2010/8/20 Anthony Messina amess...@messinet.com
perhaps we aren't exactly sure what you are trying to accomplish.
I'm quite sure about what I'm trying to accomplish but my english skills are
betraying me when explaining it.
what is
your end goal?
The whole thing is to develop one
Hi All,
I have this requirement. I have an xlite client registered with asterisk
server. And from this when i dial an extension say xxx it invokes an AGI
script which gives me a series of instructions like Welcome to this IVR
system. Press 1 to trade 2 to selland so on. I want to stop this
Hi,
Running Asterisk 1.6.2.11 and wondering what would be the best way to send an
email when a missed call has occurred ? I believe you can modify [stdexten] is
this still the case in V1.6 ?
--
Thanks, Phil
--
_
-- Bandwidth
Hi,
From time to time, I have to manually kill some frozen calls with soft
hangup commands.
(As far as I can tell, those freezes occurred after network breakdown (VPN
or ethernet link between 2 LAN switches).
So at this point, I would say I can't do much to keep those network
breakdown to
On Mon, Aug 23, 2010 at 6:14 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
From time to time, I have to manually kill some frozen calls with soft
hangup commands.
(As far as I can tell, those freezes occurred after network breakdown (VPN
or ethernet link between 2 LAN switches).
So at this
On Mon, Aug 23, 2010 at 5:35 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
Running Asterisk 1.6.2.11 and wondering what would be the best way to send an
email when a missed call has occurred ? I believe you can modify [stdexten]
is this still the case in V1.6 ?
--
Thanks, Phil
--
Do you know if OpenBTS support handoff?
Thanks
On Fri, Aug 20, 2010 at 12:32 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Fri, Aug 20, 2010 at 10:41 AM, Tim Panton t...@westhawk.co.uk wrote:
On 19 Aug 2010, at 20:59, Randy R wrote:
On Thu, Aug 19, 2010 at 12:37 PM,
Hi all,
I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server using
freetds and unixodbc, which works with 1.6.1.20.
With the same config in 1.8 I get an error when trying to start asterisk which
says:
[Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module:
Hi all,
We have an asterisk version v1.6.1.20 with a TDM400 board (2 FXS and 2
FXO).
We want to do a transfer blind and attended from a line external
connected to one FXO.
We have made configuration, and transfers from internal lines (FXS) work
fine but from (FXO) not.
We have made 2
On Monday 23 August 2010 08:14:43 unsero...@aol.com wrote:
Hi all,
I am testing with Asterisk 1.8.0 beta3 using realtime with a mssql server
using freetds and unixodbc, which works with 1.6.1.20.
With the same config in 1.8 I get an error when trying to start asterisk
which says:
[Aug 23
El 20/08/10 16:14, Kathryn Jones escribió:
Thanks for all the help, but I still can't find what's wrong.
I enabled console = notice,warning,error,debug,dtmf like Miguel
suggested. The output is attached.
I noticed that the rtp.c session never starts, which as I understand
is what catches
On Mon, Aug 23, 2010 at 2:58 AM, Janu Mukherjee janu.mu...@gmail.com wrote:
snip
system. Press 1 to trade 2 to selland so on. I want to stop this and
press 1 or talk even before the prompt finishes. How to achieve this. I was
told that this is similar to barging in asterisk but i could not
On Aug 7, 2007 'Mojo' wrote:
Nicholas Blasgen wrote:
I've got 4 SIP phone lines with a call-limit of 2 for each. I've
written a handy macro to allow my users to dial a phone number and the
macro will figure out the next available line to use by first checking
if the GROUP() is over 2 and
On Mon, Aug 23, 2010 at 9:28 AM, Gustavo Duarte gdua...@ipcomsa.com wrote:
[b] -- Unable to find extension '' in context 'from-pstn' [/b]
Please let me to know if you need configuration files.
This is a configuration problem:
*CLI dialplan show s...@from-pstn
--
Paul Belanger | dCAP
On Mon, Aug 23, 2010 at 7:14 AM, Olivier oza_4...@yahoo.fr wrote:
So, which tools are available to automatically detect that SIP channels are
up without but no RTP media is flowing in or from them ?
