I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?
Marco
2009/3/26 Grygoriy Dobrovolskyy megaho...@gmail.com
skip2pbx is the best i tryed, but nasty price ;)
___
-- Bandwidth and
I can't say it's always been like this, as I don't recall, but, Background
in 1.0 behaved like this, answering the channel if it wasn't already
answered and playing the sound file/s until they finished or an exten was
dialed...
in 1.0 the 'skip' option would cause playback to be skipped if the
2009/3/26 John Morris aster...@zultron.com
Hi, Axel.
Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later into rpmrepo.org)? That way we won't
have further fragmentation and a larger user base to test bits (which
will be
On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote:
My preferred method is to use my own TFTP server. This makes changes to
accounts/phones very fast and easy. The whole process takes me about 5
minutes to deploy an entirely new phone.
1) I modified the Grandstream template to contain my own
2009/3/27 Marco Sambo derwid...@gmail.com
I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?
Marco
Skip2pbx is based on freebsd so i dont think thank you can install it on the
same pc.
2009/3/27 Mr. James W. Laferriere bab...@baby-dragons.com
Hello Mark Miquel ,
On Thu, 26 Mar 2009, Mark Michelson wrote:
Miguel Molina wrote:
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the
On Thu, 26 Mar 2009, Andrew Hakman wrote:
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
I do. Not had a problem apart from when Digium break the protocol.
1.2 - Interweb - 1.2 - Interweb - 1.2
Hi all,
I was wondering if anyone knows how to integrate the Neospeech Text to
Speech engine with asterisk.
I have scoured the web and haven't found anything.
I think it's possible, I just don't know how to do it.
If Any body tried Neospeech with Asterisk then kindly share the experience
or
Hi, I have been trying to get a Wildcard TE122 card running here the
last couple of days.
libpri and zaptel are all installed and configured to E1 specs. The
jumper on the card is on, so configured for E1 (I'm in Norway).
When running zttool, I get 'Alarms: RED' on the single card installed.
Here in germany D-Link sells a device called the Horst-Box
Professional wich is a ADSL modem/router with WiFi and an integrated
embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind
serves me right. Size is about 180x250x50mm. Its been around for some
years so maybe it is
This sounds like you have pri_net instead of pri_cpe in Zapata.conf.
When inserting the cable going into TE122 into an ISDN phone, the
phone
works perfectly.
Any suggestions would be greatly appreciated :-)
___
-- Bandwidth and Colocation Provided
Andreas-Johann Ulvestad wrote:
When inserting the cable going into TE122 into an ISDN phone, the phone
works perfectly.
Ummm... you have a BRI, not a PRI. I've never heard of an ISDN phone
with an ISDN PRI port (E1 or T1).
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Andreas-Johann Ulvestad schrieb:
When inserting the cable going into TE122 into an ISDN phone, the phone
works perfectly.
That should not happen with an E1 line as your phone normally has a BRI
(S0) connector with only two b-channels.
Seems that your line is configured ar BRI and not PRI.
John Todd wrote:
Would it be so difficult to have perhaps two different proxies? One
would be for any SIP messages destined for IP addresses that were not
in any of the localnet= lines, and one would be for any SIP messages
destined for IP addresses that were destined for IP addresses
Try to build a local loop cable first
Loop pins 1-4 and 2-5 and connect to e1 port of your card.
You should see the green light instead of red on card physically and ur
alarm should go green too
http://wiki.sangoma.com/Cablepinouts check here for cable diagram
Hi, I have been trying to get
Hey,
The phones we receive are all on HTTP by default and point to
fm.grandstream.com by default.
So I added a hosts entry to my router pointing this to my own server and
my server automatically adds the mac address to the database.
This way, my selects the item and says what the username and
Thanks. I am forced to change servers anyways, so I'm starting from scratch,
which gives me the benefit of allowing me to plan things exactly as I want them.
I was hoping to avoid the TC400B until the server itself was almost under
strain, at which point I`d put one (or two) of those in to
I've used NeoSpeech's Java API to build a custom TTS interface that
creates sound files. I call that from Asterisk using AGI. Then I just
have Asterisk play the file I created.
From: asterisk-users-boun...@lists.digium.com
hi
for 800 you can have a complete core 2 quad server you should have many
servers and make an asterisk cluster instead of one super server.
