s/Trixbox/FreePBX/g
Please, Trixbox is a distro, the GUI is FreePBX.
Another option might be Destar. Google it up.
On 9/18/07, Matt Riddell <[EMAIL PROTECTED]> wrote:
>
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>
> Ken D'Ambrosio wrote:
> > Are there any Asterisk GUIs out there that actu
On Tuesday 18 September 2007 00:51:47 Kevin Kiely wrote:
> Per the earlier recommendation, I picked up one of the ATS X10001P to
> evaluate. I was able to configure the LAN for access, however, I don't see
> where to enter the sip credentials. I have accessed the web interface with
> root/test and
Per the earlier recommendation, I picked up one of the ATS X10001P to
evaluate. I was able to configure the LAN for access, however, I don't see
where to enter the sip credentials. I have accessed the web interface with
root/test and don't see any sip configuration information. I also accessed
v
There is need of Addons as you will need mysql module to connect to mysql and
asterisk. so you have to install addons.
Keshav.
Luís Palma <[EMAIL PROTECTED]> wrote: Hi,
Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database
backend for storing CDRs without having to re
On Monday 17 September 2007 17:19:52 Luís Palma wrote:
> Is there a way to enable the usage of UNIQUEID CDR field using a MySQL
> database backend for storing CDRs without having to recompile
> asterisk-addons as stated here
> http://www.voip-info.org/wiki-Asterisk+cdr+mysql ?
Please see configs/c
Naa Bilal, haven't got to investigate it thoroughly yet. Kinda been
occupied. Will let you know, if I do manage to do that.
bilal ghayyad wrote:
>Dear Benjamin;
>
>OK friend, things are clear. But now I came to the
>same original issue that you asked about it, which is
>the ability to stop the l
You The Man, Anselm. Thanks for the details.
Anselm Martin Hoffmeister wrote:
>Am Dienstag, den 11.09.2007, 17:11 +0530 schrieb Benjamin Jacob:
>
>
>>Thanks Anselm. This does clears a few things for me.
>>Tho, I couldnt find the patterns you mentioned in the docs(do point me
>>to the location i
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Ken D'Ambrosio wrote:
> Are there any Asterisk GUIs out there that actually parse the data files,
> themselves, instead of having some sort of metadata middle-man, which
> leads to said overwriting? I mean, I, personally, love the CLI -- always
> have
Hi Russell,
I, myself, have just gone through this same thing. Here is what you need to
do to correct this problem.
1.Download http://www.mezzo.net/asterisk/app_swift-2.0rc1.tgz, if you
haven't already.
2.Look in the source of app_swift.c for the line:
const int framesize =
Greetings,
I've recently upgraded from Asterisk 1.2 to 1.4. I've been searching for
a solution, but am also trying the easy way at the same time. I've now
got David of Cepstral now speaking using app_swift from
http://www.mezzo.net/asterisk/app_swift.html .
The problem is, he sounds way worse
Lacy's response in the thread 'Why does
everyone seem to dislike *now?', has a small
bit that caught my eye.
Chan_Skinny made a lot of progress between 1.2 and
1.4, and even more in the later 1.4.X releases.
I am curious as to which features/functions that
chan_skinny might be lacking compared t
On Mon, September 17, 2007 7:21 pm, Matt Riddell wrote:
> In the past, you could help someone sort a problem, only for the config
> files to be overwritten the next time the user did something in the GUI.
Are there any Asterisk GUIs out there that actually parse the data files,
themselves, instea
On 9/17/07, Jim Canfield <[EMAIL PROTECTED]> wrote:
>
> Greetings,
>
> Last week I began researching Asterisk for the first time. I did what most
> noobs would do; downloaded an image that seemed simple and straightforward
> and had some credibility (*now). I also downloaded the TFOT version 1 as
Jim Canfield wrote:
> SIP wrote:
>
>> Not at all relevant to your query, but I still use the mysql CLI for any
>> mysql task... and while most OSs have nice, functional tools to add
>> users (command-line tools), there are SOME (*cough* Irix *cough*) where
>> there are no CLI tools and VI is
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Jim Canfield wrote:
> console and the file editor...both are very nice IMHO. I see no reason
> to create a "second class" of community citizens.
