Hi,
try to delete old .wav ones.
Why not using a sounds/nz subfolder and set language to nz?
Giorgio
Lists wrote:
Hi all,
I am wanting to change the sound files from the standard ones to a New
Zealand voice pack.
I have copied the files into the /var/lib/asterisk/sounds directory and
On Thu, Jul 10, 2008 at 10:18:21AM +0200, Giorgio Incantalupo wrote:
Lists wrote:
Hi all,
I am wanting to change the sound files from the standard ones to a New
Zealand voice pack.
I have copied the files into the /var/lib/asterisk/sounds directory and
chowned them to
Call Flow:
1) Extension 10 calls out
2) Extension 10 transfers the call to extension 20
3) Extension 20 picks up the call.
Right when 10 transfers the call to 20 the h extension is invoked for extension
10. Why is this ? Is there any way to have the h extension not called on a
transferred call
I have a mixed PBX system with both Asterisk 1.4.21 and 1.2.27 (moving to
1.2.28).
For now I need to keep a few boxes in 1.2 and not migrate them all to 1.4.
However, I would like to have func_odbc and res_odbc on all servers.
On 1.4.21, native func_odbc seems to work fine.
On 1.2.27, the
This is what I use. The Read does have a default timeout but you should
be able to put your own.
extensions.conf:
exten =
s,n(dial),Dial(SIP/sipura2_1SIP/sipura1_1SIP/sipura2_2SIP/spa942_3SIP/aastra480_3,20,mtTM(screen))
exten = s,n(vmail),Voicemail([EMAIL PROTECTED])
[macro-screen]
Hi list,
My caller ID is not working anymore on my TDM11B (TDM400P) cards and i get
this error message on the asterisk console:
== Starting post polarity CID detection on channel 4
-- Starting simple switch on 'Zap/4-1'
[Jul 8 11:58:55] WARNING[9539]: callerid.c:219 callerid_get_dtmf:
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report bugs
for people from
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
if you still want donate t.38 development please contact me at cervajs at
Hi,
I'm interested if it's possible to configure Asterisk the following way:
user calls a huntgroup, and then when one of the hunts answers the call,
other hunts are not hung up, but Asterisk transfers the callees to some
extensions, or something else.
--
Best Regards
Alexander Olekhnovich
I just think because of the Asterisk design it can not be implemented.
On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich
[EMAIL PROTECTED] wrote:
Hi,
I'm interested if it's possible to configure Asterisk the following way:
user calls a huntgroup, and then when one of the hunts answers
Jerry Geis wrote:
What should I do now?
silly me it is 1.4.21.1 not 1.2.21.1
If you haven't already, I'd suggest reporting an issue in mantis.
http://bugs.digium.com/
--
Sean Bright
[EMAIL PROTECTED]
___
-- Bandwidth and Colocation
As RTP packets have a sequential number, is there some logging/debugging option
in Asterisk to monitor how many packets have been lost on a SIP call?
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Passo Fundo - RS - Brasil
+55 54 2104-7000
Steve Underwood wrote:
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report bugs
for people from
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
if you still want donate t.38 development please contact
Alexander Olekhnovich wrote:
I just think because of the Asterisk design it can not be implemented.
On Thu, Jul 10, 2008 at 3:16 PM, Alexander Olekhnovich
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
I'm interested if it's possible to configure Asterisk the
following
For example, here is a dial plan to call a huntgroup (111,222,333)
[reach-hunt]
exten = _X.,1,Dial(SIP/111SIP/222SIP/333|timeout|G(extra-context^s^1))
exten = _X.,2,Hangup()
Here in the dial plan if one of the numbers will answer the call it will be
transferred to extension 2, and the caller to
Tomorrow at 12 Noon EDT, we'll exchange some views and notes about
SIP, the protocol you usually can not avoid even if yiou wanted to.
- Past, present and future of SIP
- SIP greatest strengths and weaknesses.
