Well, it applies to many phones, such as the Cisco and Polycoms,
among others, but generally, there is a way to define a dialplan that
changes the amount of time you have to wait for the phone to assume
that you are done dialing. (ie: if it sees 10 digits, wait one
second, and if it sees
: Thursday, August 11, 2005 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
systemtoreplace an old PBX but using existing phone
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first
Subject: Re: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone
The difficulty is making the phone dial quickly when you dial a three
or four digit extension number, yet not having it dial so quickly
that it screws up a user who dials
Tom Rymes wrote:
On Aug 11, 2005, at 11:49 AM, Sean Rima wrote:
Tom Rymes wrote:
On Aug 11, 2005, at 10:35 AM, Sean Rima wrote:
Andrew Kohlsmith wrote:
On Thursday 11 August 2005 09:31, Sean Rima wrote:
They are standard phones but I also want them to have all the
features
that
Jonathan k. Creasy wrote:
YeahI think that every install I have done the first thing that
happens is why is there a delay before the call connects? and the
answer is you have to hit dial or wait 10 seconds.
What all phones does that apply to? I'm fairly certain it applies to the
Rima
Sent: Thursday, August 11, 2005 10:50 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
system toreplace an old PBX but using existing phone
Tom Rymes wrote:
On Aug 11, 2005, at 10:35 AM, Sean Rima wrote
Steve Gladden wrote:
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
The idea would be to get a network
Generally speaking one works against one's own best interests when he
reminds the group that he has been posting on a topic repeatedly without
anyone answering.
Yes agreed,
In this case my only intention was to acknowledge the fact that I
realized I was asking a 3rd time and hopefully not
This function is based on a non-standardized extension to SIP made
by Broadsoft. I have all the specs and are looking into this. Don't
expect anything to happen quickly though, I have to complete another
large SIP project first (SIP Transfers) and then start looking into
this. It requires quite a
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Olle E. Johansson wrote:
This function is based on a non-standardized extension to SIP made
by Broadsoft. I have all the specs and are looking into
Steve Blair wrote:
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Do you know which standard they base this on?
/O
___
Asterisk-Users mailing list
Olle E. Johansson wrote:
Steve Blair wrote:
Features like bridged line appearance are expected to be available in
release
8 of Cisco's SIP image. I do not have an ECD for this release.
Do you know which standard they base this on?
Great question ;-) No I do not for obvious
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several sipura-841 SIP phones for
Generally speaking one works against one's own best interests when he
reminds the group that he has been posting on a topic repeatedly without
anyone answering.
What you are asking for is not reasonable; it's not the way Asterisk
works, and there is in my mind (and I'll bet in the minds of
Still looking for some direction with this subject:
I think the term is called multi-line appearance
Is this something that is directly supported in Asterisk?
I can't seem to find any information on it or how to actually use it
This is where you have several sipura-841 SIP phones for
This is a very newb. question.
Been using asterisk very happily now for several months and am considering
getting some of those really 'cool' multi-button phones with LEDs and
buttons.
It's unclear to me if it is a straightforward task to actually setup a
multiline 'presence' on the phones where
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a slackware installation to talk to a standard phone line so that asterisk
can call out?
Dan
On Thu, 7 Jul 2005 10:49:32 -0700
Dan Adams [EMAIL PROTECTED] wrote:
Hi, I am sorta a newbie to the asterisk community at
least in the realm of
hardware types. I was wondering, what type of card is
used to allow asterisk,
on a slackware installation to talk to a standard phone
line so that
Take a look here:
http://www.digium.com/index.php?menu=product_detailcategory=hardwareproduct=TDM400P
MARK.
Dan Adams wrote:
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a
@lists.digium.com
Subject: [Asterisk-Users] Newbie Question: Type of card
Hi, I am sorta a newbie to the asterisk community at least in the realm of
hardware types. I was wondering, what type of card is used to allow asterisk,
on a slackware installation to talk to a standard phone line so that asterisk
Hi,
I have some problem to get this setup working. I have a CAC Channel
Banl I, with FXO and an Asterisk box ( I am using [EMAIL PROTECTED] 1.2)
and I have a TE110p installed in this box.
What I want to do is, just to be able to dial one of those lines that
already are connected to the channel
Recheck your zaptel.conf. That's not the correct setup for a T1 trunk.
