Hi,
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Headless Linux system for Asterisk
Just make sure you make sure the BIOS is set not to halt the system on
any errors.
What about make it work without a graphic card too?
If the MB has the
Sorry what did you say you had in your hand ???
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Friday, 19 December 2003 2:34 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] after hours
On Thu,
Cees de Groot schrieb:
Andrew Thompson [EMAIL PROTECTED] said:
While an exceptionally devious concept, I don't think it'd work out like you
planned. Wouldn't that mean you'd have to dial out the 900 number yourself,
meaning You would be charged for the 900 call.
At least with ISDN, you
Dan [EMAIL PROTECTED] said:
Why to pay for a graphic card from the new generation if you don't need
it...
Or you buy a decent A brand pizzabox, which has everything on-board. We
use IBM xSeries, which has everything you want built-in and comes with
management hardware so I can take over the
Hi all,
I was wondering whether any of you have experience/info on Cable and/or ADSL
modems that would come together with a SIP phone adaptor. What I am
interested in is something that would plug directly into you ISP's cable (be
it ethernet or adsl/phoneline), would combine a modem/router/nat
Hi all,
There is someone with some experience interconnecting a Panasonic digital
PBX (KX-TD1232) with Asterisk?
Thank you and best regards,
Dan
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Hello,
I`m using the Dlink DG-104s with asterisk, it works ok for incoming
outgoing calls. The problem is, when for exmple the line on the dg104s
is off hook and I dial that extension I dont get a busy but a ringing
tone. And no call waiting signals on the dg104s side. Is there a way to
detec
help reload says:
Reloads configuration files for all modules which support reloading.
Well that's all fine and dandy, but which modules actually support
reloading?
MGCP does not support reloading
Philipp
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Hi!
I was wondering whether any of you have experience/info on Cable and/or ADSL
modems that would come together with a SIP phone adaptor. What I am
interested in is something that would plug directly into you ISP's cable (be
it ethernet or adsl/phoneline), would combine a modem/router/nat
I've read through the archives and have picked up that * does not need a
gatekeeper to talk directly with an H323 handset to send and receive calls.
I'm trying to go PSTN*-H323 and all the examples that I can find
use a gatekeeper. Are there any examples or hints for doing it without
Hi,
I'm, doing the same, but with a php agi, and invoking a modified script to use
sphinx:
/usr/local/bin/decoder2
#!/bin/sh
BASE=6403
S2BATCH=sphinx2-continuous
HMM=/usr/local/share/sphinx2/model/hmm/6k
TASK=/usr/local/share/sphinx2/model/lm/${BASE}
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 19 December 2003 04:18, John Todd wrote:
An additional modifier to Dial was added specifically for Local
channel types. If you add the /n modifier at the end of a
chan_local call, then the variables will be erased upon passage
through
bam wrote:
I've read through the archives and have picked up that * does not need
a gatekeeper to talk directly with an H323 handset to send and receive
calls.
I'm trying to go PSTN*-H323 and all the examples that I can
find use a gatekeeper. Are there any examples or hints for doing
Hi Patrick,
I have exactly the same problem with diva BRI-2M. Did you solve it?
Regards,
Daniel
Patrick a écrit:
Hello,
I have an issue getting the chan_capi module to load in asterisk cvs
from today. Plain 2.4.20 kernel with melware patches for the Eicon Diva
Server Bri card.
I load the
My understanding is that if the codec type matches on each leg of the
call then
Asterisk is using pass through. The PBX simply has to relay the RTP
packets
unchanged from one device to another. If you are going from a phone using
codec tupe A to codec type B, Asterisk must perform a conversion
- Original Message -
From: Dawid Mielnik [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 6:01 AM
Subject: [Asterisk-Users] nat router + sip phone adaptor (+adsl modem)
Hi all,
I was wondering whether any of you have experience/info on Cable and/or
ADSL
modems
SW wrote:
Hi Gurus
Not a guru, but I'll see what I can do.
How do I make x100P does not answer incoming calls ?
The only thing that springs to mind is that you create an incoming
context, and have an extension like:
Exten = s,1,Wait(1000)
Dunno if it will work or not, but
On Thu, 2003-12-18 at 17:09, Juan J. Sierralta P. wrote:
Does anybody know if Polycoms has a three finger salute as Cisco 79XX
does ? I really hate to unplug ethernet cable since you have to release
the stand first.
