On Tue, Jan 27, 2009 at 01:17:59PM -0800, Michael Higgins wrote:
> On Tue, 27 Jan 2009 20:43:30 +0200
> Tzafrir Cohen wrote:
>
> > To the best of my understanding, latest Asterisk should support it
> > through chan_dahdi .
>
> Cool. Got any links to any related informations, so I can go by more
On Tue, Jan 27, 2009 at 08:35:56PM -0500, Steve Totaro wrote:
> Who said he wants HIS software to be Open Source? It is his and if he
> want to, God forbid, make a profit from his work, then who the hell
> are you to say it is not right?
His software seems to be patches to GPLed software. That m
On Wed, Jan 28, 2009 at 7:56 AM, D Tucny wrote:
> There have been a number of comments recently about a shortage of
> documentation on Asterisk, so I wanted to cover briefly the documentation
We are drowning in it compared tot he situation four years ago when
you either read the source or suffere
On Wed, Jan 28, 2009 at 12:46 AM, Karl Fife wrote:
> Is there a digium twitter 'user' to follow that only tweets important
> announcements and release information?
> If there is not, I think there should be.
None of the Digium Twitter accounts are very noisy. The 'digium'
account is kind of a bus
> www.voip-info.org
[...]
> So, the easiest way that people could contribute to improving Asterisk
> documentation right now would appear to be by improving articles on
> www.voip-info.org...
Absolutely.
What I tend to do is the make contributions to a particular page
whenever I encountered a
Hi All,
There have been a number of comments recently about a shortage of
documentation on Asterisk, so I wanted to cover briefly the documentation
options available and suggest what they are useful for and how they can be
improved...
Documentation sources:
http://www.asteriskdocs.org/
- Not much
I got notified on 22 Jan that I was about to be unsubscribed due
to "excessive bounces". I've checked my mail logs, and saw the
following bounces (that I had generated):
Jan 12 04:50:44 "437 Bad Message-ID"
Jan 16 19:56:19 "437 Bad Message-ID"
Jan 19 20:36:20 "437 Bad Message-ID"
Jan 22 18:11:35
>4. Someone said "Maybe there needs to be a beginner list...". I agree
>almost 100% with your oppinion but, we all know that the problem is not
>(at least for now, when there are lots of documentation topics to
>write) newbies questions, but bad formulated questions and people not
>wanting to
Hi Steve,
Thanks for the tip. But unfortunately it doesn't help. The Avaya is
passing on the MFC codes to the SIP phone when it answers the call. I
think the solution might be in the Avaya configuration to properly
convert the signaling.
Regards,
Steve
Steve Totaro wrote:
> Answer() is the cu
Maybe at some place of the thread, we missed the focus of it.
1. About the Wiki, totally agreed that its sometimes outdated and you
could get lost. But as we all know, if you want something to be
correctly done, you have to do it yourself; (that takes me to point 2. ,
below)
2. The feeling tha
On Tue, Jan 27, 2009 at 6:25 PM, Michael Higgins wrote:
> On Tue, 27 Jan 2009 16:55:02 -0500
> Steve Totaro wrote:
>
>> On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen
>> wrote:
>> > On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote:
>> >
>> >> Get a hold of Marcin Pyco (Former Digium
Philipp Kempgen schrieb:
> Pascal Bruno schrieb:
>> Actually I installed them after, so do you recommend I recompile asterisk?
>
> Yes. The unixodbc header files (package unixodbc-dev on Debian,
> might be called -headers/-source/-devel on other distros) have to
> be present before compiling Aster
Pascal Bruno schrieb:
> Actually I installed them after, so do you recommend I recompile asterisk?
Yes. The unixodbc header files (package unixodbc-dev on Debian,
might be called -headers/-source/-devel on other distros) have to
be present before compiling Asterisk.
cd /usr/src/asterisk-X.X.X/
ma
in my modest experience using mail list
i learned to answer a question if i know the answer and if i don't like how
the question was made i forget it.
don't start a thread about the question.
if you know the answer (or think you know and you hope you can help someone)
and you like to respond give a
Actually I installed them after, so do you recommend I recompile asterisk?If
I do so, I wont loose my current configuration files right?
