On 8/8/14, 14:05, Gergo Csibra wrote:
Hi,
back in the old analog telephony days there was "digital" PBX-es and
digital "system" phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blink
Hello
Thank you for your response. I thought it could be easier moving the old card
to the new machine and using the DAHDI driver. Unfortunately my first attempt
for this failed. The card shows up in the original machine as:
>dahdi_hardware -v
pci::00:00.0 wcb4xxp+ 1397:08b4 Jung
On 7 August 2014 21:06, Kevin Larsen
wrote:
> > back in the old analog telephony days there was "digital" PBX-es and
> > digital "system" phonesets. This phonesets have had many individual
> > illuminatable buttons connected with extensions. The PBX can show on
> > the buttons if some extension i
On 8 Aug 2014, at 06:05, Gergo Csibra wrote:
> back in the old analog telephony days there was "digital" PBX-es and
> digital "system" phonesets. This phonesets have had many individual
> illuminatable buttons connected with extensions. The PBX can show on
> the buttons if some extension is ringin
On Friday 08 Aug 2014, Gergo Csibra wrote:
> Hi,
>
> back in the old analog telephony days there was "digital" PBX-es and
> digital "system" phonesets. This phonesets have had many individual
> illuminatable buttons connected with extensions. The PBX can show on
> the buttons if some extension is
Almost all of the phones has this feature in build (Polycom,CISCO
SPA,Digium etc..) Try going through this link
https://wiki.asterisk.org/wiki/display/AST/Presence+State and setting up the
right subscribe settings on the phone buttons;
-Original Message-
From: asterisk-users-boun...
On 8 August 2014 04:50, D'Arcy J.M. Cain wrote:
> Shot in the dark here but does "core show channels" show an inordinate
> number of channels, especially channels that you know should be closed?
>
>
I get that problem sometimes, do you have a solution for it/know the cause
for it?
--
__
On Fri, 8 Aug 2014 11:01:50 +0200
Mikael Fredin wrote:
> On 8 August 2014 04:50, D'Arcy J.M. Cain wrote:
>
> > Shot in the dark here but does "core show channels" show an
> > inordinate number of channels, especially channels that you know
> > should be closed?
>
> I get that problem sometimes,
On Thu, Aug 7, 2014 at 10:50 PM, D'Arcy J.M. Cain wrote:
> On Thu, 7 Aug 2014 22:00:47 -0400
> Jerry Geis wrote:
> > :[Aug 7 21:35:24] ERROR[19582] acl.c: Cannot create socket
> > [Aug 7 21:35:24] WARNING[19582][C-0283] res_rtp_asterisk.c:
> > Unable to allocate RTP socket: Too many open f
Hi
The only way to have CD service is using:
DAHDISendCallreroutingFacility(, ,
cfu|cfb|cfnr|unknown)
or it is possible also with Dial() command before answering the call also ?
Regards
Babak--
_
-- Bandwidth and Colocation Pro
On 08-08-14 10:09, Toney Mareo wrote:
Hello
Thank you for your response. I thought it could be easier moving the old card
to the new machine and using the DAHDI driver. Unfortunately my first attempt
for this failed. The card shows up in the original machine as:
dahdi_hardware -v
pci::0
On Fri, 8 Aug 2014 06:26:52 -0400
Jerry Geis wrote:
> next to nothing...
> This is normal as I only have a partial PRI like 5 lines.
> Very low volume
Me too but I was getting every call staying in the active channels
list. I thought that you might be having the same issue. See the
thread "Cal
Using the BLFs on Digium phones does not require the use of the Digium
Phone Module for Asterisk, or DPMA. SchmoozeCom (the FreePBX guys) use the
BLFs on Digium phones independently of the DPMA.
I am not sure why a previous response refers to this module as 'toxic'. It
is a free to use module whic
> I am not sure why a previous response refers to this module as
> 'toxic'. It is a free to use module which allows a host of Digium
> phone features to be quickly implemented with Asterisk, like
> security-enhanced auto provisioning.
Without creating a large off-topic response, there is a segm
If you think it is bad then
do not use it;
else
use it;
There is no natural law that requires to publish the sources, even if the software is otherwise
free. You can always write your own modules and publish the sources. I have difficulties seeing
your point.
Without creating a large
All;
I have a customer who does some small, limited fax broadcasting. What he
wants to do is to be able to tell when a phone number is actually a human
rather than a fax machine so he can delete the number from his customer
list. Determining whether a call is a fax or not on the incoming is ea
You might be able to do it by asterisk AMD, but personally I haven't used it on
fax detection..
http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tech Support
Sent: Friday, August 08, 2
On Thu, Aug 7, 2014 at 2:53 PM, Jerry Geis wrote:
> I am using a cyberdata "sip paging adapter" and with the
> Dial(MulticastRTP/basic/IP:port) and with
> tshark I see the RTP data, my device looks like its accepting the data
> and I hear a click for my relay on my device so it would seem its
> a
Hi Shishir,
thanks your your response, would you help me about how to set up in the sip
proxy server? actually, i'm beginner on asterisk.
thank you.
Hi Jaya,
it would be nice for me if i can assist you, but i don't know to much about
asterisk. i'm sorry
On Fri, Aug 8, 2014 at 3:05 AM, Shish
Follow these tutorials for setting up the sip proxy server
http://www.opensips.org/Documentation/Install
http://www.opensips.org/Documentation/Tutorials
They have everything you need.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rafa
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