Dear Sir
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk.
Thanks
Nirukshitha Gamage
--
This e-mail and any attachments are intended for the above named re
Hi all,
I have configured two asterisk Boxes.Then I need to communicate
these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk
Thanks,
Ishanka.
- Original Message -
From: "Branko Samardzic" <[EMAIL PROTECTED]>
To:
Hi Jerry,
- Original Message -
From: "Jerry Geis" <[EMAIL PROTECTED]>
To:
Sent: Thursday, December 01, 2005 10:56 PM
Subject: [Asterisk-Users] version 1.2 with chan_bluetooth
Any body gotten this to compile chan_bluetooth under 1.2?
WHat steps did you take.
You only need to chang
Hi all,
I have configured two asterisk Boxes.Then I need to communicate these
asterisk boxes via the IAX.It is better if you can help me to configure two
boxes to communicate via asterisk
Thanks,
Ishanka.
- Original Message -
From: "Branko Samardzic" <[EMAIL PROTECTED]>
To:
Sent: Fri
Hi,
We are currently in the process of getting everything in place to offer
both unlimited inbound and outbound services in Australia. We have
partnered with some providers in Australia to provide a business grade
VoiP service with coverage in all states with low rates for DID's and
outbound
> Does anybody know, why it is not possible, to run asterisk within
> screen?
Yes, it is possible but you can't scroll up so you only see the last
~40 lines. At least I didn't work for me but I didn't research this
further.
I just run it as for testing purposes:
screen chroot /usr/local/asterisk
Hi,
I am experimenting with trunkfreq parameter.
When it is 20ms I can see both parties in IAX session sending IAX frames
every 20ms.
When I modify this parameter to 40ms then I can see that only server that
initiated
IAX connection works properly (i.e. sends IAX frames every 40ms while other
side
On 12/1/05, Art Luke <[EMAIL PROTECTED]> wrote:
> Has anyone been able to set up a sip trunk between and Avaya S8700 and
> Asterisk? I can't seem to find any good docs on the subject. Any help would
> be greatly appreciated.
>
Unless Avaya has come out with something new in the 8700 software
betw
root/myroot
ram
On 12/2/05, Goran Donev <[EMAIL PROTECTED]> wrote:
What is the default username and password for [EMAIL PROTECTED] a2billing module.
Thanks___--Bandwidth and Colocation provided by
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Good Idea. I am doing a similar thing but for a different reason:
I use the system call to bring in mp3 files for music on hold. We make
custom Music on Hold messages and we store them at our DC. I am also
using this to pull mp3 updates for holiday music.
Try doing any of this with any OTHER PBX
Every once in a while I find a nice, compact little project is good enough to
share to the rest of the user community as a single post. Here's something
that I was happy worked as planned. This is not particularly clever, but uses
some infrequently-used tricks of running system commands from
you direct the calls to a ring group where both extensions are a member of it
(etc etc)
you'll work it out from there, just stop trying to set up the same extension
twice and start thinking groups.
Cheers,
Dean
From: [EMAIL PROTECTED] on behalf of Mike McM
Queues are your friend.
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
ZlotySent: Friday, 2 December 2005 11:04 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] call center
dial plan
Hello
H
Hi!
I downloaded asterisk 1.2.0 and compiled it myself.
The default behaviour is that calling 'asterisk' will return the prompt
and calling 'asterisk -v' is returning the CLI.
I want to run asterisk within screen, however '/usr/bin/screen -L
/usr/sbin/asterisk -v' outputs:
[screen is terminating
Hello
How to write dialplan which will be doing something
like this:
I want to divide sip clients(consultants) into groups,
And when call is incoming for example for number 6604,
it will be redirected to
first free random choose sip client from group 6604.
Best regards
Robert
On 12/1/05, Mike McMullen <[EMAIL PROTECTED]> wrote:
Hi All,
I have an employee who works mostly in our office but
maybe once or twice a week has to work from home to help
care for her special needs child.
My question is how to handle setting her up so that she only has one
extension shared
Sorry, to misinterpret.
I also never tried this.
