Re: [asterisk-users] Panasonic PBX connect to Asterisk (Ikka Tirtawidjaja)

2016-09-14 Thread Mc GRATH Ricardo 
Dear ikka
Thanks, well the best way is by  SIP to E1 ISDN gateway (could use Digium, 
Audiocodes, Sangoma, etc.).
But should contemplated on KX-TDA600 ISDN PRI card KX-TDA0290 and configure as 
QSIG signalling.
I don't recommend to use ATA FXS converter, first problem you couldn't dial to 
any extension to KX-TDA600, PBX answer incoming call from analogue line through 
Voice mail configured as Automated attendant, or DISA card.
The other point is line attenuation. 

Mc GRATH Ricardo
-- 
_
-- 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

[asterisk-users] Customizing the messages for voice mail

2016-09-14 Thread D'Arcy J.M. Cain 
I know that users can create their own messages but I can't see how to
change the extension number.  Here is my scenario.

Users extensions can be any string.  Sometimes it is an actual phone
number but often it is their user ID.  Let's say that two of them are
foo and foo3.

I also provide virtual PBX extensions that accept an extension and
direct the call to a device.  A phone can be in more than one PBX so
calling 41 might assign extension 200 to foo and 201 to foo3.
Calling 416555 might assign 210 to foo and 220 to foo3.

All of this works great until a call goes to voice mail.  Now the foo
extension says "The person at extension  is not
available"...  The message for foo3, which might be extension 201 or 220
in the above example depending on which PBX was called will say "The
person at extension 3 is not available..."

So my question is this.  Is there some way in the dialplan to set the
number that is used for that message?  Is there a variable that can be
set before doing a GoTo to foo or foo3?

Thanks.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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

Re: [asterisk-users] [SOLVED] Re: Feature Request: what about "core stop panic" ?

2016-09-14 Thread Tzafrir Cohen 
On Wed, Sep 07, 2016 at 01:41:55PM -0600, George Joseph wrote:
> On Wed, Sep 7, 2016 at 11:03 AM, Olivier  wrote:

> > My system shows:
> > # ps aux | grep asteri
> > asterisk   429  7.3  2.4  59468 25088 ?Ssl  18:47   0:03
> > /usr/sbin/asterisk -U asterisk -G asterisk -g
> > ...
> > # sysctl kernel.core_pattern
> > kernel.core_pattern = core
> >
> 
> Since "core" is a relative file name, the file will be in whatever the
> working directory is for the process.  You may have to hunt it down.  For
> better debugging, you might want to set core_pattern to something like
> " /tmp/core-%e-%t".  That way all core files will have a name like
> "/tmp/core-asterisk-1473164587.7705".  "man core" should give you more info
> on constructing the file name.

But looking at the process of asterisk may help:

  ls -l /proc/$PID_OF_ASTERISK/cwd

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com

-- 
_
-- 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

[asterisk-users] Asterisk 13 externip

2016-09-14 Thread Madushan Geethanga 
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
-- 
_
-- 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

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

2016-09-14 Thread Harry McGregor 

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





--
_
-- 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







-- 
_
-- 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