywhere. But my office have three different land-line numbers and three of
them are terminating into an internal PBX ( normal matrix telephone PBX)
with more than 60 extensions. This internal PBX is the live PBX where we can
call local, STD and ISD from extensions. 

At present i have some practical difficulties to configure telephone lines
at the end of asterisk PBX. So i am trying to connect my asterisk PBX to the
normal telephone PBX.

I have installed 1 port x100p FXO card  in my asterisk PBX and detected by
my freepbx. Then i removed my normal telephone extension cable from phone
and connected to the FXO  port of my asterisk PBX. 

Ultimately my intention is that 

1) if somebody call to my normal telephone extension, that should reach to
my asterisk server, and asterisk server should send this call to my asterisk
extension. 
2) if i am calling from my asterisk extension, call should go to the normal
telephone PBX via FXO card in my asterisk server and ultimately the call
should send outside via the telephone PBX.


Is my approach is correct ? If it is wrong please somebody assist me to
connect my asterisk PBX to normal telephone PBX.

Michael.K

 

 

 

 

The approach sounds right IF your legacy PBX phone is a simple analog phone.
If it is a proprietary phone, you need to try a different approach.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

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Re: [asterisk-users] Redirecting call from one E1 to another?

[Don Kelly]

Tony Mountifield wrote:
 I don't want just to relay the call through to the second box using 
 IAX or SIP or an additional PSTN channel. What I would like to do is 
 to redirect the call in the PSTN so that it ends up connected only to 
 the second box.


Doug wrote:
If I recall correctly, it's only possible if the provider has a DMS-100

http://www.voip-info.org/wiki/view/RDNIS




I don't think Tony wants to redirect the call in that sense. His first
server has answered the call, then wants to transfer it to the other server.

I would suggest Two B-Channel Transfer (TBCT), transferring to a unique
number (received as DNIS by the other server) that would identify the call
as transferred from the first server and, perhaps, the reason for the
transfer.

It looks like TBCT may not have been implemented in Asterisk for EuroISDN.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax


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Re: [asterisk-users] Asterisk as a Condo door opener/intercom

[Don Kelly]

Continuing top posting...

The same argument could be made for any commercial solution. Why use
Asterisk when we could throw $4,000 at our problem for a commercial
solution?

I'd like to have a solution that would have the features you suggest for
$400.

--Don


On Behalf Of C F
Sent: Monday, April 11, 2011 11:43 AM

Search the lists. Some hints:
Viking electronics makes a door box that connects to any analog line
(IIRC e-20).
They also make a DTMF keypad that integrates in series with any analog
line. They might also make a door box with a DTMF keypad on it.
Sandman makes a relay that will get energized when there is a ring on
the line which could be used to unlock the door.

However, why would you use asterisk? Using asterisk for the sole
purpose of MDU entry system is like using windows for asterisk, it
works but why?
Go for the commercial solutions, it comes with a geziilion options for
your setup one of them the ability of chosing an apartment, another
add key fobs, another one is the ability of using a code for the
residence (not guests) to unlock the door. Also the interface with
asterisk you will have to build one from scratch. The commercial
solutions have em built in.

On 4/10/11, Bruce B bruceb...@gmail.com wrote:
 Hi Everyone,

 Looking to replace a condo intercom system. Apparently the current one
taps
 into the lines and dials phone numbers but needs to be changed as it's
 faulty.

 I will probably still use the same analogue dialing and back it up with a
 VoIP line and use the current cabling that is in place. But as for as the
 door opening function goes, I am not sure how to interface and how open
 these modules are usually built.

 I would appreciate it if someone with experience can throw in some
pointers
 as to what I might be facing and what challenges I have to solve to
replace
 this with a nice Asterisk system.

 Thanks,


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Re: [asterisk-users] MOH on DAHDI PRI Channels

[Don Kelly]

 

  _  

[Shariq Khan]



 

Is it possible to start MOH when calling to DAHDI Channel that has ISDN E1
connected with it. When the called party press hold on his phone then
asterisk start MOH??

