* You are viewing the archive for November, 2011

Walkie talkie to sip phone interfacere:

> I’ve been trying to find a solution that would allow our sip phones to
> communication with walkie talkies. Our setup is that we have sip phones
> setup in 2 locations, headquarters and dome. We can communication from
> headquarters and dome through sip phones, but within the dome we have
> technicians that use walkie talkies to communicate as they go about
> their work. Our hope is to allow 2 way communications from our sip
> phones at headquarters (or within the dome) with our technicians using
> their walkie talkies as they are working throughout the dome. Not sure
> if this is possible but I would appreciate any suggestions.

It’s definitely possible. How practical it is, remains to be seen and
is probably a “how well do the details work out?” issue.

The simplest approach is probably this: you’ll need a walkie talkie
“base station” which will serve as the transmit/receive point for
the dome. The most straightforward would be to use one of the
actual walkie talkie radios, with a well-filtered DC power supply
in place of a battery pack.

You would need to hook up the W/T’s “speaker out” and “mic in”, and
perhaps the “push to talk” line, to suitable audio and control
I/Os on some sort of Asterisk end-point. The least expensive way
would probably be to use the Asterisk server itself (or some PC
running a soft-phone client), and use the PC/server’s sound card
jacks. You would need some sort of level-adjusting (padding)
system for the signal being fed into the walkie talkie’s “mic”
input (these generally require much less voltage than a sound
card’s line-level output), and it would probably be a good
idea to have an audio isolation transformer in each audio path
to prevent ground loops and hum and RF pickup.

You’d need some way of keying the radio’s push-to-talk when someone
on the phone starts to speak, and then release PTT when the
voice stops. Some walkie talkies have a VOX (voice-operated
switch) which will do the job. Others do not, and you would need
a separate VOX circuit (not difficult).

One possible hardware device which might save you trouble is
the Tigertronics SignaLink USB – it’s primarily designed for and
sold to amateur-radio operators but has multiple uses. It consists
of a USB “sound card”, isolation transformers, an adjustable
VOX/PTT circuit, and a very flexible radio-interconnect-cable system
which you could adapt to the speaker/mic needs of your walkie talkie.
You’d simply plug it into the server’s USB jack, and it would become
a secondary audio interface.

On the Asterisk side, you’d want to use the ALSA channel driver,
and create an inbound extension which would simply “dial” the
ALSA channel. Or, you might decide to use one of the various
Asterisk bridging/conference applications, and have the ALSA
walkie-talkie channel be perpetually signed into a conference
bridge which one or more other users could phone into.

You’ll need to select and implement a suitable security policy,
to control who can access your dome audio link, and decide
whether someone in the dome can use a walkie talkie with
DTMF capability to dial calls or otherwise control the Asterisk
system via radio.

Finally, you need to make sure that all of this is legal in your
particular jurisdiction. Here in the U.S., there are quite
a few personal and mobile-radio services for which it is *not*
legal to create a connection to the wired telephony system.
This is probably something you should determine first, rather
than at the end.

Installing asterisk on a server vs appliance(e.g digium mypbx)

Hi,

I am looking into advising a client on the pro’s and cons of using
Installing asterisk on a server vs appliance(e.g digium mypbx). the
appliance seems cheaper initially.

From experience, what would be pro and cons for either option?

Walkie talkie to sip phone interface

Hi All,

I’ve been trying to find a solution that would allow our sip phones to
communication with walkie talkies. Our setup is that we have sip phones
setup in 2 locations, headquarters and dome. We can communication from
headquarters and dome through sip phones, but within the dome we have
technicians that use walkie talkies to communicate as they go about
their work. Our hope is to allow 2 way communications from our sip
phones at headquarters (or within the dome) with our technicians using
their walkie talkies as they are working throughout the dome. Not sure
if this is possible but I would appreciate any suggestions.

Thanks,

Ferdinand

how to find out one way latency

Hi All,

How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough for a SIP Trunk.

Please help. We are trying to sign up with a new SIP Provider.

Thanks,
Najim

Sound files with MixMonitor not playable with Media Player

Hello,

the wav sound files that are created by using MixMonitor()-command are
not playable with Windows Media Player.

I can play them with vlc-player and on my Fedora with Totem.

This is one of the files :

/var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian)
data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

What would be missing on the system (Centos 5.7) that makes wav-files
difficult for Windows Media Player ?

Issue with Polycom SPIP650 and its sidecar

Hello,

On one location, I’ve got from time to time (let say one a week) the
following issue :
the phone SoundPoint 650 works ok (can call or answer, display and sound
are ok),
the sidecar looses its display : entries on sidecar’s LCD screen are not
displayed anymore, or names are truncated, or BLF are not shown or updated.

I only have one SPIP650 on this system so I can’t compare with others.

What could be the root cause of this ?

Regards