pstn                                                           pstn

asterisk                 link between                 avaya pbx

both systems tied together by 2 pri’s
both have trunks out to the pstn
want to get rid of the avaya pstn trunk and send thru my asterisk box
avaya still has inbound calls on this trunk until late november (at&t
is dragging their feet doing the porting – 8 weeks between ) and still
has stations that we are not in a position to migrate to asterisk just
yet (about 500)
can get the call to show on the link between systems, but asterisk has
no station associate it with so it drops?
any suggestions out there?

2010/10/24 Olivier

> Hello,
>
> My setup is :
> phone —– PSTN/ISDN —– Patton SN4638 ——- Asterisk
>
> (Asterisk is in 1.6.1.18, Patton in 5.3)
>
> When I call the Asterisk, I can read from console that :
> – the call comes in,
> – the line MusicOnHold(,10) in my diaplan is reached and played,
> – I see RTP packets coming in and out
> (hundreds of lines such as:
> Got RTP packet from 192.168.102.200:4890 (type 00, seq 005360, ts
> 2343932047, len 000160)
> Sent RTP packet to 192.168.102.200:4890 (type 00, seq 036824, ts
> 082080, len 000160)
> )
> – but I can’t hear anything on my end.
>
> When I tried this system locally, I could hear the MOH (but now, I can’t
> try any local operation from my location).
>
> I was thinking of a codec issue but I can’t see why and where.
>
> Suggestions ?
>
> Regards
>
> PS: If this helps, I’m using default MOH .wav files in
/var/lib/asterisk/moh directory.

Hello people, this is my first mail to this list. I’m new to asterisk
and trying to set up an IVR. So far my dialplan works nice connecting
with softphones, but I’m having problems to detect hangups on the analog
line.
Here are the details:

-Clone pc with Ubuntu server 10.04 64 bits
-Asterisk 1.6
-Openvox A400P, lspci -v says:
01:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
Subsystem: Device b100:0001
Flags: bus master, medium devsel, latency 32, IRQ 21
I/O ports at c800 [size=256]
Memory at ff0ff000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2
Kernel driver in use: wctdm
Kernel modules: wctdm, wcopenpci, hisax, netjet

And this is the scheme:

PSTN <-----> Analog Panasonic PBX <-----> Asterisk

The problem as you already may know, is the busy/hangup tone generated
by the Panasonic PBX, connecting the asterisk boc directly to the pstn
works fine.
The Panasonic model is KX-TA616, as far as I understand, the manual[0]
says that it can detect Polarity reversion, but not generate it.
I recorded the hangup sound (using Record()), converted and opened in
audacity: it’s 260ms tone, 250ms
So I configured:
busydetect=yes
busycount=3
busypattern=260,250

But sill can’t detect the hangup.

The tone can be downloaded here http://cringer.3kh.net/web/hangup-tone.wav
And the dahdi configuration is here:
http://cringer.3kh.net/web/chan_dahdi.conf
http://cringer.3kh.net/web/dahdi-channels.conf

Let me know if you need any other details on the setup.

Thank you very much in advance.

[0] http://smarthomeuae.com/new3/product.php?productid=218&pcategory=16001

PS: please excuse my bad english.

I’m running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I’d like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.

What’s the best way to go about doing this? I’ve looked into Fax for
Asterisk, bit I’m not sure that I want it or NVFax detection.

Any pointers to share?

Hello community,

I have successfully set up asterisk free PBX server and I am also able to connect to it by softphone.

Now as next step I want to extend this and Connect Asterisk to PSTN,

My Required scenario:

I need a number which will connect outside PSTN world to my PBX and by applying extension particular softphone or connected normal phone should get connected.

Which hardware I need for it.
Also please explain a bit of dial plans.