I think it comes from the fear that without logging into an Asterisk
server and knowing what settings etc
Hi all,
does anybody have implemented asterisk on a Beowolf cluster (HPC - High
Performance Computing) using Message Passing Interface (MPI)?
Thanks,
Felipe Neuwald.
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On Mon, 2007-09-17 at 09:51 -0600, Stephen Bosch wrote:
> I posted it at least four times, and not one made it through. Perhaps
> it's a spam filter.
Yeah, I'm looking into the problem now. It seems to be related to the
spam filter sitting in front of the mailing list server. I'll post an
update
Joao Pereira wrote:
> But still, the user can choose not to answer the phone.
> I want to force the users to accept the calls.
>
You would also need a phone that did not have Do Not Disturb. I don't
think you would find that.
I think you could get a software company like Zoiper to remove the
Hi,
Is there a way to enable the usage of UNIQUEID CDR field using a MySQL
database backend for storing CDRs without having to recompile
asterisk-addons as stated here
http://www.voip-info.org/wiki-Asterisk+cdr+mysql ?
After version 1.4 it is said in release that it can be done (not sure if it
ap
SIP wrote:
> Not at all relevant to your query, but I still use the mysql CLI for any
> mysql task... and while most OSs have nice, functional tools to add
> users (command-line tools), there are SOME (*cough* Irix *cough*) where
> there are no CLI tools and VI is your only option (especially if
On Sep 17, 2007, at 11:11 AM, Joao Pereira wrote:
> But still, the user can choose not to answer the phone.
> I want to force the users to accept the calls.
Wouldn't that be the same as paging/intercom, then?
http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom
___
Stephen Bosch wrote:
> Matt Riddell wrote:
>>> Subject: Astricon 2007 -- does anybody need a ride?
>> Heh can't see any reason it would have been moderated!
>
> I posted it at least four times, and not one made it through. Perhaps
> it's a spam filter.
>
> -Stephen-
>
A ride from the airport to
Although I've never tested it with asterisk, we use Neos (
http://www.neosmt.com/ ) for in-house jabber communication on our
Windows boxes (you didn't specify required OS). It supports h.323,
webcams, and file transfer -- although I've never tested any of these
features, the rest of it seems t
What's the best way to debug what's going on within Asterisk?
I turned up the 'core debug', but that did not give me what I was
hoping to find. I'm hoping to see some kind of error that explains
why it will not pass through the g729 codec.
Thanks,
Scott
On 9/14/07, Scott Moseman <[EMAIL PROTEC
Hello,
I have a small LAN network where I am running a Jain-Sip softphone on two user
pc's. These softphones are connected through Asterisk(Trixbox). Although the
phones do work in providing an audio conversation, there is a long delay(about
20 seconds) in the initial RTP session setup. I ha
Not at all relevant to your query, but I still use the mysql CLI for any
mysql task... and while most OSs have nice, functional tools to add
users (command-line tools), there are SOME (*cough* Irix *cough*) where
there are no CLI tools and VI is your only option (especially if you're
remotely l
Am Montag, den 17.09.2007, 15:50 -0400 schrieb [EMAIL PROTECTED]:
> Hello,
>
> I have a small LAN network where I am running a Jain-Sip softphone on two
> user pc's.
> These softphones are connected through Asterisk(Trixbox). Although the
> phones do
> work in providing an audio conversation, t
Try adding canreinvite=no to the sip.conf entries for UA or UB.
Asterisk can't timeout the RTP if the RTP is not being handled by Asterisk.
Arun Kumar wrote:
> Hi All,
>
>
> UA <> Asterisk Server <-> UB
>
> if there is no rtp for a specified number of minutes / seconds then I w
Greetings,
Last week I began researching Asterisk for the first time. I did what most
noobs would do; downloaded an image that seemed simple and straightforward
and had some credibility (*now). I also downloaded the TFOT version 1 as
a guide.
As questions arose, I tossed a few
Hi,
so not too much replies yet, i would like to come and meet some asterisk-users.
what about werkzeug-h ? Nice place to drink some beer :)
cu
Chris
On 9/14/07, SIP <[EMAIL PROTECTED]> wrote:
> Curses! I just got BACK from Vienna yesterday. I should have stayed
> another week. :)
>
> N.