Let me make a simple AGI script that will do this
#!/usr/bin/env ruby
message = ARGV.shift
$stderr.puts "\n#{Time.now} #{message}"
just put the above lines in a file in agi-bin direcotry say, 'Echo'
and call it like this
exten => s,7,AGI(Echo
> Is there anyone out there who has given this outfit money and actually
> received any service from them?
I am about to give up on sixTel, because of poor customer service.
It's a shame, because they otherwise seem quite competent.
I signed up Sept. 23 at http://www.iax.cc . Outbound service w
Title: Re: [Asterisk-Users] MeetMe with the V (video) option
who's done it? and how much
money are they talking about? I've been looking to pay for something like that
for a while.
Cheers,
Dean
From: [EMAIL PROTECTED] on
behalf of Matt RiddellSent: Wed 11/30/2005 9:15 AMTo:
Asterisk U
Steve Totaro wrote:
Hi All
I am using prepaid auth (callingcards), the idea is for a
prepaid support line. It is up and running but I have a
couple of questions with regards to modifications I would
like to make.
When a user calls and they go through the process of entering
their card number.
[EMAIL PROTECTED] wrote:
> There is a review on the homepage at http://voipspeak.net
>
> It has been available for a few weeks, it is much nicer than the 841!
Who has it for sale in UK?
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Ast
There is a review on the homepage at http://voipspeak.net
It has been available for a few weeks, it is much nicer than the 841!
Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL
On 12/1/05, Mike McMullen <[EMAIL PROTECTED]> wrote:
> Hi All,
>
> I have an employee who works mostly in our office but
> maybe once or twice a week has to work from home to help
> care for her special needs child.
>
> As background we have AAH 2.0 running with 8 analog lines
> connected to two di
Mike McMullen wrote:
Hi All,
I have an employee who works mostly in our office but
maybe once or twice a week has to work from home to help
care for her special needs child.
As background we have AAH 2.0 running with 8 analog lines
connected to two digium t400P cards. We have 10 sipura-841s
as
Give here 2 SIP accounts in the SIP.conf and have the extension ring both
of them at the same time.
exten => 0042,1,dial(SIP/acc1&SIP/Acc2)
If one of the sip account is not registered it will not ring.
Best regards
jan
--On Thursday, December 01, 2005 02:14:24 PM -0800 Mike McMullen
<[EMAIL
Hi,
I have some SoundPoint IP 501 phones, running SIP 1.6.2.
I would like to configure them so that line 1 connects
directly to a SIP provider, and line 2 connects to a local
Asterisk PBX. That should be simple enough, but this
provider requires URIs like sip:[EMAIL PROTECTED] .
However, the DNS
No I think what he means is more like a text file exists at
/tmp/something for example, and you might
...
exten => s,7,System(echo "Call from ${CALLERIDNUM}" >> /tmp/something)
...
but I haven't tested it yet. Can someone confirm this?
Innocent Evil wrote:
Yes,
I have something like this i
I am having issues with transferring calls.
I can transfer outgoing calls, but not incoming calls.
* 1.2 / BRIstuff 0.3 PRE1 / 1 HFC cards
Connecting calls between the 2 cards (1 NT mode, 1 TE mode)
The caller is always able to xfer the call, the callee only sometimes
(have not yet been able to
Goran,
Yes, these are possible. You can roll your own, or use an off the shelf
system like our "ITSP in a box". See my message of a few minutes ago at:
http://lists.digium.com/pipermail/asterisk-users/2005-December/136800.html
We'll have calling cards and callerid authentication as you descri
Hi,
after installing ooh323 i tried to register the Asterisk box to a cisco
gatekeeper.
In fact the Asterisk registers with the Cisco Gatekeeper. However after 300
sec (the default TTL) the Asterisk gets unregistered and never registers
again.
What follows is the h323_log:
[EMAIL PROTEC
Hi All,
I have a really funky problem, which I can't seem to pin point.I have a
setup that looks something like this:
SS7 Networks --SS7--> Veraz IGate4000 --SIP--> Asterisk
Now, Asterisk has a second connection, that looks like this:
Asterisk --PRI--> Avaya CTI
Now, I'll describe severa
Asterisk can authenticate by CLID - it's not a good
idea, though as CLID can be spoofed
- Original Message -
From:
Goran Donev
To: asterisk-users@lists.digium.com
Sent: Thursday, December 01, 2005 10:36
PM
Subject: [Asterisk-Users] Can Asterisk do
This?
