Andrew,
Sounds like it could be a good fit for your needs. Although that
raises many questions as to how exactly you should deploy it. If you
have a good Internet connection to the office in question you could
perhaps use VOIP termination for your outbound calls instead of the
current 4 PSTN
If you are going to do hylafax why not just do it seperate from
asterisk on a regular modem and just email o ut the results. Don't see
the big bonus to using a FXS and the adding cost and point of
failures.
On Mon, 23 Aug 2004 11:13:25 +1200, Carlos Hernandez
[EMAIL PROTECTED] wrote:
About
Depending on the application the Grandstream is decent but for
prolonged use I've found it's better to not pinch the pennies and go
with something a bit more expensive but with less problems. For a
simple SOHO deployment I'm just putting a Sipura SPA 3000 on their
cordless base station
you should be able to transfer using the manager interface from 1
user's phone to another
- Original Message -
From: Ben Merrills [EMAIL PROTECTED]
Date: Mon, 16 Aug 2004 11:29:44 +0100
Subject: [Asterisk-Users] Call stealing
To: [EMAIL PROTECTED]
Hi,
How can I (through the
CVS has them
- Original Message -
From: Wiley E. Siler [EMAIL PROTECTED]
Date: Sat, 14 Aug 2004 16:50:43 -0700
Subject: [Asterisk-Users] Free MOH MP3
To: [EMAIL PROTECTED]
Hello All,
Sorry to rehash a question I am sure has shown several time but I
cannot google up the answer
That could right don't really use MOH much but I noticed there was in
CVS. Although why would it be in CVS of asterisk if not used for MOH
though?
On Sun, 15 Aug 2004 18:57:39 +0100, Kevin Walsh [EMAIL PROTECTED] wrote:
William Suffill [EMAIL PROTECTED] top-posted:
CVS has them
That hasn't
the mirrors of rc1 are also listed in the wiki as well
On Tue, 10 Aug 2004 10:43:36 -0400, Seth Remington
[EMAIL PROTECTED] wrote:
On Tue, 2004-08-10 at 09:19, [EMAIL PROTECTED] wrote:
On Tue, 2004-08-10 at 07:35, [EMAIL PROTECTED] wrote:
Hello!
Is there a asterisk mirror?
use 1 of the broadcast mail patches on bugs.digium.com so when a msg
for a shared box comes in it is copied to all the priv boxes
associated w/ that group so the mwi on all those phones goes on as
well
On Sat, 07 Aug 2004 13:32:30 -0400, Don Hughes
[EMAIL PROTECTED] wrote:
The message waiting
add noload = app_txtcidname.so to your modules.conf would be a temp
fix. I would cvsup and rebuild it if you need txtcidname
On Fri, 6 Aug 2004 21:03:57 -0500, Lyle Giese [EMAIL PROTECTED] wrote:
I am putting together my first *. I had it running with two other pc's
running xlite and setup
Give each user a voice box then use 1 of the vm broadcast patches in
the bug tracker so that 1 to all in a perticular goup can be done.
On Thu, 05 Aug 2004 15:57:36 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I got my voicemail message working,thanks but now,keep in mind I'm
i've only used chan_h323 which suggests u download the tars and
extract them in /root. Takes a while to build but I did manage to get
it working
On Thu, 5 Aug 2004 13:47:29 +0200, Asmine Ouloube
[EMAIL PROTECTED] wrote:
This is what I've done:
Take asterisk, libpri and zaptel with cvs
After
yes change your dial macro to use SetCallerID and SetCIDName
and it will use that instead
On Tue, 03 Aug 2004 19:50:26 -0700, Trevor Peirce [EMAIL PROTECTED] wrote:
Is there any way to have asterisk set CID to Private or Unknown instead
of asterisk when a call comes in that is either blocked
asterisk needs license to work w/ G729
10 USDs per channel. Once the box has licenses it can convert the gsm
to g729 on the fly for you for the g729 phones. Besides you wouldn't
want to record voicemails in g729 either since you want to be able to
play them back from any where.
- Original
Any more information than that? I have a copy here as well but haven't
had time to read through it.
P.S. Yes I know my name is mentioned in the book. No need to flame me
on that fact. I am a regular consumer like anyone. Author felt
inclined to put it in there.