On Tue, Jan 27, 2009 at 6:27 PM, Philipp Kempgen
wrote:
> Pascal Bruno schrieb:
> > I have remove the comment defor res_odbc.so and res_config_odbc.so in my
> >
Is there a digium twitter 'user' to follow that only tweets important
announcements and release information?
If there is not, I think there should be.
It would be highly utilitarian to get an SMS when there is an update to
Asterisk, Dahdi, ADA etc, but I don't want to be bothered real-time wit
It seems to me that there a lot of "it ought to work or could be made
to to work" associated with implementing US BRI into Asterisk
I agree. I have a BRI in service and have offered a couple of times to do
tests. I have an HFC card here that I intend to use with it myself, and plan
to test tha
Pascal Bruno schrieb:
> I have remove the comment defor res_odbc.so and res_config_odbc.so in my
> modules.conf, but the module is still not loading
>
> when I do:
>
> module show like odbc
> I have o module returned
>
> anybody knows why?
Did you install unixodbc and unixodbc-dev before compil
On Tue, 27 Jan 2009 16:55:02 -0500
Steve Totaro wrote:
> On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen
> wrote:
> > On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote:
> >
> >> Get a hold of Marcin Pyco (Former Digium Employee and extrememly
> >> smart guy.
> >>
> >> He has code/patch
Wilton Helm wrote:
Technical aspects aside, I have seen way too many voip providers come
and go, meantime the local phone company enjoys pretty much a monopoly
and is not in any financial trouble - to me this is the biggest reason
for wanting a pots or bri or t1, etc over voip from a smaller co
Wilton Helm wrote:
> Thanks for engaging with me on this. I picked up the book and I see
> what you mean about Appendix B. I had under-appreciated it probably
> because of a paradigm shift I need to make. I think you meant
> Appendix E rather than F for dialplan.
>
> I still am not quite on
I'm with you on this. A VoIP trunking solution is never going to equal a LEC
PSTN solution. It may be adequate for some purposes, but I'm not about to dump
my BRI for a pair of IP numbers. The trade-offs aren't worth the small cost
savings for me. Just the packetized delays (not to mention i
>There are far better resources out there for teaching Linux
>newbies. Instead, voip-info.org attempts to provide the sorts of information
>that is useful for those already familiar with Linux
I can appreciate that. And I can appreciate being at the other end of the
pipe, as I like to gloss ove
Thanks for engaging with me on this. I picked up the book and I see what you
mean about Appendix B. I had under-appreciated it probably because of a
paradigm shift I need to make. I think you meant Appendix E rather than F for
dialplan.
I still am not quite on the same page with you, though.
>Rather, can you tell me who claims to support what hardware, so I can confirm?
I don't have any notes on what I did. It was a bunch of Google searches. I
seem to recall that Digium themselves made a two port BRI that would work.
Eicon has some well respected products that I am pretty sure ar
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
___
-- Bandwidth and Colocation Provided by http
On Tue, 27 Jan 2009 09:49:41 -0800, Michael Higgins wrote:
It seems to me that there a lot of "it ought to work or could be made
to to work" associated with implementing US BRI into Asterisk. That
being the case what's called for is someone to try it, just to prove
the point.
Over the past year
On 1/27/09, Steve Totaro wrote:
> On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen
> wrote:
>> On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote:
>>
>>> Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart
>>> guy.
>>>
>>> He has code/patches for zaptel to US BRIs work
Hello,
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file. This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and need
to get the dialstatus.
Thank you.
_
On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen wrote:
> On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote:
>
>> Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy.
>>
>> He has code/patches for zaptel to US BRIs work that include SPID as a
>> variable in zap conf
Don't over think this, guys. Again, the point of having a WIKI is to
allow for customization. A landing page for Asterisk documentation
within voip-info.org is all you need, not a whole new source of
documentation.
Jai Rangi wrote:
> **
> I understand. As someone else already mentione
**
I understand. As someone else already mentioned, Voip-Info.org is for more
than just Asterisk. Perhaps if we created a single source that was just for
Asterisk...where everyone could contribute towards making the documentation
better. I would be very interested in helping sponsoring such
On Tuesday 27 January 2009 12:35:15 Wilton Helm wrote:
> >It seems to me that everything one may want to know would be contained
> >on voip-info.org
>
> My own experience is that it covers a very broad spectrum (far broader than
> Asterisk) and in a rather terse manner. I have spent an hour or two
On Tue, 27 Jan 2009 12:27:04 -0600
Jerry Jones wrote:
> Instead you could always get a SIP/IAX provider.