>
>
>
Kenneth T. Van Wie II wrote:
...
Did you mean to include an answer or a question of some type?
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
___
Sign up now for Astr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Monday, September 17, 2007 12:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Ans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris
Mason (Lists)
Sent: Monday, September 17, 2007 12:45 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Call Center SoftPhone with Auto Ans
Hi,
is there anybody using an analog TOPEX GSM gateway?
My asterisk + TDM400P does not receive the hangup signal from that gateway.
Is there anybody who can give me a hint?
Thank you!
Giorgio
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Matt Riddell wrote:
>> Subject: Astricon 2007 -- does anybody need a ride?
>
> Heh can't see any reason it would have been moderated!
I posted it at least four times, and not one made it through. Perhaps
it's a spam filter.
-Stephen-
___
Sign up now
Bryan M. Johns wrote:
> Stephen,
>
> Thanks for the heads-up on the cab ride from Phoenix to the event. I
> did not know it was that far. I will be coming in Wednesday morning and
> I may take the same route you are considering.
>
> Anybody coming in Wednesday morning that wants to split fare?
Hi Bruce,
It was not deleted, it was closed automatically when the commit to 1.4 to fix
it happened and then an additional note was added for the commit to trunk. If
you didn't get an email detailing this as you should have I will test and pass
it off to get fixed.
Joshua Colp
Software Develo
But still, the user can choose not to answer the phone.
I want to force the users to accept the calls.
Regards
Joao Pereira
Thiago Maluf wrote:
> Ola Joao,
> tem um modo do Asterisk fazer isso sim.
> Entre em contato no meu GTALK por esse e-mail e eu te dou mais informações.
> Abs!
>
> Hi Li
Hi All,
UA <> Asterisk Server <-> UB
if there is no rtp for a specified number of minutes / seconds then I want
to disconnect the call. I've tried using rtptimout and rtpholdtimeout but no
luck
pls guide.
thanks
arun
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There is for sure X-lite and other similar but you won't get file
sharing which is meaningless either way. If you need it then use msn :-) .
http://www.counterpath.com/xlitedownload.html
Anselm Martin Hoffmeister wrote:
Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel:
Dear al
On Mon, Sep 17, 2007 at 05:31:30AM -0400, Watkins, Bradley wrote:
>
> > On a side note, does anyone have the URL to the AEL example so I can
> > write out an extensions.conf version for the wiki?
>
> It's called queues-with-callback-members.txt in the /docs directory in
> the source tree.
Well,
But as mentioned in the same email, the feature set just isn't the same
and it's a /lot/ more difficult to implement.
I'll be extremely disappointed if it /does/ get removed from Asterisk
without a suitable /easy/ and /equivalent/ function being made readily
available.
Matt Riddell wrote:
Am Montag, den 17.09.2007, 05:09 -0700 schrieb satish patel:
> Dear all
>
> I have setup of asterisk 1.4.11 Now i want soft phone
> which one support file sharring + video + voice call with asterisk SIP
> is there any soft phone which support this all feature ??
Yes, there is such a
Hi Joshua,
My bad, I thought that when you monitor the bug id that you would get an
email when there were any additional notes added to the bug. I have been
receiving email messages from "[EMAIL PROTECTED]", but, there is
nothing in the message body, ie, an empty message.
If it has indeed been fi
Bruce McAlister wrote:
> Moises Silva wrote:
>> Open a bug in http://bugs.digium.com/ including all the information
>> you provided here.
>>
>
> OK, bug id 0010734 created:
>
> http://bugs.digium.com/view.php?id=10734
>
Interesting, this bug was deleted without any notification (that I'm
aware
Dear all
I have setup of asterisk 1.4.11 Now i want soft phone which one
support file sharring + video + voice call with asterisk SIP is there any soft
phone which support this all feature ?? with asterisk
Regards
Satish Patel
-
Moody fr
Stephen,
Thanks for the heads-up on the cab ride from Phoenix to the event. I did not
know it was that far. I will be coming in Wednesday morning and I may take the
same route you are considering.
Anybody coming in Wednesday morning that wants to split fare?