- What's the state of chan_sip in asterisk code
- anything else anyone wants to add
Hi All;
I do not know why when I select the codec to be G729, it keeps display it on
the Asterisk CLI during the call as following:
G729
G729
G729
...
...
Any advise how to let this stop?
Regards
Bilal
___
-- Bandwidth and Colocation
Obviously I meant JULY 11th.
On Thu, Jul 10, 2008 at 3:17 PM, randulo [EMAIL PROTECTED] wrote:
Tomorrow at 12 Noon EDT, we'll exchange some views and notes about
SIP, the protocol you usually can not avoid even if yiou wanted to.
- Past, present and future of SIP
- SIP greatest strengths
I agree. In that case people who use includes in their scripts for which they got paid should pay a portion of their pay to the writer of each include they use.
Original Message
Subject: Re: [asterisk-users] (announce) asterisk T.38 gateway
From: Rob Hillis [EMAIL
Replying to myself.
A reload isn't enough in 1.2.27. I needed to restart asterisk.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now:
When people release software under the GPL license, like Steve Underwood did
with libunicall, spandsp and so on, they were supposed to know that other
people has the right to use their code.
Atenciosamente,
Vinícius Fontes
Núcleo de Tecnologias Convergentes
Canall Tecnologia em Comunicações
Hi Mark,
I have no issue with people using something in new and interesting ways.
Adding a small wrapper around a large library, and asking for a bounty
is in a rather different category from that.
At least this is honest work. You should see some of the sleazy ways
people have made money
I've had a similar problem with a A101DX + A202DX. I was trying to bridge from
my A101 to my A202 to get faxes over my E1 line. I've done an number of things,
I'm not exactly sure which one helped, but it now works very nice for fax:
(1) Using zap show channel N on the CLI I noticed that
Vinícius Fontes wrote:
When people release software under the GPL license, like Steve Underwood did
with libunicall, spandsp and so on, they were supposed to know that other
people has the right to use their code.
The problem is that almost any licence term which tries to limit the
Hi Ryan,
On Wed, Jul 09, 2008 at 05:54:28PM -0400, Ryan M. Colbert wrote:
I'm trying to build a simple accept/reject screening app for inbound calls
that * forwards to my cell phone. Basically I want * to announce the caller
ID and then let me press 1 to accept the call or 2 to reject the
Hi,
I have an Asterisk 1.2.18 box with a TDM400P card.
If I make a call and then I hangup the phone, the call ends correctly
but if I receive a call and I hangup the phone the other party does not
get the hangup signal from Asterisk.
Is there anybody who can explain this strange behaviour?
I'm sorry to be more concrete I want to make conference with all the hunt
group and caller by just dialing the HuntGroup.
On Thu, Jul 10, 2008 at 4:18 PM, Alexander Olekhnovich
[EMAIL PROTECTED] wrote:
For example, here is a dial plan to call a huntgroup (111,222,333)
[reach-hunt]
exten =
On Wed, 2008-07-09 at 17:54 -0400, Ryan M. Colbert wrote:
I'm trying to build a simple accept/reject screening app for inbound
calls that * forwards to my cell phone. Basically I want * to
announce the caller ID and then let me press 1 to accept the call or 2
to reject the call and send the
On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED]
wrote:
Vinícius Fontes wrote:
When people release software under the GPL license, like Steve Underwood
did with libunicall, spandsp and so on, they were supposed to know that
other people has the right to use their code.
On Thu, Jul 10, 2008 at 11:43 AM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Jul 10, 2008 at 10:24 AM, Steve Underwood [EMAIL PROTECTED]
wrote:
Vinícius Fontes wrote:
When people release software under the GPL license, like Steve Underwood
did with libunicall, spandsp and so on, they
On Tuesday 08 July 2008 11:53:47 Cyril SCETBON wrote:
hi,
I'im using asterisk 4.1.21 and astrundir is configured as followed in
/etc/asterisk/asterisk.conf :
[global]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir = /var/lib/asterisk
astagidir =
I have used this [EMAIL PROTECTED] for a while now, 2 years.
