http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords
Cheers
2005/12/1, Joe Pukepail [EMAIL PROTECTED]:
Look into the findme feature, there is a patch on the bug tracker to add
this feature. I believe that someone shows how to do it in the dial plan.
I plan on
If u want kernel 2.6 dont use SMP support
I use asus with celeron,amd and it works fine.
2005/12/1, John Brookes [EMAIL PROTECTED]:
I am putting together a box to run asterisk.
Which version of linux and MB-cpu do you suggest?
TIA
John B
___
U can use AMI (asterisk management interface)
write small application that connect to ami and check voicebox status
Configure manager.conf and try telnet [asterisk's ip] 5038]
2005/12/1, Hiu Yen Onn [EMAIL PROTECTED]:
I have been using xlite client, FOC. There is no sign of image displayed
on
Hi,
I want to setup the MWI on the Optipoint 410 Standard SIP.
Until now haven't found any information about this.
Probably anyone know the right way and other tips for this phone?
Thanks and regards
Wolfgang
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Hi!
I just built Asterisk on Debian Sarge myself and it worked without any
problems.
Can you cut 'n paste the error messages?
I can't make any sense from the output ...
Greetings,
Marcus
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hagen Rode
Hi
as suggested in the group
I have downloaded the [EMAIL PROTECTED]
installed one of my PC for testing
and made 2 extentions to test
iam able to talk each other
now i have setup one Trunk
and made Out going
when ever i call to out side
i get a voice tone saying that all trunks are busy
how
Have u availability tranks ?
2005/12/1, ram [EMAIL PROTECTED]:
Hi
as suggested in the group
I have downloaded the [EMAIL PROTECTED]
installed one of my PC for testing
and made 2 extentions to test
iam able to talk each other
now i have setup one Trunk
and made Out going
when ever i
yes
i got new account from provider
and i have registered
no one using, iam using that account for testing
ram
On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Have u availability tranks ?2005/12/1, ram [EMAIL PROTECTED]
: Hi as suggested in the group I have downloaded the [EMAIL PROTECTED]
Hi Marcus,
haven't you got an Unable to initialize mISDN error during asterisk
startup?
I have a problem with chan_misdnI'm trying to understand where is
the prob...I haven't recompiled my kernel with mISDN support because
Digium claims it is inside Asterisk 1.2, maybe it's my
type in console: sip show registry
and verify status of your trunk
2005/12/1, ram [EMAIL PROTECTED]:
yes
i got new account from provider
and i have registered
no one using, iam using that account for testing
ram
On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote:
Have u
HI:
I ve downloded asterisk 1.2 and when i tried to dial
on console this message appears:
No such command 'dial' (type 'help' for help)
Despite iam not running any audio softwares.
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
Hi
here is the results
asterisk1*CLI sip show registryHost Username Refresh Statex.x.x..2:5060 xx 105 Registered
i have edited the orginals
ram
On 12/1/05, Giovanni Miano [EMAIL PROTECTED] wrote:
type in console: sip show registryand verify status of your trunk2005/12/1, ram
[EMAIL
pastme context for outgoing
2005/12/1, ram [EMAIL PROTECTED]:
Hi
here is the results
asterisk1*CLI sip show registry
HostUsername Refresh State
x.x.x..2:5060 xx 105 Registered
i have edited the orginals
ram
On
On 12/1/05, Rob Thomas [EMAIL PROTECTED] wrote:
After upgrading to 1.2.0 (from a three-week-prior CVS version), I've
suddenly had people starting to complain of lost calls. They'd be there,
and suddenly they'd drop out - they could be in a conversation, or more
often, the caller would be on
hi
iam not sure is the right answer iam posting
let me know is this correct or not
Sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)disallow=allallow=ulawallow=alawcontext = from-sip-external ; Send
Context in extensions.conf
2005/12/1, ram [EMAIL PROTECTED]:
hi
iam not sure is the right answer iam posting
let me know is this correct or not
Sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0; Address to bind to (all
Hi all,
I was trying to use G.723.1 codec for my terminator as Pass through.
But when the second party pickup phone the call is going dropted
automatically with the following error:
No path to translate from SIP/123456-fca7(1) to SIP/myterminator.com-ff11(4)
Dec 1 10:54:39 WARNING[7480]:
On 11/25/05 18:32 Olivier Taylor said the following:
Yes, beta2 works perfectly, but 1.2 released version gives me this error.
looks like you did not clean out your modules directory when you installed
1.2 over 1.2 beta. try doing that and reinstalling.
