On 20 Aug 2012, at 01:37, Noam Birnbaum wrote:
A client wants to keep their old Inter-Tel KTS analog phones for budget
reasons. Two questions:
1. How could they use these with FreePBX?
You'd need an ATA or an analogue card. And how well they work, depends on if
they are 'true' analogue, or
u need an analog telephone adaptor to connect your analog phone with switch
hub. but the device is rather expensive.
Sent from Samsung Mobile
Noam Birnbaum n...@maccentricsolutions.com wrote:
Hi folks,
A client wants to keep their old Inter-Tel KTS analog phones for budget
reasons. Two
On Monday 20 August 2012, Noam Birnbaum wrote:
Hi folks,
A client wants to keep their old Inter-Tel KTS analog phones for budget
reasons. Two questions:
1. How could they use these with FreePBX?
Via so-called Analogue Terminal Adaptors (ATAs). These are basically a SIP
phone with an FXS
On Sun, Aug 19, 2012 at 8:37 PM, Noam Birnbaum
n...@maccentricsolutions.com wrote:
Hi folks,
A client wants to keep their old Inter-Tel KTS analog phones for budget
reasons. Two questions:
1. How could they use these with FreePBX?
2. Would they be losing any features that they currently
Its really weird working with OpenSuse. I am not sure how others are using
with OpenSuse. Through Yast also I tried to install Asterisk package, it
didn't find.
Now I am clueless to work with OpenSuse.
Regards.
On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com
From the forum I understand OpenSuse 12.2 is pre-relase and better to use
OpenSuse 12.1. Lets check with OpenSuse 12.1.
Regards.
On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Its really weird working with OpenSuse. I am not sure how others are using
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA
on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP
server at the same network all with g729 codecs and rfc2833 for the DTMF.
Making calls via
On 19-08-12 21:58, Alec Davis wrote:
So I'm just looking on how to make a BLF-button blink or turn
red, to show to my customer that there are still calls inside
the queue waiting.
Can I only apply on Asterisk 1.8.5 ? Or can I apply to my Asterisk
1.8.11 also ?
It's 4 lines, plus 2 debug
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
Hi
I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of
ATA on the network who autenticate the phones: Linksys PAP2,
Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the
VoIP server at the same network all with
Hi,
I have to connect 3 asterisk servers,each of them being TLS server for
his clients and connected in both way in TLS with both others asterisk,
each having hi own Common Name. Is this possible?
I set up 2 asterik's , one server and the other client, this is OK. But
I can't deal with
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other webserver listening on 80), was
that
- Original Message -
Joshua
Can you copy and past into a wiki page for everyone's benefit? Maybe
https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
like page would be good.
If this thread has taught me anything it's that there needs to be a complete
wiki page, just
- Original Message -
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br
wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other
- Original Message -
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br
wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
jcas...@instant.com.br wrote:
I still get unauthorized from sipml5 with these modifications. I
used port 80 instead of 8088 (no other
On Mon, Aug 20, 2012 at 10:52 AM, Joshua Colp jc...@digium.com wrote:
- Original Message -
Joshua
Can you copy and past into a wiki page for everyone's benefit? Maybe
https://wiki.asterisk.org/wiki/display/~jcolp/WebRTC_Demo_Setup or
like page would be good.
If this thread has
On 20/08/12 16:23, Administrator TOOTAI wrote:
Hi,
I have to connect 3 asterisk servers,each of them being TLS server for
his clients and connected in both way in TLS with both others asterisk,
each having hi own Common Name. Is this possible?
I set up 2 asterik's , one server and the
I have been looking for the specs (format, bit rate, ect) on custom ringtones
for digium phones. Using the DPMA how would I deliver the ringtone to a digium
phone?
--
_
-- Bandwidth and Colocation Provided by
On 8/20/2012 10:14 AM, Josh Hopkins wrote:
I have been looking for the specs (format, bit rate, ect) on custom
ringtones for digium phones. Using the DPMA how would I deliver the
ringtone to a digium phone?
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Configuration
All the answers
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Rusty Newton
Sent: Monday, August 20, 2012 9:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digium Phones
On Mon, Aug 20, 2012 at 11:53 AM, Joshua Colp jc...@digium.com wrote:
- Original Message -
On Fri, Aug 17, 2012 at 7:04 PM, Juan Castro jcas...@instant.com.br
wrote:
On Fri, Aug 17, 2012 at 5:45 PM, Juan Castro
jcas...@instant.com.br wrote:
I still get unauthorized from sipml5
On 8/20/2012 10:31 AM, Josh Hopkins wrote:
[Josh Hopkins]
Thanks for the reply. I do see it now. Not sure how I missed all that before.
One thing I don't think I did see was where I should place my ringtone file?
I am guessing that if I put it in the same location as our custom imagefile
- Original Message -
The complete URL to use is http://asterisk IP address or
host:8088/ws
Note the /ws at the end. WebSocket support is only available there.
