On Mon, 15 Jun 2015, lu...@sulweb.org wrote:
I'm new here and I'm interested in building a small PBX with asterisk at
home. I have one single PSTN line and ethernet cabling in place. I
already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM
and RAID 10 SATA disks). I make and
I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php
for $30, and it works great for a home install. Very low power draw as
well.
James Cass http://goog_987864563
jcas...@gmail.com
On Mon, Jun 15, 2015 at 10:50 AM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
On Mon, 15 Jun 2015, Steve Edwards wrote:
Although, if you lose power, you've probably lost your Internet
connection as well so you could only make calls between extensions.
And you would lose the Italian equivalent of 911. In the US, everybody
over the age of 6 has a cell phone stapled to
I don't know this 'translates' to Italy, but this is what I would advise
somebody in the US to consider, assuming you have a reliable Internet
connection.
0) I hope you mean you want to run Asterisk at home instead of 'Asterisk
at Home.' A@H was an ancient distribution from around
On Mon, Jun 15, 2015 at 12:43 AM, Nathan Anderson nath...@fsr.com wrote:
What you want is called SIP call forking, and unfortunately, last time I
checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP channel
driver does not support it, and I would be shocked if Asterisk 12+
hello,
is it possible to play queue periodic-announce without stopping agents
ringing? actual situation is sequential - ring agents, play announce
(for 15 sec), ring agents , ... (i need to connect agent with caller
asap when agent is free)
is it possible with ARI?
--
On 15-06-15 08:48 PM, Matt Darnell wrote:
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally
it is all TDM so there is absolutely zero latency.
We are able to use ATA's for faxes and analog phones but
Hi List,
I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I
have installed Asterisk 13.4 with srtp,pjproject. I have configured two
numbers for webRTC clients, when i try to call from a client (sipml5) to
another client (sipml5) it throws the following error:
Jon Pounder wrote:
snip
Fax is really the only need recently, and even that has alternatives like
emailing scans that most people prefer now.
The legal and medical communities still seem to prefer faxing, in the (
mistaken? ) belief that it is more secure. In fact the medical community
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally it
is all TDM so there is absolutely zero latency.
We are able to use ATA's for faxes and analog phones but devices that use
modems, they fail 99.99% of
The mediatrix 4102s line kicks ass.
On Jun 15, 2015 8:49 PM, Matt Darnell mattdarn...@gmail.com wrote:
In the past we have used Adtran Atlas 550's to break out FXS ports for
devices like modems. The great thing about the 550 is that internally it
is all TDM so there is absolutely zero
On Mon, Jun 15, 2015 at 9:22 AM, Marek Cervenka cerv...@fpf.slu.cz wrote:
hello,
is it possible to play queue periodic-announce without stopping agents
ringing? actual situation is sequential - ring agents, play announce (for 15
sec), ring agents , ... (i need to connect agent with caller
On Mon, 2015-06-15 at 11:03 -0500, Matthew Jordan wrote:
On Mon, Jun 15, 2015 at 12:43 AM, Nathan Anderson nath...@fsr.com wrote:
What you want is called SIP call forking, and unfortunately, last time I
checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP
channel driver
James Cass wrote:
I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for
$30, and it works great for a home install. Very low power draw as well.
James Cass http://goog_987864563
jcas...@gmail.com mailto:jcas...@gmail.com
The JS-200 runs a very old ( 1.4 ) version
If I call a number from the phone of my wife, I get this warning:
[Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for
gsmtolin
I think this is related to silence suppression. Either ignore it, or find the device that does
this and disable silent suppression.
jg
--
Hi list!
If I call a number from the phone of my wife, I get this warning:
[Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for
gsmtolin
(more time per seconds).
I didn't found any help in Google with this message...
Someone wrote about turning off silence suppression,
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.10.2
DAHDI-Tools-v2.10.2
dahdi-linux-complete-2.10.2+2.10.2
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
On Monday 15 Jun 2015, lu...@sulweb.org wrote:
Hello all,
I'm new here and I'm interested in building a small PBX with asterisk at
home. I have one single PSTN line and ethernet cabling in place. I
already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM
and RAID 10 SATA
Hello all,
I'm new here and I'm interested in building a small PBX with asterisk at
home. I have one single PSTN line and ethernet cabling in place. I
already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM
and RAID 10 SATA disks). I make and receive 10 calls a day on average. I
On Mon, Jun 15, 2015 at 04:56:31PM +1000, Tim Groeneveld wrote:
On Mon, 15 Jun 2015 16:46:13 +1000 [Lucio] wrote
Hello all,
I'm new here and I'm interested in building a small PBX with asterisk at
home. I have one single PSTN line and ethernet cabling in place. I
already
On Mon, 15 Jun 2015 17:26:40 +1000 Tzafrir Cohen wrote
The SPA3102 can be found cheap on Ebay, and will be easy to setup in
Asterisk.
http://www.infoworld.com/article/2633694/data-modeling/your-pstn-and-you--linksys-spa-3102-and-asterisk.html
Once the FXS is set up, it's
Hey Lucio,
You will need a FXS port. I would recommend setting up something like Cisco
SPA3102.
The SPA3102 can be found cheap on Ebay, and will be easy to setup in Asterisk.
http://www.infoworld.com/article/2633694/data-modeling/your-pstn-and-you--linksys-spa-3102-and-asterisk.html
Once the
On Monday 15 Jun 2015, Ivan Demkovitch wrote:
Hello group!
I’m new to Asterisk but got one running finally :)
Now I’m trying to solve following problem. I have company Automated
Attendant and each employee have SIP phone at home, SIP phone in office,
cell phone.
I want all those 3
On Monday, June 15, 2015 at 9:21 AM, Nilesh Govindrajan wrote:
How about ringall strategy with a queue?
Not sure how that would help. Every SIP phone in the queue would still have to
have a unique SIP identifier/username.
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
--
On Monday, June 15, 2015 at 9:03 AM, Matthew Jordan wrote:
This is true for chan_sip. It is not true for the PJSIP stack.
The PJSIP stack does allow for multiple devices to register contacts
to a single Address of Record (AoR). You can then dial contacts
individually, or dial all contacts
On 6/8/15 1:18 AM, Luca Bertoncello wrote:
Hi list!
Another day, another problem...
I'm checking with Nagios my Asterisk at home, and since yesterday I noticed
that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours,
so that I have a new IP every day), the peer of an
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