I respond myself, hold down: Volume+, Volume-, Hold and
I ahve found the pb: there was some /dev/... device mising.
Regards,
Daniel ANDRE
Daniel ANDRE a écrit:
Hi Patrick,
I have exactly the same problem with diva BRI-2M. Did you solve it?
Regards,
Daniel
Patrick a écrit:
Hello,
I have an issue getting the chan_capi module to load in asterisk
Hi all,
My * box is registered with IAXTEL too.
The problem is that iaxtel.com [69.73.19.178] makes me a lot of traffic
(both inbound and outbound) on my external IP address (* is behind a NAT
router and only UDP port 4569 is open to the server).
More, all by browsing and download traffice is
Hello,
Is there any docmnetation on the configuration of chan_capi:
syntax of capi.conf?
Dial string configuration?
Best Regards,
Daniel ANDRE
--
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com
- Original Message -
From: Pavel Litvinenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 8:42 AM
Subject: Re: [Asterisk-Users] Asterisk to H.323 without gatekeeper
bam wrote:
I've read through the archives and have picked up that * does not need
a
Answers at the end.
On Fri, 2003-12-19 at 14:37, Daniel ANDRE wrote:
Hi Patrick,
I have exactly the same problem with diva BRI-2M. Did you solve it?
Regards,
Daniel
Patrick a écrit:
Hello,
I have an issue getting the chan_capi module to load in asterisk cvs
from today.
I think this would do it: Dlink DVG-1120M/H/S
No experience with it
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Dawid Mielnik
Verzonden: vrijdag 19 december 2003 12:02
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] nat router + sip phone
Did you ifdown the dynamic interfaces first ?
Martin
Yes, this still results in a crash on the box with the tor2.
Is there any 'controlled' way to bring down a dynamic span?
... just for fun here is zttool (spans 3 4 are unused)...
Alarms Span
OK Tormenta 2 (PCI) Quad T1
Hi all,
I have tested RxFAX application through X100P card. When Fax
arrive i obtain the next trace:
-- Starting simple switch on 'Zap/1-1'
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing SetMusicOnHold(Zap/1-1, random) in new stack
-- Executing
Would someone who has the HandyTone ready to ship reply offlist with price
with 2day and 3day shipping to 28379?
-
Andrew Thompson http://aktzero.com/
Your eyes are weary from staring at the CRT. You feel sleepy. Notice how
restful it is to watch the cursor blink. Close your eyes. The
Hi All,
I need help configuring my E100P digium card. I am testing it with a marconidigital transmission analyzer. Although everything configures fine. All channels get configured and the output of cat /proc/zaptel/1 verifies that.
problems;
1. i dont know how to use the zttool for debugging
Robert Mann wrote:
Residential Long Distance.
One of my biggest pushes towards a VoIP provider was cheap long
distance. Now in the U.S. at least with SBC they now have a plan for
Unlimited Long Distance. The price is 30.00 a month if you do not have
a couple of required features on the
Not all VoIP providers will have Vonage's 911 issues. It's perfectly
possible for a VoIP provider to provide outbound caller information to
the PSAPs if they spend the time and money to do so.
Stephen
Summary: if you're the only caller, calling only to the US, then you
might be crazy to not
That was it!
I have spent the last week dealing with Cisco tech support on this. They
even sent me a new phone. They escalated this to senior support.
They couldn't get it, I couldn't get it, but you could.
Thanks ever so much!
Merry Christmas,
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
Dawid Mielnik wrote:
Hi all,
I was wondering whether any of you have experience/info on Cable and/or ADSL
modems that would come together with a SIP phone adaptor. What I am
interested in is something that would plug directly into you ISP's cable (be
it ethernet or adsl/phoneline), would combine
--- Mauricio Nuez [EMAIL PROTECTED] wrote:
/usr/local/bin/decoder2
#!/bin/sh
BASE=6403
I changed this to BASE=turtle
$S2BATCH -nbest 1 -nbestdir /tmp -verbose 0 -adcin TRUE -adcext wav -ctlfn
${CTLFILE} -ctloffset 0 -ctlcount 1 -datadir ${TASK} -samp 8000
-agcmax TRUE -langwt 6.5
Sorry for the lack of info...here goes - We are using CVS from earlier
this week. Our phones are fxs signalling and it is connected to
asterisk via a channelbank and a TE410P card.
I have put a bug into the bugtracker, it's ID is 687
thanks,
Derek
On Thu, 2003-12-18 at 19:19, John Todd
Hey All,
I need to forward an extension to an other depending on the current
time but I could not get it done with GotoIfTime.