- Original Message -
From:
i used chan_h323 properly by Jeremey without issue and I did warn you
privately about the H323 support in Asterisk not being without issue
or configuration problems.
I'd suggest chan_h323 and follow Jeremy's docs to the letter and it should work
On Fri, 23 Jul 2004 20:21:48 -0400, Jeremy
i use a p2 400 here and it has problems with the scheduling but for 1
or 2 calls that would be ok. Depending on the volume you expect at 1
time adress the hardware according. I'd suggest atleast a 1ghz or so
On Thu, 15 Jul 2004 08:11:43 +0100, Simon Chappell
[EMAIL PROTECTED] wrote:
Hello All,
Seems quite interesting. Any suggestions of where to order one and
about how much?
On 15 Jul 2004 16:54:03 -0700, Wolfgang S. Rupprecht
[EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] (Tom Neville) writes:
; FXO port - Line from our office PBX.
[40]
...
secret=NOPE
Have you gotten
Using bison 1.35 here
- Original Message -
From: Fletcher Bonds [EMAIL PROTECTED]
Date: Wed, 14 Jul 2004 09:09:48 -0700
Subject: [Asterisk-Users] RE: [Asterisk-User] asterisk compile problem
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
From: Nik Martin
[EMAIL PROTECTED]
You need a cisco smartnet license to legally download the firmwares
for the phone. This would include the sip firemware
On Wed, 14 Jul 2004 20:26:27 +0200, xfastjackx [EMAIL PROTECTED] wrote:
Hi everybody,
I will receive my CISCO 7960G tomorrow. I've ordered it as a global
spare without any
voiptalk.co.uk
On Wed, 14 Jul 2004 16:36:51 -0700, Dameon D. Welch-Abernathy
[EMAIL PROTECTED] wrote:
On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote:
Can somebody help me with some names of good UK SIP providers?
I am looking for a UK number to connect to my asterisk server.
Normalize for Linux can tell you the levels of a wav and can be used
to adjust it according.
Been toying with using it for some of my streaming media clients since
it sucks to go from too low and having to up the volume to very loud.
On Mon, 12 Jul 2004 10:31:08 -0400, Seth Remington
[EMAIL
Just asking for abuse though unless it is restricted or grounds for
termination without a refund,
People prefer to set their CID to a proper call back number such as
myself but it has can be used for less positive uses.
On Wed, 07 Jul 2004 11:45:48 -0400, Jeremy McNamara [EMAIL PROTECTED]
Even to interface analog lines with asterisk you'd need hardware too
which perhaps will put
it out of the reach of your small organization.
$100 for a x100p (a analog port for asterisk)
On Wed, 07 Jul 2004 12:27:38 -0400, Mike Wagner [EMAIL PROTECTED] wrote:
That's all extremely way over my
well then lever it db driven and set the #'s in the db and update that
to the proper call order as needed
On Wed, 07 Jul 2004 13:51:10 -0300, Gelson Dias Santos
[EMAIL PROTECTED] wrote:
The problem is, there is no pattern. It´s not an open/close scenario.
This month I need to call
2.4 kernel? I have a RH 9 w/ 2.4 using ztdummy just fine a bit older though.
Message seems to show that the phones have trouble reaching each
other. Did Sip to Sip between the phones work fine?
On Tue, 6 Jul 2004 09:43:18 -0700 (PDT), Jack Turer
[EMAIL PROTECTED] wrote:
Any thoughts on the
line 1 is always default for calls when a line isn't selected prior to
dialing. Best bet would just be reverse the order you have them on the
Cisco line 1 as primary line 2 as secondary.
On Mon, 2004-06-07 at 12:57, Dragan Mickovic wrote:
I have a SIP phone (Cisco 7960) registered to 2 * pbx, is
ztdummy will suffice. A Zaptel interface is used as a timing device for
the conference.
On Thu, 2004-05-27 at 11:58, pesb wrote:
Hi there,
I need to implement a SIP Conference Server. I've saw that
asterisk has an application called meetme. But, it says that A ZAPTEL
INTERFACE
I just downloaded it today and the config menus just have for Firefly no
SIP or IAX2
On Thu, 2004-05-27 at 12:14, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
brian [EMAIL PROTECTED] wrote:
Just an FYI FireFly no longer works with anything but the FireFly network.