 

 

[Danny Nicholas] 

Question #1

Dial(DAHDI/1/5551212,20,m) will play moh until the other end answers

Question #2

Don't think so since you're asking Asterisk to detect on hold from outside
(this might be do-able in a SIP environment, but DAHDI tends to be copper).

Hope this is correct/helps.

 

[Don Kelly]

Looks like the call is to the DAHDI Channel from an outside caller, so the
called party is inside. This is simple MOH.

 

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[asterisk-users] Missing audio

[Don Kelly]

I have a FreePBX system with PRI trunks that's doing a number of things very
nicely, but frustrating me in one area.

 

I am using a Grandstream GXW-4008 in an off-premises location to provide
POTS service on four ports (this device worked fine in an early
application using a hardware VPN to the Asterisk server).

 

The Grandstream has a public static IP port, as does the Asterisk server.

 

Extensions 1021, 1022, 1023, and 1024 register just fine.

 

A ring group, 1020, distributes calls to these extensions and they handle
incoming calls in hunt as I'd expect.

 

Calls on the first port are consistently fine.

 

Calls on the other ports are fine for a day or more, then they lose audio or
have one-way audio.

 

One mystery for me is that the first port always continues to work.

 

I've assumed that this is some sort of UDP port problem, but I've Googled
and studied stuff on-line and haven't figured out what I should be doing to
fix it.

 

I'd really appreciate some help.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
651 842-1001 fax

 

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Re: [asterisk-users] Does Asterisk support NI-1 (DMS 100) and NI-2 forT1s?

[Don Kelly]

 Zeeshan Zakaria
 Sent: Friday, January 21, 2011 6:11 AM


 For a client I am setting up a system which will use T1 PRI from Primus,
who offer only NI-1 and NI-2 protocols for D-Channels. Previousely I have
only
 used switchtypes euroISDN and National. Although the documentation says
Asterisk does support NI-1 ans NI-2, but wanted to get your opinion if you
have  used these protocols on an Asterisk box and if there were any things
to consider. If anybody has experience with Primus, it'll be more helpful.

 

 

I'm using NI-2 with no problem (but haven't tried all features).

 

You say you've used National-wouldn't that be NI-1 or NI-2?

--Don

 

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Re: [asterisk-users] Mailing list question

[Don Kelly]

 Is the any kind of 'tag' that I can include at the end of my message 
 to make the list processing software ignore and dispose of my 
 disclaimer?

It looks like there were underscores on the same line as the --

I think the actual idea is to include '-- ' with nothing else on that line

--Don
-- 
Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax



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Re: [asterisk-users] Top Posting

[Don Kelly]

  On 01/19/2011 12:18 AM, randulo wrote:
  Although there's no requisite mention of ${Horrible_Dictator}, can't
  we pretend there was, call a Godwin and kill this subject?

 11:39 Parker said
 That would fall under Quirk's Exception: Intentionally invoking Godwin's 
 Law to attempt to kill a thread is rarely successful. :)

Didn't work this time :)



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Re: [asterisk-users] res_fax

[Don Kelly]

 There was a typo in the res_fax documentation.  Application_SendeFax
should be the correct documentation.  I don't know where Application_SendFax
is coming from - it's probably old.  When the next import happens,
Application_SendFax should be replaced by the correct version (then I'll try
to remember to remove the bogus SendeFax copy).

Am I the only one confused here? (probably) It seems like you imply that
SendeFax (which looks like a typo to me) is correct in the second sentence,
then reverse yourself in the last parenthetical statement.


I'm not confused if he means that the content of Application_SendeFax is
correct and the content of Application_SendFax is old. After the next
update, the content of Application_SendFax will be correct and
Application_SendeFax will go away

  --Don



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Re: [asterisk-users] Top Posting

[Don Kelly]

I also agree this is a pointless discussion because, clearly, nobody is
willing to budge, and it has nothing to do with Asterisk.

Amen :)

It may yet have a point - another few hundred (thousand) of these and the
board will blacklist items with the words top post and bottom post :)

And maybe If you have received this communication in error...  :)



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Re: [asterisk-users] Top Posting

[Don Kelly]

I'm top-posting this simply to be consistent with the previous couple posts.

