Quinn--
I would venture to guess that your problem is because you are using the
sound file streaming
mechanism at too high a level. At the app/agi level, you don't get any
control over the process.
You start the sound process, then you wait for the interrupt; it's all
neatly bundled into a single
Greetings all.
First off, thank you for your time on this. I have spent literally 12
hours searching every forum and article I can find, and I'm going
cross-eyed, so I need to bother everyone with this.
I am running * 1.2.37, and I am trying to get the hints working, so I
can turn one of m
I configured our SIP gateway to automatically dial extension "s" when a
phone is picked up. I want Asterisk to play a dial tone, wait for an
extension to be dialled, and hangup on timeout
This works great, but I also want Asterisk to *stop* playing the dial
tone after the first digit is pressed
S
On Monday 25 January 2010 08:52:45 Mark Hulber wrote:
> Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had
> this problem before and even when I move to back versions I have the
> issue. I did upgrade safe_asterisk and the init.d scripts a version or
> so ago but even when I
On Monday 25 January 2010 07:43:30 Kevin P. Fleming wrote:
> JR Richardson wrote:
> > Hi All,
> >
> > I'm using Asterisk 1.4 branch and checking the status of some SIP
> > Peers with the functions ${SIPPEER(101:status)} and the result is "OK
> > (48 ms)". Seems to work fine.
>
> That is a bug; the
On Monday 25 January 2010 03:12:08 Zhang Shukun wrote:
> hi, dear all
>
> MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
>
> is that the grammar is different between them?
>
> extensions.conf
>
> exten => s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
> blacklist\ where\
On Mon, Jan 25, 2010 at 2:56 PM, Danny Nicholas wrote:
> Since you're "Perling" it, why not just put the $sb_retval in a while loop
> like this:
>
> - my $response_good=0;
> - my $sb_retval=undef;
> - while (! $response_good) {
> - my $tmp_retval = $c->agi->exec('SpeechBackground', $path);
> -
Hello,
I have a situation where a remote worker dials in to the asterisk server, enters
the "secret code", then dials out via Disa on a PRI. This was all working great
until this morning. Now the calls is placed out, connected but there is no
voice from/to either phone. This is what shows on the
Since you're "Perling" it, why not just put the $sb_retval in a while loop
like this:
- my $response_good=0;
- my $sb_retval=undef;
- while (! $response_good) {
-my $tmp_retval = $c->agi->exec('SpeechBackground', $path);
-if ($tmp_retval eq 'play_next') {
$sb_retval=$tmp_retval;
On Mon, Jan 25, 2010 at 2:07 PM, Danny Nicholas wrote:
> What does your dialplan snippet to run this look like?
It's part of a Perl FastAGI, running on a separate box from Asterisk.
The Perl code is using Asterisk::AGI's exec() method to call
SpeechBackground:
my $sb_retval = $c->agi->exec('Spee
What does your dialplan snippet to run this look like?
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver
Sent: Monday, January 25, 2010 3:57 PM
To: Asterisk Users Mailing List - Non-Commercial Dis
On Mon, Jan 25, 2010 at 1:06 PM, Danny Nicholas wrote:
> I assume you've tried the $GARBAGE in your grammar to make speech run
> "tightly"?
Yes, we use $GARBAGE currently. Here's our grammar:
#ABNF 1.0 UTF-8;
language en-US;
mode voice;
tag-format ;
root $Command;
$Play_Next = [play] next
I assume you've tried the $GARBAGE in your grammar to make speech run
"tightly"?
--
Danny Nicholas
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quinn Weaver
Sent: Monday, January 25, 2010 2:58 PM
To: aster
Hi,
We've run into an interesting (to us) problem with SpeechBackground. Inside a
AGI script, we're playing some extended audio—basically, like a podcast—and we
want playback to stop if and only if the speech recognized matches something
in our grammar. If there's speech that doesn't match, we j
Dear all,
Can someone from Netherlands who has SPA3000 send me the Regional & Line1
settings.
I'm not sure what is wrong but I can call from asterisk to phone attached to
spa300 FXO, but not the other way. I tested three phones: siemens gigaset,
tiptel 160 and hpoj k80(fax). Only tiptel 160 can c
> From: "cb" Sent: Sunday, January 24, 2010 12:42
> I use the Snom 370 all day long at work. I have never had a problem
> adjusting the volume. I change it multiple times a day as I keep my
> handset on one volume and my headset on another, so I'm always going
> up and down and I've never accident
You must read the upgrade instructions. The database definitions in
res_mysql.conf have changed. The way you reference the database in
extconfig.conf is also different.
