On Sat, 20 Jun 2009, C. Savinovich wrote:
> Let me see if I get you: you inserted the installation CD, then you
> restarted the computer, and now you want to know what to do next?
How about:
1) Turn off the computer.
2) Read the installation guide for the CD.
3) Install the software.
4) Read
I have an Asterisknow.org CD. When I boot up, it seems ready for me to
choose update, console, etc. I'm assuming I need to do something at the
CLI prompt. Is there a tutorial that would take me from loading CD to
making first test call?
Computer is Dell Optiplex GX260
50GB free disk space
1.
Hello!
Sorry, I'm sure it's stupid. but I've got a simple ISDN line and a simple
ISDN-card, now finally running. :-)
I'm using application Jack and asterisk (CLI) only to do my bidding. Now I
can make calls. But how ca I setup my extensions.conf to receive a call? I've
had an example like
I wanted to say thanks to those who responded to my query. You all gave me
some good ideas to explore that I had not considered before, which is what I
was hoping for. :^)
On Tuesday 14 November 2006 10:50, Henry.L.Coleman wrote:
> By the time you purchase PCI cards for you extensions (FSO por
By the time you purchase PCI cards for you extensions (FSO ports)you would
be better off purchasing SIP phones like Grandstream GXP 2000 this will
give you a fully featured PBX IP phone for about the same cost or less
than FSO ports. Asterisk will have no problem running 25 or more SIP
phones
Perso
Jason,If you must stick with analog phones, you can find higher density channel banks that will host 8, 16 or up to 24 ports each. They communicate back to your asterisk server via your LAN. Or, as has been stated, you can purchase IP phones that also communicate back to your asterisk server via
Maybe you should try this http://www.digium.com/en/products/hardware/aadk.php .
Is very heavy loaded if 9PCI cards at a server. But is possible but not encourge. Maybe you can consider to have digital extension with IP phone. THis is my opinion. :-) good luck
On 11/14/06, Jason Flatt <[EMAIL PRO
Hello all.
My company currently has an older Executone PBX system that we are outgrowing.
Rather than wait until the last minute to make a hasty decision, I thought it
would be a good idea to do some research and compare options first. My
expertise is in computers and networking, and telephon
)
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Dean Collins" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, November 05, 2006 9:01 PM
Subject: RE: [asterisk-users] Newbie questions about Voice mail
Dean
Tha
>
> The next step should be
> 1a/ You boss decides You or someone in your team skill up in asterisk
> Or
Does the asterisk communitty have a presence at any of the IP telephony
conference?
..Brian
You just missed it check out www.astricon.net it was 2 weeks ago in
Dallas.
(but yes Digiu
On Sun, 5 Nov 2006, Dean Collins wrote:
Date: Sun, 5 Nov 2006 15:21:19 -0500
From: Dean Collins <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Newbie questions about Voice mail
Hi Brian,
Uhmmm
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
> Sent: Sunday, 5 November 2006 3:02 PM
> To: Dean Collins
> Cc: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
> Subject: RE: [asterisk-users] Newbie questions about Voice mail
>
> Dean
>
> Thanks f
necessary.
..Brian
On Sun, 5 Nov 2006, Dean Collins wrote:
Date: Sun, 5 Nov 2006 00:04:36 -0500
From: Dean Collins <[EMAIL PROTECTED]>
To: [EMAIL PROTECTED],
Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Newbie questions about Voice mai
006 11:54 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Newbie questions about Voice mail
>
>
> I am totally ignorant about actually using asterisk for any purpose. I
> have read some of the docs but not all. I am currently doing a
telephone
> audit f
I am totally ignorant about actually using asterisk for any purpose. I
have read some of the docs but not all. I am currently doing a telephone
audit for my company and one of the issues is voice mail. We are spending
quit a bit of money with our telco for voice mail services and I was
wonder
You can put the Asterisk system in front (i.e., between the PSTN and your Comdial system). This will let Asterisk choose whether the call should go out over the PSTN or the Internet using VoIP.
You would use the same for the second location, provided that is a complete Comdial system. You could
- Original Message -
From: "Ken Williams" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Wednesday, November 01, 2006 2:10 AM
Subject: [asterisk-users] Newbie Questions
I've been doing a lot of reading over
I've been doing a lot of reading over the last few weeks on Asterisk,
and will be implementing a test system this week to play with.
