HOw about :
development
Dave
On Wed, 2004-12-29 at 04:51, Alspach Family wrote:
I am getting ready to submit a list of department names to be recorded.
This is what I have so far:
Accounting
Accounts payable
Accounts receivable
Administration
Billing Collections
Complaint
Customer
On Wed, 2005-01-05 at 10:23, Jay Milk wrote:
We all mostly know that * as well as various SIP phones support SMS.
While the final setup is somewhat of a mystery, there are reports of
those lucky souls who have it working. We also know that in order to
send an SMS to a mobile phone, we need to
On Wed, 2005-01-05 at 16:03, richard wrote:
Hi,
I have the following scenario.
I have an Asterisk server running on an internal IP address behind a
firewall, and I have a remote user trying to connect to my Asterisk box
behind his firewall, but he can't seem to get a connection.
I have
If so please let me know off list and I will try to coordinate.
Dave
[EMAIL PROTECTED]
703-727-1312 Mobile
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On Mon, 2005-01-17 at 11:11, Jess Coburn wrote:
Hello I have a 800 DID setup to dial into my Asterisk server and I'm
wondering if it's possible to ID when it's a payphone or not? I
suspect it's not since I'm getting calls from someone else's SIP or
IAX box.
If I had a digium card installed
On Thu, 2005-01-20 at 09:18, [EMAIL PROTECTED] wrote:
Last concern about making my channels in a group and add that group in
my dial plan. How can I make sure it will start with channel 4 and not
pick a random one between the 3 channels as I'm pretty sure if I put in
my dial plan a group
Responses embedded below!
On Mon, 2005-01-24 at 18:49, Keith Burns wrote:
Depending on the switch they are using, there are a limited number of
D-channels (or D-channel licenses).
CAS signaling needs RBS (its the winking in this case).
-Original Message-
From:
On Sat, 2004-12-18 at 15:50, Steven Critchfield wrote:
On Sat, 2004-12-18 at 20:31 +, Antony Stone wrote:
On Saturday 18 December 2004 20:27, Rodolfo Grave wrote:
Hi and thanks once more.
I moved the card around, and it kept the same IRQ. Then I went into
setup and changed
On Mon, 2005-01-24 at 16:15, Matthew Boehm wrote:
Follow this diagram:
Many POTS lines - Many Channel Banks - Mux - DS3 ---cloud--- DS3 -
demux into T1s - Many Asterisk's
If someone calls into 1 of the asterisk boxes (via PRI or VoIP), and I send
the call back down the line above to a
How would you deliver calls to the voicemail system without the PBX
functions?
db
On Thu, 2005-01-27 at 08:22, [EMAIL PROTECTED] wrote:
HI
I would like know if it's possible to use the VoiceMail only of the Asterisk
Sytem without use the PBX part ?
Thank.
On Thu, 2005-01-27 at 16:06, Kim Lux wrote:
I've got Grandstreams (SIP devices) working behind double NATs, none the
less.
I recommend turning STUN off and make sure that your SIP devices are
generating random port numbers. If they generate static port numbers,
you'll get port collisions.
On Thu, 2005-01-27 at 17:25, Kim Lux wrote:
On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote:
Will you Please share your configuration, I was ready to give up,
thinking no one had been successful.
I am not using Asterisk, so I can only give you the Grandstream part of
things. Maybe
Why not simply delete the cdr via AGI script (ie delete cdr from table
name where number dialed ) for those calls that don't adhere to the
dialed number that you want to capture, or am I missing something ? This
would allow you to remove the cdr at the completion of the call, and
preserve
Does the system you are connecting to have a digital interface (T1) or is it
truly a 2/4 Wire EM system? If it is T-1 then you do not need a channel
bank however if it is the analog interface then you will. There are several
choices of banks available and there are also several drop and insert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Timothy R.
McKee
Sent: Wednesday, July 07, 2004 11:58 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
This has always been one of my pet peeves, even as I worked in
Of David Boyd
Sent: Wednesday, July 07, 2004 17:48
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Timothy R.
McKee
Sent: Wednesday, July 07, 2004 11:58 AM
To: [EMAIL
On Fri, 2004-11-19 at 18:40, Michael Vogel wrote:
Hi!
Today I played arround with phpagi. I hope I can use it to completely
hand over the control about outgoing calls. I don't want to use the
extension.conf for that.
At now I only found a method to call an agi-script when dialed at
On Sat, 2004-11-20 at 01:32, Gregory Junker wrote:
Add to it, my message wasn't a flame but rather a terse comment. Your
Never said it was a flame. I said it was in a tone virutally guaranteed
to make the guy consider you and everyone on the list to be a conceited
jackass.
