On Mon, October 31, 2005 8:40, KARIM MOUSLI said:
hello evryone
can somone help me get asterisk to work with outgoing calls to a voip
operator
i have tried many stings, but i cant triger the outgoing calls, calls on
the same pbx are working fine
what did i mis out ?
in advance thanks
Sip show peers includes the line:
602/602(Unspecified)D N 0UNKNOWN
However, I can call it? Should not peer means if it is reachable?
bye
Ronald Wiplinger
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Chris Bagnall ha scritto:
lower soft buttons hae labels like Pnbsp;, and apart from the single
This is a old firmware issue, upgrading the phone firmware everything is
working ok with the 7960
Sergio
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This indicates that 602 is a dynamic host. It must therefore register
with the pbx so that the pbx knows where to send data.
In this state it is unregistered so it will be unlikely you can call it.
Regards,
Mark
-Original Message-
From: Ronald Wiplinger [mailto:[EMAIL PROTECTED]
Sent:
On Mon, 2005-10-31 at 16:33 +0800, Ronald Wiplinger wrote:
Sip show peers includes the line:
602/602(Unspecified)D N 0UNKNOWN
However, I can call it? Should not peer means if it is reachable?
I dont quite understand the question, I think there
problem i can't get asterisk to dial to sip provider no matter what provider i
choose
the prefix and telephone format is the main problem and i cant figure it even
thoug i looked at example and diD not work for me
i took exmple on nufone and net2phone configs !
IF I UNDERSTAND THINGS WELL, i
I've asterisk 1.0.7 (debian package) with zaptel 1.2-beta1 (to avoid the
rmmod hangs the server problem already discussed here).
The card is a digium TE410P, configured in this way :
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
span=3,1,0,ccs,hdb3,crc4
bchan=63-77
This is a old firmware issue, upgrading the phone firmware
everything is working ok with the 7960
Sadly, that's the problem at the moment - I can't seem to get hold of new
firmware for love nor money. Even the hunting for firmware on ebay route
yielded zero results when I had a look yesterday.
Chris Bagnall ha scritto:
Sadly, that's the problem at the moment - I can't seem to get hold of new
firmware for love nor money. Even the hunting for firmware on ebay route
yielded zero results when I had a look yesterday.
Buyu the cheapest cisco smartnet contract and you will be able to
Anyone out there knows how the call-pickup works on [EMAIL PROTECTED] I
tried *8 and it did not work. Can a IAXs client also me assigned into a
call-pickup group?
Thanks in advance,
Stephen
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Anyone out there with experience connecting this? The Siemens HiPath
3550 comes with 2 BRI (S0) ports built-in and is configured as a BRI
trunk interface card.
Thanks in advance.
Stephen
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Hi all,
through oh323 i can register to my gatekeeper and make
and receive calls.
My gatekeeper routes the incoming call as well as the
outgoing.
The problem is simply that i can't ear nothing from my
SIP ipPhones. I can ear my voice from a normal
telephone in my SIP phone but no viceversa.
Anyone seen this one so far? Seems to happen in or outgoing, and even
if I just pick up the channel.
09/15 revision works fine, but the 10/31 checkout is doing this
instantly. All with HEAD zaptel and libpri
Oh and another off topic thing.
Sometimes I have a way of forgetting I have asterisk
Hi!
Is there an available implementation of Session Border Control on Asterisk?
Thanks a lot in advance!
Luca
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I am getting the following error when I click on create new ratecard
Fatal error: Cannot redeclare display_minute() (previously declared
in /var/www/asterisk2/htdocs/A2Billing_UI/lib/Misc.php:52)
in /var/www/asterisk2/htdocs/A2Billing_UI/Public/frontoffice_data/CC_var_tarif
fplan.inc on line 27
Hi Stephen, I don't think you can use fritz card
to connect to a Siemens pbx.
You have to use a card that works in NT mode for
exemple a more cheap compatible Bristuff card.