Can you please elaborate as to how this answers my question? Would getting a
SIP/IAX provider be the same as getting a working 'BRI' ISDN line coming into
the machine and properly handled i
Doesn't look like SuSE is that evolved just yet. I poked at a few other init
scripts in /etc/init.d, and they're all pretty much in the format of:
echo -n "Starting something ..."
command
rc_status -v
Some of the init scripts are downright horrific in their design because of
this. I would im
On Tuesday 27 January 2009 15:05:57 Wilton Helm wrote:
> >It actually does contain references of all applicaitons, CLI commands, and
> > such.
>
> Where? I saw some examples, but I've never found an organized list of
> commands. I'd love it.
For applications, Appendix B, and for dialplan functio
On Tue, 27 Jan 2009 12:56:46 -0500
Jon Pounder wrote:
> I barked up the same tree you are barking for a while and just gave
> up - lots of "you could buy this and try it", but no proven solution.
That's exactly what I've come up with. Thanks for your reply.
I don't see anything truly contradic
> Date: Tue, 27 Jan 2009 12:50:36 -0600
> From: "Danny Nicholas"
> Subject: Re: [asterisk-users] Muted sound on a Linksys 962
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> Message-ID:
> Content-Type: text/plain; charset="us-ascii"
>
> This worked for me
> Ext
On Tue, 27 Jan 2009 11:44:12 -0700
"Wilton Helm" wrote:
> I'm in the same boat and have been looking at this for several
> months, but haven't actually jumped in, hands-on, yet. No, I don't
> think the situation is as dismal as you paint it, although the lack
> of appropriate marketing for BRI i
On Tue, 27 Jan 2009 20:43:30 +0200
Tzafrir Cohen wrote:
> To the best of my understanding, latest Asterisk should support it
> through chan_dahdi .
Cool. Got any links to any related informations, so I can go by more than
anonymous hearsay? '-)
> No need for extra bristuff or whatever. But thi
Jared Smith wrote:
On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote:
I'm still pretty new to the mailing lists myself. I don't consider
myself a novice Asterisk user, but one of my biggest 'complaints' is
the lack of a well documented FAQ or Manual for Asterisk.
Steve Totaro wrote:
> On Tue, Jan 27, 2009 at 12:49 PM, Michael Higgins wrote:
>
>> Folks --
>>
>> First, apologies for not lurking for weeks or months to get the culture of
>> the list. I read the recent post about improvement to the quality of posts
>> with some amusement and full agreement
On Tue, Jan 27, 2009 at 03:45:34PM -0500, Steve Totaro wrote:
> Get a hold of Marcin Pyco (Former Digium Employee and extrememly smart guy.
>
> He has code/patches for zaptel to US BRIs work that include SPID as a
> variable in zap confs.
Could you please expand on that point?
Why should such p
>It actually does contain references of all applicaitons, CLI commands, and
>such.
Where? I saw some examples, but I've never found an organized list of
commands. I'd love it.
Wilton
___
-- Bandwidth and Colocation Provided by http://www.api-digita
On Tue, Jan 27, 2009 at 12:49 PM, Michael Higgins wrote:
> Folks --
>
> First, apologies for not lurking for weeks or months to get the culture of
> the list. I read the recent post about improvement to the quality of posts
> with some amusement and full agreement. The problem is a big and very
>I wonder if BRI would have gotten traction if it offered PRI functionality
I can't say for sure, and don't even know the differences in functionality, but
you may be right. When I last ordered DID I couldn't justify PRI so brought it
in as analog. At that point in time and with that LEC PRI
On Tue, Jan 27, 2009 at 12:50:42PM -0700, Wilton Helm wrote:
> I just got a very nice posting from Tzafir showing me a web domain
> I didn't even know existed.
It only includes documentation generated by 'make docs' . And is
actually linked from the README itself.
> I'm not abandoning it by a
>I'm impressed that you picked up 6502 assembly out of an even larger
>"vaccum" considering there was no 'net back then to help at all. Did
>you install a PBX on an Atari?
No, I interfaced a Rockwell AIM to a 300 station Philips electromechanical PABX
(designed and built about 100 interface ca
How pompous are we now?