I agree that top-posting is preferable for the reason that Andrew pointed
out and I prefer no trimming (other than signatures--especially legal
disclaimers, etc.) so I can delete every message except the most recent and
maintain the entire thread.

However, as pointed out a couple days ago, this list's rules specify that
we'll respond after the text being responded to, so that's what I'll be
doing.

PLONK is retro--like bottom-posting :)

--Don



On Behalf Of Vince Vielhaber
Sent: Tuesday, January 18, 2011 9:29 AM

I'm top posting this so you will see it and if you don't understand it,
look it up.


PLONK!!


On Tue, 18 Jan 2011, Andrew Thomas wrote:

 Why do I top post?  Simple.  I read every message in the thread - and if
 there are 10 messages (for example) in that thread - then why should I
 have to read them all over again on the last one?

 Top posting is here - to stay!

 Stop being so anal and 'retro'.  Bottom posting belongs in forums - top
 post belongs in e-mail lists.

 There - said it!



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Re: [asterisk-users] Sendind e-mail with Hylafax

[Don Kelly]

snip

Although I put my e-mail in /etc/hylifax/Dispatch I can't receive.

Flavio Roberto Miranda



It may be different for your Hylafax version, etc., but you may want your
email in 

/var/spool/hylafax/etc/FaxDispatch

 

And you probably want to post your questions to the Hylafax list

http://lists.hylafax.org/cgi-bin/lsg2.cgi

--Don

 

 

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Re: [asterisk-users] Top Posting

[Don Kelly]

  On Tuesday 18 Jan 2011, Don Kelly wrote:
  PLONK is retro--like bottom-posting :)
 
  --Don

  boun...@lists.digium.com] On Behalf Of A J Stiles

  Retro?  For those of us who actually know what PLONK means, it's
hilarious.  

  Now, here is a link 

  http://www.youtube.com/watch?v=R1JXYgwwDeY

  Posting answers *before* the question to which they refer breaks the flow
of 
  conversation  

It's clear from your response that you have not followed this entire
thread--depending solely on the snippets in the message to which you
responded. Thanks for illustrating one of my points.

I've been working with computers for over 40 years and don't have the
foggiest notion how the Green Day--Wake Me Up When September Ends video
applies to Top Posting.

  --Don



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Re: [asterisk-users] Top Posting

[Don Kelly]

 I've been working with computers for over 40 years and don't have the
 foggiest notion how the Green Day--Wake Me Up When September Ends video
 applies to Top Posting.


It's a reference to the Everlasting September in 1993.  AOL added
usenet access to its service, unleashing a horde of dirty, no-good
n00bs onto the interwebs.  And alas, there was much consternation and
gnashing of teeth over the new user's lack of netiquette.

Thanks for the explanation.



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Re: [asterisk-users] Top Posting

[Don Kelly]

 Paul Belanger wrote:

 It is not a matter of preference, it is actually a rule [1]. Top-posting
 is also an annoying practice [2] and NOT the general accepted way to
reply.

 [1] http://www.asterisk.org/community/rules
 [2] http://linux.sgms-centre.com/misc/netiquette.php#toppost



Thanks for pointing out the rule ([1] #5); this is a more effective way of
communicating the list's etiquette than the snide asides that we generally
see. It also conforms to the rule ([1] #1) that says posts should be
considerate and respectful.

With regard to my preference, I stand by my explanation in my first post in
this thread. I think the main argument for bottom-posting is we've been
doing it that way for decades.

That said, of course I want to follow this list's etiquette. I've posted a
couple times asking how I can interleave responses in Outlook or what other
approach can I take to make it practical to stop top-posting. Any
suggestions?

-- 

  --Don



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[asterisk-users] Top Posting

[Don Kelly]

Bruce et al.

 

I'm posting a new thread with the Top Posting subject so I won't draw
complaints about hijacking the 4-port thread.

 

Top Posting refers to the practice of sending a message with a reply at the
top and including the entire thread below the reply. I prefer this. If I'm
actively following a thread, the most-recent information appears at the top
of the message I receive. If I've missed part of the thread, I need to look
only at the most recent message and scroll down a bit to see what's been
happening.

