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Stefan-Michael. Guenther (in-put GbR) wrote:
Hi,
Am Dienstag, 8. August 2006 13:41 schrieb Matt Riddell (NZ):
Stefan-Michael. Guenther (in-put GbR) wrote:
Hi,
I'm trying to start a PHP5 script via the AGI Interface.
The asterisk version is
Hi,
Am Mittwoch, 9. August 2006 08:09 schrieb Matt Riddell (NZ):
The problem is, as you can see from the output in the CLI, that
Asterisk claims that it executes the script, but nothing happens. It
doesn't create the file /tmp/asterisk and it doesn't send an email.
When I execute the
I'm attempting to setup asterisk running real-time with mysql. Right now I
can get asterisk to start and run but a show dialplan shows basically
nothing other than parking extensions. I'm watching the full log also for
debug messages and I can see that asterisk is connecting to mysql with out
If i am not mistaken, you need to have another IAX user tables to store all the iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!On 8/9/06, Shaun [EMAIL PROTECTED] wrote:
I'm attempting to setup asterisk running real-time with mysql.Right now Ican
Hi all
Does any one experience the scenario in which an agent sits behind any trunk
and serves the queue. Bcz I have tried and the queue acts very veared as
when it tries the agent extension across the trunk it starts music onhold
and when the agent is busy/not responding then it stops music on
Hi all
Does any one experience the scenario in which an agent sits behind any
trunk
and serves the queue. Bcz I have tried and the queue acts very veared as
when it tries the agent extension across the trunk it starts music
onhold
and when the agent is busy/not responding then it stops music on
I registered 5 g729 codec and the result was , I cant use
these channels because all channels are not available even I have no call on
the system
5/0 encoders/decoders of 5 licensed channels are currently
in use
Please help
*
If I make a call to a mobile phone from an ISDN30 PRI line and the call
is A) not answered (but no voicemail) or B) the call is rejected, is
there any list anywhere of the different codes returned to the ISDN
(hangupcause)
For example, if I call an o2 mobile, and reject is pressed then I get
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
"Sharon Lim" [EMAIL PROTECTED] wrote
in message news:[EMAIL PROTECTED]...If
i am not mistaken, you need to have another IAX user tables to store all
Sorry if i am wrong. Did you add something in extensions.conf to identify your context ? Something like this http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
BTW, how come your extensions got snip? I taught extension is a number that you dial?On 8/9/06, Shaun
[EMAIL PROTECTED]
No, i cannot open a bug on this, cause i dont have a PRI that uses zap.
so if there were any questions,
you had to answer them.
But there is a similar bug, using mISDN.
http://bugs.digium.com/view.php?id=7435
and a solution for ME, dont know if it will help you:
Hi,
I have solved it (but don't understand yet, why it works)!!!
SuSE 10.1 uses different configuration files for the cli version and the cgi
version of PHP5.
When I modify the first line in the script from
#! /usr/bin/php5
to
#! /usr/bin/php5 -c /etc/php5/cli/
the php scripts gets
I want to connect two ISDN bri card directly.
Is it necessary to use a cross-cable?
I use a fritz card in TE mode and a Atlantis Card (zaphfc) in NT mode.
Thank's Matteo
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asterisk-users
James Arscott [EMAIL PROTECTED] writes:
Hi
Small progress, though combining the suggest below, enabling overlapdial and
a few other things I have got the following :
When you hit 9 on the simenes, you hear a dial tone. As soon as you hit
another number to start dialling it complains with
Olle E Johansson wrote:
7 aug 2006 kl. 14.37 skrev Rich Adamson:
You have two choices to correct the behavior. One, change the asterisk
definitions so as to show a preference (disallow=all,
allow=g729,ulaw), or, two, change the sip phone's definition to prefer
g729 as its first choice.
With respect, I don't think you understand the dynamics of growing a
business.
