On Wed, 23 Jul 2008, Mike Diehl wrote:
What you sent me is almost exactly what I had, which indicated that that
part of my code was correct. So, I moved that block of code to the top
of my program and it worked. Eventually, I found a debug print()
statement that I had forgotten to take
On Wed, Jul 23, 2008 at 11:37:01PM -0300, Felipe Trevisan wrote:
I´m installing zaptel and asterisk on the CEntos 3.9.
I´ve installed the kernel-devel which on the kernel 2.4.21 is called
kernel-source, but when I run the pre requisites test, the zaptel won´t
recognize it.
Can I rename the
On Thu, Jul 24, 2008 at 10:53:59AM +0300, Tzafrir Cohen wrote:
On Wed, Jul 23, 2008 at 11:37:01PM -0300, Felipe Trevisan wrote:
I´m installing zaptel and asterisk on the CEntos 3.9.
I´ve installed the kernel-devel which on the kernel 2.4.21 is called
kernel-source, but when I run the pre
Greetings list,
We have a client with an analogue door intercom/opening unit which we're
attempting to replace with an IP variant. The existing unit has the following
functionality:
1) Intercom - visitor hits call, talks to operator
2) Door opening - operator can open the door by dialling a
I'm quite sure there's a bug somewhere in SIP + realtime + MySQL.
To update, since last post we've integrated with our existing users
database using MySQL views. Our legacy database uses md5
password hashes, and does not store plaintext.
During testing this morning I could swear it was all
On Thu, 24 Jul 2008, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door intercom/opening unit which we're
attempting to replace with an IP variant. The existing unit has the
following functionality:
1) Intercom - visitor hits call, talks to operator
2) Door
On Thu, Jul 24, 2008 at 10:25:34AM +0100, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door intercom/opening unit which
we're attempting to replace with an IP variant. The existing unit
has the following functionality:
1) Intercom - visitor hits call, talks to
Hi all,
maybe there is no opener device at all.
Anyway take a look here :
http://www.barix.com/
On Thu, Jul 24, 2008 at 12:11 PM, Gordon Henderson
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Thu, 24 Jul 2008, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door
Siemens HC 450 Dect intercom does exactly what you want
it doesn't come cheap, but works like a dream..
Gordon Henderson schreef:
On Thu, 24 Jul 2008, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door intercom/opening unit which we're
attempting to
On Thu, Jul 24, 2008 at 11:04 AM, Walter Stanish
[EMAIL PROTECTED] wrote:
If someone could sort out this bug (or let me know if I'm missing
something 'obvious' - a hard call with realtime documentation this
sparse...) I'd be most grateful, since we require md5secret support
to integrate with
On Thu, 24 Jul 2008, Fons van der Beek wrote:
Siemens HC 450 Dect intercom does exactly what you want
it doesn't come cheap, but works like a dream..
Not avalable in the UK, and there's an intersting comment about being able
to trivially take the unit apart with a screwdriver and
I'm trying to get a MP-114 FXS/FXO gateway working with Asterisk. It
registers fine and I can call between the MP-114 and other extensions,
but I'm not having much luck with the FXO ports. syslog shows the
problem to be in the MP-114 configuration.
Can anyone help?
On Thu, 24 Jul 2008, Tzafrir Cohen wrote:
On Thu, Jul 24, 2008 at 10:25:34AM +0100, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door intercom/opening unit which
we're attempting to replace with an IP variant. The existing unit
has the following functionality:
1)
Hi,
I want to make an insertion in a communication; A et B are in
communication, an other C wants talk to A, how can i set B on
hold state and make a call to A?.
Thanks.
Rachid
___
-- Bandwidth and Colocation Provided by
We have a post this morning on VoIPInsider covering Audiocodes gateway
configuration with Asterisk and FreeSwitch, you can find it here
http://blog.voipsupply.com/technical-advice/setting-up-an-audiocodes-mp-
114118-fxo-with-asterisk-and-freeswitch
Cory J Andrews
Director, New Market
It's not super cheap, but Cyberdata makes a SIP enabled intercom that is
vandal proof and has a dry contact relay built in to actuate a door
strike.
http://www.cyberdata.net/products/voip/voip-intercom.html
Cory J Andrews
Director, New Market Initiatives
VoIP Supply, LLC
454 Sonwil Drive
When dialing using a T1/PRI with a outgoing call files
Like Channel: Zap/1/95551212
is there ever a need to delay or pause in there?
I have gotten feedback from a customer that instead of dialing the 95551212
it seems to have dialed 55512 which just happened to be an internal
extension.
So it
Hi,
I'm looking to write a dialplan for Automatic Redialing feature,How to
ask asterisk to make a automatic re-dial if a channel is busy??
A simple example will be very useful for me.
Thanks.