You need to know the signalling the channel bank uses, and specify the
voice channels (bchannel=1-24), and the signalling channel
(dchannel=25). Those numbers are bogus, as I've never worked with T1
;)
BTW, why are you using
hi Rich,
thanks for ur help..
it works..
i have found another way,
_9XXX,1,Dial(Zap/4/1800XX,5,D(${EXTEN}))
D = will send dtmf
thank a lot Rich..
best regard,
shahdan
--- Rich Adamson [EMAIL PROTECTED] wrote:
the situation here is i want when user make
outgoing
call,
:
Date: Wed, 15 Jun 2005 00:53:14 -0500
From: Jon Gabrielson [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie question about pressing a key to
be connected to the caller
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID
You don't have to use queues to use agents. Do a show application dial
and look at what he is showing you.
You can have a macro run upon answer so put your menu there.
Kevin
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
hello all,
the situation here is i want when user make outgoing
call, asterisk will call 1800XX first then after 3
or 4 sec asterisk will insert the number that user
want to call..
user don't know that the call is go to 1800XX
first..
means user just insert the number that they want to
the situation here is i want when user make outgoing
call, asterisk will call 1800XX first then after 3
or 4 sec asterisk will insert the number that user
want to call..
user don't know that the call is go to 1800XX
first..
means user just insert the number that they want to
I have a newbie question about the dialplan. I have a main menu that
picks up on a certain DID number, and gives a list of options. When an
option is selected, for instance 1 for sales, it rings a number of users
in succession until one picks up and is connected to the caller,
otherwise it
Check out ackcall=yes in agents.conf
It allows them to press # to accept, or press * to not accept.
then you can do something like:
exten = 101,1,Dial(Agent/101,20,A(presspoundtoanswer))
or if you want to get more fancy, check out queues.conf
where you can set ring orders and answer penalties.
Greetings,
I have my first asterisk installation up and running, thanks to a lot
of reading. Could anyone point me in the direction of things to read
on automated outbound dialing? NOT predictive dialing - I will not
have agents handling the calls. These calls are reminders for
appointments,
read in voip-info.org about Asterisk Call Manager API, and may be an
easier soultion are the .call files that you can pleace in
/var/spool/asterisk/outgoing/ these files have a description of the
type of call you wanna make, in the very moment that you place the
file there, a call will be
Maybe you should review these:
http://asteriskdocs.org
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
I've never seen the register line you have used, the ones I see are
mostly like this:
register = username:[EMAIL
Over the past 2 weeks I have been able to compile and get an asterisk
system up
running on a debian Linux box.
I have setup 5 internal sip clients on the lan and all works great!
I can also call from outside (PSTN) into the system and reach extensions
and
services no problem.
All is up
Steve wrote:
I have read LOTS of docs and played quite a bit to get this far
Good, keep playing!!!
(a lot of your typing time deleted)
OK here's what messes it all up (and I admit I'm clueless here)
register = 2135551212:[EMAIL PROTECTED]
Ok I'm still playing and the way it's supposed to work is making much more
sense now.
However it is still 'not working' as soon as I add this:
[sipproviderexample.com]
type=peer
host=sipprovider.com
fromuser=2135551212
secret=2135551212
fromdomain=sipproviderexample.com
to sip.conf
I also
Hi folks,
I have an asuscom ISDN BRI card in my server and was wanting to know
whether this would be good enough to use with Asterisk. I am VERY new to
this, so have no idea how to config the software, etc. But I am very
eager to learn.
Cheers
H
___
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hamish Whittal wrote:
Hi folks,
I have an asuscom ISDN BRI card in my server and was wanting to know
whether this would be good enough to use with Asterisk. I am VERY new to
this, so have no idea how to config the software, etc. But I am very
Hi we are looking to swap out an old version of Cisco Call Manager for
asterisk and are trying to work out the best way to handle the main
office number. We will have about 35 phones and we have PRI from Colt,
we are london based.
At present when a call is made to the main number and is not
: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick
Teagle
Sent: Wednesday, April 13, 2005 8:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Newbie Question on how to handle main office
number
Hi we are looking to swap out an old version of Cisco Call Manager for
asterisk
: [Asterisk-Users] Newbie Question on how to handle main office
number
Hi we are looking to swap out an old version of Cisco Call Manager for
asterisk and are trying to work out the best way to handle the main office
number. We will have about 35 phones and we have PRI from Colt, we are
london
13, 2005 8:19 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Newbie Question on how to handle main office
number
Hi we are looking to swap out an old version of Cisco Call Manager for
asterisk and are trying to work out the best way to handle the main office
number. We will have
1. I wonder Asterisk can do this (refer to the following diagram) or not ?
(Can I make a call from the SIP phone to the normal phone )
Asteriskinternet
Asteriskcall to
normal phone/ # Server 1 == Server 2
Koa CG wrote:
1. I wonder Asterisk can do this (refer to the following diagram) or not ?
(Can I make a call from the SIP phone to the normal phone )
Asteriskinternet
Asteriskcall to
normal phone/ # Server 1 == Server
i am using the sample config files and get a dial tone. i have also
gotten it to play greetings etc, but i need the phone to ring so that
i am not tieing up the one phone line, please help, i know this sounds
insanely stupid but i cant get it to work.