 

Bottom Posting requires me to scroll through all of the history before I see
the newest addition.

 

While scrolling down, I may see something new and realize that the sender
has interleaved responses, addressing multiple points with individual
responses.

 

It's been a while, but when I researched Top Posting I found this
Wikipedia description:

 

Top-posting is a natural consequence of the behavior of the reply
function in many current e-mail readers, such as Microsoft Outlook
http://en.wikipedia.org/wiki/Microsoft_Outlook , Gmail
http://en.wikipedia.org/wiki/Gmail , and others. By default, these
programs insert into the reply message a copy of the original message
(without headers and often without any extra indentation or quotation
markers), and position the editing cursor
http://en.wikipedia.org/wiki/Cursor_%28computers%29  above it. Moreover, a
bug present on most flavours of Microsoft Outlook caused the quotation
markers to be lost when replying in plain text to a message that was
originally sent in HTML/RTF. In addition, users of mobile devices
http://en.wikipedia.org/wiki/Handheld_device , like BlackBerries
http://en.wikipedia.org/wiki/BlackBerry , are encouraged to use
top-posting, because the devices only download the beginning of a message
for viewing. The rest of the message is only retrieved when needed, which
takes additional download time. Putting the relevant content at the
beginning of the message requires less bandwidth, less time, and less
scrolling for the Blackberry user.[4]
http://en.wikipedia.org/wiki/Posting_style#cite_note-3 [5]
http://en.wikipedia.org/wiki/Posting_style#cite_note-4 [6]
http://en.wikipedia.org/wiki/Posting_style#cite_note-5  For these and
possibly other reasons, many users seem to accept top-posting as the
standard reply style.

.and an explanation of why people complain about it:

Objections to top-posting on newsgroups, as a rule, seem to come from
persons who first went online in the earlier days of Usenet
http://en.wikipedia.org/wiki/Usenet , and in communities that date to
Usenet's early days. Until the mid-90s, top-posting was unknown and
interleaved posting an obvious standard that all net.newcomers had to learn.
Among the most vehement communities are those in the Usenet
http://en.wikipedia.org/wiki/Comp.*_hierarchy comp.lang hierarchy,
especially comp.lang.c and comp.lang.c++. Top-posting is more tolerated on
the  http://en.wikipedia.org/wiki/Alt.*_hierarchy alt hierarchy. Newer
online participants, especially those with limited experience of Usenet,
tend to be less sensitive to arguments about posting style.

When I post (which is rarely, as I have little to offer the list), I top
post and explain that it's my preference and I don't know how to do it
effectively otherwise. This gives everyone fair warning to delete my posts
before reading them.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

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Re: [asterisk-users] Top Posting

[Don Kelly]

Awww...that's no fair. Andrew has bottom-posted to this top-post thread.
That really confuses me.

Andrew's 'header' appears at the top of the 'stuff,' and his comments at the
bottom.

Then there's a little sniping that I would have considered really
important when I only had 16KB of memory to work with, but it leaves me
wondering whose comment was Seconded... We've lost the attribution.

What did you mean, Andrew, about Don's multiple
signatures which I think he will review?

--Don

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Friday, January 14, 2011 7:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting

 Seconded.  Although I've succumbed to bottom posting on occasion when
 following the convention of the ongoing thread.

 On 01/14/2011 07:42 PM, Don Kelly wrote:

 Bruce et al…

 I’m posting a new thread with the “Top Posting” subject so I won’t draw
 complaints about “hijacking” the 4-port thread.

snip

 When I post (which is rarely, as I have little to offer the list), I top
 post and explain that it’s my preference and I don’t know how to do it
 effectively otherwise. This gives everyone fair warning to delete my posts
 before reading them.

