Hi,
I had some experience on Asterisk 1.0.7 and 1.2.0.
Now, I want to do something on the New Release of Asterisk 1.6.xx.
I want to know wheather there are already web GUI for use now in the
release.
Or still nedd integrate some other third part GUI?
Any advice will be appreciated.
On Sat, Apr 11, 2009 at 11:52:51AM -0400, John Rogers wrote:
Most ATAs I've seen are primarily SOC (System On Chip) implementations.
I've never really taken one apart, but perhaps now is a good time. I
recently also purchased a QuickPhones QA-342 wifi rechargeable handset:
2009/4/20 Gary Li garyli0...@gmail.com
Hi,
I had some experience on Asterisk 1.0.7 and 1.2.0.
Now, I want to do something on the New Release of Asterisk 1.6.xx.
I want to know wheather there are already web GUI for use now in the
release.
Or still nedd integrate some other third part
Can this be used in the same way as Callweaver works, IE: to invoke Sendfax by
placing (using mv command) a job description file in it?
Michael
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To
On Mon, 20 Apr 2009, Yahya Mohammad wrote:
I prefer the Nokia E-series wifi enabled cell phones that have a SIP
client builtin. I have an E61i and it works great in a wireless hotspot.
In places where SIP won't work for some reason, I register the phone to
asterisk on my laptop which then
Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes:
There's already a working theory and a patch:
http://bugs.digium.com/view.php?id=14932
Would it be possible to adapt the kernel stack checker for Asterisk? I
can easily believe it would be too much work, though.
/Benny
Michael mich...@networkstuff.co.nz writes:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid
Fax over T38 is failing, on the same system it worked with Callweaver.
What do I need to post to be get further assistance please?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
I think that all existing GUIs will in time migrate to Asterisk 1.6, if they
are not already supporting it, so using your favourite one should not be
much of an issue.
l.
2009/4/20 Gary Li garyli0...@gmail.com
Hi,
I had some experience on Asterisk 1.0.7 and 1.2.0.
Now, I want to do
Benny Amorsen wrote:
Michael mich...@networkstuff.co.nz writes:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER'
On Mon, Apr 20, 2009 at 3:47 AM, Michael mich...@networkstuff.co.nz wrote:
Can this be used in the same way as Callweaver works, IE: to invoke Sendfax by
placing (using mv command) a job description file in it?
Yes. Callfiles can be used to initiate faxes.
On Mon, Apr 20, 2009 at 5:34 AM, Michael mich...@networkstuff.co.nz wrote:
Fax over T38 is failing, on the same system it worked with Callweaver.
Some people claim great success with callweaver. If Callweaver is
working great for you, why change what works?
What do I need to post to be get
On Monday 20 April 2009 05:46:17 Rob Hillis wrote:
Benny Amorsen wrote:
Michael mich...@networkstuff.co.nz writes:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1'
in extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650
Please, is there anyone who can help me with this zaptel -- Dahdi
-problem ??
Will chan_dahdi.conf work with zaptel.conf ? Will Asterisk be able to
communicate with the Digium TDM pci-card ?
Or do I need to compile dahdi and recompile Asterisk ???
Thank you for your reply.
Jonas.
On Sun, Apr 19, 2009 at 05:17:38PM +0200, jonas kellens wrote:
VoIP-wiki.org states :
Digium resources http://www.asterisk.org/zaptel-to-dahdi
/etc/zaptel.conf Becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.conf
Now, what do I have installed on
2009/4/20 jonas kellens jonas.kell...@telenet.be:
Please, is there anyone who can help me with this zaptel -- Dahdi -problem
??
Will chan_dahdi.conf work with zaptel.conf ? Will Asterisk be able to
communicate with the Digium TDM pci-card ?
Or do I need to compile dahdi and recompile
Hi,
If I want to downgrade from 1.4.24.1 to 1.4.23, can I use the latest
asterisk-addons/libpri/dadhi builds?
Regards,
Mike
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or
What is the syntax to progress with a dial plan after hangup please?
Michael
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Exten = h,1,noop(stuff after hangup)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Monday, April 20, 2009 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
You can use the extension h
exten = h,1,app()
exten = h,n,app()
.
On Mon, Apr 20, 2009 at 10:31 AM, Michael mich...@networkstuff.co.nzwrote:
What is the syntax to progress with a dial plan after hangup please?
Michael
___
-- Bandwidth and
On Tue, 21 Apr 2009 02:40:58 you wrote:
You can use the extension h
exten = h,1,app()
exten = h,n,app()
.
On Mon, Apr 20, 2009 at 10:31 AM, Michael mich...@networkstuff.co.nzwrote:
What is the syntax to progress with a dial plan after hangup please?
Thanks. That works for 1 item, but
Hello,
When a voice message is saved and e-mailed as a wav, the total time of the
voice mail does not show up in, e.g., windows media player, why is this?
I have only used wav49/wav:
; Use wav49 format for all voicemail messages
format=wav49|gsm|wav
Justin.
You might want to use AGI or DEADAGI if you have more than 1 thing to do on
hangup.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael
Sent: Monday, April 20, 2009 9:46 AM
To: Pascal Bruno
Cc: 'Asterisk
Trying to make something coherent out of indiscriminate [top|bottom]
posting...