I have a client who is looking for the proposed solution and
was wondering if any asterisk professionals know if this can be done by
asterisk.
Calling card platform.
Users calling in through local access numbers, they dial
local access numbers and make calls through the syst
Hi,
Is it possible to have an Asterisk act as a gateway to an Index PBX. I
would like to migrate users from Index to Asterisk, but need to have
some kind of mechanism for the 2 systems to communicate during the
migration.
I have read that this can be done by installing a dual port PRI card
Many people on the Asterisk lists have seen our demo of "ITSP in a box"
over the last few months. We've just uploaded the latest release, so you
may care to take a look. This version has:
- Improved menu structure.
- Voicemail access on the web.
- Fax storage.
- Customers can create IVRs and
Just signed up with Telefinity Dash911 (Dash911.com) and their solution
is pretty simple to implement for asterisk users. They basically
give you a 10 digit number that you use to route the 911 call to which
in turn passes it to their system where they perform a reverse lookup
on the caller ID to
Lots of options here. You could turn her DID into a hunt group that
rings both phones. You could write a small webapp that lets her toggle
which phone to use (the web app would edit the extensions.conf file and
then reload the configs in Asterisk)... or the easiest way is to simply
let her take the
Is the 941 available yet?
What improvements have they made.
Backlit display?
The 841 with the late firmware is a fairly nice phone, but the display
might as well not be there. And whoever decided button placement, size
and color should have been fired.
JMO
John Novack
Cory Andrews wrote:
Hi All,
I have an employee who works mostly in our office but
maybe once or twice a week has to work from home to help
care for her special needs child.
As background we have AAH 2.0 running with 8 analog lines
connected to two digium t400P cards. We have 10 sipura-841s
as handsets in the office
If anyone is in need of it, we now have a PDF copy
of the the Linksys SPA-941 admin guide. It's 92 pages, under 1MB zipped if
anyone has a public FTP or something they want to stick it in just email
me.
Thanks,
Cory AndrewsPurchasing
Manager++VOIPSupply.comA Division of
Hey all,
Anyone have shoutcast for MOH with Asterisk 1.2? If so, what should the
musiconhold.conf file look like? I found a bunch of references to older
config file syntax and have tried a few variations but can't seem to get
it to work. Any help would be greatly appreciated.
Thanks,
Da
On Thu, 2005-12-01 at 14:25 -0500, Bill Michaelson wrote:
> Just curious...
>
> Is there anyone out there who has given this outfit money and actually
> received any service from them?
>
I have no problems. Just prepaid outbound, cant comment on any other
services. They are in the process of
Now that you mention it, nice idea.
I think transcode or so can rearange 6.1 or 8.2 sound systems.
But no. It's just for recording simple inbound and outbound calls from
PSTN to a local extension. So no gang-bang ;-)
Hans
On Thu, 2005-12-01 at 10:49 -0800, Innocent Evil wrote:
> What you wanna to
Jan,
I would be really really interested (on or off list) to hear what you
are doing. Would you be able to find out if you can tell me more
details.
On 12/1/05, Jan Saell <[EMAIL PROTECTED]> wrote:
> Without going thru the ditail to much - im not shure that im allowed to
> reveal tom much - but
Yes,
I have something like this in my extension.conf
#include verizon.conf
and in verizon.conf file have verizon related dialplan
Thanks
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 15:5
Forgive me if this is old news...
http://www.spectrum.ieee.org.nyud.net:8090/oct05/1846
--
Cheers
Wayne
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This worked:
exten => NXXNXX,2,Set(CALLERID(num)=88)
>
> CLI*> show function CALLERID
>
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hello list,
inspired by a number of previous posts, I prepared this node in the
AstRecipes wiki:
http://www.oinko.net/astrecipes/index.php?n=107
It wants to be a good starting point to show low-level hardware problems
with zaptel (and PCI in general) cards.
Anybody has suggestions to be ad
Any body gotten this to compile chan_bluetooth under
1.2?
WHat steps did you take.
THanks,
Jerry
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Have you paid the $30k+ (US dollars) that it costs just to put the G723
codec on your Asterisk server?