Make sure you explicitly call Hangup() and implement max timeouts for
your channels.
--
Paul
On Sat, Aug 21, 2010 at 7:09 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
Your experience could be different and better then most, and you have
certainly complete right of your own opinion.
Speech recognition is only as effective as your grammars and they are
never 100%. The require lots
Hi All,
I've got a project installing a Digium TDM800P card with 8 FXO's in an
Asterisk box.
The computer is running Slackware 13.1 and I've installed the current
Dahdi and Asterisk 1.6.2.11.
I've installed several boxes that are pure VOIP but, I haven't installed
a Dahdi interface and
On Mon, Aug 23, 2010 at 10:03 AM, Tilghman Lesher tles...@digium.com wrote:
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at
[Aug 23 15:06:12] WARNING[7180]: loader.c:387 load_dynamic_module: Error
loading module 'res_config_odbc.so':
/usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol:
ast_odbclear_cache
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so.
Sorry for the late response.
Philipp,
I've checked the file below and also the suggested voip-info link. None of
those describe how or why Asterisk assumed that 402 should be mapped to NORMAL
TERMINATION status. Both places refer to how Asterisk status should be mapped
to SIP cause and not
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static realtime, you should not be using the 'preload' directive
at all, and everything will just naturally load in the right order.
On Monday 23 August 2010 09:34:23 Paul Belanger wrote:
On Mon, Aug 23, 2010 at 10:03 AM, Tilghman Lesher tles...@digium.com
wrote:
For this one, you need to ensure that res_odbc.so loads before
res_config_odbc.so. Actually, the load order in 1.8 is such that, unless
you're using static
Hi everyone,
I've installed asterisk 1.8.0-beta3, and found this errors related to
several modules:
[Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error
loading module 'chan_iax2.so': /usr/lib/asterisk
/modules/chan_iax2.so: undefined symbol: ast_aes_set_decrypt_key
[Aug 23
* -Original Message-
* From: Todd Reese trees...@gmail.com
* Reply-to: Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
* To: asterisk-users@lists.digium.com
* Subject: [asterisk-users] Dahdi install gone wrong
Hi,
Odd problem have just noticed in that when I call into the PBX DAHDI detects
the call and hands it off to the extension, if I then hang up it still
continues to process through the dialplan.
This is what I have in chan_dahdi.conf:
[channels]
language=en
echocancel=yes
usecallerid=yes
Has anyone successfully implemented Asterisk as a voicemail server for a
GSM/cellular system and worked out a way to send notifications of new
messages to the phones?
--
_
-- Bandwidth and Colocation Provided by
The card you installed has FXO or FXS modules in it ? are you getting
your lines directly from the telco co???
Doug D
On Mon 23/08/10 8:37 AM , Cassius Smith cass...@cassius.org sent:
* -Original Message-
* From: Todd Reese
* Reply-to: Asterisk Users Mailing List -
On Monday 23 August 2010 10:35:26 Miguel Molina wrote:
I've installed asterisk 1.8.0-beta3, and found this errors related to
several modules:
[Aug 23 08:31:54] WARNING[3883]: loader.c:429 load_dynamic_module: Error
loading module 'chan_iax2.so': /usr/lib/asterisk
/modules/chan_iax2.so:
They are FXO modules and yes, the lines are coming in from the telco.
On 8/23/2010 12:05 PM, Doug Dawson wrote:
The card you installed has FXO or FXS modules in it ? are you
getting your lines directly from the telco co???
Doug D
On Mon 23/08/10 8:37 AM , Cassius Smith
Here are the output requested.
asteriscoII*CLI dialplan show s...@from-pstn
[ Context 'from-pstn' created by 'pbx_config' ]
's' = 1. Answer()
[pbx_config]
2. Background(main-menu)
[pbx_config]
3. WaitExten()
[pbx_config]
-= 1
What is menuselect actually looking for when it blocks me from selecting
res_odbc ?
I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3 -
so I'm confused
as it is claiming these are the pre-requisites ?
How can I best track down what it _thinks_ is missing ?
(This is
I've made the system work by overlaying the old trixbox config in
/etc/asterisk. BUT this is a disaster waiting to happen with this client.