Refer to this page:
http://www.voip-info.org/wiki/view/Asterisk+zaphfc
Bye
- Original Message -
From: Stephen
Does asterisk have a module for generating tones, or a set of prerecorded GSM
tones, like 1100Hz tones et cetera?
/Obelix
This message was sent using IMP, the Internet Messaging Program.
app_milliamp is your friend
Obelix wrote:
Does asterisk have a module for generating tones, or a set of prerecorded GSM
tones, like 1100Hz tones et cetera?
/Obelix
This message was sent using IMP, the Internet Messaging
Thanks, thought that should work but had a type error which have now
corrected. One further question, how can I set up a line so that if 440 is
dialled before a number the 0 is taken out so only 44 is actually used?
Thanks again.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Go to their website and download the most up to date version and then try it
again..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraser
Sent: 31 October 2005 10:43
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] A2Billing
I am
Quoting Erik [EMAIL PROTECTED]:
Where can I download it from? I searched the lists and the web for any reference
to it and there is no mention of it.
Regards
Obelix
app_milliamp is your friend
Obelix wrote:
Does asterisk have a module for generating tones, or a set of prerecorded
GSM
One further question, how can I set up a
line so that if 440 is dialled before a number the 0 is taken
out so only 44 is actually used?
exten = _440.,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:3})
You could probably do it by playing around with different offets as well:
exten =
Obelix wrote:
Quoting Erik [EMAIL PROTECTED]:
Where can I download it from? I searched the lists and the web for any reference
to it and there is no mention of it.
Regards
Obelix
app_milliamp is your friend
Obelix wrote:
Does asterisk have a module for generating tones, or a set of
To bad that prefixes like +220 (Gambia), +230 (Mauritius), +240
(Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290 (Saint Helena),
+350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech
Republic), +500 (Falkland Island), +590 (Guadeloupe), +670 (Timor
Leste), +680 (Palau), +690
Hi folks,
I understand that IAX2 supports public key authentication. Is the
transmission also encrypted or is it possible to encrypt an IAX2 trunk
between 2 *s?
Regards,
Stefan
--
(o_ Stefan Gofferje| SCLT, MCP
//\ Reg'd Linux User #247167 | VCP #2263
V_/_ Heckler
To bad that prefixes like +220 (Gambia), +230 (Mauritius),
+240 (Equatorial Guinea), +250 (Rwanda), +260 (Zambia), +290
(Saint Helena),
+350 (Gibraltar), +370 (Lithuania), +380 (Ukraine), +420 (Czech
Republic), +500 (Falkland Island), +590 (Guadeloupe), +670
(Timor Leste), +680 (Palau),
I believe this is the latest version.
Open_A2Billing_version_Raccoon.tar.gz
as of Oct 30 2005.
On Mon, 31 Oct 2005 11:54:18 -, Sam Tam wrote
Go to their website and download the most up to date version and
then try it again..
-Original Message-
From: [EMAIL PROTECTED]
Hello,
Is there a way to put a voicemail quota to a SIP user? I mean a quota on the
user's mailbox instead
of a particular message of the user like the 'maxmessage' does currently.
Quata can be total file size of message or
total minutes of messages of a mailbox.
Thanks for your help
Hello.
I solved my problem. I got by cvs the last chan_bluetooth.c and it
works.
cvs -d :pserver:[EMAIL PROTECTED]:/home/cvs login
Best regards,
José Luis Gómez
El sáb, 29-10-2005 a las 00:24 +0300, Vlasis Hatzistavrou escribió:
Hello,
We had similar problems with chan_bluetooth and various
Mik,
Your asterisk server is another machine of your GK ? You can start verifying
if the traffic between the machines (related to RTP packets) is ok.
Do you have firewall ?
--
[ ]'s
Daniel Varella de Oliveira
Tecnologia IP Ltda
Tel.: +55 (21)3139-4091 / r. 108
Rio de Janeiro - Brasil
problem i can't get asterisk to dial to sip provider no matter what provider
i choose
the prefix and telephone format is the main problem and i cant figure it even
thoug i looked at example and
diD not work for me
i took exmple on nufone and net2phone configs !
