On 22/10/09 6:52 PM, das sandesh wrote:
There were 2 problems that we faced, one was at around 50 calls, few
calls were just dead air, and when I saw the logs I could see that it
was sent to the sip provider and after that there was no log for that
particular call that was having dead air, but
Hello,
I'm wondering if I can take benefits of long prompts to compute in the
background the next step to be performed by Asterisk.
Do you know what will be the behavior of asterisk if I send a STREAM
FILE command immediately followed by another command ? Will asterisk
stack commands or will it
/listinfo/asterisk-users
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Systemadministrator
christophorus.la...@semanticedge.de
SemanticEdge GmbH
Kaiserin-Augusta-Allee 10-11
10553
Dear all,
I fixed the issue by myself.
I have edited chan_sip.c file to avoid sdp version gettng increment.
I think this is a bug of asterisk. According to RFCs it should increment it
only it there is change on SDP message body. chan_sip.c alway increase it by
one at every SDP message. I have
On Thu, Oct 22, 2009 at 01:03:15AM +0300, B.Masoud @ SH wrote:
It's not caller ID issue,
I can make asterisk answer the line by omitting the line
answeronpolarityswitch=no ,
answeronpolarityswitch = *yes*; right?
but this will take effect on all 24 TDM
channels, I want some to have
On Wed, Oct 21, 2009 at 9:57 PM, SIP s...@arcdiv.com wrote:
Sounds like it wasn't a very interesting track. ;)
Not sure, but I guarantee the previous night was interesting :) The
VUC guys, sometimes led by Randal Happy Hour Schwartz, know how to
party. One night I got two hours sleep and was
On Wed, Oct 21, 2009 at 8:28 PM, Danny Nicholas da...@debsinc.com wrote:
Is THAT a summary :)?
As I said above (or below?) I we'll be talking about this on VUC
Friday at 12 Noon. In fact, here's the whole spa^H^H^H preview:
VoIP Users Conference (VUC) Astricon, Been there, Got the T-shirts
Sorry I was away for some time, here is dump.
Best regards,
Josip
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov
Sent: Monday, October 19, 2009 12:36 PM
To: Asterisk Users Mailing List -
On Wed, Oct 21, 2009 at 10:15 PM, Olivier oza-4...@myamail.com wrote:
2009/10/21 Leif Madsen leif.mad...@asteriskdocs.org
Olivier wrote:
Hi,
Siemens Gigaset line of products include an integrated web browser with
which firmware download is possible.
The trouble is you need to
Hi,
I am facing audio issue in my skype for asterisk setup.
*Flow of the call is like this.*
e.g.
Skype users :
test2
Sip users:
1001
1002 -- test2
This both sip users 1001 and 1002 are register in same asterisk. And also
test2 skype user is register in same asterisk.
Now 1001 is dialing
@Matt: will check that one out, thanks
@David: will do a search in the archives to see if I can find something
there :) thanks!
as soon as my setup is done and working correctly, I'll post the results
back here.
On Wed, Oct 21, 2009 at 21:19, Matt Florell astma...@gmail.com wrote:
On
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the autopause works after member failed to answer call.
However, other queues don't work for the autopause function.
queue 1000:
-- Nobody picked up in 25000 ms
-- Auto-Pausing Queue Member
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Darrick Hartman wrote:
I don't think that Maildir or a database backend solution (such as
Exchange) suffers from this same limitation.
Maildir makes sense, but the text I quoted in an earlier message is now
no longer part of the imapstorage text.
exted != exten
Ok. That was the actual error, I guess I needed some sleep.
Thanks.
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To UNSUBSCRIBE or update options visit:
On Thu, 22 Oct 2009, Patrick wrote:
I'm wondering if I can take benefits of long prompts to compute in the
background the next step to be performed by Asterisk.
I did this a few years ago so that I could process a credit card
authorization request while the Please wait... prompt was playing.
Rilawich Ango escribió:
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the autopause works after member failed to answer call.
However, other queues don't work for the autopause function.
queue 1000:
-- Nobody picked up in 25000
Dear,
I am getting this in CLI on release candidate version of Asterisk. Any
ideas, or points where to look?
-- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi
[Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite: write()
returned error: Broken pipe
--
In main/utils.c there is a line (1126) that reads like this -
ast_log(LOG_ERROR, write() returned error: %s\n, strerror(errno));
change ERROR to NOTICE -
ast_log(LOG_NOTICE, write() returned error: %s\n, strerror(errno));
and do make make install. This will remove this message.
Hi,
Most (if not all) IP phones support provisioning through DHCP/TFTP.
The trouble is some phones seem to require to store their config files in
TFTP root directory.
This makes this TFTP root directory a bit messy.
What are the best practices or tricks to manage this TFTP root directory ?