I'm having a hard time deciphering the trixbox extensions*.conf files in
order to make a simple system where the client won't muck it up.
On 8/23/2010
Hi,
I am happy with the usual GDB backtrace methods and so forth, but have
an issue that I cannot work out how to trace on 1.6.2.10.
If I use either the Bridge() app, or the manager Action: Bridge() in a
certain scenario (Basically to bridge 2 SIP channels, like an attended
transfer, resulting
Hi,
This is a real strange one and trying to phantom it out. One of our clients is
connected to our Asterisk installation, from two sites, via VPN which works
great. Every so often one of the sites VPN tunnel goes does for say a couple of
seconds. When that happens all the extensions,
On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton t...@westhawk.co.uk wrote:
What is menuselect actually looking for when it blocks me from selecting
res_odbc ?
I've got unixOdbc installed and working. I also have
/usr/lib64/libltdl.so.3 - so I'm confused
as it is claiming these are the
Last time I looked, no OpenBTS does not (yet) support handoff between base
stations during a call.
Handoff between calls can be done using SIP registrations to a central
asterisk.
Tim.
Sent from my iPhone
On 23 Aug 2010, at 13:42, equis software equissoftw...@gmail.com wrote:
Do you
At 09:26 AM 8/23/2010, you wrote:
There's already an issue open for this, AND there is a patch posted, BUT the
reporter needs to verify that the patch(es) fix the issue for him:
https://issues.asterisk.org/view.php?id=17707
And how was I supposed to now that? I being the reporter.
I hate to
Hi list,
I have a small problem happening due to call transfer.
Whenever the call gets transfered to a remote extension ( which is not
registered to asterisk ) it results in hangup().
When asterisk tries to dial the other extension it results in failure
making the call cut down :(
Is there
Steve Davies schrieb:
I need suggestions please on how to determine where it is locking, and why.
Many thanks,
Steve
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
or whatever options you want.
maybe you could
On Mon, Aug 23, 2010 at 1:19 PM, Ira i...@extrasensory.com wrote:
I hate to seem stupid, but when I got the email I looked there but
have no idea what I'm supposed to do or how to do it. What is a patch
and what do I do with it?
Not stupid, in fact I would like to write documentation about
Hi,
We are using asterisk 1.4.34, ubuntu 10.4, below is suspicious activity
recorded in our full log. Could you help us to explain what had
happened. Thanks.
=== my friend, 801, from his room did a test by dialing echo test in
freepbx, *43:
[Aug 20 14:42:46] VERBOSE[14427] logger.c: --
On Monday 23 August 2010 12:19:38 Ira wrote:
At 09:26 AM 8/23/2010, you wrote:
There's already an issue open for this, AND there is a patch posted, BUT
the reporter needs to verify that the patch(es) fix the issue for him:
https://issues.asterisk.org/view.php?id=17707
And how was I
At 11:28 AM 8/23/2010, you wrote:
In the future (and this goes for everybody, not just you), if you do not
understand a request made by a developer, PLEASE ask that question, instead
of giving us no feedback whatsoever.
I'll defer to Paul's excellent set of instructions as to how to test a
At 11:28 AM 8/23/2010, you wrote:
I'll defer to Paul's excellent set of instructions as to how to test a
proposed patch.
I found them. I don't use IAX2 and so it ended up in the recycle bin.
Ira
--
_
-- Bandwidth and
On Monday 23 August 2010 13:56:10 Ira wrote:
At 11:28 AM 8/23/2010, you wrote:
In the future (and this goes for everybody, not just you), if you do not
understand a request made by a developer, PLEASE ask that question,
instead of giving us no feedback whatsoever.
I'll defer to Paul's
On 23 Aug 2010, at 18:07, Warren Selby wrote:
On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton t...@westhawk.co.uk wrote:
What is menuselect actually looking for when it blocks me from selecting
res_odbc ?
I've got unixOdbc installed and working. I also have /usr/lib64/libltdl.so.3
- so
I'm looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an old installation of Cardiff document stuff).
Is someone doing that?
If no one has direct experience, is there a HylaFax client that emulates
WinFax print-to-fax?