What I'm trying to do is ring on the extension 1 if time is between
8:00AM and 2:00PM and on extension 2 if is between
2:01PM 11:00PM.
exten =
What about having your VoIP gateway system placing a 911 call to the 911
answering center in the appropriate region and when the 911 operator
answers, have a message say This is a 911 call from 123 Main Street,
Nowhere Nebraska then connect the caller to the 911 operator. Legal?
Maybe. Dunno.
Hi,
If you put BASE=turtle, then the script search for
/var/lib/asterisk/model/lm/turtle/turtle.ctl, and the content of this file is
a list of files to process, without the extension, because this is provided
via the parameter -adcext of sphinx2-continuous.
the turtle dir is the demo provided
about GotoIfTime you have:
show application GotoIfTime
-= Info about application 'GotoIfTime' =-
[Synopsis]:
Conditional goto on current time
[Description]:
GotoIfTime(times|weekdays|mdays|months?[[context|]extension|]pri):
If the current time matches the specified time, then branch to the
Hi!
gotoif usually takes a priority as label, not an extension! See below.
Never tried if you can also use the notation context,extension,priority
instead of just priority, but it might work. Just try it.
I need to forward an extension to an other depending on the current
time but I could not
All right...
Its working now.
thank you very much!
regards
Oz
On Dec 19, 2003, at 3:02 PM, Philipp von Klitzing wrote:
Hi!
gotoif usually takes a priority as label, not an extension! See below.
Never tried if you can also use the notation
context,extension,priority
instead of just priority,
Hi,
--- Mauricio Nuez [EMAIL PROTECTED] wrote:
Inside ${BASE}, i had a control file ${BASE}.ctl, with the line:
/var/lib/asterisk/sounds/mensaje
hardcoded.
Ok, got that.
To work with asterisk, i put the -sample option at 8000 , because that is the wav
sampling. The turtle example work at
On Thu, 11 Dec 2003, Jeremy McNamara wrote:
Chandra wrote:
Explain why you think you really need RADIUS Accounting? Why not talk
right to the database itself and save yourself that unneeded
complication and points of failure.
Jerey,
ISP's integrating Asterisk could utilize
I would like to know if anyone has come up with a script for 911 dialing
rules that put correct information on our locations. We have our office
in 3 different building one being our production shipping dock. It is
almost 2 blocks away. We are connected with Ethernet Wireless between
the
On Thu, 11 Dec 2003 [EMAIL PROTECTED] wrote:
Explain why you think you really need RADIUS Accounting?
Why not talk
right to the database itself and save yourself that unneeded
complication and points of failure.
I know this has come up before, and in a perfect world, where *
On Thu, 11 Dec 2003, Andrew Thompson wrote:
Understand* boxes are in real live actual production now. Once you
leave the vacuum of the lab, there are going to be things like this that
come up. And many will be for good reasons. Others will be for
crappy, legacy reasons. Both
On Thu, 11 Dec 2003, Andrew Thompson wrote:
- Original Message -
From: Doug Shubert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 11, 2003 11:48 AM
Subject: Re: [Asterisk-Users] Re: * with RADIUS
Hello,
We use RADIUS with a MySQL backend database server for
Hi Stephen,
Interesting
6) When, and if, the quality/reliability improves sufficiently, a DID
line in the area code of your choice, which provides 6 simultaneous call
presentations for $7.99/month, will beat any land line hands down.
I did not know that one connection can have many
James Sharp wrote:
What about having your VoIP gateway system placing a 911 call to the 911
answering center in the appropriate region and when the 911 operator
answers, have a message say This is a 911 call from 123 Main Street,
Nowhere Nebraska then connect the caller to the 911 operator.
- Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: Asterisk User List [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 4:06 PM
Subject: [Asterisk-Users] 911 settings.
I would like to know if anyone has come up with a script for 911 dialing
rules that put correct
SW wrote:
Hi Stephen,
Interesting
6) When, and if, the quality/reliability improves sufficiently, a DID
line in the area code of your choice, which provides 6 simultaneous call
presentations for $7.99/month, will beat any land line hands down.
I did not know that one connection can have
I have a question on SIP devices that are setup and working but you
change the login name and contents to them why does asterisk need to be
shut down and restarted for them to work? I have reloaded extensions
and done a reload command. But the updated sip phones do not work until
I shut down and
I've configured the DIAX phone. It registers with the * server, and I
am able to make calls from DIAX.