What happened to the 'open source community'?
There's a give and take involved; you answer questions you know how to answer
in the hopes that someone with greater experience and knowledge of the software
will answer your questions.
Yikes.
-Original Message-
Fro
On 1/27/09, Tilghman Lesher wrote:
>
> I would have said Queen's English, but that evokes Freddy Mercury.
>
...and Freddy Mercury evokes Kevin Fleming.
Perfect - we're back on topic!
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.st
On Jan 27, 2009, at 10:50 PM, Wilton Helm wrote:
>YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is
"worthless" at this point because it is only a cookbook example and
I haven't tried to modify it to meet my needs. I didn't inten
On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote:
> I'm still pretty new to the mailing lists myself. I don't consider
> myself a novice Asterisk user, but one of my biggest 'complaints' is
> the lack of a well documented FAQ or Manual for Asterisk.
Asterisk is truly an open-source communi
I wonder if BRI would have gotten traction if it offered PRI functionality
(DID's and aggregation of multiple spans). Even TODAY I would drop many of
my sip trunks for such hypothetical BRI trunks for locations where a full
PRI is too much capacity.
That's the bane of the PRI: "Welcome to Joe
>YMMV. Mine certainly did. For the better.
My comments were more negative than I intended. My installation is "worthless"
at this point because it is only a cookbook example and I haven't tried to
modify it to meet my needs. I didn't intend to imply that Asterisk is
worthless, just that I've
Thanks for the reply. I have looked at the links you provided and I think they
will be useful. I may have some issues with drivers for the HFC, but I guess I
won't know until I try it.
Wilton
___
-- Bandwidth and Colocation Provided by http://www.api
>To the best of my understanding, latest Asterisk should support it
>through chan_dahdi . No need for extra bristuff or whatever. But this
>needs some testing.
Any chance I could get some information on how to set it up and use it (keeping
in mind that I have limited Asterisk experience and no ex
> Wilton Helm wrote:
>
> [snip]
>>
>> My conclusion after installing a worthless * demo (that actually does
>> allow two SIPs to talk to each other) is that Asterisk is not of any
>> value to anyone other than a person who makes a full time career out
>> of running Asterisk systems. I've installe
Ira wrote:
> At 09:30 AM 1/27/2009, you wrote:
>
>> People are always going to ask stupid questions.
>>
>
> For me it's not so much the stupid questions as the expectations that
> we're here to solve their problems according to their needs. If that
> continues to happen and the noise leve
Wilton Helm wrote:
[snip]
>
> My conclusion after installing a worthless * demo (that actually does
> allow two SIPs to talk to each other) is that Asterisk is not of any
> value to anyone other than a person who makes a full time career out
> of running Asterisk systems. I've installed and
At 09:30 AM 1/27/2009, you wrote:
>People are always going to ask stupid questions.
For me it's not so much the stupid questions as the expectations that
we're here to solve their problems according to their needs. If that
continues to happen and the noise level gets high enough those that
have
On Tue, Jan 27, 2009 at 11:24:38AM -0700, Wilton Helm wrote:
> >New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
> >existent.
>
>
> I first looked at * about four months ago and rapidly came to the same
> conclusion. Even with the O-Reilly book, which I purchased in pap
This worked for me
Exten => s,1,Answer()
Exten => s,n,Dial(Zap/g1/w5551212)
What happens is that * doesn't go "full duplex" until it does a "Native
Bridge". The Answer Command creates a temporary bridge until the real one
can take effect.
-Original Message-
From: asterisk-users-boun...@l
We all need the Univeral Language translators from Star Trek.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Tuesday, January 27, 2009 12:42 PM
To: Asterisk Users Mailing List - Non-Commer
On Tuesday 27 January 2009 10:54:37 Philipp Kempgen wrote:
> Tilghman Lesher schrieb:
> > On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote:
> >> Minivoicemail actually has
> >> - multiple e-mail formats
> >> - locale support so you get the date in local language and format.
> >
> > Un
If you find something on a WIKI that is outdated, guess what you have an
opportunity to do . . .