No more SIP, No
check the caller id in your incoming extension before you pass to to a
end user. Reset $calleridname to unavaliable if no number is given
On Tue, 2004-05-18 at 15:18, Roger wrote:
I have a question - if a user calls up w/ blocked caller id I get the
following on my phone
Incoming call from
Thinking about it further you could set the 6th line to autoanswer and
have the pbx call you and play MOH when none of your lines on the
asterisk box are in use.
On Thu, 2004-05-13 at 10:57, Joseph wrote:
Is there any way to play background music on a sip phone
while the phone is not in use like
Sure you could even use the examples posted here and the wiki to use the
outgoing spool to make calls. Just use a crontab to place a call file in
the outgoing spool every x # of days and problem should be solved.
On Thu, 2004-05-13 at 14:41, Mark Phillips wrote:
Those of you whom have a free
Billy,
Attachment seems to be due to a GNUPG sig file
-- William
On Sun, 2004-05-09 at 12:00, Billy Huddleston wrote:
Mark,
Would you please re-config or use a different mail client as to not send
your replies back as attachments??
It's VERY kludgy, and, I'm just going to stop reading
in the context of the incoming DID assuming their Caller ID is equal to
the mailbox for their voicemail aka DID #
exten * = 1,VoicemailMain(${CALLERIDNUM})
You might want to improve this though like so:
Add all assigned DIDs to an Asterisk DB
On * check if callerid is a valid did u assigned
if
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who use hardware phones but still want to keep TXT
chat and presence and perhaps integrated into 1 of the iax soft phones
as well to
would lean toward integration to that standard as
well.
On Wed, 2004-04-07 at 21:05, Duane wrote:
William Suffill wrote:
They modified iax to include the presence packet but only works on their
customized firefly network. I was thinking along the lines of a software
app for those of us who
cell phones?
-- William Suffill
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Would it be possible to use an IAX softphone in your situation?
I know iaxcomm is available for both Windows and Linux and can handle
multiple accounts.
On Tue, 2004-04-06 at 10:26, WipeOut wrote:
Martin Mielke wrote:
Hi Markus,
Markus Miertschink wrote:
The one I know of is
wrote:
William Suffill wrote:
Would it be possible to use an IAX softphone in your situation?
I know iaxcomm is available for both Windows and Linux and can handle
multiple accounts.
yes, iaxComm works for both Linux and Windows, but the sound quality is
poor compared to SIP softphones
curl could also be used. Since people asked I'm going to write it up
tonight since I use a GS as well until my Cisco shows up.
On Sat, 2004-04-03 at 09:52, Duane wrote:
Walker Haddock wrote:
I know that you can reboot the GS phones by hitting the rs.htm URL on the phone.
But, you have to
force all the users to a meetme extension ?
On Tue, 2004-03-02 at 11:46, Daniel ANDRE wrote:
Hello,
I am faced to a problem with call transfert with a MGCP Phone. I use
this to make a consultative call transfert:
1. send flash event
2. dial the number and speak with the other person
3.
Take some pics =)
On Tue, 2004-03-02 at 21:29, Matthew Marlowe wrote:
I've converted it... :) I cut, sanded and crazy glued a plastic notch
and made a whole on the handset.. Looks like it came like it. Works
perfect.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
don't thank me it's documented in the app just remembered stumbling on
it in the network tab.
On Sun, 2004-02-29 at 15:46, asdasd wrote:
sweet, cheers
- Original Message -
From: William Suffill [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 29, 2004 8:44 PM
Subject
All the digits should already be recorded so you could easily skip that
part and play back any digit from the AGI 1-9 that it was assigned.
On Sun, 2004-02-29 at 00:03, Robert Lawrence wrote:
I would be interested in the AGI Script. As for the voice prompts, I
am having Allison record some
if u add #'s to your contact list w/ @networknameinyourclient
they are connected thru that network such as firefly or others
On Sun, 2004-02-29 at 15:05, asdasd wrote:
You know what would be nice?
If Firefly could have a Network to use assigned to a contact.