If we are all to benefit from the continued development of Asterisk then
it is in our own best interests for Digium to succeed, because their
success is for our benefit. Your posting is unfortunate as it
On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote:
I want to connect two ISDN bri card directly.
Is it necessary to use a cross-cable?
No. A standard (non-crossed) ethernet cable will work, as long as it has
all 8 wires.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
If you gave software to Digium then you helped Mark become very rich.
What's wrong with making Mark a rich man? He has come up with a great new
product and I'm sure he has risked a lot to get it to you. Asterisk is free
so he owes you nothing.
How about you take your jealousy elsewhere or
We experienced this problem with a Sangoma A104D card. With echo cancel turned on, the card was not detecting incoming DTMF digits to our IVR properly. However, when we added the line relaxdtmf=yes to zapata.conf, the problem went away. If the relaxdtmf setting is not curing the problem for you, I
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Stefan-Michael. Guenther (in-put GbR) wrote:
Yeah. That response is usually when things are not happening properly.
Matt, the ouput hasn't changed, although ist script is executed properly.
Why
do you thing that this output shows a failure?
Not only that but Asterisk and Digium has enabled ALL of us to market
and produce and support a product for businesses that would have no
other alternative but to spend even more money on the big boys or get
smaller less featured phone systems without the benefits of VOIP.
We all succeed in this
EVERYONE PLEASE DON'T FEED THE TROLL!!
That post was done only for the sake of generating responses, and we do
no one any favors by taking the bait.
thx.
B.
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asterisk-users mailing list
To
First let me just say that I am a total newbie when it comes to phones
but I have several years of Linux and it experience. I have been
tasked with offering a competing solution to our current phone
providers based on asterisk...
To show my complete ignorance I am going to try and describe out
Thanks David
But what I was more looking for was storing the configuation file eg
extensions.conf as a database file in MY SQL and then have asterisk load
the table from MYSQL as opposed the text file extensions.conf ?
Is there any benefit in this ?
Thanks all
Barry
Lol, your definition of rich and mine are obviously very different.
$13m I'd call that working capital.
He hasn't sold the company, he hasn't walk away to retire in Anguilla.
Personally I'm very thankful for the work put in by Digium, it allows me
to run a pabx in my home office with
you must add option insecure=very|yes|no in sip.conf, see http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conffor more info
by default incoming calls goes into default context
have you checked if registration has occuredin sipproxy?check debug messages in asterisk console
Tzafrir Cohen wrote:
On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote:
I want to connect two ISDN bri card directly.
Is it necessary to use a cross-cable?
No. A standard (non-crossed) ethernet cable will work, as long as it has
all 8 wires.
I disagree. If the pin layout of the
I'll feed the Troll!!!
Mark deserves this, he has given us, all of us, a way to make and/or
save money. Kudos to him and the staff at Digium. I, for one feel Mark
owes me nothing but I still feel like I owe him and the project much of
my uncompleted work.
Way to GO!! Mark. May the extra cash
Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link -
http://www.oreilly.com/catalog/asterisk/the pdf version can be found here but having one to read is much
Hi there,
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
I am trying to find out the differences between a solution using an external
ATA (like the Sipura SPA-3000) or an internal PCI card (like the Sangoma A200
with 2
Is there any difference between having asterisk as a jabber client or
jabber component ?
Does anyone know what settings need to be set (!) in order to connect as
a component to a wildfire server ?
Julian
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Stephen G wrote:
Hi there,
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
I am trying to find out the differences between a solution using an external
ATA (like the Sipura SPA-3000) or an internal PCI card (like the
Rich,
Thanks for the quick reply and your advice.
My main goal is to build a small, energy efficient, always on server that will
be able to run Asterisk and connect up to the PSTN, with FAX ability. The PCI
card makes the solution cleaner, but it is harder to find small
cases/motherboards
Hi
All
I am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.