Rachid
___
-- Bandwidth and Colocation Provided by
On Thu, Jul 24, 2008 at 09:23:44AM -0400, Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I assume I'd have to make a separate group for
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten = _71NXXNXX,1,Read(ZAPLINE|conf-getchannel)
exten =
leave the existing keypad there. as for integrating it with asterisk.
use an ata with 2 FXS ports. one FXS port connect to a viking door box
http://www.vikingelectronics.com/ and set the ATA to do hotline on it.
that door box is a regular analog phone in the shape of a door box
that when call is
Your using a Linksys right? you can use the fxo port and send DTMF.
Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door intercom/opening unit which we're
attempting to replace with an IP variant. The existing unit has the following
functionality:
1) Intercom -
On Thu, Jul 24, 2008 at 12:23 AM, John Faubion [EMAIL PROTECTED] wrote:
Well, I am not sure what is needed to interface between the
two. I hoped there was something you could use and from the
Joseph,
Now I'm pretty sure we are not talking about the same things. Let me see if
I have the
Could you explain further?
On Thu, Jul 24, 2008 at 4:13 AM, Gregory Malsack [EMAIL PROTECTED] wrote:
I resolved this problem. The key was to get the right combination of
self/callee and peer/caller. Read the instructions regarding the application
map very closely. My problem was that I
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I use:
exten =
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls out over each channel for testing?
I assume I'd have to make a separate group for each channel in the
/etc/asterisk/zapata.conf? Or could I just
On Wed, Jul 23, 2008 at 06:19:58PM +0200, Philipp Kempgen wrote:
While it may sound rude that's absolutely correct. As a software
developer in many cases you are more or less sure that an issue
has already been solved so you expect the user to upgrade to the
latest version or at least to the
On Thursday 24 July 2008 10:30:26 Jay R. Ashworth wrote:
On Thu, Jul 24, 2008 at 09:39:42AM -0400, Doug Lytle wrote:
Jay R. Ashworth wrote:
So I have these 4 new PRIs turning up tomorrow. Anyone have any
suggestions on some dialplan that I could use to allow me to manually
dial calls
Hello,
Asterisk 1.4.21.1
Well it seems like my month for questions. I have a situation where the
CallerID num shows as [EMAIL PROTECTED](the ip of the asterisk
box) on calls to any of the internal phones. This prevents the
ability to dial out from the missed call list. I have not been able to
If I understand you, then yes you can. I do this now. All our telco
lines come through our analog NEC phone switch and then through FXO/
FXS ports to my Asterisk. Asterisk handles voicemail and the menu
system so when somebody dials 6 to get my extension the asterisk
does the following:
The migration does not have to happen all at once, you can take it
slow, make it invisible to the end user, start using VoIP trunks and
all that Asterisk has to offer, and have a super flexible migration
path.
Steve,
Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1 card
This is a recovery from last week's fiasco. Tech issues prevented the
conference from having our full complement of voices.
If you are in an Asterisk Users Group, you'll want John Todd to know
about it. If nothing else, he may get you a Digium beachball or my
personal favorite, the Digium
There's not any direct way of which I am aware in a single command, but from
the shell you could do the following (and yes, this is a bit of a hack):
for i in `rasterisk -x queue show |grep wait |awk -F '{print $2}'`; do
rasterisk -x core show channel $i | grep Caller ID;done
That will return
If someone could sort out this bug (or let me know if I'm missing
something 'obvious' - a hard call with realtime documentation this
sparse...) I'd be most grateful, since we require md5secret support
to integrate with our existing users database.
Welcome to Asterisk!
It's highly unlikely
I need to replace a cisco call manager with an asterisk box. Phones
are cisco 7940 and 7910. I know the 40's can use SIP but the 7910's
have to use the skinny/sccp driver. Its been quite awhile since I did
anything with asterisk, so I am looking for some assistance with the
configuration and am
Hi,
On Thu, Jul 24, 2008 at 10:53:59AM +0300, Tzafrir Cohen wrote:
Specifically one of the many RPM packages Axel Thimm maintains is
Zaptel, and is also vs. RHEL3: http://atrpms.net/dist/el3/zaptel/ .
He reported several breakages in the past (which were fixed). I see that the
latest version
my best offer to you is to read more about the dial plan to understand what
happens.. or try to understand what does freepbx do and what does it write and
understand the applications..
Date: Wed, 23 Jul 2008 20:53:45 -0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and
Telepresence systems I have two IP patents for the VoiP Lite protocols
and have been designing and building OSS IPBXs for companies including
Google going back to 2001. I'm not mentioning any of that to be jerk I
mentioned it to
My son owns compoanyn here in San Jose and when a customers says they
want Cisco be provides Cisco phones with OSS PBX, it seems to work the
lower cost and Cisco phone on the desktop.
- Original Message -
From: Steve Totaro
To: Asterisk Users Mailing List - Non-Commercial Discussion
T G wrote:
I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and
Telepresence systems I have two IP patents for the VoiP Lite protocols
and have been designing and building OSS IPBXs for companies including
Google going back to 2001.
