date:20160913

[asterisk-users] Panasonic PBX connect to Asterisk

2016-09-13 Thread Ikka Tirtawidjaja

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


Thanks in advance,


Regards,

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

2016-09-13 Thread Mc GRATH Ricardo

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it 
only support H323 Trunk through 4 or 16 channels gateway card  and  TDM 
technology with  ISDN BRI and PRI card.

Mc GRATH Ricardo
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[asterisk-users] Wanpipe 7 Driver problem with Debian 8.5 (Sangoma B600DE)

2016-09-13 Thread Guillermo Di Donato

Hi all,

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.

If I try the same process with Debian 7 (Wheezy), installation works just
fine.

Has anybody gone trough this scenario? Has anybody managed to have Snagoma
B600DE working on Debian 8.5?

Regards,

Guillermo
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Re: [asterisk-users] Need ISDN call generator

2016-09-13 Thread Storer, Darren

Keep an eye out for older model INET Spectra call generators, with ISDN /
SS7 stacks. These days the old boxes are being sold off very cheaply on
popular auction sites.

Hammer was the other popular call generator hardware that you might find
being sold at a fraction of the original cost.

HTH

Darren

On 28 August 2016 at 10:20, Hooman Fazaeli  wrote:

>
> Hi
>
> To troubleshoot FreeBSD panics triggered by ISDN load on an asterisk
> system,
> we are looking to buy an ISDN call generator/simulator device.
>
> The minimum requirements include:
>
> - Not too expensive
> - PRI support (BRI support is a plus)
> - CCS+CRC4 farming + HDB3 coding
> - EuroISDN (DSS1) support.
> - A minimum of 4 ports (120 channels/concurrent calls)
> - Compatibility with Digium cards.
> - DUT in TE mode.
> - Reliable & stable operation.
>
> I would like to hear your recommendations for and experiences about
> such a device. Recommendations on hand crafted systems using
> Asterisk, DAHDI and PRI cards on any OS which has worked stably for
> someone are also welcome
>
> Thanks.
>
>
> --
> Best regards
> Hooman Fazaeli
>
>
> --
> _
> -- 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
>
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Re: [asterisk-users] Wanpipe 7 Driver problem with Debian 8.5 (Sangoma B600DE)

2016-09-13 Thread Julian Beach

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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[asterisk-users] Recording "Never" on extension not stopping recording

2016-09-13 Thread Ian Harding

This must be something simple.  We have recording starting on all our
inbound calls and it works fine.  I'd like to have the option to stop
recording when transferred to certain extensions.

I set extension 7077 to "Never" and the log makes it look like the
recording was stopped but it was not.

Any ideas how to troubleshoot?

[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:1] NoOp("SIP/6010-000386f2", "Exten Recording
Check between 6010 and 7077") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:2] Set("SIP/6010-000386f2", "CALLTYPE=internal")
in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:3] ExecIf("SIP/6010-000386f2",
"0?Set(CALLTYPE=)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:4] Set("SIP/6010-000386f2", "CALLEE=never") in
new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:5] ExecIf("SIP/6010-000386f2",
"0?Set(CALLEE=dontcare)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:6] GotoIf("SIP/6010-000386f2", "0?callee") in
new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:7] GotoIf("SIP/6010-000386f2", "0?caller") in
new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:8] ExecIf("SIP/6010-000386f2",
"1?Set(CALLER_PRI=0):Set(CALLER_PRI=0)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:9] ExecIf("SIP/6010-000386f2",
"1?Set(CALLEE_PRI=0):Set(CALLEE_PRI=0)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:10] GotoIf("SIP/6010-000386f2",
"1?caller:callee") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
(sub-record-check,exten,13)
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:13] Set("SIP/6010-000386f2", "RECMODE=dontcare")
in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:14] ExecIf("SIP/6010-000386f2",
"0?Set(RECMODE=dontcare)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:15] ExecIf("SIP/6010-000386f2",
"1?Set(RECMODE=never)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:16] Gosub("SIP/6010-000386f2",
"recordcheck,1(never,internal,7077)") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:1] NoOp("SIP/6010-000386f2", "Starting
recording check against never") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:2] Goto("SIP/6010-000386f2", "never") in
new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
(sub-record-check,recordcheck,14)
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:14] Set("SIP/6010-000386f2",
"__REC_POLICY_MODE=NEVER") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:15] Goto("SIP/6010-000386f2", "stoprec")
in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
(sub-record-check,recordcheck,25)
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:25] NoOp("SIP/6010-000386f2", "Stopping
recording: internal, 7077") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:26] Set("SIP/6010-000386f2",
"__REC_STATUS=STOPPED") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:27] System("SIP/6010-000386f2",
"/var/lib/asterisk/bin/stoprecording.php "SIP/6010-000386f2"") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[recordcheck@sub-record-check:28] Return("SIP/6010-000386f2", "") in new
stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[exten@sub-record-check:17] Return("SIP/6010-000386f2", "") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[s@macro-exten-vm:7] GotoIf("SIP/6010-000386f2", "1?macrodial") in new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
(macro-exten-vm,s,13)
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[s@macro-exten-vm:13] GosubIf("SIP/6010-000386f2", "0?clrheader,1()") in
new stack
[2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
[s@macro-exten-vm:14] Macro("SIP/6010-

Re: [asterisk-users] Recording "Never" on extension not stopping recording

2016-09-13 Thread John Kiniston

I'd recommend trying the support resources for the GUI that is managing
your asterisk installation.

