Re: [asterisk-users] Panasonic PBX connect to Asterisk

[C F]

Use a sip to PRI gateway or a PRI card in the asterisk system. Connect that
to the Panasonic TDA600 using a PRI card on the panasonic side (KX-TDA0290).
this will be the most worry free solution.

On Wed, Sep 14, 2016 at 2:05 AM, Harry McGregor 
wrote:

> Hi,
>
>
> You need to find out more about the configuration of this specific TDA600,
> as it could be either POTS or E1, once you know that, you can determine
> what options are best.
>
> -Harry
>
> On 09/13/2016 10:51 PM, Ikka Tirtawidjaja wrote:
>
> Dear Harry,
>
> Thx for the explanation.
>
> My team manage building's PBX that use Asterisk 13.x.
> We use Asterisk PBX for this buildings that have apartment and office
> customer.
> From my Asterisk PBX, we connect to IPPhone (yealink) or ATA Converter
> (cisco SPA112).
> Others are using PBX like panasonic analog, audiocodes SBC, etc, and we
> use ATA Converter to convert from SIP to Analog (CO Line)
>
> Now, we have a new customer (tenant) that have Panasonic TDA600.
> If we use FXS or ATA Converter, its going to have a lot of that, because
> this tenant going to use about 60 ext / sip line.
> Replacing asterisk PBX on my (company) side or replace TDA600 on my
> customer side is not acceptable.
> So we need to find a "win-win" solution for this.
>
> Thx in advance,
>
>
> Ikka
>
>
>
>
> On Wed, Sep 14, 2016 at 12:40 PM, Harry McGregor 
> wrote:
>
>> Hi,
>>
>> On 09/13/2016 06:51 AM, Ikka Tirtawidjaja wrote:
>>
>> Hi,
>>
>> Is there anyone here who has experience connecting Asterisk (ver 13.8)
>> with PBX Panasonic KX-TDA600 ?
>>
>> The architecture more less like this :
>>
>>
>> Telco Sip Trunk ---> Asterisk 13 ---> Panasonic KX-TDA600 ---> Phone / Fax
>>
>>
>> What connectivity do you currently use for the KX-TDA600?  E1, T1, POTS,
>> BRI?
>>
>> Others have suggested a SIP to E1/T1 gateway, which would let you skip
>> the asterisk box, if you don't have other uses for it.
>>
>> Another option is to use a PCI-E E1/T1 interface card in the asterisk
>> box, especially if you already have an E1 or T1 interface in the KX-TDA600.
>> I personally don't like buying smaller then a dual T1/E1 card, as the price
>> difference between a dual and a single is so small.  If the KX-TDA600 is
>> set-up for Analog/POTS, you can use a channel bank on the second T1/E1
>> port, and feed POTS into the KX-TDA600.
>>
>> For a small installation that wanted to keep their Nortel Key System, and
>> their Telco really wanted to provide a PRI instead of POTS (the Nortel
>> could only take pots), we used a dual T1 PCI card in an asterisk box, ran
>> PRI on the Telco interface, an ADIT 600 channel bank on the second
>> interface, and handed 4 POTS lines to the Nortel Key System.
>>
>> The key is to give your self the most flexibility to change later, and
>> preserve your existing investment.
>>
>> -Harry
>>
>> Thanks in advance,
>>
>>
>> Regards,
>>
>> Ikka - Jakarta, Indonesia
>>
>>
>>
>>
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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Leandro Dardini]

No, there is no Music On Hold starting and the bad thing is the call
duration reported by asterisk was just few seconds while the call duration
reported by the provider was few thousand seconds, the max allowed. So they
will be able to terminate the call on the asterisk side and have it run on
the provider side.