--
Regards,
Hi
Thanks for the replies.
Its fixed now. The problem I had was that I had a
mixed Debian Stable and Unstable system. Some of the libraries I downloaded
from Unstable caused breakages. Basically, long story short, I re-installed
Debian, got all the required packages from Stable
On Wed, Nov 30, 2005 at 01:17:33PM -0500, Sean Cook wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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 example please?
Pablo Allietti
2005/12/1, Pablo Allietti [EMAIL PROTECTED]:
hi sean you have a example please?
In your zapata.conf ensure there is context=from-pstn like this:
[channels]
context=from-pstn
--
Alejandro Vargas
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I'm using mediatrix mgcp device without problems with [EMAIL PROTECTED]
2.0 over the LAN. But now I trying one of this devices through
internet. My firs problem was nat, but I decided to leave this problem
for later and try it through a vpn. I used gvpe because it is very
transparent. The device
2005/12/1, Tejas Shah [EMAIL PROTECTED]:
I am a newbie to asterisk. I installed a asterisk server to make
communication between 2 X-Lite's SIP based phones. I made following
configuration in sip.conf :
For newbies (like me) a good start is to use amp or install directly
Thanks Mr.Miano
Thanks a lot. Now I think I wont have to bother about balming all my problems
to zapata. I have also succeeded quite a bit and installed a basic PBX system
without it.
Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
Please advice me how i can make it work?
It looks like your Phone is not compatible to G.723.1 or this codec is
disabled within sip.conf
Elmar
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I believe you have to have a sound card.
-Original Message-
From: jonny hashem [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 01, 2005 5:16 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] cannot dial on console on asterisk 1.2
HI:
I ve downloded asterisk 1.2
Is there any way to tell asterisk that ignore protocol errors instead
of dropping the call?
--
Alejandro Vargas
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I am looking about those two script because I am
not able to find them on www.generationd.com
May some one help me please?
Thanks
Rosario
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How are you updating the E911 address information?
We have literally been pulling teeth at Verizon to get access to their
PS/ALI database to make the updates that we need to.
On 11/30/05, Jan Saell [EMAIL PROTECTED] wrote:
Just a small note that we have used a cluster of asterisk to connect or
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as ACD
agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing
Hi Tim
Thanks for the info.
I see what your example is doing.
However what if I want Asterisk to call someone
that isnt on the local network?
So if someone is out and about they can be
called on a mobile to let them know something is down?
Tony
From:
[EMAIL PROTECTED]
nobody has problems like me?
Von: René Enskat [Teamware GmbH]
[mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 1. Dezember 2005
08:35An: 'asterisk-users@lists.digium.com'Betreff:
App_rxfax problem
When i load the fax
modules into the asterisk i got this errors but compile was
ok!
I have the
check /var/log/asterisk/full
2005/12/1, René Enskat [Teamware GmbH] [EMAIL PROTECTED]:
nobody has problems like me?
Von: René Enskat [Teamware GmbH] [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 1. Dezember 2005 08:35
An:
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.
They are then asked
Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08
WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so:
undefined symbol: fax_set_phase_d_handler
Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so
failed!
-Ursprüngliche Nachricht-
Von:
When I arrived a call, I would the call transfer in to another telephone number, but Asterisk show error: Executing GotoIfTime("Zap/4-1", "08:30-12:30|mon-fri|*|*?4") in new stack -- Executing GotoIfTime("Zap/4-1", "15:30-18:30|mon-fri|*|*?4") in new stack -- Executing Goto("Zap/4-1", "6") in new
nobody has problems like me?
---
== Registered application 'StartMusicOnHold'
== Registered application 'StopMusicOnHold'
[app_rxfax.so]Warning, flexibel rate not heavily tested!
Warning, flexibel rate not heavily
Matt Riddell wrote:
dashy dude wrote:
Dear All
I am trying to build a high availability cluster of
asterisk.
I am using RedHat cluster management suit on
Enterprise edition AS3
Origianally, astdb was located on native hard disk of
each server.
All my end points are configured for Reinvite=Yes
From: Jan Saell [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 30, 2005 9:32 AM
Subject: Re: [Asterisk-Users] Blind transfer question
I did a quick check on the blindxfer config parameter and i cant find
I just sent the error in full log:
Dec 1 15:01:08 VERBOSE[27950] logger.c: [app_rxfax.so]Dec 1 15:01:08
WARNING[27950] loader.c: /usr/lib/asterisk/modules/app_rxfax.so:
undefined symbol: fax_set_phase_d_handler
Dec 1 15:01:08 WARNING[27950] loader.c: Loading module app_rxfax.so
failed!