I guess the first time it didn't get through... I didn't see it appear
in the list, that is...
I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on
it. They seem to run fine inside my network, so that's OK.
Now, I want to start using a X100P to connect it to my phone
bram kortleven Wrote
Message: 6
Date: Sat, 19 Mar 2005 22:16:39 +0100
From: bram kortleven [EMAIL PROTECTED]
Subject: [Asterisk-Users] newbie question
To: asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
I guess the first time it didn't get through... I
bram kortleven wrote:
I guess the first time it didn't get through... I didn't see it appear
in the list, that is...
I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on
it. They seem to run fine inside my network, so that's OK.
Now, I want to start using a X100P to connect
I installed an [EMAIL PROTECTED] machineand configured a few SIP accounts on
it. They seem to run fine inside my network, so that's OK.
Now, I want to start using a X100P to connect it to my phone line, to
make call routing between internal SIP phones/softphones, my local
phoneline and an external
Contact me offlist and I will gice you some info
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bram
Sent: Friday, March 18, 2005 10:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie question
I installed an [EMAIL
We have also been looking at various GUI's for Asterisk... ([EMAIL PROTECTED]
being one)... can anyone recommend one that would be ideal for a business
user in a basic small / medium office environment?
Depends on what you mean by GUI
Simple GUI to edit/view Asterisk's config :
- phpconfig :
Hi all,
Just a quick question from someone who is reasonably new to
the Asterisk server.
We have ordered the hardware for a test environment, and plan
on setting it up at the start of next week.
At the moment, we have a couple of VOIP handsets, a Digium TE110P
card, an E1 line and
Callum McGillivray wrote:
Can anyone tell me from experience how long it might take to get it up
and running so that we can make some basic test calls ?
If the hardware is functional and the E1 is provisioned properly, a
decent admin should be able to have Asterisk running and making test
calls
2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question
Callum McGillivray wrote:
Can anyone tell me from experience how long it might take to get it up
and running so that we can make some basic test calls ?
If the hardware
Title: RE: [Asterisk-Users] Newbie Question
As someone who started out
using * when I was just slightly educated on Linux, I can say that you are
probably many steps ahead of the average new * user. You have people that
know Linux so that is a major plus. The actual configuration
] On Behalf Of Callum
McGillivray
Sent: Friday, 4 March 2005 2:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Newbie Question
Thanks for the quick reply Kevin (and Dean!)...
It's the kind of answer I was hoping for.
The E1 has been used on a Hardware
: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] Newbie question
Hello!
When the oprator transfers calls to internal extensions to unavailable
or busy extensions, how can I prevent these calls from going to
voicemail, and route them back to the oprator? But other calls, ie
internal
Hello!
When the oprator transfers calls to internal extensions to unavailable
or busy extensions, how can I prevent these calls from going to
voicemail, and route them back to the oprator? But other calls, ie
internal between extensions, and calls coming in via DID should get
voicemail if
PROTECTED] På vegne af Tim De Lange
Sendt: 18. februar 2005 14:25
Til: asterisk-users@lists.digium.com
Emne: [Asterisk-Users] Newbie question
Hello!
When the oprator transfers calls to internal extensions to unavailable
or busy extensions, how can I prevent these calls from going to
voicemail
Okay, I'm going to preface this by saying I'm sure I've overlooked something
really basic here. I just need someone to hit me with a clue stick and
point out what I'm missing.
I've got a TDM card with four FXO modules. I've plugged one of them into a
PSTN line. I'm working through the examples
Folks,
I've just successfully set up Asterisk (as part of the
Asterisk Management Portal installation). When I say
successfully, I mean that I have gone through all
the steps detailed for the installation of AMP and not
hit any snags there. I can connect to my asterisk
server via ssh and can also
beonice wrote:
Ouch ... error while writing audio data: : Broken
pipe
What are the messages before this?
--
Cheers,
Matt Riddell
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News -
Here is the entire output until it dies:
[EMAIL PROTECTED] asterisk]# /usr/sbin/asterisk -cp
Set to realtime thread
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Not found
(No such file or directory)
Asterisk
Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu:
beonice wrote:
Ouch ... error while writing audio data: : Broken
pipe
What are the messages before this?