Leandro

2016-09-15 19:18 GMT+02:00 Max Grobecker :

> Maybe the client just put the call on hold.
> So the call technically has not ended AND the client does not need to send
> or handle any RTP data.
> Is there any mention of "music on hold" for this channel?
>
> Greetings
>  Max
>
>
> - Nachricht von Leandro Dardini  -
>  Datum: Thu, 15 Sep 2016 18:06:14 +0200
>Von: Leandro Dardini 
> Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>Betreff: [asterisk-users] Tricking asterisk to think the call has
> ended, but it was continuing on the other side
> An: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
>
>
> I am banging my head over a simple asterisk trick I was seeing on one
>> asterisk server.
>>
>> An extension dials an international premium number, the called number
>> answers, then the extension hangups, but the call continue to run on the
>> international number side, generating an high profit for the premium
>> number
>> company and a big loss for the asterisk owner.
>>
>> I think some sort of "transfer" takes place, but I can't identify how they
>> do it and most important, how to prevent it.
>>
>
> - Ende der Nachricht von Leandro Dardini  -
>
>
>
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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Max Grobecker]

Maybe the client just put the call on hold.
So the call technically has not ended AND the client does not need to
send or handle any RTP data.
Is there any mention of "music on hold" for this channel?

Greetings
 Max


- Nachricht von Leandro Dardini  -
 Datum: Thu, 15 Sep 2016 18:06:14 +0200
   Von: Leandro Dardini 
Antwort an: Asterisk Users Mailing List - Non-Commercial Discussion

   Betreff: [asterisk-users] Tricking asterisk to think the call has
ended, but it was continuing on the other side
An: Asterisk Users Mailing List - Non-Commercial Discussion




I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.

An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.


- Ende der Nachricht von Leandro Dardini  -




pgpjNbRGpcjUL.pgp
Description: Digitale PGP-Signatur
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Re: [asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Dovid Bender]

The best is to get a PCAP so you can see exactly what is going on. Look
into voipmonitor.org or homersip to capture all of your traffic. There are
many ways that people commit fraud. If you are thinking about transfers
what they do is the call the fraudulent number then they send a 302  to
another number so they are now calling twice and the call is not on your
network.

On Thu, Sep 15, 2016 at 12:06 PM, Leandro Dardini 
wrote:

> I am banging my head over a simple asterisk trick I was seeing on one
> asterisk server.
>
> An extension dials an international premium number, the called number
> answers, then the extension hangups, but the call continue to run on the
> international number side, generating an high profit for the premium number
> company and a big loss for the asterisk owner.
>
> I think some sort of "transfer" takes place, but I can't identify how they
> do it and most important, how to prevent it.
>
> Here the relevant logs:
>
> [2016-09-08 21:00:25] VERBOSE[18771][C-066c] pbx.c: Executing
> [0021628990XXX@dialoutbound:595] Dial("SIP/201-boxoffice-0f66",
> "SIP/0021628990XXX@SBC002_VirginMedia,60,T") in new stack
> [2016-09-08 21:00:25] VERBOSE[18771][C-066c] app_dial.c: Called
> SIP/0021628990XXX@SBC002_VirginMedia
> [2016-09-08 21:00:27] VERBOSE[18771][C-066c] app_dial.c:
> SIP/SBC002_VirginMedia-0f67 answered SIP/201-boxoffice-0f66
> [2016-09-08 21:00:27] VERBOSE[18771][C-066c] bridge_channel.c: Channel
> SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
> <00bd58c3-3bce-4f1b-9d79-11eb96f37260>
> [2016-09-08 21:00:27] VERBOSE[18779][C-066c] bridge_channel.c: Channel
> SIP/SBC002_VirginMedia-0f67 joined 'simple_bridge' basic-bridge
> <00bd58c3-3bce-4f1b-9d79-11eb96f37260>
> [2016-09-08 21:00:28] VERBOSE[18771][C-066c] bridge_channel.c: Channel
> SIP/201-boxoffice-0f66 left 'simple_bridge' basic-bridge
> <00bd58c3-3bce-4f1b-9d79-11eb96f37260>
> [2016-09-08 21:00:28] VERBOSE[18779][C-066c] bridge_channel.c: Channel
> SIP/SBC002_VirginMedia-0f67 left 'simple_bridge' basic-bridge
> <00bd58c3-3bce-4f1b-9d79-11eb96f37260>
>
> Any idea?
>
> Leandro
>
>
>
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[asterisk-users] Tricking asterisk to think the call has ended, but it was continuing on the other side

[Leandro Dardini]

I am banging my head over a simple asterisk trick I was seeing on one
asterisk server.

An extension dials an international premium number, the called number
answers, then the extension hangups, but the call continue to run on the
international number side, generating an high profit for the premium number
company and a big loss for the asterisk owner.

I think some sort of "transfer" takes place, but I can't identify how they
do it and most important, how to prevent it.

