On Wed, 2003-10-15 at 15:10, rnc Info Lists wrote:
> >
> > Did you "modprobe ztdummy" before running asterisk ? I have meetme
> > running in one * box without zaptel harware.
> >
> I just tried that.
> The following messages are given:
> /lib/modules/2.4.20-8/kernel/drivers/usb/usb-uhci.o: ini
The USB card is the problem.. I should have realized that from today's
other thread... This system has no USB and unless I find a ISA USB card it
won't either since the PCI slots are full.
There is plenty else to get familar with so meetme and MOH goes on hold
(without music :-) ).
Sorry for clu
On Wed, 2003-10-15 at 15:59, rnc Info Lists wrote:
> The USB card is the problem.. I should have realized that from today's
> other thread... This system has no USB and unless I find a ISA USB card it
> won't either since the PCI slots are full.
>
> There is plenty else to get familar with so mee
Hi everyone, i need some advice, i have 3 offices with
4 telephone line in each using a normal telephone
system( rather old systems )all connected via a
clarkconnect vpn (broadband). so i have been looking
at asterisk and like all the features so i was
wondering if i could get all the telephone ca
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
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]:
/usr/lib/asterisk/m
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 f
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
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
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 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 b
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
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:
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
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
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
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 would need to change the phone service provider, because they
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 possible
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a "hunt group" with Verizon so that when a single
phone number is dial
Does * have the capability to screen calls? IOW, if someone calls in from
outside (ie. not a local extension), can * ask the calling party to state
their name, record it, ring the recipient, play the caller's name for the
recipient, then give the recipient the choice of answering or forcing the
; <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, October 01, 2003 5:21 PM
Subject: [Asterisk-Users] newbie question: MOH problem
> Just the sort of newbie question we all hate ;-)
>
> I'm a bit stuck with MOH. I think all is done right and I've read
Hi- let me start off by saying I'm a newbie to Asterisk and this list
and I'll also apologize up front for stupid questions.
I have Asterisk running and 2 SIP phones (X-Lite) plus an iaxtel
gateway set up. I used the configurations from the O'Reilly article and
I haven't even set up voice mail
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
__
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:
Matthew
- 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 Re
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 ca
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
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 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
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 abl
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 Coloc
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
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
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, 1.4.
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
I'll try.. :)
We have 4 analog lines coming into the building. These lines are simple POT
lines and we have them in a "hunt group" with Verizon so th
Chris
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, 8 January 2004 8:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Newbie Question-Looking for Feedback
M. Matt Colgin wrote:
>I've been looking at Asterisk for a replacement for our phone system an
Back in December there was a thread or remark about a Digium quad FXO
card. I would like to know when Digium will start marketing this...
M. Matt Colgin wrote:
I've been looking at Asterisk for a replacement for our phone system and I'm
hoping someone can help validate my assumptions.
We have 4
ke analog lines from *PBX)
Hard Phones: Snom 200, Grandstream ($70), Cisco ($$)
Soft Phones: X-Lite, X-Pro
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
> Sent: Wednesday, January 07, 2004 1:32 PM
> To: [EMA
On Wed, 2004-01-07 at 19:37, M. Matt Colgin wrote:
> Configuration #1
>
> - Can our telecom company provide two ISDN BRI lines. Each ISDN line
> providing 2 channels of 64kbit/s?
> - What manufacturer/model of ISDN adaptor is best supported by Asterisk?
>
> Disadvantage: Seems to
On Fri, 2004-01-09 at 20:55, Ken Alker wrote:
> Does * have the capability to screen calls? IOW, if someone calls in from
> outside (ie. not a local extension), can * ask the calling party to state
> their name, record it, ring the recipient, play the caller's name for the
> recipient, then giv
--On Friday, January 09, 2004 10:11 PM -0600 Alan Andrews
<[EMAIL PROTECTED]> wrote:
On Fri, 2004-01-09 at 20:55, Ken Alker wrote:
Does * have the capability to screen calls? IOW, if someone calls in
from outside (ie. not a local extension), can * ask the calling party
to state their name, rec
Hi!
> Does * have the capability to screen calls? IOW, if someone calls in from
> outside (ie. not a local extension), can * ask the calling party to state
> their name, record it, ring the recipient, play the caller's name for the
> recipient, then give the recipient the choice of answering o
Hi!