David
2009/3/27 Mike l...@virtutel.ca
Thanks. I am forced to change servers anyways, so I'm starting from
scratch, which gives me the benefit of
This is kinda weird, but I did a fresh install of the box, upgraded
from 1.4.18 to 1.4.24, replaced Zaptel with latest DAHDI. That kinda
worked, but it had troubles recognizing both my TE121's, so I make a
SVN checkout of DAHDI and installed that. It works fine. Not a single
PRI drop in 11 hours.
Hey All,
ATT is installing a PRI in a couple weeks and while I've been doing
homework on PRI's for the last few weeks there's something I'm still
confused about. After being asked how many digits I wanted them to send
us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked
This is the outgoing callerid. If you have 1200 DIDs in a range, you probably
only need to outpulse 4 digits (they already know the first six). If you want
to be able to make your callerid anything that may or may not be one of your
DIDs, you probably want 7 or 10. I pick 10 no matter what for
I believe she is refering to how she's going to send you your incoming calls
(on your DIDs) for example:
10 digits: 972-453-2345
7 digits: 453-2345
4 digits: 2345
so you know how to expect your incoming calls and configure your
extensions.conf accordingly.
On Fri, Mar 27, 2009 at 10:06 AM,
Dave Fullerton wrote:
Hey All,
ATT is installing a PRI in a couple weeks and while I've been doing
homework on PRI's for the last few weeks there's something I'm still
confused about. After being asked how many digits I wanted them to send
us (10) was how many digits will you outpulse to
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
Try changing
[fax-in]
exten =
Mr. James W. Laferriere wrote:
Hello Mark Miquel ,
On Thu, 26 Mar 2009, Mark Michelson wrote:
Miguel Molina wrote:
Hi all,
For those of you people that use Agents (with Agentlogin, not
AgentCallbackLogin) on a call center, I have this need: when the agent
logs in, a channel keeps
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex) softswitch and
some hardphones (Thomson ST2030).
An old draft (draft-levy-sip-diversion-08.txt) mentions this header.
ha
I'm wondering if this could be used
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex) softswitch and
some hardphones (Thomson ST2030).
An old draft (draft-levy-sip-diversion-08.txt) mentions this header.
Has this been replaced by something else ?
Regards
PS: Apologize
Olivier wrote:
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex) softswitch
and some hardphones (Thomson ST2030).
An old draft (draft-levy-sip-diversion-08.txt) mentions this header.
ha
I'm wondering if this could be used
D Tucny wrote:
2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com
Hi, Axel.
Axel Thimm wrote:
How about merging in your changes/improvements/new packages with
ATrpms (and automatically later into rpmrepo.org
http://rpmrepo.org)? That way we
Greetings list,
I'm trying to establish if there's an issue whereby certain telcos in certain
countries have not updated the London, UK numbering plan to include some parts
of the 020 3 range, despite it being in operation for some two years now.
To help with this, I'd be most grateful anyone
Can You post your solution?
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo
comun
Quoting Chris Bagnall li...@minotaur.cc:
Greetings list,
I'm trying to establish if there's an issue whereby certain telcos
in certain countries have not updated the London, UK numbering plan
to include some parts of the 020 3 range, despite it being in
operation for some two years
I've got a weird problem:
I've added a new phone and sip show peers shows a status of OK (x ms)
but when I dial it I get status is 'UNKNOWN'
Any help on how to troubleshoot this?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200
2009/3/27 Marco Sambo derwid...@gmail.com
I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?
Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning
there's no software to install on your system. In minutes the
It's pretty long and involved do to a fair amount of customization we
had to do. The NeoSpeech documentation includes the API and examples
for using it with Java, C, .Net and COM and does a better job of
explaining what you need to do than I could in a mailing list. However,
if you run into
Thins number is wrong - it has too many digits - should only be eight
after the 20. (possible you put a surplus 3 in?)How incredibly embarrassing.
You are of course correct, try +44 20 3393 7389 :-)
-Original Message-
From: Phil Reynolds [mailto:phil-aster...@tinsleyviaduct.com]
Am Freitag, den 27.03.2009, 16:35 + schrieb Phil Reynolds:
Quoting Chris Bagnall li...@minotaur.cc:
Thins number is wrong - it has too many digits - should only be eight
after the 20. (possible you put a surplus 3 in?)