Here the relevant logs:

[2016-09-08 21:00:25] VERBOSE[18771][C-066c] pbx.c: Executing
[0021628990XXX@dialoutbound:595] Dial("SIP/201-boxoffice-0f66",
"SIP/0021628990XXX@SBC002_VirginMedia,60,T") in new stack
[2016-09-08 21:00:25] VERBOSE[18771][C-066c] app_dial.c: Called
SIP/0021628990XXX@SBC002_VirginMedia
[2016-09-08 21:00:27] VERBOSE[18771][C-066c] app_dial.c:
SIP/SBC002_VirginMedia-0f67 answered SIP/201-boxoffice-0f66
[2016-09-08 21:00:27] VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:27] VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 joined 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:28] VERBOSE[18771][C-066c] bridge_channel.c: Channel
SIP/201-boxoffice-0f66 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>
[2016-09-08 21:00:28] VERBOSE[18779][C-066c] bridge_channel.c: Channel
SIP/SBC002_VirginMedia-0f67 left 'simple_bridge' basic-bridge
<00bd58c3-3bce-4f1b-9d79-11eb96f37260>

Any idea?

Leandro
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Re: [asterisk-users] Asterisk 13 externip

[George Joseph]

On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad 
wrote:

>
>
> On Wednesday, 14 September 2016, Madushan Geethanga <
> mgliyanage...@gmail.com> wrote:
>
>> Hi,
>>
>> What is the equal option for externip in asterisk 13 with pjsip. I have
>> tried
>>
>> external_media_address=XX.XX.XX.XX
>> external_signaling_address=XX.XX.XX.XX
>>
>> but asterisk 13 writes local ip to the from header. any suggestions?
>>
>
Setting 'from_domain' on the endpoint will do it.  Are you having issues
with an internal address being used in the "From" header?



>
>> Best Regards,
>> Madushan
>>
>>
>>
>
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> Sent from Gmail Mobile
>
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Re: [asterisk-users] Asterisk 13 externip

[George Joseph]

On Thu, Sep 15, 2016 at 8:38 AM, Madushan Geethanga  wrote:

> Hi,
>
> Thanks for the reply.
>
> Yes my PABX is on the cloud when I try to register to the server, the
> server  sends registration OK with public address but  OPTION method
> includes the private address of the server  in from header not the public
> address. I have include both
>
> external_media_address=XX.XX.XX.XX
> external_signaling_address=XX.XX.XX.XX
> local_net=XX.XX.XX.XX
>
>
> The client AOR is not getting registered.
>

What type of device/softphone is the client?
Is the client trying to respond back to the address in the From header
instead of the Contact header?


>
> Best Regards,
> Madushan
>
> On Thu, Sep 15, 2016 at 7:41 PM, George Joseph  wrote:
>
>>
>>
>> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad 
>> wrote:
>>
>>>
>>>
>>> On Wednesday, 14 September 2016, Madushan Geethanga <
>>> mgliyanage...@gmail.com> wrote:
>>>
 Hi,

 What is the equal option for externip in asterisk 13 with pjsip. I have
 tried

 external_media_address=XX.XX.XX.XX
 external_signaling_address=XX.XX.XX.XX

 but asterisk 13 writes local ip to the from header. any suggestions?

>>>
>> Setting 'from_domain' on the endpoint will do it.  Are you having issues
>> with an internal address being used in the "From" header?
>>
>>
>>
>>>
 Best Regards,
 Madushan



>>>
>>> --
>>> Sent from Gmail Mobile
>>>
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>>> _
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>>>
>>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>>   http://www.asterisk.org/community/astricon-user-conference
>>>
>>> New to Asterisk? Start here:
>>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>>
>> --
>> George Joseph
>> Digium, Inc. | Software Developer
>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>> Check us out at: www.digium.com & www.asterisk.org
>>
>>
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>> _
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
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Re: [asterisk-users] Asterisk 13 externip

[Madushan Geethanga]

Hi,

Thanks for the reply.

Yes my PABX is on the cloud when I try to register to the server, the
server  sends registration OK with public address but  OPTION method
includes the private address of the server  in from header not the public
address. I have include both

external_media_address=XX.XX.XX.XX
external_signaling_address=XX.XX.XX.XX
local_net=XX.XX.XX.XX


The client AOR is not getting registered.

