Hello All,
I am running Asterisk 1.4.23.1 on debian lenny and having issues with it
sending out voicemail emails. Let me preface with the following:
1. I have tested with sendmail and ssmtp (with valid smtp server)
2. Googled quite a bit to only find the above
3. The mail.log/err/info shows
On Wed, 6 May 2009, Vincent wrote:
On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Do you want to build your own?
If so, you can put togther a 1GHz fanless VIA miniITX board, case (that
will take a drive or flash IDE), memory and psu for well under
On 6 May 2009, at 08:16, Damon Brown wrote:
Hello All,
I am running Asterisk 1.4.23.1 on debian lenny and having issues
with it
sending out voicemail emails. Let me preface with the following:
1. I have tested with sendmail and ssmtp (with valid smtp server)
2. Googled quite a bit to
On 06/05/09 08:28, Gordon Henderson wrote:
snip /
One little tip: You need to compile Asterisk for an i586 processor as
the VIA processor is missing a few (mmx, etc.) instructions that a full
blown i686 has.
Hi Gordon,
I'm using a VIA C7 on a Jetway board
Hello,
in our dialplan we have some variables containing datas from our
customers calls like for instance the called number.
Now, when caller make a transfer, we would like to catch the new called
number. How to get this? Is there a return context _after_ transfer and
*before* the call is
On Wed, 6 May 2009, Alan Lord (News) wrote:
On 06/05/09 08:28, Gordon Henderson wrote:
snip /
One little tip: You need to compile Asterisk for an i586 processor as
the VIA processor is missing a few (mmx, etc.) instructions that a full
blown i686 has.
Hi Gordon,
I'm using a VIA C7 on a
Those reading the thread amy be interested in Askozia pbx
http://www.askozia.com/pbx/
Michael was at AMOOCON (great success by the way, thanks to all who
participated) and I was impressed. He will be a guest on VUC very
soon, possibly even this Friday.
/r
On Wed, May 06, 2009 at 08:28:54AM +0100, Gordon Henderson wrote:
On Wed, 6 May 2009, Vincent wrote:
On Mon, 4 May 2009 10:07:06 +0100 (BST), Gordon Henderson
gordon+aster...@drogon.net wrote:
Do you want to build your own?
If so, you can put togther a 1GHz fanless VIA miniITX board, case
Hello !
In order to chase after a problem I implemented the following dialplan to have
an
answertime of exactly one minute:
exten = xxx,1,NoOp(Test wait)
exten = xxx,n,Answer
exten = xxx,n,NoOp(Current timestamp:
${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)})
exten =
Hello,
I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
I have a couple of questions about those cheap FXO cards:
1. Are they
On 06/05/09 13:43, Vincent wrote:
Hello,
I'm looking for a dirt cheap solution for SOHO use to handle at most
a couple of POTS lines, and I notice that X10?P cards go for $15 on
eBay as opposed to $90 for an OpenVox card or over $200 for a Sangoma.
I have a couple of questions about
On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News)
alansli...@gmail.com wrote:
For a cheap backup to your VOIP service they do the job. I wouldn't use
them for a proper system though.
Thanks for the feedback. I have two more questions:
1. Can the OSLEC echo canceller run OK on an 1.6GHz Intel
Gordon Henderson wrote:
On Wed, 6 May 2009, Alan Lord (News) wrote:
On 06/05/09 08:28, Gordon Henderson wrote:
snip /
One little tip: You need to compile Asterisk for an i586 processor as
the VIA processor is missing a few (mmx, etc.) instructions that a full
blown i686 has.
Hi Gordon,
On Wed, 6 May 2009, Vincent wrote:
On Wed, 06 May 2009 14:02:20 +0100, Alan Lord (News)
alansli...@gmail.com wrote:
For a cheap backup to your VOIP service they do the job. I wouldn't use
them for a proper system though.
Thanks for the feedback. I have two more questions:
1. Can the OSLEC
On Wed, 6 May 2009 12:17:44 +0200, randulo spamsucks2...@gmail.com
wrote:
Those reading the thread amy be interested in Askozia pbx
http://www.askozia.com/pbx/
Thanks for the link.
___
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On Tue, May 5, 2009 at 9:31 PM, Steve Edwards asterisk@sedwards.com wrote:
I doubt any language is going to replace any other language for all
future developments.