Noah Miller wrote:
>> It seems to me that everything one may want to know would be contained
>> on voip-info.org
>
> Hmm. Dangerous statement. There are many things on the WIKI that are
> quite ou
I'm in the same boat and have been looking at this for several months, but
haven't actually jumped in, hands-on, yet. No, I don't think the situation is
as dismal as you paint it, although the lack of appropriate marketing for BRI
in the US has all but killed it here, making it relatively unatt
>It seems to me that everything one may want to know would be contained
>on voip-info.org
My own experience is that it covers a very broad spectrum (far broader than
Asterisk) and in a rather terse manner. I have spent an hour or two at a time
pouring over a topic there and come away little m
>New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
>existent.
I first looked at * about four months ago and rapidly came to the same
conclusion. Even with the O-Reilly book, which I purchased in paper, although
it is freely downloadable, I feel there is a huge dearth of
On Tue, Jan 27, 2009 at 09:49:41AM -0800, Michael Higgins wrote:
> What I did find left me with the impression that USA 'BRI', uh, '2B1Q'
> protocol(?) is not supported by *any* hardware vendor, at all, period,
> nor is it tested and proved in the software... stack(?), in one
> related branch o
On Tuesday 27 January 2009 10:46:40 Andrew Thomas wrote:
> -->> In many cases, this just isn't possible. While it would be nice
> to
> -->> have all
> -->> posts in the King's English, a great many users are in locales
> which
> -->> don't
>
> King's English???
I would have said Queen's Engli
Hi,
One of our customers has an issue with the callee not being able to hear them.
It seems to happen very frequently on one number in particular where
there are about 3 IVR menus to dial through
before getting to a live person. However, this does not happen on every call.
Running tcpdump on the RT
Instead you could always get a SIP/IAX provider.
On Jan 27, 2009, at 11:56 AM, Jon Pounder wrote:
> Michael Higgins wrote:
>
> At least here in Canada - DSL just seems to have killed BRI - you
> practically have to know the secret handshake to even be allowed to
> provision one any more. It kill
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Gladden wrote:
> Is 1.6 so cutting edge that I should not expect to find complete
> documentation (yet)like I seem to be expecting very easily?
Most of what is applicable to 1.4 is applicable to 1.6. I'm running 1.6
without any hiccups -- YMMV
Michael Higgins wrote:
At least here in Canada - DSL just seems to have killed BRI - you
practically have to know the secret handshake to even be allowed to
provision one any more. It killed it as an internet transport which was
its most widespread use, however its many benefits as a digital ph
Folks --
First, apologies for not lurking for weeks or months to get the culture of the
list. I read the recent post about improvement to the quality of posts with
some amusement and full agreement. The problem is a big and very real one. I
hope I'm not deepening it.
But my question isn't expl
> It seems to me that everything one may want to know would be contained
> on voip-info.org
Hmm. Dangerous statement. There are many things on the WIKI that are
quite outdated, and a great many other things that aren't there at
all.
> People don't ask stupid questions because of a lack of a FA
Hi Steve -
> New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
> existent.
Welcome to Open Source!
Seriously, look at the README files accompanying asterisk, dahdi, and
libpri. They will give you compilation/installation instructions.
You can also search this list with go
I wouldn't say that voip-info.org has everything that a person would
want to know.
This is especially true of any recent changes to dialplan applications
(and their available options)
Voip-info.org is a great place to start, and often you will find an
answer there. But not always.
People are
michel freiha schrieb:
> I would like to ask please about how I can force asterisk to send all G726
> codecs without translation...
Huh?
>
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
> g723- ---- -- -- --
> --
>
You will need to have a Nortel NRS server in your network.
Sent from my iPhone
Eric Moniz
On Jan 27, 2009, at 10:17 AM, Pablo Bernasconi
wrote:
Hi,
I need to integrate my Asterisk with a Nortel Meridian 11, but I can
´t use PRI, Analog lines, etc. It has to be via SIP protocol, and th
It seems to me that everything one may want to know would be contained
on voip-info.org
People don't ask stupid questions because of a lack of a FAQ to read,
they ask stupid questions because they're too lazy do to the footwork.
Robert Broyles wrote:
>
>> I think we'd be better off posting a r
I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG suggestion to peruse the FAQ first.