The
On Sun, 2004-11-21 at 20:53, George Burt wrote:
Thanks, but that does not actually terminate the call. The phone continues
to ring until the caller hangs up.
I have done an application with cellphones that allowed allowed me to send a
signal to the phone company to drop the call. Maybe
On Wed, 2004-11-24 at 04:14, Mike Dent wrote:
Hi,
I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and
1 SIP phone.
I've noticed some horrible buzz/rasping type of sounds! These seem to occur
when
* is trying to play back some audio or sound to me?
E.g. If I have an
See Below:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of usedcanon
Sent: Saturday, May 29, 2004 7:01 PM
To: Asterisk users
Subject: [Spam] [SpamSA] [Asterisk-Users] extracting country code from a
number
Hi
Does anyone know of an algorythm to
On Sun, 2004-10-24 at 10:24, Steve Totaro wrote:
I know she works at Digium but they probably go down the street to a real
sound stage to do the recordings via 3rd party.
A sound stage is a facility used to create and process professional
recordings. They can be used by anyone employed by
On Sun, 2004-10-24 at 17:52, dean collins wrote:
From what I read about a year ago was that it was a carrier hosted
solution that actually controlled the ss7 switching at the exchange
(basically no call costs from tromboning, and was only implemented into
an ip-centrex or hosted call centre
On Tue, 2004-11-02 at 09:18, Whisker, Peter wrote:
I have an * switch at home and one in the office. Both similar new CVS head
versions and both with chan_sip2 built in:
Asterisk CVS-HEAD-10/12/04-17:43:26
Asterisk CVS-HEAD-10/13/04-12:53:52
One is on a T1 connection and the other is on
On Sat, 2004-11-06 at 18:42, Ronald Wiplinger wrote:
Chris Foster wrote:
On Sat, 06 Nov 2004 16:42:30 +0800, Ronald Wiplinger
[EMAIL PROTECTED] wrote:
Where can I download the missing wakeup gsm files?
Ya' kinda have to make them yourself.
Do you mean NOBODY has
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you double-click to send
your messages? Just curious.
Sorry that this is unrelated but, Mojo with Horan, do you wake up each
morning and think
@lists.digium.com
Subject: Re: [asterisk-users] G729 copy protection
David Boyd wrote:
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry that this is unrelated but, Bruce, do you
double-click to send
your messages? Just curious
Bruce sorry for the top post, but your last two messages have not come
in twice Go figure...
db
On Fri, 2007-07-20 at 13:37 +0100, Bruce McAlister wrote:
David Boyd wrote:
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote:
Mojo with Horan Company, LLC wrote:
Sorry
Noah, or anyone actually,
question, can the IP address receiving the incoming call be used in
extension logic to determine call handling procedures, or maybe a better
way to ask is can asterisk provide information as to the IP address on
which a request was received?
Dave
On Mon,
Hi Robert,
which of the distros are you using as your base, dd-wrt ,
open-wrt ?
Dave
On Tue, 2007-07-24 at 17:19 -0400, Robert Augustyn wrote:
Hi all,
We have put together a firmware for a range of inexpensive routers.
It has been configured to provide optimum VoIP performance.
We have
On Wed, 2007-07-25 at 13:02 -0500, Eric ManxPower Wieling wrote:
Short Answer: No.
Long Answer: Maybe. If you can get your device to send inband DTMF and
tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should
just pass the DTMF as audio. Then if the call goes via IAX2 it
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten =
Hi Mat,
i have been working with the aa50 for a couple of weeks now. They are
slick looking devices that still have a few bugs. I tried to use the
device like an end user without previous knowledge of Asterisk or the
asteriskGUI, and can say right off that a typical person will not be
able to use
On Tue, 2007-09-25 at 10:57 -0400, Larry Costigan wrote:
Hi all,
I hope that I'm not breaking protocol too much by posting a message in
this group about a problem that I'm having with an Asterisk Business
Edition installation, but the reason that I'm posting here is
because the problem
What a waste of time...
dave
On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch wrote:
Brian West wrote:
And what was the purpose of this?
So that we would realize who we were talking to.
:)
-Stephen-
___
--Bandwidth and Colocation
Michael,
way cool.
Works in WINE also :)
db
On Wed, 2007-10-24 at 23:09 -0400, Michael Munger wrote:
Not sure if one exists, but someone had asked me for this a while ago.
Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1.
Just a concept program (but it works well).
On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote:
I need to know how fast a sip device needs to respond
to an INVITE sip message from asterisk before asterisk
retransmits the INVITE message again.