I
On Thursday 22 October 2009 09:30:58 Josip Djuricic wrote:
I am getting this in CLI on release candidate version of Asterisk. Any
ideas, or points where to look?
-- Launched AGI Script /var/lib/asterisk/agi-bin/rad-auth.agi
[Oct 22 18:21:45] ERROR[9853]: utils.c:1126 ast_carefulwrite:
Hello
I have an old Asterisk where I need to listen to Agent calls. So I
created this code:
exten = _555,1,ChanSpy(Agent)
exten = _555,n,Hangup()
But I always get:
2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No
application 'ChanSpy' for extension (default, 555, 1)
It
App_chanspy
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joao Gomes
Pereira
Sent: Thursday, October 22, 2009 10:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] ChanSpy in Asterisk 1.2.24
Olivier wrote:
Hi,
Most (if not all) IP phones support provisioning through DHCP/TFTP.
The trouble is some phones seem to require to store their config files in
TFTP root directory.
This makes this TFTP root directory a bit messy.
What are the best practices or tricks to manage this TFTP
I did run into some issues with this as well. I ended up setting
format=wav and left it at that... It wasn't so much a problem with
someone leaving a message rather when someone was forwarding messages. I
would have used wav49 but people were having problems getting wav49 to
open on their PDA's
Phew! So it's not just me! That's exactly the problem - not leaving the
message but forwarding it (I suppose the correct term rather than
transfer). Thanks - John
On Thu, 2009-10-22 at 10:29 -0500, Robert Grignon wrote:
I did run into some issues with this as well. I ended up setting
On Thu, 2009-10-22 at 08:43 +0200, Patrick wrote:
I'm wondering if I can take benefits of long prompts to compute in the
background the next step to be performed by Asterisk.
Do you know what will be the behavior of asterisk if I send a STREAM
FILE command immediately followed by another
On Thu, 2009-10-22 at 11:15 -0400, Dave Fullerton wrote:
#2 might be possible, but there's a lot of depends on factors.
The ISC dhcpd often packaged in linux distributions has the ability to
specify different dhcp options to different pools of addresses. You
can then assign clients to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Vela Sivasankaran a écrit :
| Hi,
| How can I integrate Asterisk to Nuance TTS engine instead of
| Cepstral? Has anybody done this? How is the architecture and can Java
| AGI be used to communicate between them?
I have made app_realspeak
Hi,
(I think) I followed instructions here (
http://www.voip-info.org/wiki/view/Firmware+issues+on+7940+-+7960 section
Notes added Nov 2005, revised May 2006:
at the bottom of the page) to factory reset a Cisco 7942 I wanted to
configure to SIP firmware.
When booting, I can see this requesting
Thanks a lot
The file App_chanspy was already in
/usr/lib/asterisk/modules
But I had in my modules.conf:
noload = app_chanspy.conf
Now I erased this line... but Asterisk still doesn't load this
app_chanspy...
Do I need to stop/start Asterisk? Or the reload is enough?
Thanks
Regards
Joao Pereira
2009/10/22 Jared Smith jsm...@digium.com
On Thu, 2009-10-22 at 11:15 -0400, Dave Fullerton wrote:
#2 might be possible, but there's a lot of depends on factors.
The ISC dhcpd often packaged in linux distributions has the ability to
specify different dhcp options to different pools of
I have some simple PERL scripts that worked fine with 1.4 SVN, but returned
this error when I went to 1.4.26.2. It seemed easier to change the ERROR to
a NOTIFY than try to Fix my bad code.
Here is the failing code
#!/usr/local/bin/perl
use strict;
use warnings;
sub setvar {
my ($var, $val)
Thanks. Finally, I find that it was caused by the use of the table wrongly.
On Thu, Oct 22, 2009 at 10:23 PM, Miguel Molina
mmol...@millenium.com.co wrote:
Rilawich Ango escribió:
Hi,
I have 3 queue set in the table as below.
name,autopause
1000,1
2000,1
3000,1
In queue 1000, the
I also noticed that there was a version of asterisk that had a voicemail
bug dealing with this... I am run 1.4.26.2 now and what was happening
was if IMAP forwarded (and corrupted) a voicemail, the user would try to
retrieve the message and the system would hangup on them.. The updated
code seemed
Hi there,
Has anyone Used ATCOM IAX Hard phones with any success?
Or
Has anyone found any good IAX ATA that you could recommend.
Thanks
Albert
This e-mail, as well as any other mode of correspondence, and any files
transmitted with it are intended for, and should only be
Yes I have used ATCOM-530 as an iax2 extension, without any trouble
On Thu, 22 Oct 2009 17:33:13 +0100, Albert Culleton a...@icmunicomp.ie
wrote:
Hi there,
Has anyone Used ATCOM IAX Hard phones with any success?
Or
Has anyone found any good IAX ATA that you could recommend.