I'm not mentioning any of that to be
Steve,
Lots of good info! So if I put a T1 card in an Asterisk Server, and a T1
card in the Norstar
How does a user on the Norstar dial 221 and reach a voip only user
connected to asterisk via
ip only? That assumes as you mentioned new users are added as voip users in
the future?
Have the
See ITS at www.its-tel.com The Pantel and Pancode IP are what you are
looking for.
Rupert Utteridge
Director - Sales Marketing
Digital Techniques (Australia) Pty Ltd
4 The Lee
Middle Cove, NSW, 2068
Australia
Tel: +61 2 9037 4191
Mobile: +61 424 373 516
Web: www.dtasia.com.au
T G wrote:
I'm a CCIE and CCVP. I have worked in the Cisco TSBU on both CCM and
Telepresence systems I have two IP patents for the VoiP Lite protocols
and have been designing and building OSS IPBXs for companies including
Google going back to 2001.
I'm not mentioning any of that to be
On July 24, 2008 04:42:42 pm David Cook wrote:
Have the Norstar programmer send all 3 digit, unused extensions to the PRI.
Then Asterisk will see 221, etc. and can handle at your dialplan sees fit.
Yes, this works, but you won't be able to treat those as regular extensions;
the Nortel will
On Thursday 24 July 2008 12:58:29 am Steve Edwards wrote:
The agi debug command (1.2) would have shown you where you violated the
protocol.
Nice to know...
--
Mike Diehl
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
I have a question about click to dial. Each of my users is going to
have a VOIP phone with an assigned extension. Is there a simple way to
build a web-based speed-dial list that will allow them to put in their
extension, click on the number they want to dial, and have asterisk ring
their
So you basically want a call-interrupt feature that puts the interrupted
party on hold?
rachid wrote:
Hi,
I want to make an insertion in a communication; A et B are in
communication, an other C wants talk to A, how can i set B on
hold state and make a call to A?.
Thanks.
Rachid
Hello,
On Jul/24/2008, Brent Davidson wrote:
I have a question about click to dial. Each of my users is going to
have a VOIP phone with an assigned extension. Is there a simple way to
build a web-based speed-dial list that will allow them to put in their
extension, click on the number
Yep you just build a 'log in' query (eg to identify which extension to
send Leg A to or you can just build it into the url with a unique
extension id)
and then list all the extensions you want (obviously if it's company
wide then it will be the same for all - only the Leg A will be
different).
- Fidel Garcia [EMAIL PROTECTED] wrote:
Where exactly do I have to change it?
The GUI on the AA50 generates users via users.conf. These users are added into
the dialplan automatically and are placed into the default context. Calls to
the users are made via the stdexten macro. In that
this simple php script do whatr you need
should be called by your user pc with url ike
http://ip_your_Asterisk_host/chiama_ora.php?INT=xxNOME=yyNUMERO=xxxCONTESTO=xxx
if you are interested I have developped a complete phonebook
integrated wwith asterisk
the main functions are
- multiple
for a similar project I used astandard FXS interface
connected to one extension connector (RJ) of the legacy pbx and
I could end some commands to he legacy pbx by flashing the line and then send
appropriate dtmf
bye
2008/7/23 Joseph L. Casale [EMAIL PROTECTED]:
We have an older Meridian
Hi all,
How do I set different rx and tx gains for each channel?
in my zapata.conf file I have a heading [trunkgroups] and then
[channels] under this I have information such as language context
signalling etc and also rxgain and txgain.
My assumption is that these settings are used for all
I noticed that i' m not getting any manager event for hold and unhold of a
channel.
is this normal?
Also is there any easy way through either CLI or manager to find out which
one of the channels are on hold?
I checked show channels that did not show a channel being on hold or not,
also sip show
You are mentionning very particular case here, a company with a very strict
hierarchy, where a new ideas and solutions are not advised, i think that in
the past they used cisco who has some issues from time to time, and they are
prepared for that, but new name scares them, and sometimes people use
Search someone in local area, remote configuration of server is possible but
configuring the phones is more difficult, you need someone to load
firmwares, ect
2008/7/24 Chad Whitten [EMAIL PROTECTED]:
I need to replace a cisco call manager with an asterisk box. Phones
are cisco 7940 and 7910.
I agree, No manager gets fired even if a Cisco Call Manager goes south.
that's not the case with Asterisk.
With limited experience that i have with both, i hit more bugs using
Asterisk than a CCM, but this is not relevant to your final answer.
If you can afford CCM, and you can live with less
Hello,
I want to use an arabic TTS on asterisk. Do you know any arabic TTS Open Source
supported by an amd 64? Because I found Mbrolla a free TTS that include arabic
and that you can combine it with Festival but it doesn't support and amd64
Thanks'
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