Thanks
Snip ---
7.2.1 INVITE received
When an INVITE request is received by the
On Mon, 2008-03-31 at 16:25 +0100, Steve Davies wrote:
On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote:
On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote:
The twist? We actually have far-end hangup detection working fine, and
that seems to be where the problem lies for most
On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote:
We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10,
acting as gateways from SIP to ISDN PRI interfaces. Each has one
Digium TE420 (with hardware echo cancellation) and one TC400B for
transcoding, in order to handle
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Hello ,
is it possible to control multiple legs (channels) of a call
individually, ie.
call 1 -- incoming call connected to IVR
call 2 -- outgoing call to party a made via manager interface
call 3 -- outgoing call to party b made by call-script
I would like to allow the caller on call1 to be
-Original Message-
From: ims.asuser ims.asuser [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Really WEIRD: can register but can not call!
Date: Mon, 25 Aug
is usefull to see if your _asterisk_
is
registered to some another sip server, eg. voip provider..
PJ
David Boyd wrote:
-Original Message-
From: ims.asuser ims.asuser [EMAIL PROTECTED]
Reply
Another item is the sequence for lookups!
So have you confirmed that your nsswitch.conf file has been set to look
at /etc/hosts first then dns?
Dave
-Original Message-
From: Adam Lovegrove [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, 2007-05-30 at 21:54 +0200, Olivier wrote:
2007/5/30, Matt [EMAIL PROTECTED]:
The problem with this is that if 1.2 has a bug that is making
it unstable, it should be fixed to make a stable project,
rather then steam rolling ahead to the next release. Further,
On Wed, 2007-05-30 at 15:29 -0500, Eric ManxPower Wieling wrote:
Bryan Laird wrote:
for inbound connections how does asterisk manage host=host-name
returning multiple A records... will
it allow authentication for any of the IP's returned?
I would assume that in the case of 'inbound'
What happens if you connect the fxo to the fxs and try several attempts
at completing a call? This should at least tell you if the issue is
outdialed digit issues or telco receipt issues.
Dave
On Mon, 2007-06-04 at 10:30 -0500, Rob Schall wrote:
But if this was the case, then why would the
Can anyone enlighten me as to why it takes 40 minutes or more for a
posting to the list to appear. This seems excessive, as other forums do
not take this long.
Dave
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asterisk-users mailing
On Mon, 2007-06-11 at 09:11 -0600, Steve Murphy wrote:
Gunnar--
CDR generation that covers transfers is an umimplemented feature in
Asterisk, in any version.
I have been working on a solution, but unfortunately, my solution is
radical enough that I dare not apply it to 1.2 or even 1.4. It
Murf,
you crack me up, but I totally agree with the vote or don't complain
model.
Thanks,
Dave
On Tue, 2007-06-12 at 13:05 -0600, Steve Murphy wrote:
I have created an asterisk.org blog entry:
http://www.asterisk.org/node/48358
to describe what I will shortly be committing to trunk to
Two points,
first (I believe from many previous threads, and viewing source code
) you must answer a call to place audio on the channel.
second, depending on the type of access being used by the originator of
the call, the carrier will not pass audio on the channel back to the
originator
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed on the outbound leg.
Dave
On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote:
David Boyd wrote:
Two points,
first (I believe from many previous threads, and viewing source code
) you
] On Behalf Of David Boyd
Sent: dinsdag 19 juni 2007 17:03
To: Lee Jenkins
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play dial tone withou answer
Yes Lee, he could, however he doesn't want to answer the call until the
call has been completed
On Fri, 2007-06-22 at 05:59 -0700, satish patel wrote:
can u give me example how do i create plan for this task or job
ram [EMAIL PROTECTED] wrote:
On 6/22/07, satish patel [EMAIL PROTECTED]
wrote:
Dear all
On Thu, 2007-07-05 at 15:08 -0400, Jon Pounder wrote:
Quoting Jeff Davis [EMAIL PROTECTED]:
Jon Pounder wrote:
I have a bunch of old cisco stuff with BRI ports on it but it was
never meant for voice, just purely data, so I don't think its very
useful for this purpose, but some of the
On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote:
David Boyd wrote:
I seem to remember that the wan Pipeline units supported BRI, and also
provided a couple of analog phone jacks. I will dig around in the
basement and try to find the one that I had, if I find it, who wants
Walter,
I know you just said that list users can not help with this problem,
however I must beg to differ with you. The information that you just
provided is a big help, if people take your advice about the
configuration of their internal systems. So in one way it is off the
topic of Asterisk,
On Mon, 2007-02-05 at 13:21 -0800, Michael Collins wrote:
I haven’t quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out…
The trick to using the DIALSTATUS seems to be to put it in the
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
From: Jorge Mendoza [EMAIL PROTECTED]
Funny that a digital line have a analogue pulse.
Normally the billing pulse is used on payphones. IMO you only need
the answer supervision to trigger your own billing system.