Dear all,
I'm planning to buy some IP phones with GSM audio codec support in order to
use with an Asterisk SIP server I have implemented and nowsuccessfully
running with softphones like Eyebeam and Twinkle.
A vendor offer to me the SNOM 300 IP phone, that support GSM 6.10 audio
codec. I've
On Thursday 22 October 2009 11:09:00 Danny Nicholas wrote:
I have some simple PERL scripts that worked fine with 1.4 SVN, but returned
this error when I went to 1.4.26.2. It seemed easier to change the ERROR
to a NOTIFY than try to Fix my bad code.
Here is the failing code
FYI, in case anyone else encouters this issue. The card that I had
which I could reproduce this with was hardware revision B4. I RMAed
the card with Digium support and got a newer, revision C card, and the
issue is no more.
On 20 Oct, 2009, at 3:25 PM, Chris Brentano wrote:
I've seen
On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote:
It's really simple you just read from standard input and write to
standard output.
If you tell us a programming language you'd like to use (i.e.
php/c/perl/bash etc) we can give you a link to some docs and examples.
Might I
When Asterisk establish a call through an outbound trunk, Is there any way I
can know who hang up the call first? The caller or the party called?
Thanks.
___
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asterisk-users
On Thu, 22 Oct 2009, Alejandro Cabrera Obed wrote:
Dear all,
I'm planning to buy some IP phones with GSM audio codec support in order to
use with an Asterisk SIP server I have implemented and nowsuccessfully
running with softphones like Eyebeam and Twinkle.
A vendor offer to me the SNOM
On Thu, 22 Oct 2009, jonas kellens wrote:
On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote:
It's really simple you just read from standard input and write to
standard output.
If you tell us a programming language you'd like to use (i.e.
php/c/perl/bash etc) we can give you a link to
If it was clear, I wouldn't be writing; You are suggesting something like
this?
sub setvar {
my ($var, $val) = @_;
print STDOUT SET VARIABLE $var \$val\ \r\n;
my $rv=STDIN;
while(STDIN) {
m/200 result=0/ last;
}
return;
}
-Original Message-
From:
Un-top-posting...
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
What is clearly wrong with your script is that you're failing to
retrieve all of the setup information that is sent when the script first
starts up. Each response that you're getting is
So this would actually be proper?
Here is the failing code
#!/usr/local/bin/perl
use strict;
use warnings;
sub setvar {
my ($var, $val) = @_;
print STDOUT SET VARIABLE $var \$val\ \r\n;
while(STDIN) {
m/200 result=0/ last;
}
return;
}
# turn off I/O
On Thu, 22 Oct 2009, Danny Nicholas wrote:
So this would actually be proper?
[snip]
my $envvars = STDIN;
I don't do Perl, but if that statement reads everything buffered on
STDIN -- i.e., the AGI environment, then I would guess it would work.
Proper would be to read and parse the
For my (and any other lazy watchers) benefit, would you post how you would
do my snippet in C?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, October 22, 2009 1:48 PM
To: Asterisk
Un-top-posting...
On Thu, 22 Oct 2009, Danny Nicholas wrote:
So this would actually be proper?
[snip]
my $envvars = STDIN;
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
I don't do Perl, but if that statement reads everything buffered on
STDIN -- i.e., the
Sorry about the top post (OUTLOOK) -
Thanks for the framework. It's easier to learn from a starting point than
scratch. I'm not crazy about writing 1000 lines of C to do 30 lines of
PERL, but if it makes my system fly, so be it.
-Original Message-
From:
On Thu, 22 Oct 2009, Danny Nicholas wrote:
Sorry about the top post (OUTLOOK) -
Thanks for the framework. It's easier to learn from a starting point
than scratch. I'm not crazy about writing 1000 lines of C to do 30
lines of PERL, but if it makes my system fly, so be it.
If you discard
Thanks for the update! I'm glad you got the card, kernel and dahdi
working properly again!
Chris Brentano wrote:
FYI, in case anyone else encouters this issue. The card that I had
which I could reproduce this with was hardware revision B4. I RMAed
the card with Digium support and got a
On Thu, Oct 22, 2009 at 01:30:31PM -0700, Steve Edwards wrote:
On Thu, 22 Oct 2009, Danny Nicholas wrote:
Sorry about the top post (OUTLOOK) -
Thanks for the framework. It's easier to learn from a starting point
than scratch. I'm not crazy about writing 1000 lines of C to do 30
Olivier oza-4...@myamail.com writes:
Most (if not all) IP phones support provisioning through DHCP/TFTP.
The trouble is some phones seem to require to store their config files in
TFTP root directory.
