Am 04.10.11 20:40, schrieb Esteban Cacavelos:
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in
Hi list,
How to reduce the meetme wav file size in asterisk. how can I automatically
reduce this file size.
exten =
_8600[1-2]XX,1(record),SetVar(MEETME_RECORDINGFILE=/var/spool/asterisk/meetme/conference_recording-${EPOCH}-${USER}-${TIMESTAMP}-${EXTEN});
exten =
2011/10/4 Olivier oza_4...@yahoo.fr
Hi,
Has anyone heard (or read) about an existing or emerging standard targeting
the following feature :
1. a SIP handset receives an incoming call
2. this handset starts ringing
3. then it receives an update asking to autoanswer the ringing call.
This
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set the language just with two characters.
(exten = s,n,Set(CHANNEL(language)=nl))
And in the directory /var/lib/asterisk/sounds/nl/ I placed the voicefile
beep.alaw.
But with this also I get also the same
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings/serviceline/1317737932.67.wav,0,60) in
new stack
[Oct 4 16:19:38] WARNING[13370]: file.c:644 ast_openstream_full: File beep
does not exist in any format
[Oct 4 16:19:38]
i think you can try placing the beef file in the /var/lib/asterisk/sounds
directory and not the language specific one.
and your system is calling the beep file without having it in the dialplan?
sounds strange somehow to me.
Tarek Sawah
Information Technology Adviser
Integrated Digital
Since you've changed the language (sound directory) So just as a test change
the language back to en and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@
Yes I already try this (only with language nl)
exten = s,n,Set(CHANNEL(language)=nl))
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Van: asterisk-users-boun...@lists.digium.com
hmmm...what i'm saying is this
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
*exten = s,n,Set(CHANNEL(language)=nl))*
*
*
On Wed, Oct 5, 2011 at
can you post the while dialplan? it seems cropped somewhere as i dont' see it
starting or ending anywhere.
Tarek Sawah
Information Technology Adviser
Integrated Digital Systems
CCNP, MCSE, RHCE, TELECOM
USA: +1 386 492 9993
Date: Wed, 5 Oct 2011 12:31:49 +0500
From: govoi...@gmail.com
Sorry:
*exten = s,n,Set(CHANNEL(language)=en)*
and
exten = s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/
recordings/serviceline/${UNIQUEID*}*)
NOT
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
On
Oke, I tried this, but sorry
-- Executing [s@servicelijn:91] Set(CAPI/ISDN1#02/318647615-3e,
CHANNEL(language)=en) in new stack
-- Executing [s@servicelijn:92] Set(CAPI/ISDN1#02/318647615-3e,
A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/1317800460.74)
in new stack
Hi Arjan,
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Just for double-checking, but what directory is listed as the
astdatadir in asterisk.conf?
Best regards,
Jeroen Eeuwes
--
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in advance.!
--
Esteban L. Cacavelos de Amoriza
Cel:
The alaw extension is bugging me.. can you locate the default beep.gsm
/beep.wav file in asterisk sounds directory !?
Also check the output of
*core show file formats*
*core show translation*
Also find out the codec of the established call.!
On Wed, Oct 5, 2011 at 12:50 PM, Jeroen Eeuwes
On 10/5/11 9:50 AM, Jeroen Eeuwes wrote:
Hi Arjan,
I also try to place the voicefile in the directory /var/lib/asterisk/sounds/
and /var/lib/asterisk/sounds/applications/ of but without any success.
Just for double-checking, but what directory is listed as the
astdatadir in asterisk.conf?
These are the directories which I gave in asterisk.conf
astetcdir = /etc/asterisk
astmoddir = /usr/lib64/asterisk/modules
astvarlibdir = /usr/share/asterisk
astdbdir = /var/spool/asterisk
astkeydir = /var/lib/asterisk
astdatadir = /usr/share/asterisk
astagidir = /usr/share/asterisk/agi-bin
Hi Arjan,
I try to change de astdatadir into /var/lib/asterisk/
But when I restart asterisk and I look at the settings in the CLI I still see
Data directory: /usr/share/asterisk
At least that explains why it can't find your beep-file. It is looking
in /usr/share/asterisk and not
Yes, That was the solution.
Thanks.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes
Verzonden: 05-10-2011 10:15
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re:
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same = n,Set(COMMAND=${CMD_NOOP})
same = n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =
n(COMMAND_SWITCH),GoToIf($[${COMMAND}=${CMD_DOSTUFF1}]?LBL_DO_STUFF1)
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
[blah-context]
same = n,Set(COMMAND=${CMD_NOOP})
same = n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)}
same =
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial command.
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 11:36 AM, Olivier
Hi
yes i have noticed the same result when i play a file like the default i can
hear the music but when i play another file there is no sound
about your question danny :yes i have created a file in
/var/lib/asterisk/moh1
and i configure in musiconhold.conf like below
[default1]
mode=files
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit
On 5 October 2011 10:21, Nasir Iqbal na...@ictinnovations.com wrote:
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial command.
Nasir Iqbal
ICT Innovations
What about waiting in queues?
