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 file einE. Of course an
easier way
would be to locate the 19 day
2009/1/27 Olivier oza-4...@myamail.com
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, Inc. and others.
Created
Hi,
How can view the list of RFCs and other drafts that asterisk
supports?
Basically I would like to know does asterisk supports RFC 3891?
Thanks,
Niraj Roy
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Hi to all
i'm planning the migration of a company on Asterisk, i have planned
this scenario:
2 server with
* 4 GB RAM
* 2 CPU 64 bit dual core
* RAID 1
* 2 network interfaces 1000 Mbit/s
Each server will have a virtual interface that will be switched from
one to the other in case of hardware
Am Dienstag, den 27.01.2009, 11:30 +0100 schrieb nik600:
Hi to all
i'm planning the migration of a company on Asterisk, i have planned
this scenario:
2 server with
* 4 GB RAM
* 2 CPU 64 bit dual core
* RAID 1
* 2 network interfaces 1000 Mbit/s
Each server will have a virtual
Hi all,
Is there a way to get the time that a specific queued call took to be
answered or abandoned?
Thanks,
Gabriel Ortiz
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asterisk-users mailing list
To UNSUBSCRIBE or update
If you use reinvites (the peers are all internal, right?), definitely.
Without reinvites - maybe.
On Jan 27, 2009, at 5:30 AM, nik600 nik...@gmail.com wrote:
Hi to all
i'm planning the migration of a company on Asterisk, i have planned
this scenario:
2 server with
* 4 GB RAM
* 2 CPU 64
Grygoriy,
[...] A practice that was once described in the code comments as
being nasty.
thanks for your input. My knowledge of 'hard core' programming is limited,
so I cannot judge on what is written on freeswitch.org. Though it sounds
logical to me.
But as I said, this is on a production
2009/1/27 Udo Schacht-Wiegand aster...@wiegand.name:
Grygoriy,
[...] A practice that was once described in the code comments as
being nasty.
thanks for your input. My knowledge of 'hard core' programming is limited,
so I cannot judge on what is written on freeswitch.org. Though it sounds
Dear Sir,
I would like to ask please about how I can force asterisk to send all G726
codecs without translation...
g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723- ---- -- -- --
--
gsm- -22
Bayardo Sanchez wrote:
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
'SIP/8022-b7225740'
-- Got SIP response 503 Service Unavailable back from 74.63.41.218
-- SIP/voipms4-09ab0c38 is circuit-busy
== Everyone is busy/congested at
SURE!
you can use the manager event handler and a function dump_event!
you can dump all kind off events to a text file or a database!
I did it successfully!
GOOD LUCK:)
2009/1/27 Gabriel Ortiz Lour ortiz.ad...@gmail.com
Hi all,
Is there a way to get the time that a specific queued call
Thanks for the feedback, but unfortunately, there is still no joy. If I do this:
echo -n database put FOO BAR 1 | socat STDIO
UNIX-CONNECT:/var/run/asterisk/asterisk.ctl
I get the following output on the command line:
pbx-75/2395/1.4.20.1
pbx-75 is the hostname, 2395 is the PID of the
Hello all,
I need to configure an application which let me to call from a web page.
Someone has experience using apps to make webcalls?
Which software do you use?
Thanks.
VoipCrazy.
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This conversation has been done to deathare archive search not
available.?
But the answers are www.mexuar.com
www.phonefromhere.com
and there are also a free open source versions but they take work on
your part to setup.
Regards,
Dean Collins
Cognation Inc
d...@cognation.net
The problem is I have 20 agents calling all but 3 of them get this error
when calling
On Tue, Jan 27, 2009 at 6:49 AM, Max Brooks m...@legatio.com wrote:
Bayardo Sanchez wrote:
i have a problem need help
== Spawn extension (DLPN_everything, 2095773777, 2) exited non-zero on
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 there is few information
about this type of integration.
Could someone please help me??
Thanks, Pablo
___
--
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: Tuesday, January 27, 2009 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Good morning,
I've been having some problems getting the SPA-3102 working properly in
India. Specific problem is that calls from the Asterisk server out the FXS
port is failing. When trying to make calls, I'm getting this message:
[Jan 26 23:00:31] NOTICE[2136]: chan_sip.c:13774
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/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256)
[Jan 27 16:46:40]
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
24 chanels
On Tue, Jan 27, 2009 at 9:23 AM, Danny Nicholas da...@debsinc.com 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
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:
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
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 file einE.
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
check the codecs in sip.conf
2009/1/27 michel freiha mich...@gmail.com
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
only 3
On Tue, Jan 27, 2009 at 10:03 AM, Danny Nicholas da...@debsinc.com 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:*
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 d...@cognation.net
This conversation has been done
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
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:
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,
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 ratio.
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.
Download
you can use any 1.4 how to but just use dahdi (both modules and tools)
David
2009/1/27 Steve Gladden aster...@michiganbroadband.com
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
-- 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!.
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 that
2009/1/27 Olivier oza-4...@myamail.com
2009/1/27 Olivier oza-4...@myamail.com
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)
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
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
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
pbernasc...@isbel.com.uy 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
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- ---- -- -- --
--
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
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
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
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
-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.
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 FAQ
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 killed
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
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 English, but that
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 or
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
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
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
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
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.
Unfortunately,
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 -
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:
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 paper,
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
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
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 level gets
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
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
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
___
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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
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
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
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 intend
On 1/27/09, Tilghman Lesher tilgh...@mail.jeffandtilghman.com 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
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-
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 cards,
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 any
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 12:49 PM, Michael Higgins li...@evolone.org 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
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
___
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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
Steve Totaro wrote:
On Tue, Jan 27, 2009 at 12:49 PM, Michael Higgins li...@evolone.org 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
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.
On Tue, 27 Jan 2009 20:43:30 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com 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
On Tue, 27 Jan 2009 11:44:12 -0700
Wilton Helm wh...@compuserve.com 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
Date: Tue, 27 Jan 2009 12:50:36 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Muted sound on a Linksys 962
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Message-ID: a8d2ef23f63b4a42b264bb151c5d6...@db0005
On Tue, 27 Jan 2009 12:56:46 -0500
Jon Pounder j...@inline.net 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
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 functions,
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
On Tue, 27 Jan 2009 12:27:04 -0600
Jerry Jones jjo...@danrj.com 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
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 at a
**
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
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
On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.com 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
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 1/27/09, Steve Totaro stot...@totarotechnologies.com wrote:
On Tue, Jan 27, 2009 at 4:07 PM, Tzafrir Cohen tzafrir.co...@xorcom.com
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
On Tue, 27 Jan 2009 09:49:41 -0800, Michael Higgins wrote:
snip
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
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?
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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
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,
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 over
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
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 the
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