I have configured two extensions in extensions.conf
exten = 228,1,Dial
exten = 234,1,Dial
and configured the xlite soft phone. when I am calling from 234 to
228 it is unable to establish
Hi:
First at all:
You SIP phones are right register on sip.conf file?
Cris
From: R.Linga Reddy [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk
You need to tell asterisk what to dial. Check the dial command syntax and probably the sip.conf file.On 8/9/06, R.Linga Reddy
[EMAIL PROTECTED] wrote:HiAllI am new member to asterisk mailing list.
I have complied the asterisk and it is running fine.I have configuredtwo extensions in
Yes it supports PRIOn 8/9/06, Colin MacMillan [EMAIL PROTECTED] wrote:
Brian,What you need are some sources of good information to get you started. Based on what you wrote there is a lot to cover - impossible in an email.Buy and read the book from this link -
Stephen,
In my experience setting up an office PBX, I started with several
SPA-3000's and eventually decided to go with an A200d.
There were two reasons for changing to the A200:
1. Hardware echo canceller. Despite all the configuration settings I
tried, there was always a faint echo with
George Gardiner wrote:
Digium is not being given a whole load of money - the investors will
want a slice of the company and the future profits. That's how VC
funding works.
More like selling your soul to the Devil, actually.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Yes,
I have read this part of the Diva Server Adapters Installation Guide and
I think that it is necessary use a cross cable.
Matteo
Klaus Darilion wrote:
Tzafrir Cohen wrote:
On Wed, Aug 09, 2006 at 12:05:30PM +0200, support_list wrote:
I want to connect two ISDN bri card directly.
Is it
Is anyone familiar with the Telco in Prague?
We have an issue with the connection that will be made from the Telco
demark when we do an IPT installation next week.
-jason
-
Disclaimer:
This e-mail communication and any attachments may contain
*
ESCAUX releases net.PBX Free Edition, a free and Open Source
version of its original ESCAUX net.PBX product.
*
ESCAUX net.PBX is a turnkey Asterisk solution
Im doing some research about how to deploy asterisk in small offices.
So far I have seen the soekris implementation with astlinux and it
sounds good. Please share your comments/ideas for the following
configuration:
Note: Pure PBX only, no routing/firewall functions needed.
Small Office #1
Up to
Hi,
I've been googling all over the place and have read the relevant articles in
the Digium knowledge base. I have tried all the suggestions I found in the K.B.
Spent some time on the asterisk irc, tweaking some parameters as people thereon
thought would be helpful, but to no avail.
I am
Anyone ever hear of a company...that isn't around
anymore named Tri-Link Technologies? Apparently they had some system called a
Vortex system...I have a few of the voip desk phones here and am trying to find
some info on them to see if it is possible to reuse them for anything. Actually
a
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
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Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
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Hi, i'm trying to setup virtuemart with astcc (which is already working ok),
i've seen messages from JP Carballo and he has done that, i would like to
have a little help please.
thanks.
Julio Caceres
_
Visita MSN Latino
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
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Anyone having issues with the message waiting indicator and retrieve
button on SNOM 320's and 360's.
[EMAIL PROTECTED] ast]# asterisk -rx show version
Asterisk 1.2.10 built by root @ myhost on a i686 running Linux on
2006-07-24 23:42:12 UTC
Verbosity is at least 10
Some users get calls and
Dude I am English
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, August 09, 2006 12:56 PM
Subject: Autoreply: [asterisk-users] Tri-Link Technologies?
Attualmente non sono in sede. Per richieste urgenti contattare lo 800
919299
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
___
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Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
___
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Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
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Anyone having issues with the message waiting indicator and retrieve
button on SNOM 320's and 360's.