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Wed, Oct 5, 2011 at 1:35 PM, Yaroslav Panych panyc...@gmail.com wrote:
Hello, everyone
Here part of my dialplan context
[globals]
CMD_NOOP=0
CMD_DOSTUFF1=1
CMD_DOSTUFF2=2
CMD_DOSTUFF3=2
I don't know much about queues, but if channel enter into queue it
should not change its state. I.e. not answer, no moh, no interacting
with user input(DTMF). Less I use unknown helpers, better my
configuration is.
Second issue which can appear using queues - its async state. User can
issue 2
Can you please explain what you are trying to do? What I've perceived from
this thread is that you want to put call on hold (passively as in no
resources usage) and then on the base of some User's input from UI proceed
with the call accordingly !!
On Wed, Oct 5, 2011 at 3:33 PM, Yaroslav Panych
thanks for your replay i give the permissions 777 to this file moh1 and i
still have the same issue
best regards
2011/10/5 Sammy Govind govoi...@gmail.com
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but
2011/10/5 Nasir Iqbal na...@ictinnovations.com
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial command.
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
Yes, something like that, but
hold-state should not answer channel. answer command will be given
explicitly. or call can be transfered to Dial command, etc.
2011/10/5 Sammy Govind govoi...@gmail.com:
Can you please explain what you are trying to do? What I've perceived from
this thread is that
2011/10/5 Steve Davies davies...@gmail.com
On 5 October 2011 10:21, Nasir Iqbal na...@ictinnovations.com wrote:
You can do this by an AMI based transfer (Redirect) to Local channel, and
then in local channel's dialplan you need to add your desired custom sip
header followed by a dial
So here's what I think about your scenario:
CALL-FLOW
1- Call come in to asterisk (channel not answered)
2- Event is triggered and User decides what to do with call
3- On basis of what user decided a variable is set.
4- Asterisk on the base of that variable route the call further.
If this is the
Sorry for the resend, but i don't have got any response, so i try to
re-open the same problem.
Hello all,
Form 2-3 weeks i have some problems with incoming ISDN calls, it
interrupts after 1-2 minutes of call. I have tried to debug this with
pri set debug on span 1, i have noticied much of this
Sorry for the resend, but i don't have got any response, so i try to
re-open the same problem.
Hello all,
Form 2-3 weeks i have some problems with incoming ISDN calls, it
interrupts after 1-2 minutes of call. I have tried to debug this with
pri set debug on span 1, i have noticied much
Hello list
i have one question related to meetme,i have to providers with the first one
i put the number with 9 digit 520XX and all works without issue, with
the second i put just the last 3 numbers 500 with meetme there is nothing
but when i put the last 3 numbers like below i can call
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
To
Hey list.
I am having some issues with a parking lot. I am looking for a way that
the person can blind transfer to a parking lot. it works perfectly
accept being an unattended transfer its the caller who hears the parking
lot position. anyone have a work around to this.
exten =
Sorry forgot to mention this is on an asterisk 1.6.2.13 installation
On 11-10-05 12:31 PM, Keith Sloan wrote:
Hey list.
I am having some issues with a parking lot. I am looking for a way
that the person can blind transfer to a parking lot. it works
perfectly accept being an unattended
hi,
you are using pattern matching and not using the right syntax
like that.
exten = _520,1,answer
like that.
On 5 Oct 2011 21:47, salaheddine elharit salah.elharit...@gmail.com
wrote:
Hello list
i have one question related to meetme,i have to providers with the first
one
i put the
hello,
is there some way to notify people in the same pickup group about call
from caller to callee?
i.e. i have call from 111 to 222
there are 222,333,444 in the same pickup group
333,444 see on the phone (aastra) that 111 calling to 222 and can pickup
the call with *8
siemens have this on
someone have been installed Asterisk (Trixbox) on VirtualBox which
is installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i
should on shouldn't do it.
I just installed 3 Trixbox systems in KVM on
On 10/05/2011 01:30 AM, mahesh katta wrote:
Hi list,
How to reduce the meetme wav file size in asterisk. how can I
automatically reduce this file size.
exten =
On 10/05/2011 11:31 AM, Keith Sloan wrote:
Hey list.
I am having some issues with a parking lot. I am looking for a way that
the person can blind transfer to a parking lot. it works perfectly
accept being an unattended transfer its the caller who hears the parking
lot position. anyone have a
Depending on hardware and number of parking lots, could hints be used to let
everyone know that a parking lot was just put into use by blind transfer?
(I have a PERL web interface that does this kind of check for 1.4 but that
probably wouldn't help OP).
-Original Message-
From:
Does anyone know if there is a resource to see what changes were made
between different versions of Digium cards? For example, how
different is a TE410P revision C when compared to a TE410P 5th
generation card?
I know there were changes made to the architecture to address IRQ
issues, etc.. but
Hi,
I commented the option callerid in the file dahdi-channels.conf without
success, My SIP phone still ring after 4-5 secondes :-(
; Span 1: WCTDM/0 Wildcard TDM410P (MASTER)
;;; line=1 WCTDM/0/0 FXSKS (EC: MG2 - INACTIVE)
signalling=fxs_ks
;callerid=asreceived
I am living in Canada, so I
I have naive question. I do not have any hardware on my asterisk host. All I
have are either SIP trunk for DID or hardware ATA which bridges the asterisk
to PSTN. Do I need Dahdi install? Do i have ztdummy for timing issue? I
encounter problem in this when I try to install Dahdi latest but I found
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