Hello!
Im new to Asterisk configuration and I have few questions regarding its
configuration.
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free
calls from each of 4 pstn lines... Can I configure Asterisk to call thru
pstn line that has free minutes? For example
On Mon, 31 Jan 2011, Piotr Górski wrote:
I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of
free calls from each of 4 pstn lines... Can I configure Asterisk to call
thru pstn line that has free minutes? For example
Outgoing calls are going through PSTN 1 for 60 minutes.
Hi,
I've recently installed Asterisk 1.6.2.13. I'd like to connect GSM Trunk to
it. I purchased a few Mobigater ProOpen gateways. It states that I should
use chan_celliax module to it. On the gsmopen site I see a comment in the
documentation that I can install the module on Asterisk 1.2.x,
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
Thanks,
Bill
: Tuesday, November 17, 2009 10:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie question
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] newbie question
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Bill Shaw
Sent: Tuesday, November 17, 2009 11:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] newbie question
When typing 'help
On Tue, Nov 17, 2009 at 09:09:39AM -0800, Steve Edwards wrote:
On Tue, Nov 17, 2009 at 11:33:45AM -0500, Bill Shaw wrote:
Hi All,
[snip]
2. Run from the external shell prompt:
asterisk -rx 'help whatever' | less
Or, you can use the script command to capture the output to a file
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
my default scroll back buffer is set to around
On Tue, 17 Nov 2009, Noah Miller wrote:
You could also make it much simpler and just set your verbosity very
low or just turn it off, so there are very few messages coming across
your screen. Unless you're on a really busy machine, you should be
able to read most of the help screens.
core
-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I
On Mon, 15 Jun 2009, Shiva Kumar wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a PSTN call with terrible failure for about two days
now.
On Windows using asteriskwin32:
I have a
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote:
Hello Asterisk-users,
I am new to Asterisk. I got SIP Calls to work between two computers using a
soft phone and asterisk in the middle. Since then, I have been trying to get
my soft phone to make a
Am Samstag, den 15.12.2007, 16:55 -0800 schrieb Philip Prindeville:
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
Anselm Martin Hoffmeister wrote:
In most cases it seems to end at
the fact that providers correct caller-ids they get from the calling
party: If you send any number which is assigned to the PRI (or SIP
trunk), that is fine; if you send another number, it will be changed to
the (first) number
Philipp Kempgen wrote:
Anselm Martin Hoffmeister wrote:
In most cases it seems to end at
the fact that providers correct caller-ids they get from the calling
party: If you send any number which is assigned to the PRI (or SIP
trunk), that is fine; if you send another number, it will be
Philip Prindeville wrote:
Philipp Kempgen wrote:
Do you know of any GSM providers/contracts where faking
for a valid reason is possible?
I can think of some... in rural Idaho, cell coverage is sparse. I
might check my voice mail of my cell phone via a land line, and want to
call back
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
Hi PaulH, thanks for your answer!
Now, another question.. Every E1 card has support for pri_net/pri_cpe or only
some of them has?
Can you tell me at least one card that can do that?
Thanks again.
--
Luar Roji
On Fri, Apr 20, 2007 at 03:32:45PM +1000, Paul Hales wrote:
Your best bet is a
Hi everybody. I'm about to ask a newbie question, be warned!
I have a NEC 2000 IPS PBX connected to a E1.
Now I want to set up an asterisk, with some digium card connected to that E1.
(suggestions about the card? I'll have maybe another E1 more).
The newbie question is.. How can I connect the
Your best bet is a dual port E1 card - set one side to pri_net and the
other to pri_cpe.
PaulH
On Fri, 2007-04-20 at 01:52 -0300, Luar Roji wrote:
Hi everybody. I'm about to ask a newbie question, be warned!
I have a NEC 2000 IPS PBX connected to a E1.
Now I want to set up an
On 3/8/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Thanks for the responses.
iptables on the * box has no rules and all tables default to 'accept.'
I have not got to the point of placing calls out across the internet
yet. The issue here is no audio back from the * box when running
through
On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Ok. I have not been able to setup the box to call outside, however,
watching the packet traffic I see plenty of data flowing from the
xlite client to the * server, but never any packets from the server to
the client. (That is, during the
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote:
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
I use it on Linux and it does.
-HJC
___
Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite familiar with RH
Linux. I have configured it and am using Xlite to connect and learn to
move around the conf files. I have a problem, however. The client
connects and dials ok, but there is no audio. In
5060
and 10,000-20,000 UDP to the asterisk server.
- Original Message -
From: Chris Nighswonger [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 09, 2007 1:16 AM
Subject: [asterisk-users] Newbie Question
Hi all,
I'm new to Astrisk so bear with me.