 --Don

 Don Kelly

 PCF Corp
 People Come First
 651 842-1000
 888 Don Kell(y)
 651 842-1001 fax

As mentioned in the past, trimming your post is the best first step on
mailing lists.  Many of the top post vs bottom post comments happen on
the 5+ post on a thread when the size of the email becomes an issue.
I have blindly replied in the past and was unable to understand my own
email.  Take a moment and trim out the messy bits.  Use a (snip) or
snip to note huge missing areas. As you will note in Don's post
there is a history to the argument.  Also note Don's multiple
signatures which I think he will review after he sees it in action. :)

Above all, be polite...

~~~ Andrew lathama Latham lath...@gmail.com ~~~



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Re: [asterisk-users] Top Posting

[Don Kelly]

I can agree that the entire signature is not relevant to [the] list. but I
hope you won't find an example of it adding eight lines to every post. I
generally try to include it in only one post in case someone wants to get in
touch with me.

With regard to the trimming and snipping, I'd prefer to see the entire
history in a single message, allowing me to delete all previous posts
without losing any information.

--Don



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Rymes
Sent: Friday, January 14, 2011 9:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting

On Jan 14, 2011, at 8:52 PM, Don Kelly wrote:

I have nothing to add to the nascent flame war that I thought we had so
narrowly avoided when I sent my last message. However:

 What did you mean, Andrew, about Don's multiple
 signatures which I think he will review?
 
 --Don

[snip]

Andrew meant these multiple signatures, and implied that, once you looked at
them, you would realize it's a little redundant and not relevant to list
users:

 --Don
 
 Don Kelly
 
 PCF Corp
 People Come First
 651 842-1000
 888 Don Kell(y)
 651 842-1001 fax


It's a free country, but given that you prefer a top-posting style where you
don't trim previous messages (not judging here, just saying), you might
consider omitting your signature for list posts, as it adds an additional
eight lines to each message you send, which can really add up. Will it end
hunger or bring about world peace? No. Will it be that little bit easier on
everyone's eyes? Yes.

I think the main lesson from Andrew's post is that top or bottom posting
doesn't matter anywhere near as much as trimming posts, so that only that
portion of a previous message that you need for context is included, making
the entire message compact and nicely legible.

Tom



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Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K South London

[Don Kelly]

45K GBP would probably cover breakfast in South London. It's about 70 USD.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich
Sent: Wednesday, December 22, 2010 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45K
South London

 


45K ?

With 45K I can barely pay for gas, tolls, and breakfast.  If you guys are
such a fast growing company, probably you can pay better salaries.

CS


On December 22, 2010 at 9:23 AM Jess Hart j...@langleyjames.net wrote:




Job Description:  Asterisk MySQL Support Engineer

Fast Growing Global Telecoms Company requires a very experienced engineer
who has a variety of skill levels. The role would suit someone who has
worked at switch level and fully understands how calls are to be handled to
and from a VoIP platform, using a MySQL data base. Must be able to
understand and had experience in dealing with, CLI, PDD, ACD issues arising
from suppliers or customers.

MySQL, Administration of Database, MySQL knowledge has to be at a very
advanced level, stored procedures/triggers, replication and a strong
knowledge of AGI Scripting preferably in PHP (AGI-PHP scripts are used for
calling stored procedure from MySQL server)

Must have experience in using either SIP Express Router or OPEN SER, as we
will be deploying Kalamino throughout our Global network.

You will need skills in configuration, installation and integration of
various Asterisk applications like dial plans, IVR. Call recording,
voicemail etc. and experience troubleshooting *One way voice-path, NAT
issues, registration, etc. *


Analytical thinking and ability to adapt quickly to fast changing
requirements.

Required Skills  Qualifications:

Candidate must have good knowledge of setting up SIP and IAX Trunks.

Must have experience in installing and configuring SIP Express Router or
OPEN SER.

Installation and trouble shooting of  Asterisk Servers using Centos.

Installation and configuration PRI / E1s and Analogue cards mainly using
Digium Cards.

Good knowledge of Asterisk Dial Plans, maintaining and updating current dial
plans using   extension.conf as well as extensiosn.ael. 

Being able to write, maintain and update PHP pages linked to the MySQL data
base would be useful.

Scripting / Bash scripting would be useful.

Expert knowledge in Configuring, Maintaining and querying MySQL.