I've got two questions in regards to the ideal implementation for our
company. First, has anyone written any drivers to interface with
proprietary phones? Specific
Sorry for the long email but I am having all sorts of
probs
I basically have a number od sip phones in the house
I have 3 incoming numbers (sipgate) and one outbound service (sipdiscount)
I want all extensions to be able to call out using the outbound lines
On Fri, 2005-11-18 at 01:09 +0100, [EMAIL PROTECTED] wrote:
> You can put analogue phones in series, but I am not sure how many phones
> a FSX connection will drag and I don't think the cards are designed for
> it - don't know. Sounds to me as you would benefit from downloading
> Asterisk and pl
You can put analogue phones in series, but I am not sure how many phones
a FSX connection will drag and I don't think the cards are designed for
it - don't know. Sounds to me as you would benefit from downloading
Asterisk and play around with a few softphones first and maybe buy
Digiums starter
Hi fred,
For the branch office you could consider a 2 (or more) port E1/T1 card.
You can utilize one port for an incoming E1 from the telco (BT?) and
then the second port run to a T1 Channel Bank such as the Rhino 24 port
FXO http://www.myphonecall.co.uk/voip/channelbanks/rhino/default.aspx -
On Fri, November 18, 2005 0:02, Chris Wade said:
> Fred Blaise wrote:
> Sorry, but there is really no such thing as a "hub" for telephone lines.
> Each analog phone must be plugged into its own FXS port.
>
Unless you are willing to put telephones in parallel, like you do when
connecting multiple
Fred Blaise wrote:
On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote:
[EMAIL PROTECTED] wrote:
hi,
My second question: for a branch office of about 20 people, which E1
card do you advise? Would the TE210P be a good choice? (number of
concurrent calls would be max 10 for now) Why?
An E1 h
On Thu, 2005-11-17 at 14:48 -0600, Chris Wade wrote:
> [EMAIL PROTECTED] wrote:
> > hi,
> >
> >> My second question: for a branch office of about 20 people, which E1
> >> card do you advise? Would the TE210P be a good choice? (number of
> >> concurrent calls would be max 10 for now) Why?
> >>
>
[EMAIL PROTECTED] wrote:
hi,
My second question: for a branch office of about 20 people, which E1
card do you advise? Would the TE210P be a good choice? (number of
concurrent calls would be max 10 for now) Why?
An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an
E1 to a
hi,
My second question: for a branch office of about 20 people, which E1
card do you advise? Would the TE210P be a good choice? (number of
concurrent calls would be max 10 for now) Why?
An E1 has 30 lines, so you would be perfect with a TE110P. Connecting an
E1 to a company PABX is however a
Hi all
I am new to this whole field, being it PSTN or voIP. I am currently
reading the "Switching to VoIP" and "Asterisk: The Future of Telephony",
so hopefully, I will be less clueless soon :)
My first question: if I buy a Wildcard TDM400P, with one X100M and three
S100M modules, I would be able
Hello Hiu,
Monday, November 7, 2005, 4:51:35 AM, you wrote:
HYO> i am pretty new to asterisk. hope to learn more.
HYO> i have this notice from the console. when i was doing the echo testing
HYO> by putting the context=default. then, i called out 600 to get the echo
HYO> test, i can hear the opera
i am pretty new to asterisk. hope to learn more.
i have this notice from the console. when i was doing the echo testing
by putting the context=default. then, i called out 600 to get the echo
test, i can hear the operator talking, but i cant really hear the playback.
i am trying to dig around fr
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
I've some questions about asterisk, and in general about voip, please
help me :)
1. I've SIP accounts on external servers, and I would that my local
server will connect with those and redirect all calls from those to an
internal SIP accoun
On Tue, 2005-03-08 at 16:56, Brian Nehring wrote:
> Xlite for OS X actually.
bummer, I've been wanting to get it running under Linux.
>
>
> On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> > On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> > > I actually got X-
Xlite for OS X actually.