This looks like it might be FreePBX dialplan logic to me, Most likely it
won't be something that the list will be able to help you modify.

http://community.freepbx.org/ is the FreePBX Community support forum.

On Tue, Sep 13, 2016 at 4:32 PM, Ian Harding 
wrote:

> This must be something simple.  We have recording starting on all our
> inbound calls and it works fine.  I'd like to have the option to stop
> recording when transferred to certain extensions.
>
> I set extension 7077 to "Never" and the log makes it look like the
> recording was stopped but it was not.
>
> Any ideas how to troubleshoot?
>
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:1] NoOp("SIP/6010-000386f2", "Exten Recording
> Check between 6010 and 7077") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:2] Set("SIP/6010-000386f2", "CALLTYPE=internal")
> in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:3] ExecIf("SIP/6010-000386f2",
> "0?Set(CALLTYPE=)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:4] Set("SIP/6010-000386f2", "CALLEE=never") in
> new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:5] ExecIf("SIP/6010-000386f2",
> "0?Set(CALLEE=dontcare)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:6] GotoIf("SIP/6010-000386f2", "0?callee") in
> new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:7] GotoIf("SIP/6010-000386f2", "0?caller") in
> new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:8] ExecIf("SIP/6010-000386f2",
> "1?Set(CALLER_PRI=0):Set(CALLER_PRI=0)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:9] ExecIf("SIP/6010-000386f2",
> "1?Set(CALLEE_PRI=0):Set(CALLEE_PRI=0)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:10] GotoIf("SIP/6010-000386f2",
> "1?caller:callee") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
> (sub-record-check,exten,13)
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:13] Set("SIP/6010-000386f2", "RECMODE=dontcare")
> in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:14] ExecIf("SIP/6010-000386f2",
> "0?Set(RECMODE=dontcare)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:15] ExecIf("SIP/6010-000386f2",
> "1?Set(RECMODE=never)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:16] Gosub("SIP/6010-000386f2",
> "recordcheck,1(never,internal,7077)") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:1] NoOp("SIP/6010-000386f2", "Starting
> recording check against never") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:2] Goto("SIP/6010-000386f2", "never") in
> new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
> (sub-record-check,recordcheck,14)
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:14] Set("SIP/6010-000386f2",
> "__REC_POLICY_MODE=NEVER") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:15] Goto("SIP/6010-000386f2", "stoprec")
> in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Goto
> (sub-record-check,recordcheck,25)
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:25] NoOp("SIP/6010-000386f2", "Stopping
> recording: internal, 7077") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:26] Set("SIP/6010-000386f2",
> "__REC_STATUS=STOPPED") in new stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:27] System("SIP/6010-000386f2",
> "/var/lib/asterisk/bin/stoprecording.php "SIP/6010-000386f2"") in new
> stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [recordcheck@sub-record-check:28] Return("SIP/6010-000386f2", "") in new
> stack
> [2016-09-13 14:42:46] VERBOSE[30146][C-0002c851] pbx.c: -- Executing
> [exten@sub-record-check:17] Return("SIP/6010-0003

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

2016-09-13 Thread Ikka Tirtawidjaja

Dear Mc Grath R,

Thx for the information.

If that TDA-600 can not use SIP Trunking, can we use "SIP to E1
converter/gateway" between my Asterisk Server and Panasonic KX-TDA600 ?

or from asterisk we convert the sip phone line with ATA Converter / FXS
Gateway, and then we connect the output (RJ11) to C0 on Panasonic TDA600 ?

Any recomendation for SIP to E1 gateway ?

Thx in advance

Ikka

On Wed, Sep 14, 2016 at 12:13 AM, Mc GRATH Ricardo 
wrote:

> Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP
> technology it only support H323 Trunk through 4 or 16 channels gateway
> card  and  TDM technology with  ISDN BRI and PRI card.
>
> 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
>
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Re: [asterisk-users] Panasonic PBX connect to Asterisk

2016-09-13 Thread Harry McGregor


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] Panasonic PBX connect to Asterisk

2016-09-13 Thread Ikka Tirtawidjaja

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

2016-09-13 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 
mailto:hmcgre...@biggeeks.org>> 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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2016
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[asterisk-users] Panasonic PBX connect to Asterisk

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

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