When the MWI light is not lit on a 360, and the user hits the voicemail
button, the Snom phone dials the extension 'unknown', 'default' or
'asterisk'. If you don't have an unknown etc extension
It's a horrible, horrible autonotice that this person is unavailable. Expect to see lots of these.To contribute to the topic, I also can't find much on this phone ;)Dave
On 8/9/06, Don [EMAIL PROTECTED] wrote:
Dude I am English- Original Message -From: [EMAIL PROTECTED]To:
I am trying to get my Asterisk server to talk to a Panasonic D500 PBX
using an E1 connection. The card for the Panasonic uses MFC/R2 and I
have installed Unicall. Calls from the Asterisk server to the Panasonic
go through without a hitch and I can call any extension I want. The
problem
It says on the bottom for use only with vortex
system...but...I was just hoping to possibly find a way to use it for something
else since the company is long gone and no way to contact them.
- Original Message -
From:
David
Freeman
To: Asterisk Users Mailing List -
I really don't understand the complaint. Fonality gets a $5 mil.
investment for building its own system on top of Asterisk -- no
complaint. But Mark Co. can't get VC for their own business /
enterprise / support architecture?
Everything that we-all is contributing is part of the open source
Hey guys,
I'm currently investigating solutions about High Availability solution,
I've found out about this webpage on voip-info.org:
http://www.voip-info.org/wiki/view/Asterisk+High+Availability+Solutions
But that's cool for the voice and stuff. But what about the recording.
If I don't
I'm running Asterisk 1.2.7.1 using entirely SIP connections, but I have a
problem with DTMF signaling.
In the features.conf, I have set up sequences using * and # followed by a
single digit for transfers etc. But when I then press '*' or '#' during a call,
only each other is passed on. All
Hi folks,
I'm currently trying to get some early audio, i.e. audio without a
connection, to the caller to give some cost-free info while the alerting
phase.
The WIKI's info on the Progress() application says, that just Progress()
before e.g. a Background(soundfile|n) should work but it
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Having spent some time chasing up the importers today, it appears that
there is no real shortage of the cards yet. However they are seen a
legacy card now with people switching to broadband.
MRi claim to have 100-200 cards available, and their
Hi all,
I need to setup 6 phones about 3/4 of a mile from the main box, (can't
do it with VoIP yet because of networking issues), does anyone knows if
the boards can resist such a length for FXS ports.
Right now there is a Dialogic MSI160 working fine.
The actual length in a straight
I had a similar problem with a Siemens, most probably you are
specifying the wrong number of expected ANI digits. Try with mx,0,4
as protocolvariant, that will tell Unicall to expect 0 ANI digits, but
of course, in Asterisk environment you wont be able to get callerid.
Play around incrementing
I echo (pun intended) Rich's response. The Spa3k is ~ok~ but echo has
always been a problem for my home office. The A200D works flawlessly.
I'm looking to set up a home-office PBX/Asterisk lab using a VIA EPIA
motherboard as an always on, low powered solution.
I have seen an A200D in a
I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a
call, I get a prompt to transfer the call. This, of course, interferes
with any IVR system I'm using, as many systems will ask me to enter a
number, then press the # key. Is there any way I can get Asterisk to
ignore this key when
Hi guys,I am fighting to get a Wildcard TE405P working but it always start and put all channels in use. 14 TE4/0/1/14 Clear (In use) 15 TE4/0/1/15 Clear (In use)
16 TE4/0/1/16 HDLCFCS (In use) 17 TE4/0/1/17 Clear (In use)I've tried to downgrade zaptel and asterisk but it didn't solve the
Hi Carlos,I had the same problem and spent a lot of time studying the MFC/R2 protocol but the problem is in the libmfcr2 package version!!Try using the packages in:
http://www.soft-switch.org/downloads/unicall/unicall-0.0.3pre7And not in pre9.Both pre7 and pre9 have
libmfcr2-0.0.3.tar.gz
Yep, there are analogue line boosters for your requirement.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Manrique Feoli
Sent: Wednesday, 9 August 2006 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial
[EMAIL PROTECTED] wrote:
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299
o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
PLONK!