I have
-users@lists.digium.com
Sent: Friday, March 09, 2007 1:16 AM
Subject: [asterisk-users] Newbie Question
Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite familiar with RH
Linux. I have configured it and am using Xlite to connect and learn to
move around
.
- Original Message -
From: Chris Nighswonger [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, March 09, 2007 1:16 AM
Subject: [asterisk-users] Newbie Question
Hi all,
I'm new to Astrisk so bear with me.
I have just installed AsteriskNOW and am quite
Hi, any one test rtp packetization in 1.4?___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hello,I know little to nothing about asterisk. I am currently reading info online but was looking for other links as to good places to start. I basically just need to use asterisk to create a prototype. I need to demo a system that will allow the user to call a number, enter in some data, the
On 22:48, Sat 14 Oct 06, Rajeev Natarajan wrote:
yes to ztdummy: but you may have trouble when you try and run multiple
simultaneous meetme sessions.
On 10/5/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
omar parihuana wrote:
Is possible use meetme feature without Zaptel
yes to ztdummy: but you may have trouble when you try and run multiple simultaneous meetme sessions.On 10/5/06, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:omar parihuana wrote:
Is possible use meetme feature without Zaptel card? (ztdummy will be the solution? )Yup. :P Thanks in
Hi Folks,
I'm reading about meetme feature, but in accordance to voip-info it
say: A zaptel interface must be installed for conferencing to work.
Unfortunately I don't have a Zaptel Card, I'm using Asterisk like SIP
server then I would like to implement meetme function. What can I do?
Is
yep,
# modprobe ztdummy
You need some special routines compiled in the kernel, google around a
bit to find wich ones.
Other solution may be use app_conference, is not included in asterisk
sources, that app does not require zaptel timing.
Regards
On 10/4/06, omar parihuana [EMAIL PROTECTED]
omar parihuana wrote:
Is possible use meetme feature without Zaptel card? (ztdummy will be
the solution? )
Yup. :P
Thanks in advanced..
--
Mojo [EMAIL PROTECTED]
Office Manager, Horan Company, LLC
(907) 747- x112
___
--Bandwidth and
Thanks for the help!
What I have gathered mentally so far is that asterisk can't do
exactly what I am asking/expecting it to do.
Problem being that I am trying to get multiple inbound contexts
from multiple peers ( 3 of them in sip.conf) from one single provider.
What happens is that it matches
Hi,
If you don't specify a host= statement in sip.conf and you have a
section that includes a username and secret plus type=peer, it will
match on username and secret. (That implies that if you have three
different numbers registered with your sip provider all under one
username, calls for all
Steve Gladden wrote:
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
Have you looked at the sample configs in /usr/src/asterisk/configs?
Yes I have and my own configs are pretty much copies of them.
They do not detail, do
Hi,
I'm not an expert, but as far as i know, your incoming calls will
arrive with DID in ${EXTEN}
so the only thing you need is:
exten = 1234,1,GoTo(context1,1234,1) ; example for context extension
and priority
exten = 2345,1,GoTo(context2,2345,1)
exten = 3456,1,GoTo(context3,3456,1)
Be sure
What I do is the following and keep in mind I only use one register
statement with my provider:
exten = 18665551234,1,SetVar(FROM_DID=18665551234) ;
exten = 18665551234,2,Goto(from-pstn,s,1) ;
exten = 5185551234,1,SetVar(FROM_DID=5185551234) ;
exten =
Hello!
I've been struggling with the documentation for months on this simple
subject...
I still have not been able to get this concept down...
I have 3 sip accounts (PSTN DID's) that come into my asterisk box
and give me phone service from my itsp via SIP.
I for the life of me have not been
I've been struggling with the documentation for months on this simple
subject...
I still have not been able to get this concept down...
I have 3 sip accounts (PSTN DID's) that come into my asterisk box
and give me phone service from my itsp via SIP.
I for the life of me have not been able to
What version of asterisk? (been lots of changes happening to the sip
code over the last year)
SVN-branch-1.2-r9156
I think what I am trying to do is pretty basic and should not have changed
much in the past year.
I got started in July of 2005 and I upgrade about once per month.
In all this
Hi there,I would like to connect an Aasterisk Server with a Panasonic PBX (has E1extension). I only need 4 Lines. So I thought I could use an Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1 card which is more expensive.I dont now which card to take.Please tell me what you think
On Wed, 15 Feb 2006 08:59:22 -0800 (PST)
housi mueller [EMAIL PROTECTED] wrote:
Hi there,
I would like to connect an Aasterisk Server with a
Panasonic PBX (has E1extension).