On Tue, 08 Mar 2005 15:00:24 +1100, Howard Lowndes <[EMAIL PROTECTED]> wrote:
> On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> > I actually got X-Lite talking to the server, finally. I didn't have to
> > change any of my Asterisk servers... I just kept fooling around
On Tue, 2005-03-08 at 14:57, Brian Nehring wrote:
> I actually got X-Lite talking to the server, finally. I didn't have to
> change any of my Asterisk servers... I just kept fooling around with
> X-Lite and watching the diagnostics log and it finally worked. I can't
> really say what fixed it, I do
I actually got X-Lite talking to the server, finally. I didn't have to
change any of my Asterisk servers... I just kept fooling around with
X-Lite and watching the diagnostics log and it finally worked. I can't
really say what fixed it, I don't even feel like I changed anything.
Oh well, thanks for
PROTECTED] On Behalf Of Brian
> Nehring
> Sent: Monday, March 07, 2005 2:26 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] newbie questions
>
> I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and
> could use some help. I'
>I've read through a good amount of documentation on voip-info.org, but
>hadn't found a solution, so I thought this list might help. I'm not
>great with linux, and I suspect there might be a port problem... maybe
>Asterisk isn't listening for SIP clients. How would I go about
>checking this? X-Lite
On Mon, March 7, 2005 22:50, Brian Nehring said:
> I've read through a good amount of documentation on voip-info.org, but
> hadn't found a solution, so I thought this list might help. I'm not
> just give it username/password and point it at a SIP proxy. However,
> as far as I can tell it isn't abl
L PROTECTED] On Behalf Of Brian
> Nehring
> Sent: Monday, March 07, 2005 2:26 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] newbie questions
>
> I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and
> could use some help. I'm i
ering...
site:lists.digium.com
Als try the documentation link at digium.com
Regards,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Nehring
Sent: Monday, March 07, 2005 2:26 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] n
I'm sorry, I'm sure you get alot of this, but I'm new to Asterisk and
could use some help. I'm investigating migrating our small business
phone system over to Asterisk and VOIP. Eventually we'll have around 4
incoming SIP (or IAX if I can find one) accounts for PSTN
incoming/outgoing, then SIP hard
limitations ...
G.Hendershot
-Original Message-
From: Jean-Francois Theroux [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 02, 2005 3:14 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] newbie questions
Hello,
At the office we have a Lucent PBX, which has 3 lines comin
Jean-Francois Theroux wrote:
Ok, so far I know I would need a 4 ports FXO card for the incoming phone
lines. I was thinking a Digium TDM04B. Then, I would need a card that
would connect to a Lucent 306EC expension modules (3 incoming lines, 8
phones) that goes in a Partner type of PBX system. We
Ok, so far I know I would need a 4 ports FXO card for the incoming phone
lines. I was thinking a Digium TDM04B. Then, I would need a card that
would connect to a Lucent 306EC expension modules (3 incoming lines, 8
phones) that goes in a Partner type of PBX system. We would like to keep
those ca
First tip, use a descriptive subject line.
On Wed, 2005-03-02 at 15:14 -0500, Jean-Francois Theroux wrote:
> Hello,
>
> At the office we have a Lucent PBX, which has 3 lines coming from the
> CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones.
> All those are connected
Jean-Francois Theroux wrote:
Hello,
At the office we have a Lucent PBX, which has 3 lines coming from
the CO. 2 are used for phones, 1 for fax. In the office, we have 16
phones. All those are connected in the PBX. We do not have an automated
system nor voicemail system for now. But this is s
Hello,
At the office we have a Lucent PBX, which has 3 lines coming from the
CO. 2 are used for phones, 1 for fax. In the office, we have 16 phones.
All those are connected in the PBX. We do not have an automated system
nor voicemail system for now. But this is something we would like to
have
EMAIL PROTECTED] On Behalf Of Ken Panco
Sent: Tuesday, February 08, 2005 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] newbie questions
i installed it the other day but from some reason can only get one of
my budgetone 100's to register...any th
: Tuesday, February 08, 2005 4:44 PM
Subject: [Asterisk-Users] newbie questions
Hi,
I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards)
1. the distro
I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem
missing
(some C or C++ or python ...)
(buy the
sk Users Mailing List - Non-Commercial Discussion"
Sent: Tuesday, February 08, 2005 4:44 PM
Subject: [Asterisk-Users] newbie questions
Hi,
I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards)
1. the distro
I downloaded a "free mandrake 10.0 - 3 CD's) but some
I run Debian, and it's not hard to get a base install running. If you
want a GUI and such, then it'll be more than "follow the screen
prompts." I've been writing some Debian documents, if you're
interested, email me off-list.