Due to cost issues, Asterisk does not come the G723.1 codec. Simularly,
it does not come with the G729 codec, but you can download and license
that one for a very reasonable $10/connection (a one
Hi,
Anyone has a way to write (append) to text file from the dial plan?
Thanks,
Andre
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You can put them in after each other:
exten => s,4,BackGround(to-compose-a-message)
exten => s,5,BackGround(press1)
and if you whant some space you can add a wait inbetween.
If you whan to make it better you ahev to marge the files with a sound
editor outside asterisk.
Best regards
jan
--On
Bill Michaelson wrote:
Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
I have accounts with roughly 5 providers, with sixtel being one of the
primary ones. I have been using them since 2/2/2005. While there have
been occ
how i do to install 1 rpm from my cd rom?
i accessed with the root password but i navigate for all the directories
and
dont enough the cdrom directory, how i do that??
i need install the webmin program
thanks!
- Original Message -
From: "Henri Herscher" <[EMAIL PROTECTED]
If you want to do 100% recording, you can also do this using oreka
(http://www.oreka.org). It will mix both sides and optionally compress
the call to GSM as well. Also, if you are worried about the
stability/load of your asterisk when doing the recording, you can run
oreka on a separate box.
Cheer
What is the default username and password for [EMAIL PROTECTED]
a2billing module.
Thanks
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I have two servers connected via IAX2; one is connected to PRIs where I
receive CallerID along with CID text while the other is located over my
network connected to some channel banks providing analog dialtone. Relevant
output of "show channel" on the PRI box for one call is here:
CDR Variables
I get a similar warning with 1.2b1
Anyone have a clue as to what this means??
John Novack
asterisk183 wrote:
Why Asterisk show this message:
Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600
handle_response_register: Got 200 OK on REGISTER that isn't a register
Thanks
__
How is your agents.conf ? How is your login in
extensions.conf?
- Original Message -
From:
gc
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 2:53
PM
Subject: Re: [Asterisk-Users] Error on
using queue.
Thank you Kerry. I was able to download the firmware.
Does anybody know what files need to reside on the
tfpt server. If someone is willing to help get my
7970 phone functional again, I would really appreciate
it.
-John
You have to have a login to the Cisco site to download
the firmware.
-Kerry
Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
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Has anyone been able to set up a sip trunk between and Avaya S8700 and
Asterisk? I can't seem to find any good docs on the subject. Any help
would be greatly appreciated.
Thanks!
=A=
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Or put everyone in a Meetme room and record the conversation in the
meetme room -- just an idea.
- Waldo
On Dec 1, 2005, at 2:00 PM, Dave Walker wrote:
Use sox to make a quadriphonic (4 channels) audio file. Any more
than 4 in a call would be silly ;-)
Innocent Evil wrote:
What you wanna
Hi,
They are any succes stories with chan_bluetooth and one of the
following phone models?
- Ericsson R520m
- SonyEricsson T68i
- SonyEricsson W800i
I have tried with all of them with different kind of errors...
Thank you and best regard,
Dan
__
On Thu, December 1, 2005 17:09, Don Fanning said:
> I ended up buying a second 1 euro account because of this. But it does
> work fine.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
> Vargas
> Sent: Thursday, December 01, 2005 7:21 AM
>
> >
> >
> > Hi All
> >
> > I am using prepaid auth (callingcards), the idea is for a
> > prepaid support line. It is up and running but I have a
> > couple of questions with regards to modifications I would
> > like to make.
> >
> > When a user calls and they go through the process of entering
> >
Hugh L. Johnson wrote:
It worked, but...
SetCIDName() & SetCIDNum() are depreciated;
I figured SetCallerID() is on the way out, too.
I'd rather just touch part than have to mess with the whole.
On Thu, 2005-12-01 at 13:35 -0500, Tom Hayden wrote:
Why aren't you using the SetCallerID() cmd?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of scott
> Sent: Wednesday, November 30, 2005 11:52 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] prepaid application
>
>
> Hi All
>
> I am using prepaid auth (callingcards), the i
It worked, but...
SetCIDName() & SetCIDNum() are depreciated;
I figured SetCallerID() is on the way out, too.
I'd rather just touch part than have to mess with the whole.
