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. W
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.
>
>
>> 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
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 ' | less
>
> Or, you can use the "script" command to capture the output
> -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
0:59 AM
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 th
> 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
>>
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
> mak
You can "tee" your CLI screen (google for it) so your output is in a file
that you can use more|less|vi or some other controlled viewing method on.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Shaw
Sent:
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.
com
Subject: Re: [asterisk-users] Newbie, Question on making a PSTN call..
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, 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 th
Need help pls..Anyone?
On Mon, Jun 15, 2009 at 10:58 PM, 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 wit
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
> ca
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, i
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)
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 hardphon
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
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 a
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 co
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 3/9/07, 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
___
--
[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 UNS
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 the demo routine.
I'll try to set it up to make a call o
Don't forget about 4569 UDP port (IAX protocol) forwarded to your Asterisk box.
Best Regards;
Leonardo Kamache
On 3/8/07, Dovid B <[EMAIL PROTECTED]> wrote:
If both the asterisk server and the softphone are on the same LAN then I
would look at your firewall settings on the box. Make sure yo
If both the asterisk server and the softphone are on the same LAN then I
would look at your firewall settings on the box. Make sure you have 5060 and
10,000 - 20,000 UDP open. If the phone is connecting to the server over the
internet and the server IS behind NAT then you need to forward ports 5
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 witho
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
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 Coloc
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]>
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 al
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
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 => 5185551234,2,Goto(custom-ca
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
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 o
> 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 thi
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 fig
That is a good argument. But I am not sure yet. 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 lik
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 wh
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 cl
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 t
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
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
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, un
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
ast
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, K
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 seems
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 so
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 somethin
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 b
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
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: http://www.voip-i
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
Rog
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 woul
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
ailto:[EMAIL PROTECTED] 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(defau
Dave Morrow
To: Asterisk Users Mailing List -
Non-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 wo
MorrowSent: 18 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 promp
nSubject: Re:
[Asterisk-Users] Newbie Question: Help with incoming dial
plan
This is how I do it.
[default-incoming]exten =>
2691,1,Goto(extensions,3212,1)exten =>
2692,1,Goto(extensions,3204,1)exten =>
2693,1,Goto(extensions,3207,1)exten =>
2694,1,Goto(extensions,3212,1)exten =>
26
Title: Newbie Question: Help with incoming dial plan
This is how I do it.
[default-incoming]exten =>
2691,1,Goto(extensions,3212,1)exten =>
2692,1,Goto(extensions,3204,1)exten =>
2693,1,Goto(extensions,3207,1)exten =>
2694,1,Goto(extensions,3212,1)exten =>
2695,1,Goto(extensions,3205,1)ex
List - Non-Commercial 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
-
From: [EMAIL 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
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.
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 wr
> 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 appli
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 al
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 who
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. Creasy wrote:
YeahI think that every install I have done the
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 11
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 3
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
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
me last
year.
-Jonathan
-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
syste
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
Polyc
cial 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, 2005, at 10:35 AM, Sean
ED] On Behalf Of Tom Rymes
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:
> T
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 provide
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 hou
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
Sta
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
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-w
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 t
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 th
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 Sa
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
a
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
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 cards, there are also Dig
Here is what I use:
http://www.digitnetworks.com/store/product_info.php?cPath=22&products_id=28
I have used it with Slack, but now I am running it with FC4.
-Original Message-
From: Dan Adams [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 07, 2005 12:50 PM
To: asterisk-users@lists.digiu
Take a look here:
http://www.digium.com/index.php?menu=product_detail&category=hardware&product=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 slackwa
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 tha
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 su
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
http://lists
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,
> 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
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.
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 Originate
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 PROTEC
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 l
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]
[sipprovidere
-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 ver
Kerry thanks, do you have an example of a ring group config ? Nick
That is really the beauty of a good IVR menu design. In a good design not
only do you eliminate the "everyone can answer every call" it also benefits
the caller because they get directed to the person/dept they need to get to
faster
1 - 100 of 198 matches
Mail list logo