Only recently, I am trying Festival and on invoking festival --server I get
these errors :
/usr/share/festival/bin/festival: /usr/lib/libstdc++.so.5: version
`CXXABI_1.2' not found (required by /usr/share/festival/bin/festival)
So, I've been asked if this is possible.
Someone wants to actively monitor the duration of a call, while the call is
still in progress. Obviously, in Asterisk, once the Dial() application starts,
you lose dial plan control until after the call has ended, successful or
otherwise.
Anyone know
Hi.
We are building an application that will provide users with the ability to
call in and report an absence. The caller will have to validate themselves
and the call tree will be dynamic, based on data in a MySQL database. We
will have many customers, each calling a separate phone number, each
On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED] wrote:
Hi.
We are building an application that will provide users with the ability to
call in and report an absence. The caller will have to validate themselves
and the call tree will be dynamic, based on data in a MySQL
Yes,asterisk can do that
On Thu, Jul 10, 2008 at 12:25 PM, Mark Carpenter [EMAIL PROTECTED] wrote:
Hi.
We are building an application that will provide users with the ability to
call in and report an absence. The caller will have to validate themselves
and the call tree will be dynamic,
Hey,
I am doing a similar project , which we will be integrating mysql db and a
ivr, maybe we can work on this together since we will be sharing components.
This should save us both some time.
On Thu, Jul 10, 2008 at 12:25 PM, Mark Carpenter [EMAIL PROTECTED] wrote:
Hi.
We are building an
I'd probably be a little pissed if I were Steve Underwood if somebody
pocketed over 10k $USD for taking credit for a product that my free library
did the bulk of the work for.
I don;t think i'd feel that the entire bounty should be mine - after all
there would of been nothing stopping me from
Admittedly I have not used the ExternalIVR app. Is it any good?
I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, it can do
it, but boy it is UGLY. There's also the fact that you can't call Backgound()
in a macro, which forces you to use Read() which won't accept a timeout of
From what I can tell Read allows for a floating point input which uses
ast_waitfordigit that accepts milliseconds as input.
Douglas Garstang wrote:
Admittedly I have not used the ExternalIVR app. Is it any good?
I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure,
it can do
Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the input in
seconds. If you try and use 0, it seems to drop back to a default of 5s.
- Original Message
From: MFH [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I have a system that is driving me nuts. My customer is running
Asterisk 1.4.20.1 on a CentOS 5.2 server. It is a purely SIP and IAX2
service with no cards installed and it uses ztdummy from Zaptel 1.4.11.
They use Teliax for calls to the USA and Protel for calls in Mexico.
The
marek cervenka wrote:
hi,
there is T.38 fax gateway for asterisk
http://bugs.digium.com/view.php?id=12931
please test it and report bugs
for people from
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
if you still want donate t.38 development please contact me at cervajs at
I only did the 420 because thats what the original files looked like?
r-- -w- ---
Should I change this to 644?
Kate
Tzafrir Cohen wrote:
On Thu, Jul 10, 2008 at 10:18:21AM +0200, Giorgio Incantalupo wrote:
Lists wrote:
Hi all,
I am wanting to change the sound files from the standard
-Original Message-
Subject: [asterisk-users] Diagnosing dropped calls...
I have a system that is driving me nuts. My customer
is running Asterisk 1.4.20.1 on a CentOS 5.2 server. It is a
purely SIP and IAX2 service with no cards installed and it
uses ztdummy from Zaptel
My customer has a 10mpbs fiber connection to the Internet so we have
always assumed that the connection is not really a problem. We will
look into it. Thank you.
On Thu, 2008-07-10 at 17:49 -0500, John Faubion wrote:
-Original Message-
Subject: [asterisk-users] Diagnosing
Hi all,
I am very new to asterisk and I am just looking through the config files
to try and understand them a bit.