On Thu, 2005-12-01 at 13:35 -0500, Tom Hayden wrote:
> Why aren't you using the SetCallerID() cmd?
Use sox to make a quadriphonic (4 channels) audio file. Any more than 4
in a call would be silly ;-)
Innocent Evil wrote:
What you wanna to do if there have more than 2 parties in the conversation ? !!
--
You don't have any choice, you already made it before you came here.
-Original
Didn't work.
On Thu, 2005-12-01 at 10:36 -0800, Innocent Evil wrote:
> Set(CALLERIDNAME="Innocent Evil")
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What you wanna to do if there have more than 2 parties in the conversation ? !!
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 19:27:45 +0100
> To: asterisk-users@lists.digium.com
> Subject:
Youch. That's quite the switch! I'm surprised you couldn't HEAR the
difference. :)
--
Tom
On 11/30/05, Steve Totaro <[EMAIL PROTECTED]> wrote:
> I think if you type show codecs in the CLI you can see what codecs are
> what by the number. It shows that you tried for g728 but got iLBC.
>
> > -
Try,
Set(CALLERIDNAME="Innocent Evil")
Thanks,
--
You don't have any choice, you already made it before you came here.
> -Original Message-
> From: [EMAIL PROTECTED]
> Sent: Thu, 01 Dec 2005 12:48:13 -0500
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Altering Incomi
Why aren't you using the SetCallerID() cmd?
--
Tom
On 12/1/05, Hugh L. Johnson <[EMAIL PROTECTED]> wrote:
> What do I need to do to alter incoming CallerID? The below isn't
> working...
>
> Running Asterisk 1.2 CVS HEAD
>
> exten => NXXNXX,1,Wait(1)
> exten => NXXNXX,2,Set(CALLERID(name)
Hi,
On 19:27, Thu 01 Dec 05, Hans Witvliet wrote:
> Hi all,
>
> Perhaps a newby question, perhaps something impossible.
> While waiting for my HW to arrive, i've been studying the wiki's and
> TFOT to be preparred when it comes. Info is overwhelming. It seems that
> anything is possible...
>
> I
Hi all,
Perhaps a newby question, perhaps something impossible.
While waiting for my HW to arrive, i've been studying the wiki's and
TFOT to be preparred when it comes. Info is overwhelming. It seems that
anything is possible...
Is it possible to record allways from begin to end an entire
convers
Hi Scott,
Yes, its possible
pass 'm' option to Dial command for MusicOnHold
If destination is unreachable, you need to get the return value of Dial
and from that value you will know whether a call was connected or not. Based
on that value you can execute Dial again or not.
You can put everything
sorry, this is mistake
--
[EMAIL PROTECTED]
> http://www.mujtelefon.com
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On Fri, 2005-12-02 at 01:52 +0800, Steve Underwood wrote:
Ho Carlos,
When you build the software specify an install directory explicitly on
the command line, like:
./configure --prefix=/usr/local
There is an error in the configuration files when you let the
installation default to /u
Without going thru the ditail to much - im not shure that im allowed to
reveal tom much - but we are using a webservice to update their database.
Best regards
jan
--On Thursday, December 01, 2005 08:21:23 AM -0500 Matt <[EMAIL PROTECTED]>
wrote:
How are you updating the E911 address informa
On 09:47, Thu 01 Dec 05, Giovanni Miano wrote:
> If u want kernel 2.6 dont use SMP support
Why not ?
Seems to workout pretty nice here.
Intel 865 board with HyperThreading P4 3Ghz.
Linux 2.6.10 SMP PREEMPT HIGHMEM
Haven't seen any trouble here.
--
Michiel van Baak
http://michiel.vanbaak.info
[E
Ho Carlos,
When you build the software specify an install directory explicitly on
the command line, like:
./configure --prefix=/usr/local
There is an error in the configuration files when you let the
installation default to /usr/local. If you specify it, things work. The
next revision wi
What do I need to do to alter incoming CallerID? The below isn't
working...
Running Asterisk 1.2 CVS HEAD
exten => NXXNXX,1,Wait(1)
exten => NXXNXX,2,Set(CALLERID(name) = "Fred")
exten => NXXNXX,3,NoOp(${CALLERID(name)})
-- Executing Wait("IAX2/A-9", "1") in new stack
-- Executing
Hi:
I want to use the same phone number for the fax and voice conversations.
