Make it
Goto(s-${DIALSTATUS})
cheerz
- Ben.
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS})
On Tue, Feb 06, 2007 at 11:58:01PM -0800, Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts
in the s extension. Goto() is used in examples. Is the prefix s-
mandatory? Is it related to the original extension s? (Apparently
Goto(${DIALSTATUS}) won't
Hi there,
Is there a way to program asterisk to dial an extension Monday to Friday
at a specific time and then read a specific string? eg: Kids, go to
the bus stop now, you're about to miss the bus!
Many thanks,
Pierre
___
--Bandwidth and
On Tue, 6 Feb 2007, Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in
the s extension. Goto() is used in examples. Is the prefix s- mandatory?
Is it related to the original extension s? (Apparently Goto(${DIALSTATUS})
won't work for me.)
s- is
On one installation (* 1.2.13) Pickup doesn't work. This is what I have in
extensions.conf
exten = _**2X,1,Pickup(${EXTEN:2}8${EXTEN:3}tuevents)
exten = _**2X,n,Hangup
This is what I get on CLI
-- Executing NoOp(mISDN/3-1, incoming-beronet 80 - dolazni poziv s broja
270248) in new stack
On 7 Feb 2007, at 03:59, Jim Duda wrote:
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax.
I have the firewall configured to prioritorize port 4569 for IAX2.
I have the
every few days my ADSL connection gets dropped for a few seconds. When
it does I find my SIP connection to one of my providers does not timeout
and retry. Does the following give some clues?
Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.
(note this is the debian etch/testing
At 05.23 07/02/2007, you wrote:
Yuan LIU wrote:
After reading through several recent threads, I started to wonder
why the Cisco document (and other VoIP documents) appears to
present this issue as VoIP gateway specific. Don't (plain old)
PBX' face the same issue if they use analogue
Hi,
Am 07.02.2007 um 09:53 schrieb Pierre du Plessis:
Is there a way to program asterisk to dial an extension Monday to
Friday at a specific time and then read a specific string? eg:
Kids, go to the bus stop now, you're about to miss the bus!
Write a cronjob which creates a call file.
Hi. I was using asterisk 1.2 on a box with sip phones attached and a
long distance T1 line as the phone provider. We did a successful test
of *1 allowing one-touch recording as set in the features.conf.
Because of deadlock issues I decided to try 1.4 (latest svn as of
yesterday) and the deadlock
Hello,
A few weeks ago I enabled the dnsmgr. A few days ago I noticed we
could not reach any IAX2 peers in the USA. I did everything I could
think of including a full reboot to no avail. Re-commenting the enable
in dnsmgr.conf and restarting asterisk made things work again.
Have there been
exten =h,2,System(cp /etc/asterisk/callback.info
/var/spool/asterisk/outgoing)
You could run a script instead of the cp command in system and add the
wait in that.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
I changed from using a recent asterisk system standalone to a Trixbox
install and now I get clicks and minor dropouts on the voicemail
prompts. System load is non-existant on this machine, interrupts
*appear* to be fine, and as near as I can tell the glitch is at the same
point in the prompt
From: Yuan LIU
Sent: Tuesday, February 06, 2007 8:11 PM
After reading through several recent threads, I started to wonder why
the
Cisco document (and other VoIP documents) appears to present this issue
as
VoIP gateway specific. Don't (plain old) PBX' face the same issue if
they
use
Many SIP phones can use the SUBSCRIBE/NOTIFY mechanism of RFC-3265 to
subscribe to hints in Asterisk. This can be used to show e.g. parking
bay and/or agent status.
Have a look at
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions .
-Original Message-
From: [EMAIL
I am experiencing audio ticks when doing calls from SIP or console to
ISDN. Calls. Everything appears fine when doing ISDN-ISDN or
SIP-SIP. Console calls results in 5-8 ticks a second, SIP calls are
dependent on buffer size - 16ms are 1 tick a second, 8ms are 2-3 ticks
a second.
I recently moved
Is there an Asterisk command, app, AGI (or other) that can be called
with a phone# (or list) that will lookup somewhere definitive and report
whether the phone# is registered to a mobile phone or not? How about
other data, like its home city/district etc?
--
(C) Matthew Rubenstein
Take a look at smartCID (at www.generationd.com) Does a reverse lookup for
name/location/etc. Based on phone number.
MD
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Rubenstein
Sent: Wednesday, February 07, 2007 8:30 AM
To: Asterisk-Users
Hello all,
I want to use asterisk as protocol converter, H323 to SIP. I am using
Asterisk 1.2.14 with chan_h323 and the free version of g729.