The day one religion replaces all other religions, it may happen
because languages = religions = distros = platforms = your
Ok - after a lot of playing I'm still a bit stuck.
I'd like to accomplish the following - can't get it to work as it should (at
least in my head! LOL)
I've got an app that initiates an AMI call for Originate. I want to click a
number onscreen, send the Originate, then (this is the part I can't
I believe you can specify a default context:
sip.conf
[general]
context=mysipcontext
extensions.conf
[mysipcontext]
; be careful allowing calls that could incur toll charges here, especially
if this box is exposed to the internet
exten = s,1,Answer()
;direct your call
exten = s,n,Hangup()
The
Send your call to a different extension that will set the header before calling
your phone.
-Original Message-
From: pallet...@gmail.com
Sent: Wed, 6 May 2009 10:51:30 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AMI + AGI for outbound click to dial
Ok
Ping-ponging the call?
That's a good idea..
Now, to try to accomplish that in an AGI script.
Thanks Jim!
PB
On Wed, May 6, 2009 at 11:18 AM, Jimmy Godbout s...@inbox.com wrote:
Send your call to a different extension that will set the header before
calling your phone.
-Original
I may or may not be experiencing the behavior described in:
http://bugs.digium.com/view.php?id=14241
I'm using asterisk-1.6.0.6, Bridge(), and I'm having a hangup context
executed when the caller is still on the line. These channels are all
SIP.
I want a group of expert callers who can
Formerly on a thread called [asterisk-dev] Where to find the code of
application Bridge
On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Can someone please tell me in which file the code for the application to
be found? I was not able to find a file named
On Tue, 2009-05-05 at 17:00 -0500, Danny Nicholas wrote:
Please elaborate; obviously ?? the dialplan is the simplest route to solve
any problem.
Dialplan is not the simplest route to solve ANY problem. It is the
simplest route to solve simple problems. Writing a while loop in AEL or
lua is
Where are some good lua references? I don't think the O'reilly book even
mentions it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew
Nicholson
Sent: Wednesday, May 06, 2009 11:31 AM
To: Asterisk Users
- David Backeberg dbackeb...@gmail.com wrote:
Formerly on a thread called [asterisk-dev] Where to find the code of
application Bridge
On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen
tzafrir.co...@xorcom.com wrote:
Can someone please tell me in which file the code for the
application to
Try the patch on this bug
http://bugs.digium.com/view.php?id=15042
I don't get that error with my setup, but others have seen it. I am
fairly sure of what is causing it. Still working on a fix.
On Tue, 2009-05-05 at 22:52 -0400, Carlos Ruiz Diaz wrote:
I get a lot errors from chan_mobile
In your AMI portion, you set the outgoing call first, then the extension you
want to be reached at:
Action: Originate
Channel: Zap/g2/8135551212
Context: default
Exten: 101
Priority: 1
Timeout: 3
In the dialplan:
[default]
exten = 101,1,SIPAddHeader(...
exten = 101,n,Dial(...
exten =
Hi,
I'm having problems with an asterisk server that's not offering Codecs
for ulaw and alaw as it should.
I've three servers in total: a1, a2 and b
A1 and A2 have pretty much the same config files, except IP address info
changes
Server B is configured to accept all inbound invites.
Forgot to add: sip.conf for both A1 and A2 has the following global
codec definitions:
disallow=all
allow=clear
allow=amr
allow=ulaw
allow=alaw
The Asterisk build is a private build that adds the clear and AMR codec
setups.
The two servers are running Fedora, though A1s on 6 and
Hi,
I am using ASTCC and trying to setup a calling card platform.
The problem that I have is that astcc does not hangup calls correctly:
1. If I try to dial a number, call goes through fine. When I hang up
the call from my side I get this:
-- Called
Hi.
I have a working internal Asterisk setup with 35 phones. Around 5-10 of
these phones are physically located in a remote office via a VPN. I am
completely happy with Asterisk and would be able to set up external
calls but for one serious problem. After a period of time (perhaps a
couple of
Excuse me if this is in the archives or on the net somewhere - I really
did search. ;-)
We are wondering if there is any way to shut down a single PRI without
having to down Asterisk and/or interrupt other running PRI circuits.