I.E. I use 800 to check my
There are many options for remote support including Digium directly or
3rd party consultants that are on this list
On Thu, 2004-02-26 at 10:09, John Benson (Solutios Ltd) wrote:
Dear Mark
We have a customer who would like an Asterisk server setting up. Do
you provide this service,
why not load a client on their system they are using? There are quite a
few iax soft phones for both linux/win32
On Wed, 2004-02-25 at 13:58, [EMAIL PROTECTED] wrote:
You may be right here. I was thinking of an ActiveX plug-in. I don't expect them to
use public internet kiosks so they should be
are you on a machine that is slow or running alot of stuff? The ongoing
answer is the thread that is run by asterisk can't complete it's task
fast enough due to lack of system resources so it creates the notice
below.
On Wed, 2004-02-25 at 20:55, Carl Lougher wrote:
When I call Voicemail I get a
From Posts on this list on Sat. w/ the subject Voicemail brought to
light that there is a patch for some more advanced VM features after a
message is left.
http://bugs.digium.com/bug_view_page.php?bug_id=156
On Mon, 2004-02-23 at 12:56, Walt Reed wrote:
Looking through the Wiki and mailing
A customer is looking to change to VOIP but he wants a local incoming #
where he lives. Anyone know a provider that offers them via SIP/IAX.
I'll be running Asterisk to run all the features.
Sincerely,
William Suffill
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Asterisk-Users mailing list
that would require a transfer to the centralserver and
possibly back again. Maybe someone that has worked closely with the vmail
code can comment?
-Original Message-
From: William Suffill [mailto:[EMAIL PROTECTED]
Sent: Monday, February 09, 2004 9:02 AM
To: Darren Martz
Subject: RE
i saw something about that on the voip-info wiki
On Mon, 2004-02-09 at 11:23, Matthew B Marlowe wrote:
The newest firmware from grandstream supports configuration by mac address.
Simply upload a file cfgmac address.txt
Does anyone know the format of a cfg.txt?
use call files there is should a sample in the asterisk src
On Mon, 2004-02-09 at 12:21, John Chambers wrote:
Newbie question coming up ...
Is it possible to use the asterisk to initiate a call to a phone?
What I'm trying to determine is ways for software to connect to a
phone and send
u probably should upgrade to 0.7.2 but as far as the caller id that
would be from your sip.conf being set improperly add to your sip.conf
callerid=Caller Name # for each sip entry and that should clear it
up.
On Sat, 2004-02-07 at 00:23, John Fraizer wrote:
I'm running Asterisk 0.5.0 and using
I've been considering deploying an IAX softphone for some remote users
that want to interface with my PBX. It seems as though IAXcomm just
prints that it was rejected if they dial an extension unassigned on the
PBX. Firefly on the other hand crashes if you dial an extension that
doesn't atleast
i search them just fine in Evolution. Filters to a different folder than
my other mailing lists and works quite well. Different pop3 acc from my
isp too =) Why use bandwidth on my colo'd boxes when I can use something
I already paid for =)
On Sat, 2004-02-07 at 10:30, Eric Wieling wrote:
On Sat,
more information on how I do it you can reach me at
[EMAIL PROTECTED]
-- William Suffill
On Tue, 2004-02-03 at 20:47, Joshua Colp wrote:
Hi Folks,
I recently setup an asterisk system in order to provide a telephone
phone system for my web hosting business at a very low expense. My
problem
other options would be overkill.
Sincerely,
William Suffill
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I use Saww.net and Nufone for IAX2 to PSTN at a per min basis. So if i
pushed 5 calls i'd be charge per min for each call. Granted both the
companies above cater to * quite heavily.
On Wed, 2004-02-04 at 01:40, Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage,
Looks interesting I will check it out and see what I can do with it =)
On Sat, 2004-01-31 at 08:17, Brancaleoni Matteo wrote:
since I was annoyed this morning, I
wrote this simple php script to output
channel status from asterisk manager.
disclaimer
that's very bad written, nor commented...
I'd be interested in the patch as well
On Thu, 2004-01-22 at 13:51, Bill Hamel wrote:
Hi Chris,
This sounds what I am looking for, many thanks !
Also, I do not see an attachment, the patch that is :)
I dont know if the list strips attachments, perhaps send it to my email address
[EMAIL
i think nufone and xvoip are based on a per min basis prepaid perhaps
but no monthly fee there is probably others as well
On Mon, 2003-12-22 at 10:34, Hector Q.-datafull wrote:
Hi,
anybody knows a VOIP-PSTN company that doesn't charge monthly for making PSTN calls?
thanks.
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