sean darcy wrote:
> Mark Michelson wrote:
>> sean darcy wrote:
>>> Danny Nicholas wrote:
You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message
Here is what I use:
; Agent login logout
exten => *20,1,Answer()
exten => *20,n,wait(.0.5)
exten => *20,n,Read(AgentNumber,agent-user)
exten => *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten => *20,n,GotoIf($["${UserID}"=""]?NOUSER)
exten => *20,n,Set(AgentStatus=${DB(users/${
Mark Michelson wrote:
> sean darcy wrote:
>> Danny Nicholas wrote:
>>> You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
>>> you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
>>> Grep for it.
>>>
>>> -Original Message-
>>> From: asterisk-us
Steve Edwards wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
>> Sent: Friday, May 15, 2009 12:39 PM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] meetme dies look
Hello list,
I just updated to the last release my asterisk-addons copy and the DTMF
works almost perfect. I have the same situation that I used to have using
the stable version 1.6.1. The tones gets detected but only one keypress in a
given time. I don't know if this is a configuration problem wit
Agentcallbacklogin was deprecated in Asterisk 1.4 and eliminated from
1.6 so you now need to use Dynamic Agents. Although they claim that is
is simple enough to replace that functionality with dial plan code I
have yet to see a one line example that replaces everything the
agentcallbacklog
Hi everybody
Did anybody by any chance ever work out how to log in and out agents on
Asterisk 6+?
I used to have it working perfect in Asterisk 1.2 but since I upgraded to 6
the agent login functions are gone and the readme file that came with it
made no sense to me.
I noticed somebody on the ne
In article ,
sean darcy wrote:
>
> Well, I installed gsm:
>
> ls /var/lib/asterisk/sounds/en/conf-getconf*
> /var/lib/asterisk/sounds/en/conf-getconfno.gsm
> /var/lib/asterisk/sounds/en/conf-getconfno.ulaw
> /var/lib/asterisk/sounds/en/conf-getconfno.wav
>
> but same result :(
Perhaps it is lo
Have you tried
Exit 1
??
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hristo Benev
Sent: Friday, May 15, 2009 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] change AGI
On Friday 15 May 2009 03:49:05 pm Hristo Benev wrote:
> I came up to this solution, but is there a way to change the AGISTATUS
> variable to FAILURE -> We have it always SUCCESS
if the script you use exits successfully (without an error), AGISTATUS will
always be SUCCESS even if it didn't do what
Thanks for the answer.
I came up to this solution, but is there a way to change the AGISTATUS
variable to FAILURE -> We have it always SUCCESS
Hristo
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Nicholas
Sent: Friday, May 15, 2
Set a variable in the dialplan like this
Exten => s,1,set(AGI_RET=good)
Exten => s,n,AGI(test.agi)
exten => s,n,Gotoif($["${AGI_RET}" = "GOOD"]?good)
exten => s,n,Gotoif($["${AGI_RET}" = "BAD"]?bad)
in the AGI, put this line to generate the return
print STDOUT "SET VARIABLE AGI_RET \"BAD\" ";
How I can change AGI script return status to failure from within the
script?
It always return AGI Script completed, returning 0
Thanks,
Hristo
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing l
I call the firm from my portable at home (zoiper softphone). I have
internal extension 60, and I call the internal SIP-client 10 at the firm
via an IAX-connection over internet.
My colleague at phone 10 answers my call. I ask him to transfer me with
my colleague at extension 50. He then presses "t
Hi everybody,
I don't have any issue(for now) but I wrote a small python script to
help search through the asterisk messages file.
The script will parse 'messages' and then put the relevant info into a
database(I chose mysql) so you can run any queries you like, instead of
having to troll t
sean darcy wrote:
> Danny Nicholas wrote:
>> You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
>> you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
>> Grep for it.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>>
Hi,
This is my first experience with a mediant 1000 and an Asterisk Trixbox.
the mediant has 12 FXOs and 12 FXSs, and I want to use it them all.
I will have extensions connected to the FXS ports, and lines to the FXO.
Can anyone guide me, please?
regards,
--
Guillermo Garron
"Linux IS user f
sean darcy wrote:
> Danny Nicholas wrote:
>> You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
>> you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
>> Grep for it.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
> Sent: Friday, May 15, 2009 12:39 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] meetme dies looking for conf-getconfno
>
>
amit salunkhe wrote:
> Hello,
>
> I am using OpenSIPS to register all the users and planning to use
> asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge.