We use Asterisk servers with 4 port Digium PRI cards. In
On Wed, May 6, 2009 at 1:04 PM, James Van Vleet
james.vanvl...@verety.com wrote:
We are wondering if there is any way to shut down a single PRI without
having to down Asterisk and/or interrupt other running PRI circuits.
Somebody who knows Zaptel better could tell you whether this is a bad
On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk i...@comtek.co.uk wrote:
Can anybody provide any suggestions to help debug this? If I'm unable to
isolate/resolve the problem then its likely we'll have to drop the
Asterisk solution and I've already grown rather attached to it.
I have a
On Wed, May 6, 2009 at 12:48 PM, i...@comtek.co.uk i...@comtek.co.uk wrote:
Hi.
I have a working internal Asterisk setup with 35 phones. Around 5-10 of
these phones are physically located in a remote office via a VPN. I am
There are a number of other reasons you want a remote phone server at
Hi all,
I need to know the SIP response code from within the dial plan, someone
could point me on how to?
Gabriel Ortiz
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asterisk-users mailing list
To UNSUBSCRIBE or update
Thank you very much!
I'll try with the patch and post the results.
On Wed, May 6, 2009 at 12:37 PM, Matthew Nicholson mnichol...@digium.comwrote:
Try the patch on this bug
http://bugs.digium.com/view.php?id=15042
I don't get that error with my setup, but others have seen it. I am
fairly
I think I misunderstood your mail.
There is no patch available yet, right?
I went to the page you linked but I did not found a patch file.
On Wed, May 6, 2009 at 2:45 PM, Carlos Ruiz Diaz
carlos.ruizd...@gmail.comwrote:
Thank you very much!
I'll try with the patch and post the results.
On
I have a number of ideas of what could be happening, and most involve
routing issues over your VPN, or your VPN dropping packets. Here's a
suggestion:
* put another asterisk server on the remote side, and have the two
asterisk servers do SIP or IAX trunks back and forth.
Thanks for the
I've googled for way too long, where are the 2 letter language values
defined?
I know:
en = English
es = Spanish
fr = French
but what about Croatian, Russian, Serbian, Vulcan, etc?
Is there a list documented for Asterisk or is it just use the 2 letter
country code Internet TLD?
Thanks in
They are 2-letter ISO country codes.
http://www.iso.org/iso/english_country_names_and_code_elements
On Wed, May 6, 2009 at 1:05 PM, Steve Edwards asterisk@sedwards.com wrote:
I've googled for way too long, where are the 2 letter language values
defined?
I know:
en = English
es =
Also check out:
http://www.w3.org/International/questions/qa-lang-2or3.en.php
On Wed, May 6, 2009 at 1:22 PM, Steve Johnson stevej...@gmail.com wrote:
They are 2-letter ISO country codes.
http://www.iso.org/iso/english_country_names_and_code_elements
On Wed, May 6, 2009 at 1:05 PM, Steve
I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650.
I attempted to simply reuse the existing config files for the old phone
on the new phone, but the new phone would lock up on the 4th digit when
attempted to dial out certain numbers. So, I downloaded the newest
firmware and config
Would this return during the ring or only after the remote party has picked
up?
On Wed, May 6, 2009 at 12:51 PM, Jimmy Godbout s...@inbox.com wrote:
In your AMI portion, you set the outgoing call first, then the extension
you want to be reached at:
Action: Originate
Channel:
In this case, Placing a call from an outgoing channel to a local extension,
this will cause the local extension not to ring until the Zap channel has
picked up -
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate
-Original Message-
From: pallet...@gmail.com
On Wednesday 06 May 2009 12:04:08 James Van Vleet wrote:
We are wondering if there is any way to shut down a single PRI without
having to down Asterisk and/or interrupt other running PRI circuits.
We use Asterisk servers with 4 port Digium PRI cards. In the last few
days we ran into a
On Wednesday 06 May 2009 14:05:58 Steve Edwards wrote:
I've googled for way too long, where are the 2 letter language values
defined?
I know:
en = English
es = Spanish
fr = French
but what about Croatian, Russian, Serbian, Vulcan, etc?
Is there a list documented for Asterisk or is it
On Wed, May 6, 2009 at 1:05 PM, Steve Edwards
asterisk@sedwards.com wrote:
Is there a list documented for Asterisk or is it just use the 2
letter country code Internet TLD?
On Wed, May 6, 2009 at 1:22 PM, Steve Johnson stevej...@gmail.com
wrote:
They are 2-letter ISO country codes.