On Mon, Apr 20, 2009 at 10:31 AM, Michael
mich...@networkstuff.co.nzwrote:
What is the syntax to progress with a dial plan after hangup please?
On Tue, 21 Apr 2009 02:40:58 you wrote:
You can use the
Hi
My setup : Trixbox 2.6.1 TE410P running well ..I've 2 design issues to
consider :
1. I need to store the CallerId of the PSTN caller with his language preference
so that next time he is played the prompt in his language that he chose the
first time.What would be better - storing his
Tzafir,
When I did 'make menuselect' in 3. channel drivers, I selected [*] chan
dahdi...
But the only thing I have of dahdi is chan_dahdi.conf...
http://svn.digium.com/svn/dahdi/linux/tags/2.1.0.4/UPGRADE.txt states :
The system configuration file has moved from /etc/zaptel.conf
to
Converting is actually pretty straightforward:
Bare minimum:
/etc/zaptel.conf -- /etc/dahdi/system.conf
/etc/asterisk/zapata.conf -- /etc/asterisk/chan_dahdi.conf
Any reference to ZAP/* becomes DAHDI/* in your asterisk conf files (i.e.,
extensions.conf).
Granted, all I use Asterisk for is a
Perhaps my thinking on this issue is wrong. Do we need to keep up with all the
latest upgrades? I am thinking to install this thing and let it run for a long
time (I would hope with very little need to touch it.) We are not a feature
hungry shop, just looking for basic phone system for 60
Each of these arguments has validity. Here is my .02
1. There are a lot of 1.2 install's still up and running
2. After 1.4.22 you change ZAP to DAHDI and lose or change a lot of
references.
3. I don't think there will be a free book about 1.6 installs.
4. Simple is VERY good. There's a lot to
On Mon, 20 Apr 2009, Jimmy Ezell wrote:
Perhaps my thinking on this issue is wrong. Do we need to keep up with
all the latest upgrades? I am thinking to install this thing and let it
run for a long time (I would hope with very little need to touch it.)
We are not a feature hungry shop,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: April-20-09 2:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AstDB MixMonitor queries
On
Hi All,
We just bought a sip based PA setup here with the intention of hooking
it into our existing asterisk (1.4) setup. It works as expected when I
dial it's extension, but I want to have system generated speech played
based on some action (using cepstral, which is already installed and
On Mon, 20 Apr 2009, Sriram wrote:
2. I need all your help in writing an IVR program in Asterisk - i prefer
AGI and PHP. The caller calls into the system - he is asked to record
his Name (IVR plays it back and confirms) next he is aked to record his
age (IVR plays back and confirms) and
Here is one way
Create a call file
Channel: SIP/100
CallerID: SIP/104
MaxRetries: 1
WaitTime: 60
retryTime: 5
Application: background
Data: /tmp/systemisup
Have your dialplan create and send the call file for each person you want to
get the Cepstral file.
_
From:
On Mon, 20 Apr 2009, ContactTel Business wrote:
On Mon, 20 Apr 2009, Steve Edwards wrote:
I'm just finishing a system where for each step in my recipe, there
is a long prompt and a short prompt. The caller gets the long prompt
the first time. After hearing their response (voice or DTMF)
That works great - Thanks Danny!
-Justin
From: Danny Nicholas [mailto:da...@debsinc.com]
Sent: Monday, April 20, 2009 12:23 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk PA system with cepstral
Hi
On Mon, Apr 20, 2009 at 11:23:59AM -0700, Jimmy Ezell wrote:
Perhaps my thinking on this issue is wrong. Do we need to keep up
with all the latest upgrades?
Not directly, but there are reasons for that.
Asterisk is a codebases that changes. New features keep pouring in. We
strive to
On Monday 20 April 2009 03:08:09 am Benny Amorsen wrote:
Tilghman Lesher tilgh...@mail.jeffandtilghman.com writes:
There's already a working theory and a patch:
http://bugs.digium.com/view.php?id=14932
Would it be possible to adapt the kernel stack checker for Asterisk? I
can easily
Hi All,
I'm having a strange problem and I'm not able to understand what's happening.
I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine.
They are linked together through localhost. I've turned qualify on for the
iax peer. Periodically I've this message:
[Apr 20 23:47:46]
Alternatively look into the M() option to Dial to execute a Macro upon
connect. You could have your macro setup to call the cepstral app.
-Brent
Justin Killen wrote:
That works great -- Thanks Danny!
-Justin
On Mon, Apr 20, 2009 at 6:02 PM, Marco marcota...@libero.it wrote:
They are linked together through localhost. I've turned qualify on for the
iax peer. Periodically I've this message:
[Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer:
Peer 'iaxfax' is now UNREACHABLE!
Marco wrote:
I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine.
They are linked together through localhost. I've turned qualify on for the
iax peer. Periodically I've this message:
[Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049 __iax2_poke_noanswer:
Peer 'iaxfax' is
Same here, it doesn't remain unreachable for long, but it's annoying if
a fax arrives when it is. I'm going to try Fax for Asterisk instead:
http://www.digium.com/en/products/software/faxforasterisk.php
Ian
Marco a écrit :
Hi All,
I'm having a strange problem and I'm not able to understand
43 matches
Mail list logo