Expert level troubleshooting skills in inbound and outbound call flows.

 

 

 

Kind Regards
Jess

08451249555

 

Jess Hart
__
Langley James IT Recruitment

145-157 St John Street Clayton House
Clerkenwell59 Piccadilly
London  Manchester
EC1V 4PY   M1 2AQ

0845 124 95550845 225 5189
0207 788 66000161 660 7969


E-mail: j...@langleyjames.net mailto:ja...@langleyjames.co.uk 


 




Christian Savinovich
Telecom  Telephony Consulting
646.982.3572
c.savinov...@itntelecom.com

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[asterisk-users] Astribank Configuration Issues

[Don Kelly]

I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX
2.8.0.2.

 

We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we
had it working for a while with one NT PRI and one TE PRI and, in the
process of trying to configure another PRI, I ran into a couple problems.

 

(1) As my configuration changes didn't seem to affect the Astribank, I
power-cycled it. I found that it doesn't reload firmware automatically when
it's connected. I can force it to load, but am missing something to reload
automatically.

 

(2) I would appreciate a step-by-step suggestion of how I can make
configuration changes that propagate properly to the Astribank.

 

(3) I'd like to know if it's possible to determine what configuration has
been loaded into the Astribank without visiting the site and looking at the
lights.

 

I've spent quite a bit of time Googling, but haven't come up with the right
combination of stuff.

 

Thanks for any help you can give,

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

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Re: [asterisk-users] differential billing

[Don Kelly]

(Caution-top posting. Delete Before Reading if that's a problem for you.)

 

As I see it, real-time billing is only necessary if you have multiple people
making simultaneous calls against a prepaid balance or limited credit limit.

 

In other situations, you can simply make a determination at the beginning of
the call that it's prepaid or otherwise valid for a given period of time-4
minutes, 17 minutes, 24 hours+ or whatever. Then you can let it go 'til you
want to give a 2-minute warning, cut it off or whatever. The actual billing
would be after the call is terminated. Not much overhead.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Abdul Basit
Sent: Saturday, September 25, 2010 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] differential billing

 

Yes. you are right. I was thinking to avoid reinventing the wheel.

Will write AGIs. Trick is how to charge at 3min 59 sec or 4 min 01 sec
during live call.

 

We can monitor channel variables over AMI. But this will be a CPU overhead
(say for 100 or 200 calls) if we monitor channel variables on every second.
I want some thing to push channel details on each transition (or events like
IVR level changed, call duration updated to next minute) rather than i
request on AMI. Don't know if this logic is workable.

 

Just want a right direction.

 

-- 
Regards,

Abdul Basit | +92 32 1416 4196

 

 

 

 

   

 

On Sat, Sep 25, 2010 at 11:37 PM, Tarek Sawah tareksa...@hotmail.com
wrote:


if you are deploying your own system.. then you can build a small
application (AGI) that would do the math for you .. will devide the call
duration into the stages you want .. and does the calculation.. i think
MYSQL already can do that.. but a PHP script will do it faster and easier..
or like our billing system.. C# application interacting with Asterisk doing
all the math. after all it's all SQL and Asterisk working. you can do that
with a dial plan i believe.. so why not build an AGI to do it for you?



-- Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP USA: +13864929993









 From: basit.e...@gmail.com
 Date: Sat, 25 Sep 2010 23:27:56 +0500
 To: asterisk-users@lists.digium.com

 Subject: Re: [asterisk-users] differential billing


 Tarek,

 I already tested this feature with a2billing.

 This is difficult to extract the working code from a2billing.
 Also we are developing billing system so this is not a good idea
 to deploy another billing system in parallel.

 Any idea or link might help full.




 On Fri, Sep 24, 2010 at 9:30 PM, Tarek Sawah

  wrote:

 A quick answer? A2billing.

 It has what you call it differential billing.. but they call it
 progressive billing.. 3 steps .. for 3 different rates ..

 Go for it.. easy to setup and quick to learn and use.

 Regards



 From:
 asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Danny Nicholas
 Sent: Friday, September 24, 2010 4:19 PM

 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] differential billing



 

 From:
 asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of Abdul Basit

 Sent: Friday, September 24, 2010 8:13 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] differential billing



 Hi All,



 How can we develop a differential charging setup using asterisk like
 for 1st min we charge 1 cent, for 2nd min we charge 0.5 cent, for next
 30 sec charge @15cent, etc?



 Any idea, suggestion.

 --
 Regards,

 Abdul Basit | +92 32 1416 4196