Matt I think that is something related to mpg123...
--
D e n i s G a l v ã o
iSolve - Solve Is Our Business
Av. Candido de Abreu,
Ouch ... error while writing audio data: : Broken pipe
Did you run make samples from /usr/src/asterisk?
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To UNSUBSCRIBE or update
Denis wrote:
--- Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Em Ter 18 Jan 2005 20:43, Matt Riddell escreveu:
beonice wrote:
Ouch ... error while writing audio data: :
Broken
pipe
What are the messages before this?
Matt I think that is something related to mpg123...
--
Colin Anderson wrote:
Ouch ... error while writing audio data: : Broken pipe
Did you run make samples from /usr/src/asterisk?
Yeah he has config files.
What is the response you get running mpg123?
I.E. what are the first four lines?
I.E. what version?
:)
--
Cheers,
Matt Riddell
Em Ter 18 Jan 2005 21:27, beonice escreveu:
That _seems_ to be a possibility. But I'm not really
sure. I made sure that there is a symbolic link in
/usr/bin to mpg123 ... the actual version is in
/usr/local/bin.
Thanks. By the way, I accidentally created a new post
with the details of the
--- Colin Anderson [EMAIL PROTECTED]
wrote:
Ouch ... error while writing audio data: : Broken
pipe
Did you run make samples from /usr/src/asterisk?
Hmm. I thought I had, but I've attempted this Asterisk
installation so many times that I seem to have skipped
it on this incarnation. I just
--- Matt Riddell [EMAIL PROTECTED] wrote:
Colin Anderson wrote:
Ouch ... error while writing audio data: : Broken
pipe
Did you run make samples from /usr/src/asterisk?
Yeah he has config files.
What is the response you get running mpg123?
I.E. what are the first four lines?
--- Denis Galvão - iSolve [EMAIL PROTECTED] wrote:
Did you install mpg123 from source!? Or you're using
a distro native
version!?
You have to get the mpg123 from its website and then
get it compiled to your
suystem.
I believe I have a freshly compiled version. I've got
version 0.59r,
Put /usr/local/lib in /etc/ld.so.conf then run ldconfig.
[app_rxfax.so]Jan 18 15:46:05 WARN
ING[7952]: loader.c:258 as
t_load_resource: libspandsp.so.0: cannot open shared
object file: No su
ch file or directory
Jan 18 15:46:05 WARNING[7952]: loader.c:
440 load_modules: Loading module
--- Brian Dingman [EMAIL PROTECTED] wrote:
Put /usr/local/lib in /etc/ld.so.conf then run
ldconfig.
Hmm. I don't understand what that did, but THANKS,
Brian. It seems to have at least got Asterisk to
successfully run. Now I can have fun with configuring
it!
Thanks a bunch!
BeOnIce.
It has to do with spandsp and receiving incoming faxes. This should
probably be updated in the documentation.
On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice [EMAIL PROTECTED] wrote:
--- Brian Dingman [EMAIL PROTECTED] wrote:
Put /usr/local/lib in /etc/ld.so.conf then run
Thanks, Brian. I'm now a happy camper!
Cheers,
BeOnIce
--- Brian Dingman [EMAIL PROTECTED] wrote:
It has to do with spandsp and receiving incoming
faxes. This should
probably be updated in the documentation.
On Tue, 18 Jan 2005 17:09:04 -0800 (PST), beonice
[EMAIL PROTECTED] wrote:
Hello,
is it possible to route a phone call by Asterisk to a Skype user?
Scenario:
-
Incoming phone call
|
My telephone system
---
| |
---
|
Internal call routing
to extension # with a modem
connected to Asterisk Linux Box
Hi Mike -
its my first time to post here, im in the process of building asterisk
based telephone system (just small). i already installed asterisk
server, i just wanted to test 2 sip softphones to get working before i
move on, is it possible to have 2 softphones talk to each other
without
any
hi all,
its my first time to post here, im in the process of building asterisk
based telephone system (just small). i already installed asterisk
server, i just wanted to test 2 sip softphones to get working before i
move on, is it possible to have 2 softphones talk to each other without
any
Hello, I'm really new on Asterisk.
Is it possible to use a telephone machine connected to a modem as an asterisk
voice input output device? I do not need PSTN connection.
The scheme i'm thinking about is;
user - phone - modem - asterisk - ip - vice versa.