Bryan M. Johns
Partner
Shelton
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Hello,
I've been using for a long time asterisk-perl-0.08 for prepaid card
applications, and I've identified a problem with the last releases of
asterisk-1.2, installed with Trixbox.
The command get_variable() raises a signal SIGPIPE when it is calle
In article <[EMAIL PROTECTED]>,
fateme fatah <[EMAIL PROTECTED]> wrote:
> Can I set 1 extension(i.e.6000) in extensions.conf file for several room for
> conference call
> service ? Or for every room I should set 1 special extension.
It doesn't matter. It all depends on what logic you want to
Hi Tzafrir,
Tzafrir Cohen wrote:
> On Mon, Sep 17, 2007 at 10:23:53AM +0200, gincantalupo wrote:
>
>> Hi Tzafrir,
>> I'm currently using genzaptelconf for digium cards but I have Sangoma
>> cards too and I have to load digium drivers after Sangoma one.
>> That's why I need to understand why wc
> On a side note, does anyone have the URL to the AEL example so I can
> write out an extensions.conf version for the wiki?
>
> - --
> Kind Regards,
>
> Matt Riddell
> Director
It's called queues-with-callback-members.txt in the /docs directory in
the source tree.
- Brad
_
On Mon, Sep 17, 2007 at 10:23:53AM +0200, gincantalupo wrote:
> Hi Tzafrir,
> I'm currently using genzaptelconf for digium cards but I have Sangoma
> cards too and I have to load digium drivers after Sangoma one.
> That's why I need to understand why wctdm is automatically loaded.
> I want all the
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> Subject: Astricon 2007 -- does anybody need a ride?
Heh can't see any reason it would have been moderated!
:)
- --
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end
Hi Tzafrir,
I'm currently using genzaptelconf for digium cards but I have Sangoma
cards too and I have to load digium drivers after Sangoma one.
That's why I need to understand why wctdm is automatically loaded.
I want all the drivers disabled during boot so I can load them in the
order I want wi
On Mon, Sep 17, 2007 at 09:36:27AM +0200, gincantalupo wrote:
> Hi Tzafrir,
>
> Tzafrir Cohen wrote:
> > On Thu, Sep 13, 2007 at 05:32:25PM +0200, gincantalupo wrote:
> >
> >> Hi,
> >> I've installed Asterisk with a TDM400P on a Debian Etch distro.
> >> When I reboot the server I get zaptel and
On Sun, Sep 16, 2007 at 03:21:04PM -0700, bilal ghayyad wrote:
> Dear Benjamin;
>
> OK friend, things are clear. But now I came to the
> same original issue that you asked about it, which is
> the ability to stop the log/debug messages into
> /var/log/messages.
/var/log/messages comes from the sy
>I read your postings about faxdetection with CAPI. Interesting feature
>but I'm not sure how to implement it on incoming calls (for example fax
>dialed wrong number). Can you please show me a dialplan example how you
>use this feature (incoming fax calls into Asterisk server).
I did'nt test incom
Hi Tzafrir,
Tzafrir Cohen wrote:
> On Thu, Sep 13, 2007 at 05:32:25PM +0200, gincantalupo wrote:
>
>> Hi,
>> I've installed Asterisk with a TDM400P on a Debian Etch distro.
>> When I reboot the server I get zaptel and wctdm automatically loaded.
>> I'd like to avoid this behaviour.
>>
>
>
It is possible, kind of. The only way I was able to get this to work
for me (and only for two email addresses) was to specify one email
address as normal, and the second email address as the "pager" email
address. Possibly not what you're looking for, but the best solution
(short of email gro
Dear Benjamin;
OK friend, things are clear. But now I came to the
same original issue that you asked about it, which is
the ability to stop the log/debug messages into
/var/log/messages.
Same like your situation, the messages is comment (;)
and even the logges are written to the
/var/log/messages
On Mon, 2007-09-17 at 01:39 -0500, Tilghman Lesher wrote:
> On Monday 17 September 2007 00:48:56 Paul Hales wrote:
> > On Mon, 2007-09-17 at 07:25 +0200, Adam KOSA wrote:
> > > Paul Hales wrote:
> > > > Is there a way to specify multiple email addresses in voicemail.conf
> > > > for a specific user
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