Anyhow, on pretty much any distro, you can make your own packages
(RPM,
Voip to voip (in whatever form that takes sip-sip sip-iax iax-iax) is
what I am using asterisk for.
I would have thought mandrake would have been ok - but haven't used it
for a while. I'm running FC2 (fedora core2) and asterisk complies and
runs without any problems.
Dont fear make. Apps, for
Hi,
I want to use asterisk as a VoIP-VoIP pbx (just IP, no PSTN/ISDN cards)
1. the distro
I downloaded a "free mandrake 10.0 - 3 CD's) but some packages seem missing
(some C or C++ or python ...)
(buy the full version )
maybe the latest fedora is more complete ?
or easier to complete w
I currently subscribe to acedsl for voip service. I have ast. running on an old
compaq with 2 clone fxo cards. Everything is going good (thanks to lurking
around here). The box answers and dials over the analog. I want to bring in the
2 digitil from acecape. They currenty go to a cisco ata 186.
I'm slowly but surely bringing up my first asterisk system, plowing
through the wiki and the asterisk documentation project's book as I go,
trying to understand it all as I go.
Needless to say, I'm getting myself quite confused at times. What is the
appropriate venue to report my confusion, and p
, 2004 3:41 AM
Subject: [Asterisk-Users] Newbie questions from South Africa: Initial setup
> Hi All,
>
> I have been researching Asterisk for a few days now and have read
> hundreds of web pages and other documents. While some things are getting
> clearer, others are not.
>
>
Hi All,
I have been researching Asterisk for a few days now and have read
hundreds of web pages and other documents. While some things are getting
clearer, others are not.
I have managed to install Asterisk 1.0.1 on Debian testing (simple as
'apt-get install asterisk'). I understand FXS/FXO, code
You can plug several phones into an FXS port, but they look like the
same phone/extension to Asterisk - so they will all ring together, if
one is in use the others will be as well, etc.
The big question I see here is whether or not you want each individual
phone instrument (or group of instruments
>
> From: Matt G <[EMAIL PROTECTED]>
> Date: 2004/07/28 Wed PM 08:50:03 GMT
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Newbie Questions
>
> Hi everyone,
>
> I'm going to be helping to set * up for the company I work for, and in
> doing all my
Hi everyone,
I'm going to be helping to set * up for the company I work for, and in
doing all my research about it, have found it to be a very viable
solution for my SOHO side business at home. I do however have a few
questions, forgive me if they're stupid but I'm new to all of this.
1. I want
Hi,
I am new to asterisk. And I have some newbie questions :-)
I like to use asterisk I our office (around 20 phones) but we need to
see if a user is at the moment using his phone so he can not get a
second call from someone. Sometimes this is called a telephonecenter
where I can see the used "l
On Fri, 4 Jun 2004, Stefan-Michael. [iso-8859-15] Günther (in-put GbR) wrote:
> 1. I'm using a Fritz!Card with the i4l driver - no problem at all, my
> Grandstream BT 100 rings when I diall a regular phone number.
> Is there any need for me to configure the zapata.conf for the ISDN card to get
Hi,
I spent some hours working my way through the WIKI and a number of other
documentations, but after all, three questions are still left:
1. I'm using a Fritz!Card with the i4l driver - no problem at all, my
Grandstream BT 100 rings when I diall a regular phone number.
Is there any need for
Darren Sessions wrote:
I’ll apologize right away for asking stupid questions. J
System Setup:
SER = Proxy
Asterisk = Voicemail
All sip based setup.
1. What Is required to make asterisk –NOT- accept inbound
calls/signaling from an unknown host? I tried the peers in
sip.conf but
I’ll apologize right away for asking stupid questions.
J
System Setup:
SER = Proxy
Asterisk = Voicemail
All sip based setup.
What Is required to make
asterisk –NOT- accept inbound calls/signaling from an unknown host?
I tried the peers in sip.conf but it stil
> Third, are there any VoIP providers that I can have Asterisk
> talk to natively (i.e. via IAX or SIP, not the way I have
> Vonage set up now)? I'd be looking for a Chicago land number
> (630 specifically).
Check out www.iconnecthere.com - I think they've got pretty good coverage
nationally.