 Since the CDR records the call duration in seconds, this should be a
 relative no-brainer, assuming you are billing post-call. If you are
 wanting to generate the charges during the live calls, AMI would be
 your best option for getting a running duration of the connection.


 --
 Regards,

 Abdul Basit | +92 32 1416 4196



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Re: [asterisk-users] Record() Cmd and My SQL

[Don Kelly]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, September 24, 2010 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd and My SQL

On Thu, Sep 23, 2010 at 11:23 PM, Govind, Mahesh (NSN - IN/Bangalore)
mahesh.gov...@nsn.com wrote:
 The reason is when doing a load balancing  , We  cannot confine the
 recording to a particular asterisk machine ( If we have more than one
 asterisk machine in the topology ).

Yes you can. You can record the file wherever the call takes place. In
fact, you can make the recording on any network segment the packet
traverses as well.

 So a centralized mechanism might be better . So that any machine can
 access the recording .
 Regards
 Mahesh

Recordings are formatted data, typically stored as files. You can put
them into a database, but you haven't provided a reason why that would
be a good idea.

There are these things called shared filesystems. You should take a
look at them. They work well. Options include NFS, iscsi, sans, etc.

Or you can record the file in-place, and when the recording completes,
copy it off to your shared filesystem. That's what I do.

Or you can take a look at something like OrecX, which let's you do
network spanning on your entire subnet, and it doesn't matter where
your call takes place because all RTP streams get written to disk.

None of what you've explained would be a good reason to put your
recordings into a database. Don't do that.

Don sez: I don't know how to make Outlook indent. I usually top-post, but I
don't like getting yelled at.

Why do you say Don't do that? Is there a real reason that it would be bad?

I'd like to put the recordings in a database so they are available to
another application that has no other relationship to the Asterisk server.
The application uses the database to determine if the recording has been
listened to, by whom and if it needs additional attention.

  --Don



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Re: [asterisk-users] Record() Cmd and My SQL

[Don Kelly]

I hadn't considered writing to the db real-time; was actually planning on
recording locally and moving it to the db.

Thanks for the suggestions.

  --Don


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, September 24, 2010 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Record() Cmd and My SQL

On Fri, Sep 24, 2010 at 1:32 PM, Don Kelly d...@donkelly.biz wrote:
 Don sez: I don't know how to make Outlook indent. I usually top-post, but
I
 don't like getting yelled at.

 Why do you say Don't do that? Is there a real reason that it would be
bad?

Performance is a real reason. Multiple simultaneous write streams into
a database sounds like a disaster. While trying to read from the db
and use it to listen to recordings sounds like a bigger disaster.

/path/to/the/recording

is a short varchar string

the actual recording is a massive, usually multi-megabyte, potentially
multi-gigabyte blob.
http://en.wikipedia.org/wiki/Blob_(computing)

If you're not actually taking advantage of the recording being in the
database, doing computing that is easier because of the database, such
as nearest neighbor searches, indexing, and the like, you're just
slowing down your ability to store and retrieve recordings.

 I'd like to put the recordings in a database so they are available to
 another application that has no other relationship to the Asterisk server.

Sounds like a filesystem. I can store my pdf file with my web browser,
and read it on another computer after I store it to my shared
filesystem.

 The application uses the database to determine if the recording has been
 listened to, by whom and if it needs additional attention.

Database can maintain metadata (as can a filesystem, owner, creation
date, access date), but you could still just store a pointer to the
actual file in the db. If you were paranoid about the filesystem and
db getting out of step you could do referential integrity checks in
the application.

If you want to do something wholesale to all the recordings, like
carve off the first five seconds, it's quite straightforward with a
batched sox call against the filesystem. If you want to do that in a
db, it's a select, write output to a file, convert the file, and
replace on the value to store it back into the db.

-- 
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Re: [asterisk-users] Help me Out!!!!

[Don Kelly]

He's fortunate that the hotel insists he stay there until his situation
improves.

--Don



Rough area. Consider yourself lucky you haven't been ripped apart :P

Pete wrote:
 I hope someone has helped poor Rob, I would as I am just over the bridge 
 in Bristol, UK but some evil internet scammer has stolen all my money! ;)
 
