Hi,
This is probably outside what Asterisk is intended for, but I'm hoping it can
help.
I need to make and receive calls through a Cisco Call Manager server that I
have no control over. I have to use a Cisco soft phone (Cisco IP
Communicator), which only runs on Windows. But I'm on Linux. CCM
On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote:
When I dial the extension of a meetme conference room, I get a message that
states is not a valid conference. The meetme app was working before.
I am getting this error on the CLI:
app_meetme.c:800 build_conf: Unable to open
On Fri, 10 Apr 2009, Shocky wrote:
Hi,
This is probably outside what Asterisk is intended for, but I'm hoping it can
help.
I need to make and receive calls through a Cisco Call Manager server that I
have no control over. I have to use a Cisco soft phone (Cisco IP
Communicator), which only
Hi Shocky. It is possible. You should use SIP trunk in CCM and
configure some prefix to point to the Asterisk Box.
On the Asterisk BOX use SIP peer configuration to make calls trough CCM.
You can use some prefix from the both sides and strip it when call
arrive at the each side.
If you have
Disucssions: Following the Michael Robertson session, this week a
little more about Google Voice and SIP.
Digium has announced support for the Open Source Asterisk versions.
More from them on this if John Todd is available.
Also finally, the Polycom giveaway: http://tr.im/iyEc
The info you need
Hello,
I opened bug #0014707 concerning audio missing between Asterisk and
Gtalk. I would like to know if someone uses successfully Gtalk with
Asterisk 1.4.24?
Regards
--
Daniel
___
-- Bandwidth and Colocation Provided by
Alright I know how to do basic IVR in *. But what I'm working on trying
to do now is a survey. I've found very little things out there on google
or the archives for how to do surveys with the * ivr.
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
On Friday 10 April 2009 03:33:36 Gordon Henderson wrote:
On Fri, 10 Apr 2009, Shocky wrote:
Hi,
I need to make and receive calls through a Cisco Call Manager server that
I have no control over. I have to use a Cisco soft phone (Cisco IP
Communicator), which only runs on Windows. But I'm
Shocky wrote:
This is probably outside what Asterisk is intended for, but I'm hoping it can
help.
I need to make and receive calls through a Cisco Call Manager server that I
have no control over. I have to use a Cisco soft phone (Cisco IP
Communicator), which only runs on Windows. But I'm
On Fri, 2009-04-10 at 11:04 -0500, James A. Shigley wrote:
But I’m clueless as to how to combined the recordings into one file. I
don’t want the questions in the recordings, Only the caller’s side of
the conversation without the dead space while they listen to the
Qs/Think on their response.
Well considering php is the only language I know and even that not much. I'm
not sure how I would accomplish the recordings and combination from within the
php agi script.
Better question might be, how would you do it (bear in mind if AGI it would
have to be php) and do you have a code
On Friday 10 April 2009 10:53:17 Dan Austin wrote:
Shocky wrote:
This is probably outside what Asterisk is intended for, but I'm hoping it
can help.
I need to make and receive calls through a Cisco Call Manager server that
I have no control over. I have to use a Cisco soft phone (Cisco
On Fri, Apr 10, 2009 at 01:02:10PM -0400, Jared Smith wrote:
On Fri, 2009-04-10 at 11:04 -0500, James A. Shigley wrote:
But I’m clueless as to how to combined the recordings into one file. I
don’t want the questions in the recordings, Only the caller’s side of
the conversation without the
ls /usr/include/dahdi returns:
r...@pbx:~# ls /usr/include/dahdi
fasthdlc.h kernel.h tonezone.h user.h wctdm_user.h
r...@pbx:~#
On Fri, Apr 10, 2009 at 3:07 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Apr 09, 2009 at 04:59:41PM -0400, John Rogers wrote:
When I dial the
To answer your other questions:
Which version of each, exactly?
Both DAHDI and Zaptel?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
asterisk-1.4.23.1
asterisk-addons-1.4.7
libpri-1.4.7
zaptel-1.4.11
I corrected by rebuilding zaptel and downgrading to asterisk-1.4.21.2.
I now have my conference
Hi James,
I have done a similar implimentation with Asterisk. And its quite easy
to accomplish. The voice files (questions) have an entry in a MySQL DB
and then I wrote an AGI in BASH which actually picks a number for the DB
and dials the extension. The asterisk plays the voice file using STREAM
On Fri, Apr 10, 2009 at 02:03:11PM -0400, John Rogers wrote:
To answer your other questions:
Which version of each, exactly?