>
> I have a scenario where the signaling does not happen properly:
>
> 1) A user from Opensips dials an extension 700
Kill the WAVE!!!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, May 15, 2009 1:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] meetme dies looking for conf-getco
Danny Nicholas wrote:
> You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
> you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
> Grep for it.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-bo
Gsm is what I've been using. The problem could be a "duplication" problem -
if there are 2 or 3 foo.* in the v/l/a/s dir, Asterisk "flakes" trying to
process them all. I've had that happen with a foo.wav and foo.gsm in the
same dir. Also check permissions.
-Original Message-
From: aster
Try without first. If no joy, insert and reload dialplan.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Friday, May 15, 2009 12:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I have changed the features.conf file, yes.
And I put this in my extensions.conf :
include => parkedcalls
Is it better to put "exten => 90,1,park()" into my dialplan ?
Greetingz,
Jonas.
On Thu, 2009-05-14 at 16:08 -0500, Danny Nicholas wrote:
> Did you change 700 to 90 in features.conf? I’d
You "lost" conf-getconfno.gsm . Asterisk is trying to play that file to let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
cat meetme.conf
[rooms]
conf => 600
extensions.conf:
[meetme]
exten => 2663,1,MeetMe(,D)
exten => 2663,n,Hangup()
exten => 2666,1,MeetMe()
exten => 2666,n,Hangup()
What I'm expecting is to dial 2663, get a co
Are these attempts to scam SIP calls through my Asterisk server:
[May 13 22:50:41] NOTICE[30888]: chan_sip.c:17295 handle_request_invite:
Call from '' to extension '084312297134' rejected because extension not
found.
[May 14 13:36:35] NOTICE[30888]: chan_sip.c:17295 handle_request_invite:
Call
On Fri, May 15, 2009 at 11:17 AM, Danny Nicholas wrote:
> AFAIK we are on PRI with no alarms.
I'm thinking the common denominator with your arrangement is that call
progress info isn't getting relayed properly, and / or there are line
signaling issues.
To me that means a problem with your DAHDI/
Here's some more info
1. We join these conferences that our clients host. I'd love to host them,
but that's not an option.
2. If I'm listening to joe and try to interrupt him, he doesn't hear it.
3. Most of these clients are on land-lines. These calls work fine on our
old PBX.
4. Just Zapata
On Fri, May 15, 2009 at 10:33 AM, Danny Nicholas wrote:
> It Mostly works for me; Due to internal security concerns, we use POTS with
> TDM400 and TDM410P cards for incoming/outgoing service. Here is the
> short-list of my most common concerns:
> 1. Some numbers (especially AT&T conferences) ne
Hello,
I am using OpenSIPS to register all the users and planning to use asterisk
for Auto Attendant, Queues, Voicemail and Conference Bridge.
I have a scenario where the signaling does not happen properly:
1) A user from Opensips dials an extension 7000 which is an
auto-attendant extension
It Mostly works for me; Due to internal security concerns, we use POTS with
TDM400 and TDM410P cards for incoming/outgoing service. Here is the
short-list of my most common concerns:
1. Some numbers (especially AT&T conferences) never bridge the call, so I
have to Answer, then Dial these numbers
Yehavi Bourvine wrote:
> You check for BUSY. Check for IN_USE instead. That's what I do here (on
> 1.4, but I guess that 1.6 behaves similarly).
>
> When an extension is in IN_USE state I have a decision tree after
> consulting a database:
>
>
> * If the user wants waiting call - dial hi
On 15 May 2009, at 15:18, Tiago Durante wrote:
> On Fri, May 15, 2009 at 9:35 AM, Danny Nicholas
> wrote:
>> Greetings listers,
>> I have been running 1.4.21 for about 7
>> months now,
>> but have been told I have to move up the 1.4 food chain or into the
>> 1.6
>>
Danny,
On Fri, May 15, 2009 at 9:35 AM, Danny Nicholas wrote:
> Greetings listers,
> I have been running 1.4.21 for about 7 months now,
> but have been told I have to move up the 1.4 food chain or into the 1.6
> chain because 1.4.21 is too flaky for our POTS line handling
In article ,
DHAVAL INDRODIYA wrote:
> hi Tony,
>
> Thanks for your reply
>
> I tried with the same but it dont work for me and I used DAHDI channel and
> PRI following are
>
> mine chan_dahdi.conf setting
>
> [channels]
> language=en
> context=from-pstn
> switchtype=euroisdn
> pridialplan=loc
Greetings listers,
I have been running 1.4.21 for about 7 months now,
but have been told I have to move up the 1.4 food chain or into the 1.6
chain because 1.4.21 is too flaky for our POTS line handling (does funny
things with echo, doesn't connect to external conference c
In order to more closely align the services that Digium provides to the
Asterisk open source community with the Asterisk project itself,
we've recently renamed many of the servers that provide these services.