In my zapata-auto.conf file I have
; Span 2: WCTDM/1 Wildcard TDM400P REV I Board 2
;;; line=5 WCTDM/1/0 FXOKS (In use)
signalling=fxo_ks
callerid=Channel 5 6005
mailbox=6005
Matt Watson wrote:
That being said... i;m also quite pleased to see T.38 support being
worked on for Asterisk... its a pretty important area to further
develop IMHO.
I absolutely agree. It's been a notable omission for some time.
Unfortunately getting it written isn't the major part of
Why can't you call Background() from a MACRO ?
Isn't is just an Application like any other ?
Curious minds want to know !
Quote There's also the fact that you can't
call Backgound() in a macro,
Douglas Garstang wrote:
Don't know about Asterisk 1.4, but in Asterisk 1.2 it expects the
input
Yes , you could easily do this with asterisk.
If you have formal specs for this project, I would be interested in exactly
what you are trying to do. Email me off-line.
Steve Totaro wrote:
On Thu, Jul 10, 2008 at 1:25 PM, Mark Carpenter [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Fri, Jul 11, 2008 at 09:56:29AM +1200, Lists wrote:
I only did the 420 because thats what the original files looked like?
r-- -w- ---
Should I change this to 644?
Yes!
--
Tzafrir Cohen
icq#16849755 jabber:[EMAIL PROTECTED]
+972-50-7952406
It's a known problem.
If you call Background() in a macro, then Asterisk will look for the extensions
to jump to in the CALLING Macro/context and NOT the Macro that the Background()
app was called in.
Eg:
[macro-DoSomething]
exten = s,1,Background(Prompt)
exten = 1,1,NoOP()
[context1]
exten
On Fri, Jul 11, 2008 at 11:36:49AM +1200, Lists wrote:
Hi all,
I am very new to asterisk and I am just looking through the config files
to try and understand them a bit.
In my zapata-auto.conf file I have
; Span 2: WCTDM/1 Wildcard TDM400P REV I Board 2
;;; line=5 WCTDM/1/0 FXOKS (In use)
Try dropping the IAX2 and only use SIP. Don't ask why? Just give it a
try and see if things improve for you.
Also when you assume, you make and ass out of you and me (just a little
joke, get it? ass-u-me.)
You could be hitting an overloaded router or whatever along the way, 10mbs
fiber does
On Thursday 10 July 2008 18:48:15 Rob Hillis wrote:
Matt Watson wrote:
That being said... i;m also quite pleased to see T.38 support being
worked on for Asterisk... its a pretty important area to further
develop IMHO.
I absolutely agree. It's been a notable omission for some time.
On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
It's a known problem.
If you call Background() in a macro, then Asterisk will look for the
extensions to jump to in the CALLING Macro/context and NOT the Macro that
the Background() app was called in.
I wouldn't call it a known
On Thu, Jul 10, 2008 at 9:07 PM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Thursday 10 July 2008 19:13:50 Douglas Garstang wrote:
It's a known problem.
If you call Background() in a macro, then Asterisk will look for the
extensions to jump to in the CALLING Macro/context and NOT the
Another update on the latest hookup attempt. I can make it work
reasonably well with callprogress=yes, it detects the hangup but only
after about 7-9 seconds. My config files are the same as the last
time I posted (apparently last time I wasn't waiting long enough for
callprogress to
I´ve managed to put it to work, very simply.
Just created an A DNS entry pointing to my system. This procedure validates
the reverse lookup, gmail and others do, before accepting the mail in their
inboxes.
All my sendmail emails gets delivered with no need of smarthosts, therefore,
no need to SSl
Try dropping the IAX2 and only use SIP. Don't ask why?
Well in our case we were NOT using IAX at all. Strictly SIP.
You could be hitting an overloaded router or whatever along
the way, 10mbs fiber does not mean low latency or lost packets.
So true, hence the reason I suggested using mtr to
Hi! I am a newbie using Asterisk. I am developing an IVR using perl from AGI
and Cepstral as voices
The AGI is this
use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
#
$AGI-say_number('9865');
60 matches
Mail list logo