> Does * have the capability to screen calls? IOW, if someone calls in from
> outside (ie. not a local extension), can * ask the calling party to state
> their name, record it, ring the recipient, play the caller's name for the
> recipient, then give the reci
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Christopher Raper
> Sent: Thursday, 8 January 2004 10:06
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Newbie Question-Looking for Feedback
>
> Greetings al
--On Saturday, January 10, 2004 11:45 AM -0700 Steve Murphy
<[EMAIL PROTECTED]> wrote:
Hi!
Does * have the capability to screen calls? IOW, if someone calls in
from outside (ie. not a local extension), can * ask the calling party
to state their name, record it, ring the recipient, play the ca
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhancements compiled and installed. My
one question is, how do I
On Wed, 2004-03-10 at 20:30, Steven M. Sokol wrote:
Or, you could apply my patch, that I've been upgrading on the asterisk bug
site.
Check http://bugs.digium.com/bug_view_page.php?bug_id=752
Steve (and anybody else who may know about this code),
I have the code for your privacy enhanceme
Hi!
I have an H.323 network. I am trying to do a SIP-H.323 gateway to be able to
accept all the calls that come from other networks (SIP).
I have some questions:
- Must I use IAX? If so, how? I have not SIP servers, I only have a H.323
gatekeeper.
- If not, where is that I say that one call tha
Hi all;
I am planning a PBX/Voice mail system for a small business (approx 12
employees with phones). They have an inbound ISDN PRI, which is
probably irrelevant because all inbound calls are routed first to
receptionists which rarely route the calls on (client is a medical clinic).
Any idea
Thanks for the reply. All of the SIP phones and the Asterisk server are
on the local network (192.168.1.x) on the same side of the router (and
yes the router does have a NAT firewall). I would think the XLite to
XLite would work but it doesn't (yet). I am seeing errors on the
console when the S
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 als
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
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
sers@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 aste
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 cal
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 frontend/scr
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 frontend/scr
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'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
> 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
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
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 => 5185551234,2,Goto(custom-ca
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 al
>>
> look at the rtc driver then. you do have a rtc chip already on the
> system.
>
>>
I looked back in the list and looks like the message that mentioned who
wrote ztrtc I deleted. Can someone please let me know where to obtain
ztrtc? I did a google on it and came up empty.
Thanks,
Robert
_
On Thu, 2003-10-16 at 12:32, rnc Info Lists wrote:
> >>
> > look at the rtc driver then. you do have a rtc chip already on the
> > system.
> >
> >>
> I looked back in the list and looks like the message that mentioned who
> wrote ztrtc I deleted. Can someone please let me know where to obtain
> zt
http://www.junghanns.net/asterisk/downloads/zaprtc.0.0.1.tar.gz
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of rnc Info
Lists
Sent: Thursday, October 16, 2003 19:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] newbie question: Meetme (looking for
>> Seems you used my abreviation. It is really known by zaptelrtc. It seems
> to be written by Klaus-Peter Junghanns <[EMAIL PROTECTED]> and is
> distributed at http://www.junghanns.net/asterisk/.
>
Thanks for the info Steve. I got it but the "make" didn't work. Will work
on it over the weekend.
On Thu, 2003-10-16 at 16:29, rnc Info Lists wrote:
> >> Seems you used my abreviation. It is really known by zaptelrtc. It seems
> > to be written by Klaus-Peter Junghanns <[EMAIL PROTECTED]> and is
> > distributed at http://www.junghanns.net/asterisk/.
> >
>
> Thanks for the info Steve. I got i
Is there any one out there using an MSI 240 Global Station with Asterisk? I didn't
see it listed on the hardware page but figured I would ask just in case.
Thanks,
Patrick
<>
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 CVS-v1-0-12/12/0
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 A
>Ouch ... error while writing audio data: : Broken pipe
Did you run "make samples" from /usr/src/asterisk?
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update o
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
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
--- 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.
--- 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
--- 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
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 m
--- 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 PROTECTE
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'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 ha
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 so
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
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
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
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 han
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP) (h
I am new to the Telephony world and am trying to get a basic idea of how
things work.
Can I use Asterisk to connect two soft phones on a LAN to communicate
with one another without any additional hardware (besides the
sound-card)?
If this is possible, I seem to be doing something wrong. I have
i
Nice work Andy.
Filled in some issues I didn't quite understand. Looking forward to more.
Chris
> Tielman,
>
>You can take a look at the quick and dirty guide I'm slowly putting
together if you like...
>
>http://www.automated.it/guidetoasterisk.htm
>
>I'd appreciate any feedback you have on it
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
101 - 200 of 283 matches
Mail list logo