--Don
Don Kelly
PCF Corp
People Come
On Mon, 23 Aug 2010, Don Kelly wrote:
I’m looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an
old installation of Cardiff document stuff).
Is someone doing that?
If no one has direct experience, is there a HylaFax client that emulates WinFax
print-to-fax?
On Mon, Aug 23, 2010 at 1:03 PM, --[ UxBoD ]-- ux...@splatnix.net wrote:
This is a real strange one and trying to phantom it out. One of our clients
is connected to our Asterisk installation, from two sites, via VPN which
works great. Every so often one of the sites VPN tunnel goes does for
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Odd problem have just noticed in that when I call into the PBX DAHDI detects
the call and hands it off to the extension, if I then hang up it still
continues to process through the dialplan.
It is common for telco not
On Mon, Aug 23, 2010 at 11:56 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Odd problem have just noticed in that when I call into the PBX DAHDI detects
the call and hands it off to the extension, if I then hang up it still
continues to process through the dialplan.
It is common for telco not
On Mon, Aug 23, 2010 at 11:57 AM, Matt mhop...@gmail.com wrote:
Has anyone successfully implemented Asterisk as a voicemail server for a
GSM/cellular system and worked out a way to send notifications of new
messages to the phones?
Yes, look at externnotify within voicemail.conf
--
Paul
I looked at http://www.hylafax.org/content/Desktop_Client_Software and
visited several websites before posting this.
Nothing I saw said they would emulate WinFax print-to-fax.
I'd really appreciate hearing about any direct experiences.
-Original Message-
From:
On Monday 23 August 2010 14:58:31 Tim Panton wrote:
On 23 Aug 2010, at 18:07, Warren Selby wrote:
On Mon, Aug 23, 2010 at 11:34 AM, Tim Panton t...@westhawk.co.uk wrote:
What is menuselect actually looking for when it blocks me from selecting
res_odbc ?
I've got unixOdbc installed and
Don Kelly wrote:
I looked at http://www.hylafax.org/content/Desktop_Client_Software and
visited several websites before posting this.
Nothing I saw said they would emulate WinFax print-to-fax.
Not being familiar with WinFax, we currently use Winprint HylaFAX: It
creates a fax printer
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
Steve Davies schrieb:
I need suggestions please on how to determine where it is locking, and why.
Many thanks,
Steve
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace
Thanks, Doug,
The Cardiff Teleforms application is unattended. The Winprint link that I
looked at shows that it uses a print dialog. Does it have an automagic mode,
too?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Don Kelly wrote:
Thanks, Doug,
The Cardiff Teleforms application is unattended. The Winprint link that I
looked at shows that it uses a print dialog. Does it have an automagic mode,
too?
Not that I'm aware of, but it's been a while since I've looked to see if
there has been updates.
If anyone has any info on this it'd be much appreciated - haven't found much
about this topic anywhere. We are setting up sip probe monitor to make sure
that our Asterisk boxes are up and functional (or at least responding to the
sip protocol) and we need to determine the appropriate probe syntax
Matt Kershnar wrote:
If anyone has any info on this it'd be much appreciated - haven't
found much about this topic anywhere. We are setting up sip probe
monitor to make sure that our Asterisk boxes are up and functional (or
at least responding to the sip protocol) and we need to determine the
Greetings all-
Here's an odd question. Supposedly, IAX2 now has the ability to operate with
signaling and media in separate streams, very much like SIP. I've read about
this feature here[1] and there[2], but I have yet to see how to actually
implement or test it. There are no options in the
- Tim Nelson tnel...@rockbochs.com wrote:
Greetings all-
Here's an odd question. Supposedly, IAX2 now has the ability to
operate with signaling and media in separate streams, very much like
SIP. I've read about this feature here[1] and there[2], but I have yet
to see how to actually
On Mon, Aug 23, 2010 at 5:06 PM, Infra m...@waste.org wrote:
On Aug 7, 2007 'Mojo' wrote:
Nicholas Blasgen wrote:
I've got 4 SIP phone lines with a call-limit of 2 for each. I've
written a handy macro to allow my users to dial a phone number and the
macro will figure out the next available
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