Regards,
Austin.
signature.asc
Austin Denyer wrote:
PLONK!
Regards,
Austin.
and you sent this to a public list? you're a fucking idiot.
-A
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To UNSUBSCRIBE or update options visit:
Hi,
On Wed, 2006-08-09 at 20:50 +0200, Stefan Gofferje wrote:
Hi folks,
I'm currently trying to get some early audio, i.e. audio without a
connection, to the caller to give some cost-free info while the alerting
phase.
Many (if not all) telco does not allow sending inband audio
to the
Austin Denyer wrote:
[EMAIL PROTECTED] wrote:
Attualmente non sono in sede. Per richieste urgenti contattare lo 800 919299 o
inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
PLONK!
Regards,
Manrique Feoli wrote:
Hi all,
I need to setup 6 phones about 3/4 of a mile from the main box, (can't
do it with VoIP yet because of networking issues), does anyone knows if
the boards can resist such a length for FXS ports. Right now there is a
Dialogic MSI160 working fine.
The actual
On 7/25/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
What is your net mask? 255.255.255.0? You can try in sip.conf:
externip=212.xxx.xxx.xxx
localnet=192.168.0.0/255.255.0.0
A bit late answer, but I haven't got around to test it until now.
No, this doesn't work. Maybe it's my Asterisk (1.0.x)?
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patrick wrote:
I'm using Asterisk 1.2.7.1, and if I press the # key when I'm on a
call, I get a prompt to transfer the call. This, of course, interferes
with any IVR system I'm using, as many systems will ask me to enter a
number, then press the #
You can set the transfer feature to be whatever key press you want. If you want to go that route, you can set it up so that # doesn't initiate a transfer, and Asterisk will just pass the # as DTMF to the other side of the call. My transfer is *#. Use
features.conf to set this up.AlexOn 8/9/06,
Andrew D Kirch wrote:
Austin Denyer wrote:
[EMAIL PROTECTED] wrote:
Attualmente non sono in sede. Per richieste urgenti contattare lo
800 919299 o inviare una mail a [EMAIL PROTECTED] oppure a [EMAIL PROTECTED]
Cordiali Saluti
Giuseppe Parlato
Area Network
mailto:[EMAIL PROTECTED]
On 8/9/06, Yaakov Menken [EMAIL PROTECTED] wrote:
I really don't understand the complaint. Fonality gets a $5 mil.
investment for building its own system on top of Asterisk -- no
complaint. But Mark Co. can't get VC for their own business /
enterprise / support architecture?
Everything that
There is no doubt Asterisk is a nice peiece of software...and it does a nice
job on our prepaid/postpaid apps...
But it will need to evolve even more quickly to keep up with what freeswitch
will have within the next few months...unless they plan to stay mainly under
the idea of relatively small
Hello all !
I am looking for a way to use an ata or ipphone mac address as a part of the
sip registration.
I've tried using arp tables, but it only works on local networks not thru
the internet..
I know that vonage uses some kind of mac auth, but I think that this is an
special feature of their
Thanks Rich,
I was expecting that. I was worried because on other type of cards if
you set the phone too far you'll burn the port of the card, mainly
because of the lack of capacity to keep such a long line up.
Now when you say 2 or three ringers, you mean 2 or three ring events or
Thank You Patrick,After some minor problems in some file paths I had success compiling.The only problem was the codec_g726 witch does an illegal call and Asterisk doesn´t come up. But I only use g729 or g723 so I´ve deleted it from the modules directory and asterisk came up.
I´m going to test it
Ron Wellsted [EMAIL PROTECTED] writes:
So while there HFC cards out there, it seems that they are going to get
harder to find.
We got a few of these from Conrad. They are in Germany and I am not
sure if this one is the same as ours. But at EUR 24.95 per card you
cannot loose to much.
CA == Colin Anderson [EMAIL PROTECTED] writes:
Anyone having issues with the message waiting indicator and
retrieve button on SNOM 320's and 360's.