I only need 4 Lines. So I thought I could use an
Dignum TDM04 Card with 4 FXO or a Dignum TE110P E1/T1
card
That is a good argument. But I am not sureyet. Do you know if there are big voice quality differences between the Digital and the Analog card?HousiRobert Webb [EMAIL PROTECTED] wrote: On Wed, 15 Feb 2006 08:59:22 -0800 (PST)housi mueller <[EMAIL PROTECTED]>wrote: Hi there, I would like to
Hi Jason. It seems your doing things right whatever that means. I
think the problem is more hardware related. Sure you have line in the
FXO?? have you tried dialing directly from some IP Phone?? I have
several applications that relay on automatic call generation with
Asterisk Manager and a PHP
Hello,
Somehow I've missed something here, so hopefully I'll be able to provide
enough of my setup to get some help. I feel I'm very close to getting
it, but missing something none the less...
1. I have a digium TDM400 with (2) FXO modules on channel 3 and 4 hooked
to two POTS lines.
2. I
I dont need to configure zaptel device, you dont use it :)
2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Hello friends,
I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My
question is I am using a Welltech FXO box and ip phones by Welltech. Do I
still need to
try [EMAIL PROTECTED]
http://asteriskathome.sourceforge.net/
2005/11/29, bram kortleven [EMAIL PROTECTED]:
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on
On 00:24, Tue 29 Nov 05, bram kortleven wrote:
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Or a simple way of configging through a
Hello friends,
I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My
question is I am using a Welltech FXO box and ip phones by Welltech. Do I
still need to configure zapata.conf and zaptel.conf which I read in the
documentation from asterisk pdf file downoladed from
Are there any example configs? Or does anybody have a default config
for this setup:
1 analog digium clone card for an analogue line (my home line)
Several sip phones (a few of them on the outside of my lan (NAT fw
between) and 2 insde my lan)
Or a simple way of configging through a
Hi all :
My first posting to the group - please be gentle!
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production server
Hi Roger,
Following this instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
I was able to install and run Asterisk several times without problems.
See also: http://www.voip-info.org/wiki/view/Asterisk+Linux+Fedora
Regards,
Vassil Kolarov
www.ittconsult.com
Thanks Vassil - I'll try those pointers and report back.
Roger
Vassil Kolarov wrote:
Hi Roger,
Following this instructions:
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3
I was able to install and run Asterisk several times without problems.
See also:
Hi All:
I've been through the compile/install procedure pointed out by Vassil: I
still crash on startup. Can anyone else give me some pointers, please?
Roger
Roger Hill wrote:
Thanks Vassil - I'll try those pointers and report back.
Roger
Vassil Kolarov wrote:
Hi Roger,
Following this
Asterisk runs just fine on fc3. Best guess on your problem is that you've
got come default config parameters in /etc/asterisk directory that it is
not liking at all. You might try starting asterisk with 'asterisk -cvd'
and watch the output for errors.
Hi All:
I've
Rich: Thanks.
I tried that, with and without any config files in /etc/asterisk. It
still falls over instantly, no messages other than 'Illegal Instruction'.
Asterisk is running on other machines for me quite happily, but just
does not want to play nice on this box.
I'm sure I'm doing
Well... the next best guess is the binary package that you downloaded
has some dependencies that are not on your system, or, the package
simply wasn't intended for your distro (for one reason or another).
Does the system have a developement environment that would allow you
down download the cvs
Rich:
Sorry if I did not make myself clear.
I was trying to give some history, which is where the downloaded package
came from.
On this box (FC4), I am currently downloading the 1.2.0 source from
asterisk.org (but not the CVS), and trying to compile and build from
scratch.
The build
On Fri, Nov 18, 2005 at 09:12:46AM +, Roger Hill wrote:
Hi all :
My first posting to the group - please be gentle!
Please use a more descriptive subject line.
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram,
Roger,
Can you try with a fresh Fedora installation on this box?
Vassil
Roger Hill wrote:
Rich:
Sorry if I did not make myself clear.
I was trying to give some history, which is where the downloaded
package came from.
On this box (FC4), I am currently downloading the 1.2.0 source from
Roger Hill wrote:
I've been messing with Asterisk for a couple of weeks now.
1.0.9 is running fine on an old laptop (300MHz, 128MB ram, Kubuntu),
downloaded the binary package.
Now I'm trying to put the working installation on my production server
along with HTTP etc.
( 700MHz, 256MB ram,
On Nov 14, 2005, at 6:21 AM, Markos Paraskevopulos wrote:
Hello everyone,
I’m new to VoIP and despite a lot of reading, I’m kind of more
confused than before.
I have following question – we currently have hardware Alcatel PBX
and approx. 50 phones in the company. I was wondering if we
Hello everyone,
Im new to VoIP and despite a lot of reading, Im
kind of more confused than before.