Good guess, indeed +44 20 3393 7389 has an answering machine as
Can anyone give me any idea on where to start looking for this ? 1.4
svn (ish) It has appeared twice in the last hour on a system that gets
numerous inbound calls to the same number
TIA
Julian
[Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error:
The problem with this seems to be that when you make a distro, you want it
to be many things to many people (easy to use, lots of features, support
lots of hardware, you name it). When you build a medium/large call-center,
you usually want to keep it lean and mean, as you need a high uptime and do
On Thu, Mar 26, 2009 at 10:22 PM, David fire ddf...@gmail.com wrote:
you can use the asterisk Manager or AMI.
there is a very good java project asterisk-java but there are librarys for
almost every languaje.
look for Asterisk Manager and AMI www.voip-info.org is a good place to
start
This should be engraved in stone. IMHO, doing so even with a traditional
telco solution would be extremely risky, if one does not have an adequate
skill set and experience.
Thanks
l.
2009/3/26 Matt Riddell li...@venturevoip.com
If you are doing an install for a call centre with 100-200
On Fri, 2009-03-27 at 17:33 +, Julian Lyndon-Smith wrote:
Can anyone give me any idea on where to start looking for this ? 1.4
svn (ish) It has appeared twice in the last hour on a system that gets
numerous inbound calls to the same number
[Mar 27 17:21:07] WARNING[3239]:
In case any of you were wondering why there has been a fairly notable
upswing in the attacks happening on SIP endpoints, the answer is
script kiddies. In the last few months, a number of new tools have
made it easy for knuckle-draggers to attack and defraud SIP endpoints,
Asterisk-based
2009/3/27 Mark Michelson mmichel...@digium.com
Olivier wrote:
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex) softswitch
and some hardphones (Thomson ST2030).
An old draft (draft-levy-sip-diversion-08.txt) mentions this
The Asterisk development team is pleased to announce the release of Asterisk
Core Sounds version 1.4.15, Extra Sounds 1.4.8, and Freeplay Music On Hold
sound files. These sound files are available at
http://downloads.digium.com/pub/telephony/sounds/. Future versions of
Asterisk will do this
(Note: This announcement originally went out with an incorrect version number
mentioned for the Extra sounds. It should have went out as Extra Sounds 1.4.9
and has been corrected in this announcement. Thank you for your understanding.)
The Asterisk development team is pleased to announce the
Olivier wrote:
2009/3/27 Mark Michelson mmichel...@digium.com
mailto:mmichel...@digium.com
Olivier wrote:
Hi,
Is anyone aware of SIP Diversion header ?
It seems currently supported by Comverse (formely NetCentrex)
softswitch
and some hardphones
Hi
Had an issue today where all channels connected to the telco when dialed
returned
WARNING[15366] chan_zap.c: Call specified, but not found?
in the logs,
when I removed the isdn cable and reinserted everything was fine
any ideas?
software Versions
asterisk-1.4.21.2
zaptel-1.4.12.1
Does anyone know if the TE122 is recognized by any of the 1.2 zaptel
drivers? It seems that 1.2.16 knows it not... :)
Cheers,
j
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To UNSUBSCRIBE or
Jeff LaCoursiere wrote:
Does anyone know if the TE122 is recognized by any of the 1.2 zaptel
drivers? It seems that 1.2.16 knows it not... :)
I know the TE122 is supported in Zaptel 1.2.27. Possibly supported in a
few earlier versions as well.
Cheers,
Shaun
--
Shaun Ruffell
Digium, Inc. |
Excellent! Muchos gracias.
j
On Fri, 27 Mar 2009, Shaun Ruffell wrote:
Jeff LaCoursiere wrote:
Does anyone know if the TE122 is recognized by any of the 1.2 zaptel
drivers? It seems that 1.2.16 knows it not... :)
I know the TE122 is supported in Zaptel 1.2.27. Possibly supported in a
On Fri, 27 Mar 2009, John Todd wrote:
Seven Easy Steps to Better SIP Security on Asterisk:
Six/seven -- who's counting...
Thanks for this checklist.
Looking forward to discussion and additions.
While not specifically related to SIP, how about using autoload = no in
modules.conf and only
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