Is there any documentation for the complete removal of
Asterisk from a Linux/Unix system? I want to install a fresh copy of asterisk.
Thanks,
Chip
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Hi,
I am using Asterisk Business Edition A.1.6 (but I
guess it is the same logic for 1.2)
I am running the show queue
command for a queue that had a 36 calls and the C: parameter is growing up very
fastly, no reflecting the real calls to this queue.
lv09*CLI show queue cobranca
cobranca
I find the transfer functions a little lacking.
Examples:
I get a call
I do an attended transfer, but the called extension never answers/I get
impatient/I discover I have dialed the wrong extension.
I can not get the call back.
If I hangup, the caller is also hung up. I'd prefer the caller to
your distro shoudl supply uninstaller, something like emerge unmerge
asterisk in gentoo. Tough that would not remove your configuration
files.
Best RegardsOn 12/1/05, cp [EMAIL PROTECTED] wrote:
Is there any documentation for the complete removal of
Asterisk from a Linux/Unix system?
On Thu, 2005-12-01 at 15:50 +0100, Leif Neland wrote:
I find the transfer functions a little lacking.
Examples:
I get a call
I do an attended transfer, but the called extension never answers/I get
impatient/I discover I have dialed the wrong extension.
I can not get the call back.
Iirc
On 1 Dec 2005, at 13:33, Tony Spencer wrote: Hi Tim Thanks for the info.I see what your example is doing.However what if I want Asterisk to call someone that isn’t on the local network?So if someone is out and about they can be called on a mobile to let them know something is down?Just put a
Hi,
Is there any documentation for the complete removal of Asterisk
from a Linux/Unix system? I want to install a fresh copy of asterisk.
Depend of your distro,
You can use emerge for Gentoo or rpm with CentOS/RHEL/Fedora. Debian
also have dpkg command. If you have installed from
I was testing voipbuster. With a new account, with no credit, I can
make calls perfectly but of 1 minute.
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
Hi guys I have a question, im trying asterisk realtime in 2 servers.
Im trying to make calls from one server to another, example I call a
sip registered in sip server 1 with a phone register in sip server2
and both using the same database and family both use canreinvite=yes
but still cant
- Original Message -
From: Patrick [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 01, 2005 4:00 PM
Subject: Re: [Asterisk-Users] Better transfer
On Thu, 2005-12-01 at 15:50 +0100, Leif Neland
If you are using 1.2, it might be the joinempty and
leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 11:27
AM
Hi Trey,
It is done automagically by the system - see the setting named announce
in the queue definition.
Hope this helps
l.
On Wed, 30 Nov 2005 20:38:20 +0100, Trey Blancher
[EMAIL PROTECTED] wrote:
I want to play a file for an agent that answers a queue call, before
the agent is
On Thu, 2005-12-01 at 16:30 +0100, Leif Neland wrote:
[snip]
Unless disconnect above really means abort transfer
Yup and you could have found that out easily by trying it :)
Regards,
Patrick
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Since upgrade to BE A.1-6I get the following
messages on my console...
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-2-3 format: sln,
0x9e454b8
And several .sln files are saved on
/var/spool/asterisk/meetme/
What do this mean?
Thank you
Dov
From the asterisk source directory a
make uninstall
should also do it.
On 12/1/05, Joel Vandal [EMAIL PROTECTED] wrote:
Hi,
Is there any documentation for the complete removal of Asterisk
from a Linux/Unix system? I want to install a fresh copy of asterisk.
Depend of your distro,
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
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
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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
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 example
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,
It
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 hangup
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 to
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:
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,
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
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
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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,
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.
But
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
but over
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]: ***
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
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
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
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
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
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
sorry, this is mistake
--
[EMAIL PROTECTED]
http://www.mujtelefon.com
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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
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
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...
Is it
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) = Fred)
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 Incoming
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.
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:
Didn't work.
On Thu, 2005-12-01 at 10:36 -0800, Innocent Evil wrote:
Set(CALLERIDNAME=Innocent Evil)
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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.
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 idea is for
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?
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.
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
To:
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
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
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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Just curious...
Is there anyone out there who has given this outfit money and actually
received any service from them?
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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.
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.
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
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:
What is the default username and password for [EMAIL PROTECTED]
a2billing module.
Thanks
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