Ch
Hi gang,
I've just set up my Asterisk server with a X100P (talking to a Vonage Motorola
do-dad), and a Cisco IP7960 SIP phone. All is working quite will with
outbound and inbound calling. However, I have a few questions.
First, regarding call waiting on Vonage/X100P, how do I "click over" to
Look into www.digium.com.
Digium's cards are you best choice.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brez
Sent: Monday, November 03, 2003 4:03 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Newbie Questions
hello,
I am completely n
I built something very similar using:
- Adtran TA750 bought off Ebay for around $400 (you can do much better,
I was in a hurry.)
- A Digium Wildcard T100P
- A 4 port FXO card for the TA750 (I searched Google for "Adtran FXO"
and clicked one of the sposored links.)
You might have to pick up so
hello,
I am completely new to things but was wondering if some one could steer
me in the right direction [i.e. i was volunteered to get a PBX running
with little or knowledge] good news is, i got a lot of experience with
open source / linux / etc. anyhow. we have 4 lines coming in and need 16
Integrated Solutions
Majsmarken 9 DK2680 Solrød Strand
Tlf : 56 13 11 83 Mobil 28 66 28 48 [EMAIL PROTECTED]
> -Oprindelig meddelelse-
> Fra: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] På vegne af
> Jeremy McNamara
> Sendt: 28. juni 2003 23:20
> Til:
Hi Chris
I've done a lot of things with Cisco AVVID solutions in the past.
> CallManager).am I right in saying that Cisco phones using
> Skinny will
> not work with asterisk? Is it ever likely too?
Cisco own Skinny Protocol is not supported directly. But Cisco SE always
told me that Skinn
Check to see if you can get a IOS code leverl that supports SIP on the
6500. then maybe you can use your E1 card directly. you can also get a
SIP version of the code for the 7960's etc
Dave
>>> [EMAIL PROTECTED] 6/28/2003 2:56:12 PM >>>
Hi Chris
I've done a lot of things with Cisco AVVID solu
Johnny Witt wrote:
CallManager).am I right in saying that Cisco phones using
Skinny will
not work with asterisk? Is it ever likely too?
Cisco own Skinny Protocol is not supported directly. But Cisco SE always
told me that Skinny is a subset of H.323. But I would'nt count on it to be
fun
On Sat, 2003-06-21 at 11:02, CSTe wrote:
> Hi.
>
> I am new to this software, and I want to implement a client (SIP or
> IAX) with PHP or at least to pass the main functions (connection,call,
> transfer, hangup, call id etc) to a CRM.
> Does anyone know if I could achive a project like that wi
Hi.I am new to this software, and I want to implement a
client (SIP or IAX) with PHP or at least to pass the main functions
(connection,call, transfer, hangup, call id etc) to a CRM.
Does anyone know if I could achive a project like that with AGI ? Any
example using AGI with PHP ?
Do I ha
[snip]
Yes. The manual should explain further details.
I am not a voice comms expert and our Cisco IP Telephony system was
installed by an outside company
Are you referring to the Asterisk manual here?
Yes.
Just looking at the manual, it seems to me that I could use a Zaptel
E1 card and c
Hi
Thanx for the info.sorry to hassle you, but I have follow on questions
below.
I seem to recall that there is a Cisco 79xx administration tool in the
http://www.vovida.org/ pages somewhere.
Had a look at this.this tool will certainly make managing Cisco SIP
phones easiert
Thanx for the infounfortunately, I think we would need an Communications
Media Modulewhich we don't have
Chris.
From: <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] Newbie questions.
Date: Fri, 20 Jun 20
>>* .or could we connect Asterisk to the 6509 over IP and so make
>>it part of the main phone system?
>I don't know. Does the 6509 talk SIP?
It doesn't appear to. I would love to be wrong. It does support MGCP,
though.
___
Asterisk-Users mailin
Hi.
I have just successfully setup Asterisk with 2 Cisco 7940 phones
(converted for SIP) and a SIP softphone on a W2K box.and it all
seems to work very well.to those who wrote this software, it is
really cool.
Anyway, I am new to this software, and I have a lot of questions
which
Hi.
I have just successfully setup Asterisk with 2 Cisco 7940 phones (converted
for SIP) and a SIP softphone on a W2K box.and it all seems to work very
well.to those who wrote this software, it is really cool.
Anyway, I am new to this software, and I have a lot of questions which I
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