When calling from SIP to H323 everything is fine. But when calling from H323
to SIP, the phone using SIP doesn't hear the other party.
The phones and
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I worked with Cisco and HP and they should do what you are looking for.
I even worked with cheap unmanaged switches ~20 Euro and they work with
VoIP.
Do you know for switch that can tell me that on port 7 there are two active SIP
calls.
Hi everyone!
I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and
I'm having some issues with the Chanspy application. All the agents are
on SIP channels with g711 and all the communications are inside a LAN.
When I'm spying a SIP channel, the audio from one of the ends
Christoph Fürstaller schrieb:
Can someone explain what the parameters pridialplan and prilocaldialplan
are? What do they do and do I need them?
I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx.
The pbx technican complains about the format of the nr asterisk sends.
Look at here
http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Johann
Steinwendtner
Inviato: mercoledì 7 febbraio 2007 14.56
A: Asterisk Users Mailing List - Non-Commercial
Stefan Wintermeyer wrote:
Hi,
Am 07.02.2007 um 09:53 schrieb Pierre du Plessis:
Is there a way to program asterisk to dial an extension Monday to
Friday at a specific time and then read a specific string? eg: Kids,
go to the bus stop now, you're about to miss the bus!
Write a cronjob
Hi all,
I'm new posting here, though not to perusing. I'm having an issue
with attended transfer and was wondering if anyone had heard of the
problem/had any suggestions... Apologies in advance if this post is
excessively newb-oid.
- An incoming call C is passed to A, a POTS telephone
Hello there all.
i have an agi-bin python script that calls out when a file is dropped into
the /var/spool/outgoing
the script seems to work, and the call is placed, but the script runs
without knowing when the phone is picked up.
i mean, the call is made, and the script begins to run. So by the
I'm still not seeing chan_zap in menu option three.
I copied the source directories from /root/downloads/asterisk (where I had
put them) to /usr/src/ and then did what you suggested below and I got the
same result.
I'm going to try make uninstalling all the packages deleted all source
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad
(gigaset.siemens.com).
C450IP costs less than 100 USD (in Italy at least), S450 is slightly
more expensive.
I have Siemens C450 IP for two days and it seams weary good.
I'm
the ./configure thing requires the sources of zaptel, not asterisk.
Are you sure they're passing the zaptel sources?
Well... i'm out of ideas. If it doesn't work you might want to re-post
your thread (specifically say you don't see chan_zap in make menuconfig)
and start with a new message
Hello,
I've discovered that in Italy ISDN lines can be
programmed to generate a billing pulse every n
seconds (it dipends from the pricebook). The pulse has these figures:
frequency
12 kHz ± 1%
level
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the prefix
s- mandatory? Is it related to the original extension s? (Apparently
Goto(${DIALSTATUS}) won't work for me.)
Goto(${DIALSTATUS}) won't
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need the
answer supervision to trigger your own billing system.
Jorge Mendoza
Stefano Corsi wrote:
Hello,
I've discovered that in Italy ISDN lines can be programmed to generate
a
At 16.22 07/02/2007, you wrote:
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need
the answer supervision to trigger your own billing system.
Yes, it's strange. But I find no mention on answer supervision in the
NT1Plus manual
'export MYIP' in the startup script for Asterisk.
Larry Alkoff wrote:
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll
Jim,
I too am a Teliax user. Talk to their technical support. IAX2 is NOT
preferred. They'll tell you to use SIP.
Jim Duda wrote:
Thanks for the reply Lacy.
Yes, I know that I am using IAX2 and not SIP for my connection to
teliax. IAX2 is the preferred protocol for connection to teliax.
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should have used other drivers.
The installation procedure looked difficult so I decided to get one from
Digium - it's not that expensive, my time is much more expensive.
Made the
If you are using the add on console then yes you can control it, but if you are
just using the phone then it will be in alphabetical order and not in the order
that you want. (I had this issue a month ago and as of then there was no fix
for this).
- Original Message -
From: Bryan
(Sorry for top-posting)
I'm making good progress. However, so as not to clutter the list I will
post my solution on the wiki in the next few days. I'll send out the
link as soon as I've got something substantial for you to review.
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi
I set up call back functionally thru AMI (local channel).
The two calls are bridged and the call is established.
But when I hang up the local channel (the first extension that rang), the
other leg of the call *is not released*
Time events:
0) Socket communication(AMI)
1)extensionA
Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should have used other drivers.