Best Regards,
Madushan

On Thu, Sep 15, 2016 at 7:41 PM, George Joseph  wrote:

>
>
> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad 
> wrote:
>
>>
>>
>> On Wednesday, 14 September 2016, Madushan Geethanga <
>> mgliyanage...@gmail.com> wrote:
>>
>>> Hi,
>>>
>>> What is the equal option for externip in asterisk 13 with pjsip. I have
>>> tried
>>>
>>> external_media_address=XX.XX.XX.XX
>>> external_signaling_address=XX.XX.XX.XX
>>>
>>> but asterisk 13 writes local ip to the from header. any suggestions?
>>>
>>
> Setting 'from_domain' on the endpoint will do it.  Are you having issues
> with an internal address being used in the "From" header?
>
>
>
>>
>>> Best Regards,
>>> Madushan
>>>
>>>
>>>
>>
>> --
>> Sent from Gmail Mobile
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
>>   http://www.asterisk.org/community/astricon-user-conference
>>
>> New to Asterisk? Start here:
>>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> George Joseph
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
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> _
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Re: [asterisk-users] Asterisk 13 externip

[Faheem Muhammad]

On Wednesday, 14 September 2016, Madushan Geethanga 
wrote:

> Hi,
>
> What is the equal option for externip in asterisk 13 with pjsip. I have
> tried
>
> external_media_address=XX.XX.XX.XX
> external_signaling_address=XX.XX.XX.XX
>
> but asterisk 13 writes local ip to the from header. any suggestions?
>
> Best Regards,
> Madushan
>
>
>

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Re: [asterisk-users] mysql phonebook

[A J Stiles]

On Thursday 15 Sep 2016, tux john wrote:
> hi. i am running asterisk 11 and i am using astdb to store all my contacts
> and their numbers. so everytime they call me, i can see their name on the
> screen of the phone. i am making use of the following to retrieve the name
> from the astdb exten =>
> WhatEverIsMyDID,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
> exten => WhatEverIsMyDID,n,Answer()
>  
> in the same machine i have mysql. i would like to make use of mysql to
> store and retrieve phonebook as well create blacklist of numbers. i
> thought of creating 2 databases
> -phonebook, will contain name, number
> -blacklist, will contain name, number
>  
> once i update all the database fiels how can i see the names whenever
> someone calls? regarding the blacklist, i would like to send them to hear
> a sound (eg tt-monkeys) or simply hangup. 

Write an AGI script that expects a phone number as its parameter, performs a 
database lookup on the number and sets some channel variables with the 
caller's name and whether or not they are blacklisted.  You probably need only 
one table, really; use VARCHAR() fields for the number and name and something 
like a TINYINT(1) for indicating whether or not the number is blacklisted.  
After the script exits, the dialplan will see any variables it set.  So you 
can do something like this;

exten => s,1,Set(from=${CALLERID(num)})
exten => s,n,(Incoming call from ${from})
exten => s,n,AGI(lookup_caller.agi,${from})
; /var/lib/asterisk/agi-bin/lookup_caller.agi sets variables `blocked` to true
; if the caller is blocked, and `callername` to the caller's name
exten => s,n,GotoIf(${blocked}?unwelcome:permitted)
exten => s,n(permitted),NoOp(This call is allowed)
exten => s,n,Set(CALLERID(name)=${callername})
; we can maybe do something else funky with callername here
exten => s,n,Dial(${ALL_EXTS})
exten => s,n,Hangup()
; tell unwanted callers where to stick that phone
exten => s,n(unwelcome),MP3player(/songs/kevin_bloody_wilson/dicktaphone.mp3)
exten => s,n,Hangup()


I used some simple example code to implement a little daemon on users' 
workstations; which listened on a UDP port, and created system notifications 
informing the user of an incoming call.  As this was all on the inside of a 
firewall, I also included the capability to open up a web page.  The AGI script 
was able not only to notified the workstation adjacent to the phone of the 
incoming call; but if the number was recognised as belonging to a user within 
our system, would bring up their details on screen  (all the work was done 
through a custom web application).

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
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Re: [asterisk-users] mysql phonebook

[Doug Lytle]

I do this.  I connect back to the mysql server via odbc, and as you, I have two 
databases for this, one called blacklisted and the other called speeddials.