I have following question we currently have
hardware Alcatel PBX and approx. 50 phones in the company. I was wondering if
we would need to change the phone service provider, because they
On Mon, 2005-11-14 at 12:21 +0100, Markos Paraskevopulos wrote:
Hello everyone,
I’m new to VoIP and despite a lot of reading, I’m kind of more
confused than before.
I had an asterisk system up and running then read some dox and becuase
what I read at that time wasnt well written it has that
Title: Newbie Question: Help with incoming dial plan
Hi all. I just got Asterisk installed with a Digium TE110P T1 card. Have it working for outbound calls so I know that all the hardware is functioning.
Since all inbound calls come through my T1, I would like to setup a dial plan that
hree.
Thanks,
Steve Totaro
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just go
Thanks,
Steve Totaro
- Original Message -
From:
Dave Morrow
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:26
AM
Subject: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
Hi all. I just go
October 2005 16:41To: Asterisk Users
Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Newbie Question: Help with incoming dial
plan
I do not use any DID, all calls come in on the same
number 111222 so what I would like to do is simply prompt the caller
-Commercial Discussion
Sent: Tuesday, October 18, 2005 11:41
AM
Subject: RE: [Asterisk-Users] Newbie
Question: Help with incoming dial plan
I do not use any DID, all calls come in on the same
number 111222 so what I would like to do is simply prompt the caller to
enter
] On Behalf Of
asteriskSent: Wednesday, October 19, 2005 11:51 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Newbie Question: Help with incoming dial
plan
add this context
[default-incoming]exten =
111222,1,Goto(default-incoming,s,1)
exten = s,1
Michael Boger Jr wrote:
Sean,
What kind of hotel do you have? Some PMS vendors require the call accounting
and check-in interfaces to their system. I am not aware that asterisk
supports these serial interfaces.
No they have no call accounting etc as such everything is done manually.
I
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, August 11, 2005 8:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk
systemtoreplace an old PBX but using existing phone
Jonathan k
Discussion
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
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Basically, I can build the system but an looking for a card that will
allow for upto 20 extensions to be wired into the back of the PC.
@lists.digium.com
Subject: [Asterisk-Users] Newbie Question: Building an Asterisk system
to replace an old PBX but using existing phone
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Basically
Well, it's unlikely you're going to find a PCI card that can handle
twenty analog lines, however I suggest you look at purchasing a call
bank such as the adit 600. You then can link up your * server with
the call bank using a T1 card and control and route calls using that
method.
--
Tom Hayden
On Thursday 11 August 2005 08:34, Sean Rima wrote:
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Can you plug one of the phones into a REGULAR telephone line and get dialtone
and
Chad Osmond wrote:
To use the old phones and existing wiring you'll need some E1/T1 FXS
Channel banks and a T1/E1 Card. Each bank will handle 30/24 phones and
pipe them into a single E1/T1 connection.
You can connect up to 4 (or 8 soon from Sangoma) T1's per card. I really
like the Sangoma
Tom Hayden wrote:
Well, it's unlikely you're going to find a PCI card that can handle
twenty analog lines, however I suggest you look at purchasing a call
bank such as the adit 600. You then can link up your * server with
the call bank using a T1 card and control and route calls using that
Andrew Kohlsmith wrote:
On Thursday 11 August 2005 08:34, Sean Rima wrote:
I have a brief from a local hotel to build a PBX using Asterisk but they
want to use their exisiting telephones and wiring from an old PBX that
no longer works.
Can you plug one of the phones into a REGULAR telephone
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 Asterisk does provide, so I may build a bos for my house and show
them that as well
Standard phones can still do MWI (if they have a light), call transfers,
three-way
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 Asterisk does provide, so I may build a bos for my house and show
them that as well
Standard phones can still do MWI (if they have a light),
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 Asterisk does provide, so I may build a bos for my house and
show
them that as well
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 Asterisk does provide, so I may build a bos for my house and show
them
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 Asterisk does
Sent: Thursday, August 11, 2005 3:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
systemto replace an old PBX but using existing phone
On Aug 11, 2005, at 11:49 AM, Sean Rima wrote:
Tom Rymes wrote:
On Aug 11
: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
systemto replace an old PBX but using existing phone
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
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
Polycom
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew M
Stemen
Sent: Thursday, August 11, 2005 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Newbie Question: Building an Asterisk
systemtoreplace an old PBX
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew M
Stemen
Sent: 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
You are right.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarpo,
Louie
Sent: Thursday, August 11, 2005 5:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Newbie Question: Building anAsterisk
You write out a dialplan, then when you match a pattern in the dial
plan, the Polycom will initiate the call immediately. This way you can
have 4 digit internal extensions dial immediately, or have it wait for a
long distance or international number.
Ah... OK. Sounds like it's similiar to the
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