Jorge Mendoza
Stefano
On Wed, 2007-02-07 at 14:49 -0800, Yuan LIU wrote:
From: David Boyd [EMAIL PROTECTED]
Date: Wed, 07 Feb 2007 15:24:04 -0500
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote:
From: Jorge Mendoza [EMAIL PROTECTED]
Funny that a digital line have a analogue pulse.
Normally the billing
Hi Stefano,
I have a question, how would you go about using the billing pulses to
generate an invoice/bill. Also can you provide an ascii drawing of the
layout of the equipment as you intend to use it, they say a picture is
worth a thousand words:)
db
On Thu, 2007-02-08 at 15:13 +0100,
On Thu, 2007-02-08 at 16:48 -0800, Yuan LIU wrote:
From: Stefan Wintermeyer [EMAIL PROTECTED]
Date: Thu, 8 Feb 2007 21:56:11 +0100
Am 08.02.2007 um 18:39 schrieb Forrest Beck:
Does anyone have some method, or AGI scripts that will automatically
call a list of numbers from a database and
On Wed, 2007-02-28 at 13:05 +0100, nik600 wrote:
On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote:
nik600 wrote:
In the last months i've developed a web application for the use of an
asterisk call center.
Yuo can
- make calls from a web interface
- login/logout in queue
On Thu, 2007-03-29 at 12:16 -0400, Brad Stockdale wrote:
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of an Asterisk
On Fri, 2007-04-06 at 09:30 -0600, David Thomas wrote:
A start would be to get the contact information and actually CONTACT
the person about it. Come on now.
Maybe I'm confused... Isn't that what Dovid did when he replied to Tim's post?
Regards,
David
Could someone please remove this person from the list. It seems that the
person is saying they will be away for (9) nine months, with their
auto-reply set.
dave
On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote:
Je suis absent du 2/04/2007 au 11/04/2007.
Je répondrai à votre
On Wed, 2005-04-27 at 23:22 -0400, Steven Kalcevich wrote:
I think its a win win situation. Cisco has tons of money to throw at
them to get a better product with more features. I dont believe they
would aquire them and not put money in them to make a better product.
I guess the
Run nmap against the ip address and see what ports are active for tcp
service. Maybe you can connect via a different port (I know it should be
80), and see if the configuration is different between voice and web
dave
On Tue, 2005-05-10 at 21:50 -0400, Steve Prior wrote:
I just got a refurb
What code set is the 500 PRI configured for?
Dave
On Fri, 2005-06-10 at 17:05 -0400, [EMAIL PROTECTED] wrote:
Hi Guys,
I have several * servers connected to T1 PRI's from various service providers
in multiple locations the US. All the * servers use the same hardware with
the same OS and
Subject: [Asterisk-Users] HELP! With Postresql
I am having some real problems with getting CDR records to go to
a Postresql
database. I think I have followed every post and instruction available and
Asterisk still happily writes to a text file. Postresql is installed and
working on a
- Original Message -
From: Bincy K. Philip [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, November 08, 2007 2:13 PM
Subject: [asterisk-users] asterisk and installing chan_h323.so rpm
Hello,
When I tried to install chan_h323-1.0.1-module.i386 RPM i got
On Mon, 2007-11-26 at 08:08 -0600, Erik Anderson wrote:
On Nov 26, 2007 7:51 AM, Dovid B [EMAIL PROTECTED] wrote:
Hi,
I am currently playing with DD-WRT and I like it. I am looking for something
that is more SIP Aware. Anyone know one those that are out there ?
Dovid - what exactly are
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote:
On Saturday 15 December 2007 10:02:23 Rob Hillis wrote:
One of the biggest barriers to upgrading are the number of little
gotchas in syntax changes that can make an upgrade from 1.2 to 1.4
quite painful. After the pain I went
Thanks for your thoughtful response.
Dave
On Sun, 2007-12-16 at 10:43 -0600, Tilghman Lesher wrote:
On Saturday 15 December 2007 12:14:29 David Boyd wrote:
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote:
Of course, all of these deprecations should be covered in UPGRADE.txt, so
On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote:
the attached log with verbose=6 and debug=6 refers.
we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each
other. we're trying to have txfax send out on one of those pri ports with
rxfax listening on the other side,
On Wed, 2008-02-20 at 17:34 -0600, Tilghman Lesher wrote:
On Wednesday 20 February 2008 16:42:59 Steve Totaro wrote:
On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote:
Hi all, I am a huge fan of
You also need to specify the port so telnet mx1.datagrama.net 25 return
is the command to use.
db
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni
Terre
Sent: Monday, August 24, 2009 10:25 AM
To: Asterisk Users Mailing
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