A lot of IP phones support HTTP instead of TFTP. This helps, because it
is fairly easy to
On Thu, 22 Oct 2009, Tzafrir Cohen wrote:
On Thu, Oct 22, 2009 at 01:30:31PM -0700, Steve Edwards wrote:
On Thu, 22 Oct 2009, Danny Nicholas wrote:
Sorry about the top post (OUTLOOK) -
Thanks for the framework. It's easier to learn from a starting point
than scratch. I'm not crazy about
On Thu, 22 Oct 2009, Benny Amorsen wrote:
A lot of IP phones support HTTP instead of TFTP. This helps, because it
is fairly easy to write a script which dynamically generates the
configuration.
Someone really ought to write a TFTP daemon with the same feature... Or
a TFTP plugin for apache
I'm doing some quick research on how to get our videos from AstriCon
available in a reasonable format that allows easy viewing, reduces
our bandwidth costs, and allows good tracking for who/where/what is
viewing the videos.
YouTube seems to have a very nice set of tools and statistics
I like viddler. Nice stats and as far as i know, no limit in length.
Also a nice customizable player.
On 10/23/09, John Todd jt...@digium.com wrote:
I'm doing some quick research on how to get our videos from AstriCon
available in a reasonable format that allows easy viewing, reduces
our
We currently use asterisk 1.4.x with two Zaptel cards connected to POTS
lines. So we make outbound calls from their softphones (using ulaw
format), which go over a dedicated DSL line to the asterisk server in
our office, which then converts the calls to POTS.
This all works fine, assuming
http://video.google.com/
Free, no length limit, and they seem to have plenty of bandwidth...
Regards,
Ron Arts
NeoNova BV
John Todd schreef:
I'm doing some quick research on how to get our videos from AstriCon
available in a reasonable format that allows easy viewing, reduces
our
I thought google pulled uploading to that site after they bought youtube.
On Thu, Oct 22, 2009 at 4:05 PM, Ron Arts ron.a...@neonova.nl wrote:
http://video.google.com/
Free, no length limit, and they seem to have plenty of bandwidth...
Regards,
Ron Arts
NeoNova BV
John Todd schreef:
Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura
adapters?
I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal
cards not external units like Linksys.
I have one Linksys 3K that with so echo so bad that it is pretty much useless.
--
Joseph
On Thu, Oct 22, 2009 at 05:19:14PM -0600, Joseph wrote:
Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura
adapters?
I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal
cards not external units like Linksys.
I have one Linksys 3K that with so
You're so right. Sorry about that.
Ron
Op 23 okt 2009 om 01:14 heeft Kyle Kienapfel doctor.w...@gmail.com
het volgende geschreven:\
I thought google pulled uploading to that site after they bought
youtube.
On Thu, Oct 22, 2009 at 4:05 PM, Ron Arts ron.a...@neonova.nl wrote:
On Thu, 2009-10-22 at 16:04 -0700, Robert L Mathews wrote:
We currently use asterisk 1.4.x with two Zaptel cards connected to POTS
lines. So we make outbound calls from their softphones (using ulaw
format), which go over a dedicated DSL line to the asterisk server in
our office, which then
Calling all members of the asterisk community,
I am posting about an old issue that has been reported many places and
times online, To my amazement, there has yet to be anyone that has
reported any solutions to the following problem.
Initially when putting callers on hold, it plays between 30
I had to restart Asterisk, and now the module is loaded.
Thanks a lot for the help
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
Danny Nicholas wrote:
Try module load app_chanspy.so from CLI. If that doesn't work,
On 10/23/09 01:40, Tzafrir Cohen wrote:
On Thu, Oct 22, 2009 at 05:19:14PM -0600, Joseph wrote:
Will the DAHDI with OSLEC have any effect one ECHO in Linksys or Sipura
adapters?
I know Dahdi is a replacement for Zaptel. And Zaptel is mostly for internal
cards not external units like
Abdulmnem Benaiad
Almontaha CTO
Almontaha IT Co.
cell: +218 92 5200025
fax: +218 21 4835263
www.almontaha.com.ly
On Wed, Oct 21, 2009 at 11:57 PM, das sandesh sandesh...@gmail.com wrote:
Hi Matt,
I already used the tuning-primer.sh script to enhance the values for the
parameters,
On Thursday 22 October 2009 18:54:54 OrangeCell Center Inc. wrote:
Initially when putting callers on hold, it plays between 30 and 60
seconds of old audio that was on the stream in the past. Then after
that 30-60 seconds, it does a hard cut into what is currently playing
(which sounds pretty
Hello guys,
Thank you for your answers
I've seen in the ExternalIVR command :
If the child process dies, ExternalIVR() will notice this and hang up
the channel immediately (and also send a message to the log).
That's not what I'd like, I want that if the process finish gracefully
that the AGI
Jeff LaCoursiere schreef:
Working with a new client that has a ton of these phones, and in a new
installation the phone is registered, can place and receive calls with no
issues, but has a locked picture of a phone in the upper right corner.
Any Linksys experts know what this means? I have
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