The installation procedure looked difficult so I decided to get one from
Digium - it's not that expensive, my time is much more
On Wed, Feb 07, 2007 at 04:46:46PM +0200, Cosmin Prund wrote:
the ./configure thing requires the sources of zaptel,
Actually, it requires zaptel.h in the pointed place, or in the default
place (as installed by the install target). Note that zaptel = 1.2
installs zaptel.h to /usr/include/linux,
First, I didn't realize I hijacked another thread! Please accept my
apologies.
Now the problem:
Asterisk isn't compiling chan_zap. chan_zap also doesn't appear in the list
of channels when you make menuconfig
I have read all the replies and specifically Cosmin's and Tzafrir's emails.
zaptel.h
So simple... I'm doing that right now, I've sent them an email.
I didn't find that email address on Digium's support page...
Thanks.
Bruce Ferrell wrote:
Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should
I had a typo in my last email. I meant --with-zaptel where I wrote
--with-zap.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by
Hi,
have a look at: http://www.aussievoip.com/wiki/index.php?page=freePBX-Centos
This is based on Centos, but there is not a great difference between this
and Fedora.
It runs through all the requirements and installation for Zaptel and
Asterisk in addition to the FreePBX web based config
I captured the output of ./configure and found the following lines:
lines snipped
checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes
lines snipped
checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes
lines snipped
So it seems
All digital lines (BRI or PRI) provides answer and release supervision.
The drivers will send to * this information, and this information will
be registered into the CDR automatically. You only need setup your
billing system.
As said before you do not need to intercept the billing pulse.
Jorge
On 7 Feb 2007, at 15:54, Cosmin Prund wrote:
Hello: I got into a trap. As far as I know I do not need to pay any
royalties to use G.729b in Romania, so I should have used other
drivers. The installation procedure looked difficult so I decided
to get one from Digium - it's not that
From:"Trevor G. Hammonds" [EMAIL PROTECTED] From: Yuan LIU Sent: Tuesday, February 06, 2007 8:11 PM After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific.Don't (plain old)
Hello all,
I want to use asterisk as protocol converter, H323 to SIP. I am using
Asterisk 1.2.14 with chan_h323 and the free version of g729.
When calling from SIP to H323 everything is fine. But when calling from H323
to SIP, the phone using SIP doesn't hear the other party.
The phones and
From:"Eric \"ManxPower\" Wieling" [EMAIL PROTECTED]Yuan LIU wrote:In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in the s extension.Goto() is used in examples.Is the prefix "s-" mandatory? Is it related to the original extension "s"? (Apparently Goto(${DIALSTATUS}) won't
same for me, however today I started receiving the same amount as usual
On 2/6/07, C F [EMAIL PROTECTED] wrote:
Since Monday I didn't see much traffic.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
C F wrote:
Since Monday I didn't see much traffic.
gmail is having some sort of problem. I haven't gotten hardly any
messages from any of the digium lists in my gmail account.
___
--Bandwidth and Colocation provided by Easynews.com --
From:Jorge Mendoza [EMAIL PROTECTED]Funny that a digital line have a analogue pulse.Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system.Jorge MendozaStefano Corsi wrote:Hello,I've discovered that in Italy ISDN lines can be
In most cases, if I follow these steps, I get a working asterisk with
zaptel:
in asterisk, 1.4 and trunk:
make distclean
rm /usr/lib/asterisk/modules/*
then, get the zaptel source that corresponds to your version of
asterisk.
configure, make, make install it as root. If you try to use a 1.4
Greetings list,
We have an issue with call quality at 2 sites where the users (4 Elmeg
IP290s at one site, 2 SPA942s at the other) do not have an asterisk box
on-site. Each site has an 8mb down/448k up ADSL connection and the phones
connect via SIP to an asterisk box in a datacentre using g729.
From:"Robert DeVries" [EMAIL PROTECTED]
I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context:[callback]exten= 501,1,Congestion() exten= 501,2,Hangup()
exten =h,1,System(cp
please send me more info
thanks!
Tim Panton [EMAIL PROTECTED] wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our lightweight java softphone (Corraleta
SDK) - it can be
Has anyone got any hints of how to setup the OpenSuSE Firewall2 with VOIP
friendly traffic shaping? The only bit I see is in the config file regarding
how to setup a simple HTB. I come from Shorewall, and am finding this
firewall to be different. Any help is appreciated.