However, when I try to call the DIAX phone from another phone, I get a
busy signal.
My extensions.conf:
exten = 70,1,Dial(IAX/mike/mike,30,tr) # IAX Mike
exten = 70,2,Voicemail(u70)
exten =
Hello All,
My SNOM 200 phone keeps generating the following message on the *
console:
Notice [11127005368] : File chan_sip.c Line 5394 (handle_request) :
Unknown sip command 'Publish' from '192.168.1.17'
What does this mean and how do I remedy the problem?
Thanks,
Michael
--
Michael Graves
As far as I know, iconnect
explicitly disallows multiple call presentations. The iconnect thing was
discussed on the list a month or two back.
I red that discussion, it was more on multiple outgoing calls. I noticed
with Iconnect, the sip invite message always comes as the
[EMAIL PROTECTED] So
Dan wrote :
Subject: [Asterisk-Users] Interconnecting Panasonic KX-TD1232 digital
PBX and *
Hi all,
There is someone with some experience interconnecting a Panasonic digital
PBX (KX-TD1232) with Asterisk?
Ehh, what exactly do you want to do? I've got * 'interfaced' via the ISDN S0
bus of
Hi all,
I am looking at setting up a TDMoE link between * boxes and am having a
rough time locating and documentation or configuration examples. I have
gotten far enough to get the dynamic link up between boxes, but not sure
where to go from here. I'm not even sure which modules need to be
We often hear about the problems with vendors, but not as
often when a vendor does more than expected.
Last night Iplaced an order for a Sipura ATA with http://www.voxilla.com/
Today theycalled me to explain that they had just run
out of inventory, and would be happy tocancel/refund my
Anyone investigated the new service offerings from Level(3) in the
last few months? They claim to be using ENUM and SIP - see
http://www.level3.net/2192.html for details. Any idea of their
pricing model for mid-sized enterprise applications or call centers
for origination/termination? More
At 3:42 PM -0800 12/19/03, SW wrote:
As far as I know, iconnect
explicitly disallows multiple call presentations. The iconnect thing was
discussed on the list a month or two back.
I red that discussion, it was more on multiple outgoing calls. I noticed
with Iconnect, the sip invite message
Hi All,
I wish to connect * to a Cisco using a E100P board.
When I load the driver I got this error message:
-bash-2.05b# modprobe wct1xxp
ZT_CHANCONFIG failed on channel 1: Function not implemented (38)
/lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed
I am new to the sip side of things and have a question regarding
ringback. I don't hear ringback when using the sjphone softphone when
dialing internal extensions. It's fine when dialing outside over the
pstn.
Is this a issue of the softphone, configuration or sip in general?
Thank you,
-gcc
After searching the archives for a while, I couldn't find any easy way to
get everything loaded on startup. So I decided to take a stab at writing
some notes on what I've found. If everyone chips in, maybe we can make that
part easier for new users!
Both the Zaptel and Asterisk packages have a
At 5:26 PM -0500 12/19/03, Andrew Thompson wrote:
I would like to know if anyone has come up with a script for 911 dialing
rules that put correct information on our locations. We have our office
in 3 different building one being our production shipping dock. It is
almost 2 blocks away. We
It's certainly not _illegal_ in any way that I can think of, and I
expect that anything is better than no information at all. Sounds
like a good idea. The only shortcoming I can think of is the lack of
ability for the PSAP to hear the address more than once without
making an unacceptable
Andrew Thompson wrote:
- Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: Asterisk User List [EMAIL PROTECTED]
Sent: Friday, December 19, 2003 4:06 PM
Subject: [Asterisk-Users] 911 settings.
I would like to know if anyone has come up with a script for 911 dialing
rules that
On Fri, 19 Dec 2003, Nick Bachmann wrote:
I don't know how big of a customer you are for your phone company, but
if you have more than a token number of lines they'll hopefully go for it.
Another option is to call the non-emergency number of the dispatch center
and explain this one
On Saturday, 13 December, 2003 11:16, Tilghman Lesher wrote:
...
Directory does not need an escape condition. If you fail to enter
anything within the allotted time (see ResponseTimeout), you jump
to the t extension.
That makes for a rather ill solution for the poor fool (like me, often)
WARNING[1232119104]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect process 256 frames
Do not try to do inband DTMF on G.729
Can we wiki-fy this?
Regards,
Andrew
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