CA When the MWI light is not lit on a 360, and the user hits the
CA voicemail button, the Snom phone dials the extension 'unknown',
CA 'default'
Hi,
same by me, the patch affects app_rxfax, app_txfax and the G.726 codec from
the spandsp. However, it doesn't link it to the libspandsp properly, asterisk
complains: undefined symbol: g726_encode. I added to modules.conf the line
noload = codec_g726.so and asterisk comes up again.
Thanks for
Daniel Botelho P. Moraes wrote:
Hello all !
I am looking for a way to use an ata or ipphone mac address as a part of the
sip registration.
I've tried using arp tables, but it only works on local networks not thru
the internet..
I know that vonage uses some kind of mac auth, but I think that
Don't forget to do lightening and surge protection.
On 8/9/06, Manrique Feoli [EMAIL PROTECTED] wrote:
Thanks Rich,
I was expecting that. I was worried because on other type of cards if
you set the phone too far you'll burn the port of the card, mainly
because of the lack of capacity to keep
You stinking slefish pig, you use asterisk for free and this is what
you say in return.
How does it go (Cleaning out my closet):
http://www.loglar.com/song.php?id=10620 look at the end of verse 3
thats for you.
On 8/9/06, Randall H. [EMAIL PROTECTED] wrote:
If you gave software to Digium then
Thanks Rana,
We got a prompt response from Sangoma and gave them SSH access to
troubleshoot the issue. Modifying relaxdtmf on its own did not help but
if used in combination with rxgain it made all the difference. The
system is working pretty good now with these two modifications:
When Background or Playback is used in a dial plan how many digits are
collected and what variable are they returned in?
I'm trying to do a simple auto attendant and having very little luck
--
One day at a time, one second if that's what it takes
When you use playback no digits are collected. When you use backround
the digits go to an available extension in that context, for example:
[ivrcontext]
exten = s,1,Background(testfile)
exten = _X,1,Noop(user pressed ${EXTEN})
this will gotot Noop for any single digit that is pressed, you will
On 8/9/06, Randall H. [EMAIL PROTECTED] wrote:
If you gave software to Digium then you helped Mark become very rich.
http://abcnews.go.com/Technology/wireStory?id=2290152
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Congrats
Now they can buy Sangoma with cash and still have a couple million left over :)
(remember it's a Canadian company USD$1 = CAN$0.89)
http://tinyurl.com/p4epp
Company cash value of Sangoma USD$10.3 million.
MATT---
On 8/9/06, Michael Graves [EMAIL PROTECTED] wrote:
On 8/9/06, Randall H.
Is anyone else having problems with them? Order placed online 13
days ago. voiplink.com charged my cc for the product 11 days ago.
They can't seem to ship Linksys spa-942 that they claim to have in
stock. Order is still pending on their web site. Calls to them
confirm no shipment but also
Good morning, all,I am in immediate need of configuring an Asterix to act as wake up call system.I need:1. user calls in and at the prompt enters his room number.2. Asterisks then checks DB to ensure that that room number exists3. Asterisk then prompts user to enter time4. user enters time he
I have had the same experience with a Grandstream order from them - 7
days and no product.
They even told me it was shipping Monday, but couldn't produce a
tracking number on Tuesday.
Pretty lame.
On 8/9/06, Tom [EMAIL PROTECTED] wrote:
Is anyone else having problems with them? Order placed
Vic wrote:
I am in immediate need of configuring an Asterix to act as wake up call
system.
Amazing:
http://www.google.com/search?q=asterisk+wake+up+call+site%3Avoip-info.org
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asterisk-users
I have two trunks to the same machine (x.x.x.2), one is type=friend, other is
type=peer. Asterisk seems to choose which trunk to use by the order by which
they are set out in sip.conf.
When a incoming call comes into Asterisk, it always uses the last trunk. My
understanding was that a peer
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