Doing otherwise would have required core HTTP server changes,
which I wanted to avoid. Depending on what you are
From: Noam Birnbaum n...@maccentricsolutions.com
Sent: Sunday, August 19, 2012 8:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] using analog phones
Hi folks,
A client wants to keep their old Inter-Tel KTS analog phones for budget
Thanks for your answer.
The logs where posted at pastebin, here the links:
- Working Phone: http://pastebin.com/q3pHcwna
- Not working phone: http://pastebin.com/iiCHPMmn
2012/8/20 Rusty Newton rnew...@digium.com
On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote:
Hi
I've got a little issue
Thanks to all who applied. I found out that their Inter-Tel phones are
connected to an Inter-Tel Axxess box-thingamadoodle (that's a technical term).
Inter-Tel was purchased by Mitel in 2009 so there's no support left. I think
I'm gonna recommend they use IP phones or a different VoIP
I currently run an Asterisk 10 system with hotdesking functionality set up.
Several of the users have worked with a system in the past that supported BLF
on their IP phones, and would like their current phones to behave in a similar
fashion. Right now I have a really kludgy system that mostly
Hoo-hah. It registers. Progress!
Now... media. Or not.
On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote:
- Original Message -
The complete URL to use is http://asterisk IP address or
host:8088/ws
Note the /ws at the end. WebSocket support is only available
On Mon, Aug 20, 2012 at 2:58 PM, Juan Castro jcas...@instant.com.br wrote:
Hoo-hah. It registers. Progress!
Now... media. Or not.
On Mon, Aug 20, 2012 at 12:51 PM, Joshua Colp jc...@digium.com wrote:
- Original Message -
The complete URL to use is http://asterisk IP address or
This is all nice and good but the documentation all assumes that you are on a
Debian box and use MYSQL. What about us SUSE/Postgresql folks?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Pocock
In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF
updates to SIP peers that have subscribed to a hint looking at a Custom device
if that Custom device state is RINGING or RING_INUSE. All other states seem to
be working correctly.
The hint section of the dialplan
I currently run an Asterisk 10 system with hotdesking functionality
set up. Several of the users have worked with a system in the past
that supported BLF on their IP phones, and would like their current
phones to behave in a similar fashion. Right now I have a really
kludgy system that mostly
Put my sipml5 changes there. By the way, this is what happens when I
try to call a X-Lite extension from a sipml5 extension:
jcvmasterisk1*CLI
== Using SIP RTP CoS mark 5
[Aug 20 17:24:02] ERROR[22737][C-0009]: chan_sip.c:32140
setup_srtp: No SRTP module loaded, can't setup SRTP session.
Hi,
How to define a extension mask as global variable in Ast 1.8? For example:
[globals]
MYVARIABLE = _15[7-9]X
I tried this way but it did not work.
Thanks
Att,
Rafael Saraiva
--
_
-- Bandwidth and Colocation Provided by
I don’t fool with 1.8 but I don’t think this is supposed to work. The
extension mask and variable processing are two different routines. I assume
you are trying to do “real-time” filtering where you would set a variable to
control a group of extensions?
From:
On 20/08/12 21:11, Danny Nicholas wrote:
This is all nice and good but the documentation all assumes that you are on
a Debian box and use MYSQL. What about us SUSE/Postgresql folks?
They are both good questions, and there are partial answers:
SUSE:
reSIProcate can be built from source on a
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Pocock
Sent: Monday, August 20, 2012 3:49 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk as TLS server as well as TLS client
On 20/08/12 22:53, Danny Nicholas wrote:
I'm fond of the tar-config-make method that Asterisk uses. Is this possible
for reSIPprocate? If so can you provide a link?
http://www.resiprocate.org/ReSIProcate_1.8_Release
You can access the download directory (use the 1.8.5 tarball) or SVN
On 08/20/2012 03:20 PM, Noah Engelberth wrote:
And after 303 tries to call 302 while 301 302 are still on a call
(301 302 on a call, plus 303 calling 302):
-= Registered Asterisk Dial Plan Hints =-
_3XX@hints : Custom:${EXTEN} State:Idle
Watchers 0
Hi,
you need to build Asterisk with SRTP support...
*wget http://sourceforge.net/projects/srtp/files/latest/download -O
srtp-latest.tgz
tar -zxvf srtp-latest.tgz
./configure --prefix=/libsrtp
make make install*
*And for Asterisk...*
*./configure --with-srtp=/libsrtp*
*
*
*this should work...*
*
So I can just add these 4 lines to app_queue.c and this will give me
the ability to use : exten = 566,hint,Queue:voipq1 ??
Yes, then I assume you know that you need to compile etc.
./configure
make menuselect
make
make install
Alec
--
2012/8/19 Alec Davis siva...@paradise.net.nz
Do you also know why it hasn't been accepted ? Seems like this
functionality is asked for on different forums. Wanting
to watch a
queue for calls is not that strange.
Not sure why?
Maybe I didn't promote it enough.
Ok Thanks Bryant, let me try with OpenSuse 12.1.
Regards.
On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote:
I have the current version of 8.x and 10.x on systems. I am using OpenSuse
12.1, We are working on getting a 12.2 boxs up just running out of time.
Asterisk on
42 matches
Mail list logo