My dialplan code below:


exten => _5X,1,Answer()
exten => _5X,n,Gosub(check_blacklist,s,1)
exten => _5X,n,Gosub(get_callerid,s,1)


[check_blacklist]

exten => s,1,GotoIf($["${CALLERID(number)}" = "" ]?2:3)
exten => s,n,Set(CALLERID(all)=Restricted <0>)
exten => s,n,Set(ARRAY(flag,note)=${ODBC_BLACKLIST(${CALLERID(number)})})
exten => s,n,GotoIf($["${flag}" = "YES"]?blacklisted,s,1)
exten => s,n,NoOP(Caller not blacklisted)
exten => s,n,Return

[blacklisted]

exten => s,1,NoOP(Caller: ${CALLERID(number)} is on the black list)
exten => s,n,NoOP(NOTE: ${note})
exten => s,n,Set(CDR(userfield)=Blacklisted)
exten => s,n,Zapateller(answer)
exten => s,n,Hangup(2)


The ODBC query is:

[BLACKLIST]
dsn=MySQL-blacklisted
readsql=SELECT flag, note FROM [putyourdatabasenamehere] WHERE 
phone=${SQL_ESC("${ARG1}")}


[get_callerid]

exten => 
s,1,Set(ARRAY(speed.dial,speed.name)=${ODBC_GET_CALLERID(${CALLERID(num)})})
exten => s,n,Set(CALLERID(name)=${speed.name})
exten => s,n,Return()

The ODBC query is:

[GET_CALLERID]
dsn=MySQL-speeddials
readsql=SELECT phone, name, code FROM [putyourdatabasenamehere] WHERE phone = 
${ARG2}

Doug

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Re: [asterisk-users] Wanpipe 7 Driver problem with Debian 8.5 (Sangoma B600DE)

[Guillermo Di Donato]

Thank you very much for your feedback Julian. I agree with you, Wanpipe is
a pain to install. For the moment I will stick to Debian Wheezy (kernel
3.2), but it bothers me not to be able to go for a complete distro update.
Let´s see if somebody else had luck with this process.

Best regards,
Guillermo


On Tue, Sep 13, 2016 at 7:51 PM, Julian Beach  wrote:

> Hello Guillermo,
>
> On Tuesday, September 13, 2016, 7:10:29 PM, you wrote:
>
> > I have tried to install Wanpipe 7.0.20 in a fresh-installed Debian
> > 8.5 (Jessie) server, but when I reach to the point where I have to
> > enter the Linux Headers directory (installed in default with
> > “apt-get install linux-headers-$(uname -r)”), I get an error, the
> > package refuses to compile and installation aborts.
>
> Wanpipe installs fill me with dread! Sangoma say that they only
> support CentOS 6.x installs, and after many struggles to get Wanpipe
> to work on Fedora versions, that is what I run it on (CentOS 6.7).
> Even then, install issues are not uncommon, and it seems to be
> particularly sensitive to kernel issues - I avoid kernel updates on my
> Asterisk box!
>
> I am currently running with kernel 2.6.32. Sangoma says that Wanpipe
> will work on 4.3 kernels, but if your kernel is newer than 4.3
> (November 2015) then that might be your problem.
> "setup_drv_compile.log" is the place to look for clues. You should
> find it in the directory where your Wanpipe source files are.
>
>
>
> --
> Best regards,
>  Julianmailto:jb_s...@trink.co.uk
>
>
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>
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>
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[asterisk-users] mysql phonebook

[tux john]

hi. i am running asterisk 11 and i am using astdb to store all my contacts and their numbers. so everytime they call me, i can see their name on the screen of the phone.
 i am making use of the following to retrieve the name from the astdb

exten => WhatEverIsMyDID,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => WhatEverIsMyDID,n,Answer()

 

in the same machine i have mysql. i would like to make use of mysql to store and retrieve phonebook as well create blacklist of numbers.

i thought of creating 2 databases

-phonebook, will contain name, number

-blacklist, will contain name, number

 

once i update all the database fiels how can i see the names whenever someone calls?

regarding the blacklist, i would like to send them to hear a sound (eg tt-monkeys) or simply hangup.

 

Some advice please?

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