Effective immediately:
1) http://bugs.digium.com has moved to https://issues.asterisk.org
On Fri, May 15, 2009 at 5:06 AM, Khaled W. Chehab wrote:
> Dears I installed digium fax and followed the instruction at
> http://downloads.digium.com/pub/telephony/fax/README,And as you can see
> above that t38 is loaded
Step 1: This is an asterisk-users question.
Step 2: This is not an asteris
On Fri, May 15, 2009 at 12:12:23PM +0300, Tzafrir Cohen wrote:
> On Fri, May 15, 2009 at 02:47:30PM +1000, Alex Samad wrote:
> > Hi
> >
> > I have a fxs (tdm410 ) connected to a pstn that is primarily used for
> > faxing, it is meant to be a just in case line.
> >
> > How do I tell asterisk to ig
Hi,
I'm wondering if someone could tell me if DNS names in Asterisk .conf files
(eg. iax.conf, sip.conf) are supposed to be resolved only once at * startup or
periodically, according to DNS timeouts.
I'm asking because I defined host=voipb1.mydomain.local in a iax.conf trunk and
it was workin
Hi Carl,
In addition to what was stated below, you will need to configure ODBC on
the operating system. So in other words, to get Asterisk to start
talking to ODBC, you need to configure 4 things:
1. /etc/odbcinst.ini needs to be configured. This depends on your OS and
which database type your are
Hi,
I tried to call transfer from Zap to Zap channels using Manager Api. How to
call transfer
between Zap channels. Please give any sample code to do this work.I have
able to call transfer
from sip to sip using Manager api.
This is my code for call transfer from sip to sip
Originate
Can You try to fire command dialplan show On-net
what it shows if it does not show anything then it not properly loaded .
regards
Dhaval
On Fri, May 15, 2009 at 12:30 PM, michel freiha wrote:
> Dear Sir,
>
> As I'm using asterisk in real time mode, On-Net is defined in extensions
> table...The
hi Tony,
Thanks for your reply
I tried with the same but it dont work for me and I used DAHDI channel and
PRI following are
mine chan_dahdi.conf setting
[channels]
language=en
context=from-pstn
switchtype=euroisdn
pridialplan=local
prilocaldialplan=local
callerid=asreceived
signalling=pri_cpe
u
Hi,
is there a possibility to tell zaptel or Asterisk to modify the DTMF
sensibility?
The problem what i have is that the Asterisk don't get all Numbers which the
analog-FAX dial, let say the FAX dial 123456789 the Asterisk get to number
24679. I think that can be to DTMF Tone duration or the
Dears I installed digium fax and followed the instruction at
http://downloads.digium.com/pub/telephony/fax/README,And as you can see
above that t38 is loaded
I am using a call file to send fax1.tif file as fax to the gateway named
add
The problem that Addpac send always Receive 488 Not acce
On Thu, May 14, 2009 at 10:55:42PM -0400, sean darcy wrote:
> Tzafrir Cohen wrote:
> > On Thu, May 14, 2009 at 06:37:53PM -0400, sean darcy wrote:
> >> I have two internal analogue extensions off a TDM400P. If the first is
> >> busy, I'd like to ring the second. So:
> >>
> >> [incoming]
> >> exten
On Fri, May 15, 2009 at 02:47:30PM +1000, Alex Samad wrote:
> Hi
>
> I have a fxs (tdm410 ) connected to a pstn that is primarily used for
> faxing, it is meant to be a just in case line.
>
> How do I tell asterisk to ignore the line completely - ie don;t pick up
> when it rings ?
Simply don't A
Dear Sir,
As I'm using asterisk in real time mode, On-Net is defined in extensions
table...The structure of the On-Net context is listed in the link below:
http://pastebin.com/d50b2ba42
Regards
On Fri, May 15, 2009 at 1:23 AM, Darryl Dunkin wrote:
> What does your ‘On-net’ context look like?
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