On Mon, 2010-01-25 at 09:33 +, Ishfaq Malik wrote:
> What happens when you try the command
>
> mysql -uroot -proot
On Mon, 2010-01-25 at 05:49 -0500, Doug Lytle wrote:
> Kingsley Tart wrote:
> > Hi,
> >
> > Does anyone know what it means when I've got an incoming fax routed
> > through to iaxmodem+hylafax and then I see this in the asterisk log:
> >
> > DEBUG[18902] chan_dahdi.c: Detected digit 'f'
> >
>
>
Glad we could (not) help.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Monday, January 25, 2010 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
On Mon, 2010-01-25 at 07:50 -0800, Lee Howard wrote:
> Kingsley Tart wrote:
> > DEBUG[18902] chan_dahdi.c: Detected digit 'f'
> >
> > This happens just after the initial fax negotiation has started and
> > seems to correspond with the sending fax machine giving up.
>
> Turn off fax detection.
Hi
This is crazy. Something about writing to the list gives me ideas ;)
What I am looking for is
show manager command pausemonitor
;)
Thanks anyway, all.
Julian
2010/1/25 Julian Lyndon-Smith
> Oh, crap. the second I send, I realize I use features.conf, right ? ;)
>
> Is there any other way o
Oh, crap. the second I send, I realize I use features.conf, right ? ;)
Is there any other way of getting this into the dialplan ? I would rather
not have to have the users pressing a key, but for software to intercept the
appropriate point and perform some AMI command
Julian
2010/1/25 Julian Lyn
Yeah, was looking at this - my "issue" is that the dialplan is already
running (the channel is already bridged to a SIP phone), so how do I tell it
*which* channel to pause ?
Julian
2010/1/25 Danny Nicholas
> Check this link
>
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor
>
>
>
> D
On Mon, Jan 25, 2010 at 6:10 AM, bhrugu mehta wrote:
> Hi, all
> Is ther any way to pass channel queue such a way
> Queue(SIP/1001&SIP/1002&SIP/1003)
>
> thanks,
>
> Bhrugu Mehta
>
>
You would define those SIP peers as members in queues.conf:
[queue_name]
member => SIP/1001
member => SIP/1002
me
Kingsley Tart wrote:
> DEBUG[18902] chan_dahdi.c: Detected digit 'f'
>
> This happens just after the initial fax negotiation has started and
> seems to correspond with the sending fax machine giving up.
Turn off fax detection.
Thanks,
Lee.
--
___
Hi,
I'm experiencing some strange problems with out Digium card.
First the details abount hardware and software:
Digium, Inc. Wildcard B410 quad-BRI card (rev 01)
Asterisk 1.6.0.20
dahdi-linux-complete-2.2.1-rc2+2.2.1-rc2
libpri-1.4.10.2.tar.gz
The problem now is that there are a number of cli
Recently safe_asterisk is failing to pick up ASTSBINDIR. I've never had
this problem before and even when I move to back versions I have the
issue. I did upgrade safe_asterisk and the init.d scripts a version or
so ago but even when I try older ones I still have the problem. When I
hard code
On Sat, 2010-01-23 at 21:19 -0500, Alex Balashov wrote:
> > What is the situation with Asterisk and fax over IP ? Can Asterisk
> > receive a fax over a POTS or ISDN line ?? Do I then need a Digium
> > TDM-card and an FXO-module or a T38-gateway ?
>
> Despite what anyone may say about Fax over IP
>From what I read, queue is agent-specific, not channel (I've only been
playing with this for two days, so don't jump too hard, gurus.)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta
Sent: Monday, January 25, 2
Yes, you can using SIP
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-Soporte PostgreSQL
*-www.jqmicrosistemas.com
*-809-849-8087
*---*
> From: qu...@vega.com.vn
> To
thanx... a typo... the routers local ip is 10.26.208.253
yves
Tim Nelson schrieb:
> - "Yves Arikoglu" wrote:
>
>> Hi
>>
>> My System is:
>> Asterisk 1.6 running on a Dell Server with two network interfaces.
>> eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
>>
>> t
Check this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor
Depending on your release, you can "pause" and "un-pause" monitoring.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian
Lyndon-Smith
Sent: Mon
Something similar along the lines of a previous email - has anyone
developed, or is using, something similar to this
http://www.veritape.com/wp-content/uploads/2009/11/veritape-call-tagging-module-description.pdf
Julian
--
_
--
During a telephone conversation with a customer, they sometimes give card
details over the phone. under the pci-dss regulations we are not allowed to
record the conversation where the details are being given. Is there a "mute
command" or pause that can be sent to MixMonitor ?
How has anyone else s
- "Yves Arikoglu" wrote:
> Hi
>
> My System is:
> Asterisk 1.6 running on a Dell Server with two network interfaces.
> eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
>
> the local ip 10.26.208.252
> and the external ip 89.244.x.y
>
Either a typo or you have an IP con
thanks, i tried this already but unfortunately no change.
any further suggestions or answers concerning my other questions?
thanx, yves
Cary Fitch schrieb:
> As a guess, they can both talk to the server, but can't talk to each other.
>
>
> What is common to that is they may be trying to reinv
JR Richardson wrote:
> Hi All,
>
> I'm using Asterisk 1.4 branch and checking the status of some SIP
> Peers with the functions ${SIPPEER(101:status)} and the result is "OK
> (48 ms)". Seems to work fine.
That is a bug; the function should be returning OK without the
calculated lag value.
--
K
Hi
My System is:
Asterisk 1.6 running on a Dell Server with two network interfaces.
eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has
the local ip 10.26.208.252
and the external ip 89.244.x.y
eth0 of the server is configured to 10.26.192.107
The Problem:
SIP registration wor
As a guess, they can both talk to the server, but can't talk to each other.
What is common to that is they may be trying to reinvite each other, and
there is no path through the respective routers/firewalls to the other.
So if reinvite is set to yes, set it to no, in both phone profiles on the
s
Hi, all
Is ther any way to pass channel queue such a way
Queue(SIP/1001&SIP/1002&SIP/1003)
thanks,
Bhrugu Mehta
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRI
joern wrote:
>
> I'm trying to setup Web-Meetme 4.0 and I always get the following
> warning when I open the default page http://localhost/web-meetme
>
> Warning: session_start() [function.session-start]: Cannot send session
> cache limiter - headers already sent (output started at
> /var/www/
Kingsley Tart wrote:
> Hi,
>
> Does anyone know what it means when I've got an incoming fax routed
> through to iaxmodem+hylafax and then I see this in the asterisk log:
>
> DEBUG[18902] chan_dahdi.c: Detected digit 'f'
>
This may be related:
http://www.trixbox.org/forums/trixbox-forums/help
Hi,
I'm trying to setup Web-Meetme 4.0 and I always get the following
warning when I open the default page http://localhost/web-meetme
Warning: session_start() [function.session-start]: Cannot send session
cache limiter - headers already sent (output started at
/var/www/web-meetme/locale.php:3
Hi,
Does anyone know what it means when I've got an incoming fax routed
through to iaxmodem+hylafax and then I see this in the asterisk log:
DEBUG[18902] chan_dahdi.c: Detected digit 'f'
This happens just after the initial fax negotiation has started and
seems to correspond with the sending fax
What happens when you try the command
mysql -uroot -proot asterisk
Ish
Zhang Shukun wrote:
> hi,all
>
> when i upgrade from 1.4.28 to 1.6.2.1, i can't connect to mysql
> database anymore, error as follow:
>
> [Jan 25 15:38:25] WARNING[3003]: res_config_mysql.c:325
> realtime_mysql: MySQL RealTim
hi, dear all
MYSQL commands work well in 1.4.28 edition, but not in 1.6.21
is that the grammar is different between them?
extensions.conf
exten => s,2,MYSQL(Query resultid ${connid} SELECT\ callerid\ from\
blacklist\ where\ companycode = ${ARG2} and callerid=${ARG1} and
blockenabled = 1)
cli:
> The problem 'I can place calls but no one can reach me'
> is our number one support question. Advising the user to check the DND
As a general comment, the DND button on a decent phone should LIGHT UP
when it's in use. On the Polycom 650, it is very clear on the LCD
screen with flashing icons, bu
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