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 result
2011/10/4 Olivier
> 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 feature w
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 => _8600[1-2]XX,2,Meetme,${EXTEN}|
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
I've noticed on our system the sound files have to be in an exact format for
Asterisk to play them.
Bit Rate: 128kbps
Audio sample size: 16 bit
Channels: 1(mono)
Audio Sample rate: 8kHz
Audio format: PCM
I actually downloaded a program and remixed the audio files to match these
settings. Before
have you tried with MYSQL command?
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Oct 4, 2011 at 11:25 PM, Bryant Zimmerman wrote:
> Hey all
>
> I wanted to get some input on what you all think is the best way to lookup
You need to make sure your (DAHDI or ZAPTEL ) is set up properly for your
country's CLID protocol
In the US CLID is sent between the first and second rings, and with a proper
configuration Asterisk waits a ring before processing the call
Other parts of the world use different methods and protoco
You have files in /var/lib/asterisk/moh1?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, October 04, 2011 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-us
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: 0981 22
Hey all
I wanted to get some input on what you all think is the best way to lookup
database data from asterisk dial plan.
This is a two fold question.
1. I am using fun_odbc to pull settings and values back and it works good
but is there a better way. I want to maintain performance and sim
i configure new music on hold like below in order to play music for outbond
calls
i want tp play a music until answer form customer
[default1]
mode=files
directory=/var/lib/asterisk/moh1
exten => 0678XX,1,Set(CALLERID(number)=520XX)
exten => 0678XX,n,MixMonitor(zap_g1_${EXTEN}_${UNIQ
On some analogs systems caller id is sent after first ring, so
removing "callerid=asreceived"
may help
Nasir Iqbal
ICT Innovations
http://www.ictinnovations.com/
On Tue, Oct 4, 2011 at 4:38 AM, neo haux wrote:
> Hi
>
> I am testing a degium TDP400P (2fxo+2fxs) on my asterisk
>
> I configure
My favorite is didww.com ,
another one is ipcomms.net (not very prompt with their customer service)
Hope this helps..
On Sat, Oct 1, 2011 at 12:51 AM, amit mehta wrote:
> Hello members,
>
> I am looking for USA incoming DID which can be registered on softphone/IP
> Phone/ Pap2 devices.
>
> T
I see two "problems" here. Problem 1 is that you are using the alaw codec, so
it seems to me that you need this file to exist -
/var/lib/asterisk/sounds/nl/fvdb/beep.alaw. problem 2 is possibly just in my
head as I am still avoiding Asterisk 1.8 like the plague; AFAIK (or this is
just a 1.4
Hi,
Has anyone heard (or read) about an existing or emerging standard targeting
the following feature :
1. a SIP handset receives an incoming call
2. this handset starts ringing
3. then it receives an update asking to autoanswer the ringing call.
This feature would help to build software panels c
Yes,
Copy past error in mail.
In the code it is correct.
sorry
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jose P. Espinal
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discu
Yes,
In the code I use set the language
exten => s,n,Set(CHANNEL(language)=nl/fvdb)
So therefore I try also to place the file in the directory
/var/lib/asterisk/sounds/nl/fvdb/
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.d
On 10/04/2011 10:37 AM, Arjan Kroon | Mobillion wrote:
exten =>
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
Hello Arjam,
Did you notice that there's a missing '}' around the end of the line (on
the UNIQUEID part)?
--
# Jose P. Espinal
# http://w
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
wrote:
> This is my complete CLI logging
>
> -- 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
This is my complete CLI logging
-- 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
[
Usually this message is received because you did something like
playback(beep.gsm) or playback(beep.wav) instead of playback(beep). It is
(IMO) somewhat confusing because you have to do record(foo.gsm) but you have
to playback using playback(foo).
-Original Message-
From: asterisk-users-b
Hi,
I'm using the functionality Record in asterisk 1.8.5.
But when I want to record something I get the following error message:
file.c:644 ast_openstream_full: File beep does not exist in any format
Could anybody tell me where I have to place the beep.gsm file?
I already tried the following dire
On Tue, Oct 4, 2011 at 6:06 AM, Tzafrir Cohen wrote:
> On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote:
>> Is there any reason not to run Asterisk on an Intel Atom board?
>
> Only if it's not strong enough. Note that "Atom" may mean some different
> things. So consider taking various
Hi,
Using Asterisk 1.6.2.13
We are now starting to use *call transfer (patching) function.*
Call flow is as follows:
---
John Calls me and requests him to be connected to Nancy.
I place John's call on Hold
I dial Nancy and speak with her about John
I then patch the call betwe
On 04/10/11 10:21, � wrote:
Am 04.10.2011 10:33, schrieb Sebastian Arcus:
Hello list,
I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with "register =>" statement in
sip.conf).
If I restrict the number of ports used in rtp.conf (to 1
Am 04.10.2011 10:33, schrieb Sebastian Arcus:
> Hello list,
>
> I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
> sipgate.co.uk as a sip agent/client (with "register =>" statement in
> sip.conf).
>
> If I restrict the number of ports used in rtp.conf (to 1-10005 for
> exampl
On Fri, Sep 30, 2011 at 10:50:12AM -0400, Adam Moffett wrote:
> Is there any reason not to run Asterisk on an Intel Atom board?
Only if it's not strong enough. Note that "Atom" may mean some different
things. So consider taking various reports with a few grains of salt.
--
Tzafri
Hello list,
I use Asterisk with one sipgate.co.uk trunk. Asterisk connects to
sipgate.co.uk as a sip agent/client (with "register =>" statement in
sip.conf).
If I restrict the number of ports used in rtp.conf (to 1-10005 for
example) - will that affect the sip sessions to sipgate.co.uk a
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