Actually, either
(I apologize if this is a dupe, but I never saw my first copy)
I captured the output of ./configure and found the following lines:
lines snipped
checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes
lines snipped
checking for
David Ruggles wrote:
I captured the output of ./configure and found the following lines:
lines snipped
checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes
lines snipped
checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes
Hi
Do you know if the SIP protocol is compatible with semi-private calls.
I can contruct a private call by putting the SIP Privacy header to id and
then sending the call to my SIP-Pri box and it works
This tell my Pri provider that the call is private.
How can I tell my Pri provider that the
David Ruggles wrote:
I captured the output of ./configure and found the following lines:
lines snipped
checking zaptel/tonezone.h usability... yes
checking zaptel/tonezone.h presence... yes
checking for zaptel/tonezone.h... yes
lines snipped
checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes
(Previous reply got garbled in Hotmail)
From:"Trevor G. Hammonds" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 04:49:08 -0800 From: Yuan LIU Sent: Tuesday, February 06, 2007 8:11 PM After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents)
From:"David Ruggles" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 12:15:37 -0500I captured the output of ./configure and found the following lines:lines snippedchecking zaptel/tonezone.h usability... yeschecking zaptel/tonezone.h presence... yeschecking for zaptel/tonezone.h... yeslines snippedchecking
hi!
anyone please recommend/guide me of purchasing a resonably high performance
server system regarding processor(s) motherboard (+ other compulsary
peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be
more helping
I've to use Digium TE411p Quad E1 card
signalling on
Tim,
What sort of 'poor' quality are we talking about - when folks complain what
words do they use?
On the other end, folks complain that the voice drops out. Words are lost.
It's very frustrating to communicate.
Which codec(s) are you using?
ULAW
How many channels do you want to use
Yes, I had seen something in various posts about using SIP instead of IAX2. I
have been switching back and forth
between IAX2 and SIP, however, I haven't seen any noticeable difference. I
will try a switch back to SIP again and see
how that goes.
Jim
James Fromm [EMAIL PROTECTED] wrote in
I saw the same thing, but got a huge flood of messages today. A Gmail issue
perhaps?
Alex
On 2/6/07, C F [EMAIL PROTECTED] wrote:
Since Monday I didn't see much traffic.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Hi all,
Is anyone aware of any progress on this bug?
http://bugs.digium.com/view.php?id=8763
Not only is the channel randomly disappearing during idle periods, it vanishes
during a call as well. No indications in dmesg, syslog, asterisk or anything.
Only cure is to rmmod and modprobe again.
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back from
list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I
thought it was Hotmail, so I saved one outgoing mail and checked
Found my answer for those who would like to know:
Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg
GPP A: urtopsecretultrasecureaesencryptionkey
GPP B: OddBallDirectory123098
Hope that helps someone!
Curt
-Original Message-
From: Curt Shaffer [mailto:[EMAIL PROTECTED]
Sent:
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back from
list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I
thought it was Hotmail, so I saved one outgoing mail and checked
Yuan LIU wrote:
From: /Eric \ManxPower\ Wieling [EMAIL PROTECTED]/
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently
From: Lacy Moore [EMAIL PROTECTED]
Date: Wed, 07 Feb 2007 12:10:01 -0600
C F wrote:
Since Monday I didn't see much traffic.
gmail is having some sort of problem. I haven't gotten hardly any
messages from any of the digium lists in my gmail account.
It's the list, not gmail. Check dates
Hi
Billing Pulses only apply to analogue lines. You need special hardware in
the PBX interface to detect them and pass them on to the Billing software.
To my knowlege there is no Asterisk compatible hardware that does this.
George
- Original Message -
From: Stefano Corsi [EMAIL
Here's an interesting issue we're facing...
We would like users to be able to use softphones from home/work and to
use their same extensions they do at work.
The first step of getting the phones to log in as their same extensions
as work is easy and works. However, on the database side, once the
Chris Bagnall wrote:
Greetings list,
We have an issue with call quality at 2 sites where the users (4 Elmeg
IP290s at one site, 2 SPA942s at the other) do not have an asterisk box
on-site. Each site has an 8mb down/448k up ADSL connection and the phones
connect via SIP to an asterisk box in a
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I
have confirmed this on multiple phones. When the called person answers
and tries to transfer the call to another extension, the call
successfully
(Hotmail garbled reply again)
From:"David Ruggles" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 12:15:37 -0500I captured the output of ./configure and found the following lines:lines snippedchecking zaptel/tonezone.h usability... yeschecking zaptel/tonezone.h presence... yeschecking for
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
From: Jorge Mendoza [EMAIL PROTECTED]
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need
the answer supervision to trigger your own billing system.