I agree with this 100%
I'm still pretty new to the mailing lis
2009/1/27 Olivier
>
> 2009/1/27 Olivier
>
> Hi,
>>
>> I carefully followed instructions in README file lasting with :
>> /root/register
>> ... blabla
>> asterisk -r
>> CLI> restart now
>>
>> Then asterisk -r fails with :
>> # asterisk -r
>> Asterisk 1.6.1-beta4, Copyright (C) 1999 - 2008 Digium,
Tilghman Lesher schrieb:
> On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote:
>> Minivoicemail actually has
>> - multiple e-mail formats
>> - locale support so you get the date in local language and format.
>
> Unfortunately, it's using setlocale(3), which is not thread-safe. Note t
-->> In many cases, this just isn't possible. While it would be nice
to
-->> have all
-->> posts in the King's English, a great many users are in locales
which
-->> don't
King's English???
Anyway - to quote Ralph Wigham "Me fail English? That's unpossible!".
you can use any 1.4 how to but just use dahdi (both modules and tools)
David
2009/1/27 Steve Gladden
> I meant digium.com.
>
> Yay for messups!
> It's been one of those weeks.
> Really.
>
>
>
>
> > New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
> > existent.
> >
> > You
I meant digium.com.
Yay for messups!
It's been one of those weeks.
Really.
> New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
> existent.
>
> You go to the main Asterisk page (digium.org) and really just old install
> instructions for 1.2 are in the examples.
>
> Downlo
On Tuesday 27 January 2009 09:57:54 Steve Edwards wrote:
> The -user and -dev mailing lists are a valuable resource -- when they are
> not cluttered by posts unrelated to the "charter" of the lists.
>
> In my limited memory, this last weekend represents a new low in the
> "relevant subject to noise
Hi Michel,
it seems there is a codec translation in between, have you tried to
avoid it setting the codec from g729 to ulaw?
I personally make Asterisk use alaw/ulaw codecs when sending faxes
without any kind of codec translation and it seems to work.
Giorgio
michel freiha wrote:
> Dear All,
>
Yes, but is it agents a,b,c or a,b,etc? If huey, dewey and louie always get
in, but Donald never does, something may be wrong with how Donald is set up.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo
Sanchez
Sent:
The higher you raise the barrier for entry to the mailing list, the more you
decrease the amount good the mailing list is actually capable of doing.
(barrier height is inversely related to how much help we can provide to the
people that need help the most)
I agree with you regarding the subject
search for the correlata thread there is a java applet work over iax so you
will not have any problems whit the routers/firewalls nats and that stuff.
at the end of the thread there is an example made by wolfgang.
David
2009/1/27 Dean Collins
> This conversation has been done to deathare ar
only 3
On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas wrote:
> Is it the same 3 or the first 3?
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
> *Sent:* Tuesday, Janua
check the codecs in sip.conf
2009/1/27 michel freiha
> Dear All,
>
> I'm trying to send Fax using T.38 protocol but the FAX is not going
> through..I'm getting the following error om /var/log/messages
>
> [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from
> SIP/80.169.210.181
New to Aserisk 1.6 and find the 'installation tutorials' seem low to non
existent.
You go to the main Asterisk page (digium.org) and really just old install
instructions for 1.2 are in the examples.
Download links only give you asterisk itself and not dahdi or libpri
which also are needed to run
On Tuesday 27 January 2009 02:03:37 Johansson Olle E wrote:
> 27 jan 2009 kl. 01.14 skrev Tilghman Lesher:
> > On Monday 26 January 2009 08:21:10 am Danny Nicholas wrote:
> >> Did you read the source for app_voicemail? Line 239 says you have
> >> to set
> >> locale in the config and have the sound
Hi Gabriel,
Yes this information is shown in real-time and also in historical
reports with the OrderlyStats system.
OrderlyStats is now available as a Server Edition you can download and
install yourself, as well as the FREE managed service.
You can get it at http://www.orderlyq.com/statistics
Is it the same 3 or the first 3?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bayardo
Sanchez
Sent: Tuesday, January 27, 2009 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-us
24 chanels
On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas wrote:
> What is your call-limit set to in sip.conf?
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bayardo Sanchez
> *Sent:*
The -user and -dev mailing lists are a valuable resource -- when they are
not cluttered by posts unrelated to the "charter" of the lists.
In my limited memory, this last weekend represents a new low in the
"relevant subject to noise ratio."
Replying to requests with meaningless, misleading, or
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