IF I UNDERSTAND
Anyone seen this one so far? Seems to happen in or outgoing, and even
if I just pick up the channel.
Nope. cvs-head from yesterday and the last several days are working
just fine on fc3. What distro are you using?
09/15 revision works fine, but the 10/31 checkout is doing this
instantly.
Hello,
For the past week or
so, I've been looking everywhere for information about disconnect supervision
and have come to the following conclusion: there is a plethora of
information available on RECEIVING disconnect supervision signaling via an FXO
port on analog cards like Voicetronix
Hello!
Lately, I've been keeping a close eye on an Asterisk box by staying logged
into the console for long periods of time. However, it can be very
difficult to know how long a telephone call lasts when this is all you
see:
-- Executing Dial(SIP/SIP105-8e34,
Zap/g2/Number|60|t) in new stack
On Monday 31 October 2005 08:37, David Stude wrote:
The upshot of this is that I'm trying to connect a Norstar MICS system,
which has FXO analog ports, to our new Asterisk system, using FXS ports.
The Norstar only recognizes disconnect supervision and, otherwise, will
not free up the line
On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote:
Is there a way to add timestamps to each line in the console so you know
exactly how long a call took? Or is there another way of telling directly
within the console?
Of course it's possible, but you'll be maintaining the patch
Title: Re: [Asterisk-Users] TDM01B vs. X100P
On the IP Office, try making sure that
fast start is off on the h.323 trunk links. Also, look in Monitor on the IP
Office, see what errors are coming up.
Kind regards,
Chris Clauss
Avaya Certified Expert; Cisco CCDA; Microsoft MCSE
On Mon, Oct 31, 2005 at 09:30:58AM -0500, [EMAIL PROTECTED] wrote:
Hello!
Lately, I've been keeping a close eye on an Asterisk box by staying logged
into the console for long periods of time. However, it can be very
difficult to know how long a telephone call lasts when this is all you
Are u serius?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jonathan k. Creasy
Sent: Friday, October 28, 2005 9:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Geneys
Anyone using the Genesys
On Monday 31 October 2005 05:11, [EMAIL PROTECTED] wrote:
Sometimes I have a way of forgetting I have asterisk running, and do a
module unload. As you can expect, this causes an EIP and kills the
server. The server will then stay stuck at the EIP, but does anyone
know of a way to do an
[EMAIL PROTECTED] wrote on 10/31/2005
08:53:35 AM:
On Monday 31 October 2005 09:30, [EMAIL PROTECTED] wrote:
Is there a way to add timestamps to each line in the console
so you know
exactly how long a call took? Or is there another way of
telling directly
within the console?
Of course
Rich Adamson wrote:
Just update cvs-head again at 7:45pm CST. Seems the issue still exists.
Any thoughts on me opening a bug tracker item on this?
Always a good idea (and cheap too!)
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Hi Andrew,
Thanks for responding. Yes, I noticed that some code *seems* to support
this.
I started with a Voicetronix Openswitch12, which has, even in its driver
code, no support for anything resembling kewlstart. After figuring out that
the MICS seemed to respond to opening the circuit for a
O'reilly had a book out before the docs team wrote theirs.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Leif
You could always just add some
exten =
NUM,PRIO,VERBOSE(LEVEL|${DATETIME} -- THIS EVENT HAPPENED)
type commands in your dialplan to force output of the date
time, and you can even reduce the amount of verbosity to the CLI by using it
liberally to signify events, so you don't have to
Please download the last release (http://areski.net/a2billing/),
I corrected some bugs and this was one of them.
Rgds, Areski
On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
I am getting the following error when I click on create new ratecard
Fatal error: Cannot redeclare display_minute()
So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to voicemail, then Allison
Joining two conferences together over a LAN should be possible, at least
theoretically. I am not sure how the performance would be over a WAN or the
public internet. I am currently working on joining two meetme conferences
together using IAX2 trunking and will post my results after the trial.