 Cheers!
 
 
 On 15/09/10 12:14, Rob Fugina wrote:
 It is with deep sorrow and broken heart that am sending you this mail. 
 Am in deep need and  my situation is lamentable.  my family and I 
 decide to come visit Wales,United Kingdom for  a short vacation. To 
 our greatest dismay we were attacked and ripped apart at the park of  
 the hotel where we were lodging,all cash,credit cards and cell phone 
 were forcefully robbed  off us at gun point but we still have our 
 passports with us.

 We've seek help at embassy and high commission,the Police too, 
 unfortunately they have  been unable to help or offer any reasonable 
 support whatsoever. Our flight leaves in  couple of hour from now but 
 we are being held to ransom by the hotel management because we  cannot 
 settle the hotel bills. It is clear we would not be allowed to leave 
 until pay the bill. Word cannot explain the anguish in my heart now. I 
 am in need of immediate assistance.

 Rob





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Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya - What are their current cost?

[Don Kelly]

He is looking for competitive information...what are prospects paying for
Avaya when they could be saving lots of money with Asterisk systems.

Probably a better question for the biz list, but he doesn't deserve the
responses he's getting.

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Thursday, September 02, 2010 12:11 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya -
What are their current cost?

El 02/09/10 11:32, bruce bruce escribió:
 I am not interested in open source solutions. 
Then what are your doing here?
 I want to know how much the propriety systems cost in terms of 
 licensing. Specially Avaya now a days per extension. Exclusive or 
 Inclusive of the hardware for 10 agents as noted.
Go to your Avaya daddy...

-- 
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center


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Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya- Whatare their current cost?

[Don Kelly]

It could be that I'm entirely confused, but I think he asked what people are
paying for Avaya solutions--so he'd know what competitive pricing would be
for the open source solution he's prepared to offer.

When someone replied with open-source suggestions, he pointed out that that
was not the information he was looking for. He did not say that he's not
interested in providing open source solutions for his clients.

--Don




-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, September 02, 2010 1:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Fw: [asterisk-biz] To compete with Avaya-
Whatare their current cost?

He doesn't deserve the responses, but it seems that boundaries are being
pushed in both sides of the response.  If he thinks he's on the biz list,
that's one thing, but in the purely open discussion, don't be dissing open
source either.


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[asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

I'm looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an old installation of Cardiff document stuff).

 

Is someone doing that?

 

If no one has direct experience, is there a HylaFax client that emulates
WinFax print-to-fax?

--Don

Don Kelly

PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

I looked at http://www.hylafax.org/content/Desktop_Client_Software and
visited several websites before posting this.

Nothing I saw said they would emulate WinFax print-to-fax.

I'd really appreciate hearing about any direct experiences.



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Monday, August 23, 2010 3:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff


On Mon, 23 Aug 2010, Don Kelly wrote:

 
 I?m looking for a way to use our implementation of HylaFax on Asterisk
with Cardiff (an
 old installation of Cardiff document stuff).
 
 Is someone doing that?
 
 If no one has direct experience, is there a HylaFax client that emulates
WinFax
 print-to-fax?


Lots!

http://www.hylafax.org/content/Desktop_Client_Software

j


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Re: [asterisk-users] Asterisk, HylaFax and Cardiff

[Don Kelly]

Thanks, Doug,

The Cardiff Teleforms application is unattended. The Winprint link that I
looked at shows that it uses a print dialog. Does it have an automagic mode,
too?

--Don



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, August 23, 2010 3:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk, HylaFax and Cardiff

Don Kelly wrote:
 I looked at http://www.hylafax.org/content/Desktop_Client_Software and
 visited several websites before posting this.

 Nothing I saw said they would emulate WinFax print-to-fax.


Not being familiar with WinFax, we currently use Winprint HylaFAX:  It 
creates a fax printer under Windows that allows printing directly to 
HylaFAX+

http://winprinthylafax.sourceforge.net/

Doug

-- 

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.


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