If it is a fax calling, I don't want any interactive menu, I just want
to
redirect the calling to the iaxmodem extension, and if is a normal
calling
the interactive menu will be deployed. How can I detect that is fax
call
I am installing a new Asterisk server that needs mfcr2 on a machine running Fedora Core 4. I have compiled both asterisk and spandsp without any problem. On the last step, compiling libmfcr2-0.0.3 I get the following error:
libtool: link: only absolute run-paths are allowed
make[1]: ***
http://www.mujtelefon.com
--
[EMAIL PROTECTED]
> Alejandro Vargas wrote:
> btw. does anyone have a definitive list of all the finarea VOIP
> companies? I can think of:
> call1899
> call18866
> voipbuster
> sipdiscount
> voipcheap (note: this one uses a proprietary protocol, similar to IAX
>
Benoît Mérouze wrote:
Hello,
On http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi,
it's said that stream_file() might returns "-1 on error or hangup, 0
if playback completes without a digit being pressed, or the ASCII
numerical value of the digit if a digit was pressed".
Bu
Hi,
My IP Phone is using well G.723.1 because when i am testing it with
another SIP GK, working well with G.723.1.
But the problem is only accuring in Asterisk, my sip.conf is already
having the configuration of this codec.
[123456]
disallow=all
allow=g723
--
Thank You,
Cod
Tony Hoyle a écrit :
[...]
call1899
call18866
voipbuster
sipdiscount
voipcheap (note: this one uses a proprietary protocol, similar to IAX
but over different ports and not compatibile).
NetAppel.fr
--
Daniel
___
--Bandwidth and Colocation provided b
On Thu, 2005-12-01 at 11:40 -0500, C F wrote:
> On 11/28/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
> > Enforcement Bureau Outlines Requirements of November 28, 2005
> > Interconnected Voice Over Internet Protocol 911 Compliance Letters
> > http://www.fcc.gov/eb/Public_Notices/DA-05-29
Thanks. I made change to joinempty=yes. And now I
can hear the music on hold. But it would not ring the agent even if I login
agent in. When I run show queue command under CLI, I got these
messages:
queue1 has 1
calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2,
I guess this one answers some questions, and it also gives me someone
bigger than me (for now anyhow :) ) that will fight it for me:
https://www.stanaphone.com/index/news_Nov2205.html
On 12/1/05, C F <[EMAIL PROTECTED]> wrote:
> On 11/28/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
> >
On 11/28/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote:
> Enforcement Bureau Outlines Requirements of November 28, 2005
> Interconnected Voice Over Internet Protocol 911 Compliance Letters
> http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html
I'm just trying to clarify this, according
Hello,
On http://www.voip-info.org/tiki-index.php?page=Asterisk%20perl%20agi,
it's said that stream_file() might returns "-1 on error or hangup, 0 if
playback completes without a digit being pressed, or the ASCII numerical
value of the digit if a digit was pressed".
But actually when I hangu
announce is exactly what I'm looking for. I had originally thought
that meant playback for the caller, not the agent who answers the
call. If I had time I'd add that to the wiki, since it needs to be
there, and not buried in an example.
On 12/1/05, Lenz <[EMAIL PROTECTED]> wrote:
>
> Hi Trey,
>
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Pablo Allietti wrote:
> On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote:
>
> Looks like your zap channels are droping into the default context...
> better to set up a from-pstn context and start there.
>
>
>
>
>> hi sean you have a e
Alejandro Vargas wrote:
But I tried the username and passwrord of an account with credit, and
the registration is refused. With the voipbuster propietary software
it works ok (I sniffed the packets and I think it is not using
standard iax or sip ports). Are the acconts with credit blocked for
avo
My incoming BV has been intermittant for the last two days as well. It
has gone down somewhere around 4:30 PM Eastern two days in a row, then
been back up in the morning. In the 10:00 AM hour today, it was down for
about ten minutes.
Jason Schafer writes:
> I have been trying on and off for a coup
I ended up buying a second 1 euro account because of this. But it does
work fine.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alejandro
Vargas
Sent: Thursday, December 01, 2005 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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