I have a beta1 gateway with a 4 port card in PRI mode, the gateway sends
DTMF via sip info packets to another beta1 box. The peer is set to
receive info. What I get is a click sound and a very very short tone.
Sound like to me that I get the first part of the tone before it is
captured and put
Why'd your mailing software fake your From: to be my email address?
On Monday 31 October 2005 09:35, [EMAIL PROTECTED] wrote:
I started with a Voicetronix Openswitch12, which has, even in its driver
code, no support for anything resembling kewlstart. After figuring out
that the MICS seemed to
I added versioning information (version.release) in the application
(ie agi ./a2billing --version).
It will be more easy to know which version, release you downloaded and
check if a new one is available.
# Last release have an ACL user support also advanced filter to
select the cards.
Rgds, A.
Hi,
would I have to go through the entire installation again?
Thanks
John Fraser
On Mon, 31 Oct 2005 09:46:44 -0500, Areski K wrote
Please download the last release (http://areski.net/a2billing/),
I corrected some bugs and this was one of them.
Rgds, Areski
On 10/31/05, John Fraser
Brent Torrenga wrote:
So within the /var/lib/sounds/voicemail structure are the greeting files
recorded by the person at each extension (busy.wav, greet.wav). If I need to
get rid of the customized recording, it is trivial to simply delete both of
those files. At that point, if a call goes to
Is there anyone who knows where to find rugged IP phones?
Rugged in this case means that need to be installed on a ship's deck, so it
must be water resistant, anyway compliant with IP 65 specification
(protected against dust and jets of water).
Regards
++
|
Hey everybody.
I have an Adit 600 that I'm not able to get working properly with
Groundstart. The Adit (BootCode 1.23), 1 TDM (Version 6.1.2) and 1 FXO
card (Version 1.12).
The Adit is setup: ESF,B8ZS.
1st port is set as signal gs type voice.
2-8 is setup signal ls type voice.
The FXO
Does anyone have an example of a lucent
TNT h323 config to work with asterisk ?
I'd like to use sip but it's not supported in the
TAOS we have, if anyone has TAOS 10.x or later
that would be awsome as well, we have the examples
for a sip config.
thx
- Armand
Hi,
Is it possible to dynamically add contexts to the dial plan in any way?
Extensions can be added from the console and therefore also from MAPI but
their doesn't appear to be anyway to add a new context apart from reloading
the configuration files.
The reason I ask is my dialplan is getting
Hi
I have inserted the lines you suggested but Asterisk keeps the 0 when
dialling with all alternatives. Also, I am unsure what the phrase
${EXTEN::2}${EXTEN:3} does? Could you explain this abit?
My extensions.conf is:
[default]
exten = _0[1-9].,1,Dial(IAX2/[EMAIL PROTECTED]/44${EXTEN:1})
No you won't !!! You just have to copy again the AGI Web Interface
You don't have to change anything in your configuration files or
in your Database. It should be really fast to do!
FYI - areski.net/a2billing
## (ADMIN UI V1.0.3 - CUST UI V1.0.1 - AGI V1.0.1 - DB V1.0.0)
this will keep you
I don't receveid e-mail with voicemail.
When I dial 2 with telephone, Asterisk record message
but don't send a e-mail at the mailbox. Why?
I have configuration this file.
In the voicemail.conf
[general]
attach=yes
format=wav
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
Olle E. Johansson [EMAIL PROTECTED] wrote:
Adam Moffett wrote:
does anyone know when 1.2 will no longer be beta?
The quick answer is: When it's ready for release.
Open Source software doesn't really follow a set agenda.
I don't think that is an accurate statement. It is certainly true of
Why Asterisk show this message?
WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device
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http://it.messenger.yahoo.com
did u set the mailserver?
Bruno.
Fabio Montemaggiore wrote:
I don't receveid e-mail with voicemail.