On Wed, 6 May 2009, Tilghman Lesher wrote:
In trunk (1.6.3 and later), we have a command pri service disable channel
for doing this exact procedure. Support for this has been a long time coming,
as the patch came from issue 3450 (http://bugs.digium.com/view.php?id=3450).
Will this work
Hi,
Did anyone tried speech recognition using Sphinx ? I used sphinx
using this website (http://scribblej.com/svn/) but when i run
astsphinx i am getting the following error. Any clue what might have
caused this problem ?
Thanks
-Azher
INFO: s2_semi_mgau.c(1080): 1 mixture Gaussians, 256
Hi,
I am using Cisco 7960G with asterisk and it works perfect but it
needs a dhcp/tftp server for ip address and configuration files. Is
there any way i can config the phone with static configuration i.e.
without dhcp/tftp ?
Thanks
-Azher
___
--
On Wednesday 06 May 2009 16:23:08 Jeff LaCoursiere wrote:
On Wed, 6 May 2009, Tilghman Lesher wrote:
In trunk (1.6.3 and later), we have a command pri service disable
channel for doing this exact procedure. Support for this has been a
long time coming, as the patch came from issue 3450
On Wed, May 06, 2009 at 11:04:08AM -0600, James Van Vleet wrote:
Excuse me if this is in the archives or on the net somewhere - I really
did search. ;-)
We are wondering if there is any way to shut down a single PRI without
having to down Asterisk and/or interrupt other running PRI
Yes (I assume you have it configured for SIP)
Press the Setting button (square with check)
Scroll down to the Unlock Config
Enter the password of cisco
Press the accept softkey
Now you can scroll up to Network Configuration / SIP Configuration and manually
change settings.
Jimmy M. Ezell
It works, Thanks.
Jimmy Ezell wrote:
Yes (I assume you have it configured for SIP)
Press the Setting button (square with check)
Scroll down to the Unlock Config
Enter the password of cisco
Press the accept softkey
Now you can scroll up to Network Configuration / SIP Configuration and
I use these cards and they work pretty well. FWIW when Digium sold
them they were also just winmodems with a resistor removed to change
the PCI device ID. Later on the Zaptel driver included the device ID
of the winmodem.
I used to be able to get the winmodem itself for under $10, but I
think
Do you know if there is a way to have an script run whenever a user has
deleted a voicemail message?
I want to have multiple users, all with there own passwords, share the same
mailbox. When any one of them deletes a message I want it deleted from
everyone's mailbox. I believe that I can do this
scribb...@scribblej.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Azher Mughal
Sent: May-06-09 6:19 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk with Sphinx
Hi,
Did anyone
I'd say in life you get what you pay for.. and sometime you even pay for
stuff that should be free..
These knockoff cards, can be built in-house for 20$ or less using an old
walkie talkie, a rope, some standard matches, and an old MCgyver Tv
episode..They do just that, echo the sound back to the
I sent email twice, but no reply :(
ContactTel Business wrote:
scribb...@scribblej.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Azher Mughal
Sent: May-06-09 6:19 PM
To:
On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
li...@contacttel.com wrote:
I'd say in life you get what you pay for.. and sometime you even pay for
stuff that should be free..
I have to agree.
I have a few of these cards I started out with. They were great for
the wow, I finally got
Hi to All, I need to implement an automatic telephone messaging system that
works like this:
-the system generates the call based on mysql records or any database
-when the client answer the phone, the Asterisk PBX playback a recorded
message
-when finish, hang up the channel.
Only for
Jonathan Moore wrote:
On Wed, May 6, 2009 at 8:47 PM, ContactTel Business
li...@contacttel.com wrote:
I'd say in life you get what you pay for.. and sometime you even pay for
stuff that should be free..
I have to agree.
I have a few of these cards I started out with. They were
On Wed, 6 May 2009, Ricardo Melendez wrote:
Hi to All, I need to implement an automatic telephone messaging system that
works like this:
-the system generates the call based on mysql records or any database
-when the client answer the phone, the Asterisk PBX playback a recorded
message
John Novack ha scritto:
Not sure how you would do that, as the X100 card is an FXO card,
won't provide either battery or dial tone to the cordless. What you
will want for that is an FXS card or ATA. The X100 card will
connect to a central office line, and with the later software echo
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