Both DAHDI and Zaptel?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
asterisk-1.4.23.1
asterisk-addons-1.4.7
libpri-1.4.7
zaptel-1.4.11
I corrected by rebuilding
Hi James,
I guess in your case, you can use an AGI which mix the recorded voice
files alone using 'soxmix'. I am not sure, how u ll be able to
accomplish without the help of AGI.
Regards,
Kurian Thayil.
On Fri, 2009-04-10 at 12:33 -0500, James A. Shigley wrote:
Well considering php is the
Anyone want to talk briefly about one-button call parking/pickup on
Asterisk with Polycom phones? Does anyone use it or know to do it?
On many phone systems there are 2 or 3 park buttons, and you can park
a call onto an unlit park button, and then the light flashes. You can
go to any other
On Fri, 10 Apr 2009, James A. Shigley wrote:
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
2. Starts Survey
3. Ask Q1, Record the answer (voice responses) repeat this step for
each Question
4. Combined the recorded responses into one file.
5.
these packages are source packages downloaded direct from Digium. Maybe
that was the problem. I tried installing both (at one point I tried
upgrading the GUI and I remember seeing something mentioning that the new
GUI needed dahdi. I haven't yet upgraded the gui, but I guess it's
pointless now
See comments inline:
Steve Edwards wrote:
On Fri, 10 Apr 2009, James A. Shigley wrote:
Here is more or less what I'm trying to accomplish
1. Call comes in Plays Greeting
2. Starts Survey
3. Ask Q1, Record the answer (voice responses) repeat this step for
each Question
4.
From: John Rogers j...@wizworks.net
Subject: [asterisk-users] Looking for good IAX ATA
To: asterisk-users@lists.digium.com
Message-ID:
b621c35e0904091400w434f0614p716b82d26b94d...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
I've been told by someone at Digium that
On Fri, 10 Apr 2009, Julian Lyndon-Smith wrote:
See comments inline:
Steve Edwards wrote:
On Fri, 10 Apr 2009, James A. Shigley wrote:
But I'm clueless as to how to combined the recordings into one file. I
don't want the questions in the recordings, Only the caller's side of
the
I've got a challenge (or clarification request if I am mistaken) for the
group.
I have a non-profit customer on asterisk 1.4 that has multiple
volunteers that work from home. The volunteers are willing to take calls
to help out the organization.
So, a formal queue is out. They don't want
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rony Ron
Sent: April-09-09 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] New ViciDial Call Center Suite
Thank you for the links! Of course if anyone else knows of other IAX ATA
offerings, please *DO* share. Really looking for a good solution. I will
buy one of each of these offerings to test and I'll share my findings with
the group.
Thanks again!
On Fri, Apr 10, 2009 at 4:35 PM, Giuseppe
On 4/10/09, ContactTel Business li...@contacttel.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rony Ron
Sent: April-09-09 11:02 PM
To: Asterisk Users Mailing List - Non-Commercial
On Fri, Apr 10, 2009 at 7:46 PM, John Rogers j...@wizworks.net wrote:
Thank you for the links! Of course if anyone else knows of other IAX ATA
offerings, please *DO* share. Really looking for a good solution. I will
buy one of each of these offerings to test and I'll share my findings with
I *was* using the X100P FXS ATA but they discontinued it last year in late
October. Several inquiries into when they will re-release/replace it with
another IAX ATA have gone unanswered.
Atcom.cn is a MFG in china with no USA point of presence - I looked into
them, but they don't have any
hi
how i can give the control of a digium card to the virtual machine? i am
using XEN
do you recomendo other virtual machine? VMWare openVZ etc...?
Thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
Is there any documentation that explains res_config_curl?
Specifically, the format of realtime calls made to the web server and
what the return string for each call should look like?
--
Eric Chamberlain
___
-- Bandwidth and Colocation Provided
Hi Moises and Steve,
I tried with all protocol variants for Openr2 (AR, BR, CN, CZ, CO, EC, ITU,
MX, PH, VE) and setting mfcr2_skip_category=yes, but the problem persists.
I tried with Unicall and, in this way, I could make and receive calls
without problems, using protocol variant BR or CO (I
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