When I dial 2 with telephone, Asterisk record message
but don't send a e-mail at the mailbox. Why?
I have configuration this file.
In the voicemail.conf
[general]
attach=yes
format=wav
I have inserted the lines you suggested but Asterisk keeps
the 0 when dialling with all alternatives. Also, I am unsure
what the phrase ${EXTEN::2}${EXTEN:3} does? Could you
explain this abit?
The syntax is {EXTEN:initial offset:length}
So EXTEN:3 chops off the first three digits and
In article [EMAIL PROTECTED],
James Steven [EMAIL PROTECTED] wrote:
Hi
I have inserted the lines you suggested but Asterisk keeps the 0 when
dialling with all alternatives. Also, I am unsure what the phrase
${EXTEN::2}${EXTEN:3} does? Could you explain this abit?
My extensions.conf is:
I don't set the mailserver.
What can I do?
I use Debian
Thanks
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http://mail.yahoo.it
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In article [EMAIL PROTECTED],
Chris Bagnall [EMAIL PROTECTED] wrote:
exten = _44N.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
(N should be the same as [1-9] I think)
N is [2-9], Z is [1-9]
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL
cannot overwrite directory `/var/www/asterisk2/htdocs/./A2Billing_UI
cannot remove `/var/www/asterisk2/htdocs/./A2Billing_UI': Is a directory
Ill just start from scratch.
could you put a version number and date on the web page please.
Might save me and others some trouble
On Mon, 31 Oct 2005
Hi,
I seek solution for hotel management and billing solution. but I do not
know which to choose between Astbill or Asterbill ? if you have council.
Thx
David
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Asterisk-Users mailing
Darren,
An example how to call that callback.agi script? The script iself does
not have usage info.
Thanks
Anthony
Darren Wiebe wrote:
Hello. You should not need any special hardware for callback. You will
(obviously) need card to connect your box to the pstn. Do you have
something
Yes, or this for example:
[macro-rhangup]
exten = s,1,NoOp(DIALSTATUS=${DIALSTATUS})
exten = s,n,NoOp(TIME=${DATETIME})
exten = s,n,Hangup
I also output the date and time prior to dialing out.
MARK.
Sherwood McGowan wrote:
You could always just add some
exten =
Hi,
Thanks for the clarification. I had seen that the two options
existed, but the docs for the dial() command didn't state the
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got
I have a codeless language module for BBEdit, if anyone's interested
- it's not complete yet (I'm adding to it as I go along), but I will
post it to the wiki, if I could just figure out where it should
go...
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North
you should use an analog made for ships with a ATA.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: Monday, October 31, 2005 7:13 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Rugged VoIP phones for use with
Hello,
I am using Cisco 7940/7960 phones and can not get the calling name to
display on incoming calls. The names and numbers do display in the
Missed Calls and Received Calls menus, but not on incoming. The caller
id number displays fine on incoming, just not the name. Anybody know
what might
Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.
Thanks.
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In his Outgoing context, should it not be 9|1NXXNXX, to strip the 9 from
being sent to the provider?
Kevin
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: October 31, 2005 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Is there a resolution to this problem. It was posted a few weeks back.
But just chiming in again to see if someone has had any luck:
Problem:
Incoming call to a Sipura 2000, 1000, 3000 ATA.
I use the SetCallerID(name)=blah blah blah
SetCAllerId(number)=1234567890
However, On the handset, in
tengo un asterisk, alguien conoce algun proveedor que brinde el
sistema de linkar mi asterisk a su servicio para tener tarifa plana a
eeuu.
para llamar por 4 conexiones al miamo tiempo desde mi asterisk?
me parece haber visto que se configuraba con una troncal iax2
2005/10/31, [EMAIL
On 10/31/05, John Fraser [EMAIL PROTECTED] wrote:
cannot overwrite directory `/var/www/asterisk2/htdocs/./A2Billing_UI
cannot remove `/var/www/asterisk2/htdocs/./A2Billing_UI': Is a directory
:-/
Try with cp -rf
Ill just start from scratch.
could you put a version number and date on the web
Is there a command line for discovery of
Asterisk and Zaptel Versions?