If it is possible can a user dial
I am just learing some Linux and have been able to setup Asterisk samples
and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1
analouge phone connected to port 4 of Digium TDM-400 with appropriate cards
installed to dial out on.
Leo Salas wrote:
I am just learing some Linux and have been able to setup Asterisk
samples and channels fxo card on ch.1 and fxs on ch 4.
I have an Internet Polycom phone to use to test to/from internet and 1
analouge phone connected to port 4 of Digium TDM-400 with appropriate
cards installed
]
Sent: Saturday, November 13, 2004 6:30 AM
Subject: Re: [Asterisk-Users] Newbie question
First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine
First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine !).
I was trying to connect my * box through FWD using SIP
but it is not working an
Hello:
First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine !).
I was trying to connect my * box through FWD using SIP
but it is not working
i think u missing the defaultip of the phones.. e.g.
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow=all
allow=ulaw
register = 500460:[EMAIL PROTECTED]/500460
[fwd]
type=friend
secret=cuco99
username=500460
host=fwd.pulver.com
[2000]
Hi,
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
my office, where I'll pickup that call. Or I want to
On Thu, 4 Nov 2004, Luciano Macedo Rodrigues wrote:
That's problaby a easy question to solve but I couldn't figure out how to do
what I need.
My PSTN line is connected to a phone and a FXO card. What I need is when
someone calls me, and I don't answer in 3 or 4 rings, * makes a VoIP call to
I have a TDM31B with one FXO port. The phone company provides call waiting
and the ability to switchover to that party. My problem is that now that
I'm running it through asterisk, if I'm on a call and I get another call, I
get the caller id info and the tones, but I can't switchover (via
nevermind. I was looking too far back. The answer is *0
___
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Hi All,
Is my setup possible? Or maybe the right question is Is this correct?
1 test server: installed ser and asterisk(didn't really understand much
of it yet).
1 CISCO 1750: with 2 FXO.
2 UA's: X-Lite
Using SER only, I can make calls between extensions.
Using Asterisk alone, can I also
- Original Message -
From: mihai iancu [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 10, 2004 9:05 PM
Subject: [Asterisk-Users] newbie question - app_realtime.so failed
Hello,
Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18
Everything was running nice
Hello,
Here are my info: asterisk version 1.0 with Redhat 8.0 kernel 2.4.18
Everything was running nice and clean with an old version from Aug
2004.
Cleaned all source code and binaries - download and install version 1.0
and this is what I get:
Oct 10 22:44:36 WARNING[8192]:
Because realtime isn't in 1.0 or 1.0.1 its ONLY in cvs-head.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of mihai iancu
Sent: Sunday, October 10, 2004 9:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] newbie question
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my home office all the demo ivr seem to work. But I have one question
regarding * using /dev/dsp0. I only have one sound card on my system and
it
On Tue, 14 Sep 2004 06:47:53 -0400 (EDT), Wayne Veilleux
[EMAIL PROTECTED] wrote:
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my home office all the demo ivr seem to work. But I have one
Thanks Marconi, that solve my problem.
Bye.
Wayne
On Tue, 14 Sep 2004 06:47:53 -0400 (EDT), Wayne Veilleux
[EMAIL PROTECTED] wrote:
Hi,
I'm new to *. I just installed my X101P card with * from the source on
Mandrake 10.0 and I test it. Everything seems to work fine. When I call
at my
Hi,
sorry for annoying question;
i read http://www.voip-info.org/tiki-index.php?page=PBX%20Call%20Pickup
without understanding:
1. how to add an ext. to a pickup group (ie:. how to populate pickup group)
2. how 'Directed pickup' does work?
You dial the pickup number and your extension, and the
Yes too all, but the features you're talking about are more phone
related than asterisk related.
If your phone can log in multiple lines, then asterisk will send calls
destined for those lines to that phone.
I think most decent sip phones can do this.
On Wed, 2004-08-18 at 15:28, me peaceout
On Wednesday 18 August 2004 18:39, Joshua McClintock wrote:
If your phone can log in multiple lines, then asterisk will send calls
destined for those lines to that phone.
Actually you can do this even with Zap interfaces (standard phones) -- it'll
take some extensions logic and some DB work
...and send the appropriate extension to the appropriate ZAP
channel...
Or SIP or MGCP or H323 if you prefer...
-Chris
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: Wednesday, August 18, 2004 3:39 PM
Subject: Re: [Asterisk-Users] Newbie Question
Yes too all, but the features you're talking about are more phone
related than asterisk related.
If your phone can log in multiple lines, then asterisk will send calls
destined for those lines to that phone.
I think
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