Jorge Mendoza
Stefano
On 10:43, Wed 07 Feb 07, chester c young wrote:
please send me more info
thanks!
Tim Panton [EMAIL PROTECTED] wrote:
On 5 Feb 2007, at 21:46, chester c young wrote:
Need to deploy between 50 to 300 lightweight Linux - only browser
and softphone.
You might want to consider our
I've been trying to snip message to keep them from getting too large, maybe
I over did it. :)
chan_zap.c is in /usr/src/asterisk-1.4.0/channels
But doesn't show up in the list of channels in make menuconfig
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
The advertised datarate (8mb/448k) are the speeds at which the circuit
between the customer and the central office is clocked and has no
relationship with *effective* throughput. At the central office are
*shared* facilities than connects each DSL connection with the network,
and over
Menuselect-tree does have a member entry for chan_zap. I has two depend
subnodes and one use subnode.
The depends are: zaptel and tonezone
The use is: pri
(I've installed libpri also)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 [EMAIL
that I have! :)
Have a single X100P in the system and ztcfg configures the board no problem.
zttool confirms the board is there and shows RED when the phone line is
removed and OK when the phone line is plugged in.
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
Which codec do you plan to use?
Henk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of umar tarar
Sent: woensdag 7 februari 2007 20:05
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CPU motherboard for 100+ simultaneouse calls
On Wed, Feb 07, 2007 at 11:45:30AM -0800, Yuan Liu said:
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back
from list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html)
I thought
Is there any sort of friendly interface installed on that box?
[]'s
MM
-Original Message-
From: José Pablo Fernández [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Cc:
Sent: Tue, 6 Feb 2007 17:32:59 -0300
Delivered: Tue, 06 Feb 2007 16:44:11
On Wed, 2007-02-07 at 21:00 +0100, George Camilleri wrote:
Hi
Billing Pulses only apply to analogue lines. You need special hardware in
the PBX interface to detect them and pass them on to the Billing software.
To my knowlege there is no Asterisk compatible hardware that does this.
ISDN
From: David Boyd [EMAIL PROTECTED]
Date: Wed, 07 Feb 2007 15:24:04 -0500
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
From: Jorge Mendoza [EMAIL PROTECTED]
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need
the answer
Hi all
I'm looking for a softphone that works well under terminal services
environment,
we need to set up 24 to 32 phones for a call center,
also, does any one knows if it will actually work fine under load?
___
--Bandwidth and Colocation
Hi Michiel,
Yes it's a commercial app; all the info you need is on the wiki
including pricing and installation guide.
http://www.voip-info.org/wiki/view/Mexuar
Feel free to send me an email if you are the USA with any questions or
Tim if you are in the UK (talk about round the world support -
Asterisk is getting red alarms on my T1, sometimes once or twice a
day, but today it happened 5 times. Even once is too many. Every
call in progress is dropped. Please help! What do I need to do?
What can I try? I've googled and searched this list and can't find
anything. Here's an example
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote:
I have discovered an issue on my system after upgrading from 1.2.13 to
1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone.
I have confirmed this on multiple phones. When the called person
answers and
I just tried what you suggested - it executes the sleep for 10 seconds, then
skips down to the hangup, without copying the call file to begin the
callback.
However, I then broke the system command into two lines like this:
exten =h,1,System(sleep 10)
exten =h,2,System(cp
Someone has worked with any test to speech software with aceptable
quality in spanish? Probably in english the text to speech quality
will be better.
Witch test to speech software gave you the best results in spanish?
Hi Andres,
Check www.loquendo.com out... They have a nice web front
Eric said:
This should be a FAQ. Set the RTP packet size on the SPAs to .2
instead of .3
Thanks for the suggestion. I've logged into the offending devices and set
both to .2. I'll see how it goes for 48 hours or so.
I've looked at the Elmeg ip290's and they are set to 20ms from factory, so I
In a dialplan, after i set an autohangup (with AGI) , how could i send a
sound (stream a sound ) into an open channel at X seconds before the
autohangup time get to 0 for that channel?
(Like public phones, that gives u a 'beep!!!' before ur time runs out, just
like that...)
Check the L
The first step of getting the phones to log in as their same
extensions as work is easy and works.
By definition, I guess that automatically logs out their office phones?
Has anyone tried anything like this? I would like the phones to
regrab their spot once the softphone is logged out.
1 - 100 of 116 matches
Mail list logo