Bart
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
On 10/31/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.
This question was answered *yesterday* by Mr. Olle E. Johansson:
And I
[EMAIL PROTECTED] wrote:
Does anyone know where the official release of Asterisk 1.2 is? Do we have
a time-frame of when this version will be released and how much longer it
will be in BETA.
'Where' it is? It's in the future :-)
There is no time frame, as has already been discussed on this
Hello everyone,
We are experimenting a really bad sound quality with a Digium card and the
technical support from Digium found out that we have a motherboard
incompatible with the card.
If that can help anyone, here are 2 motherboards that we have tested with
very bad zttest results :
Intel
CVS HEAD as of two days ago.
We have 50 agents (All SIP, with inbound/outbound via ISDN32 using a
TE405P with revision 2 firmware), logging in via agentcallback. At the
start of every day I restart * (service restart) At the end of today
(and most other days) we have the following problems:
Hi,
I wonder if anyone can help. I've built an asterisk instance against
the latest 1.1 CVS version. Can anyone help me out with a makefile for
rx_fax?
I'm following:
http://soft-switch.org/installing-spandsp.html
I have built spandsp OK, but get errors when I'm applying the patch.
Rich Adamson wrote:
Assuming you are using the TDM card, there is no code in asterisk to detect
whether a pstn line is connected/disconnected, nor does it listen for dialtone
before dialing.
And for some reason this isn't considered a SEVERE defect?
If the battery on the line
On 10/31/05 23:51 Fabio Montemaggiore said the following:
Why Asterisk show this message?
WARNING[14792]: chan_iax2.c:9355 load_module: Unable
to open IAX timing interface: No such device
it's just a warning. without a timing device, you couldnt use IAX2
trunking, which would greatly
There is the -T option when running the CLI but I think it only works in 1.2
Regards
Jorge
El Lun 31 Oct 2005 12:30, [EMAIL PROTECTED] escribió:
Hello!
Lately, I've been keeping a close eye on an Asterisk box by staying logged
into the console for long periods of time. However, it
On 10/31/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Is there a way to add timestamps to each line in the console so you know
exactly how long a call took? Or is there another way of telling directly
within the console?
I must say its something I would really like to see on the console
Yes, using and analog with ATA is an option, but one of the requirements is
to avoid eletric power cabling, and there is an explicit request for Power
Over Ethernet phones (which adds another not-so-common feature)... so a
native VoIP phone would be welcome.
Francesco Pellegrini
Hi;
what's card do u use 5 v or 3.3 v?
u can find motherboard in :
http://www.digium.com/index.php?menu=compatibility
http://64.233.183.104/search?q=cache:UqnvELBs-AUJ:www.voip-info.org/wiki/view/Asterisk%2Bhardware+motherboard+for+digium+cardhl=fr
I hope that can help u
2005/10/31, Ken
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
Yeah, show versions in the CLI will give you the version of your asterisk build
Also you can do the following in the CLI:
show version files filename
where filename is a valid file name.
As always in Linux you can press TAB to get a list of available
commands in the CLI, for example you can type:
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Yes, this is
Scott schrieb:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Do it in the dialplan by branching
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
Scott wrote:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Is this possible?
Thanks.
Scott.
Scott wrote:
Currently I have a queue that has dynamic members of which I would
like to set a schedule for. From say 8am to 5pm the queue would ring
the phones of queue members but after 5pm the caller get's VM.
Of course it is... don't send the call into the queue if the call
arrives
Excellent guys. This has to be the quickest responce I have ever had
on this list!
Scott.
On 10/31/05, Peer Oliver Schmidt [EMAIL PROTECTED] wrote:
Scott schrieb:
Is it possible to schedule dymanic queues?
Currently I have a queue that has dynamic members of which I would
like to set a
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