Re: [asterisk-users] Start asterisk on boot

2009-01-26 Thread Mustapha bayrouni 
How did you installed asterisk?

2009/1/26 Ralf Träskman r...@adlibris.com:
 Hi



 We runs asterisk 1.6 on a ubuntu 8.04 server.

 How can I get asterisk to start at boot?

 I have created an file named asterisk in /etc/event.d and put in this



# This service maintains Asterisk from the point the system is
# started until it is shut down again.

description Asterisk daemon

start on runlevel-2
stop on shutdown

respawn
exec //usr/sbin/asterisk –f



 But it doesn't work.



 Regards

 /ralf



 

 Ralf Träskman, IT
 AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
 Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
 r...@adlibris.com www.adlibris.com
 P Please consider the environment before printing this e-mail



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Re: [asterisk-users] CentOS and BAT File

2009-01-25 Thread Mustapha bayrouni 
In linux, there is a what is named shell script, for example bash script.
It is a powerful util allowing you to do almost everything with the system.
Combined with awk and sed , bash become more than powerful.
google about bash doc and you'l find all you need.

2009/1/25, David @ULC ucoms2...@gmail.com:

 In windows, we use BAT file to execute few series of command , which help us
 in not writing each command manually everytime we want to execute those
 commands.

 In CentOS, I want to do the same thing.


 Any Advice ?
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[asterisk-users] [asterisk_voip] asterisk and ogg files

2007-03-06 Thread Bayrouni 
Hello,
Is it possible to use ogg stream with asterisk as moh?
I have an icecast2 ogg streamer, but cannot use it with asterisk 1.4

The moh with files works
icecast2 works

but not icecast2+asterisk.

I think I need something like (see below) in music on hold config file:
mode=custom
application=/usr/bin/some_player http://icecast_server:8000/mount

Thank you

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Re: [asterisk-users] read write or only read fields in cdr?

2007-03-01 Thread Bayrouni 
Mike Lynchfield a écrit :
 try not using dst.. maybe its a regex on te fieldname that matches for
 reserved keywords..
 
 try pre_dest instead
 
 On 2/28/07, Bayrouni [EMAIL PROTECTED] wrote:

 Hello,


 I created a new field named pre_dst of type varchar(80) exactly like dst
 field in cdr table.

 In the dialplan I put:
 exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

 and when I call, all goes fine except that pre_dst has always NULL value
 in cdr.

 Do you know why?
 Is something wrong I did?


 I know that original fields in cdr are only readable, but in this cas
 pre_dst is one I created myself !!!

 Thank you.



I tried yet pre_dst un cdr table.
I even change it to foo or coco or whatever, but nothing is written in
this custom field except NULL.


+-+++--+
| clid| dstchannel | coco   | dst
   |
+-+++--+
| 2007-02-28 13:31:43 | SIP/x-081b1fd8 | NULL   x |

thanks.

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Re: [asterisk-users] read write or only read fields in cdr?

2007-03-01 Thread Bayrouni 
Edgar Luna a écrit :
 Hi,
 
 On Wed, 2007-02-28 at 23:43 +0100, Bayrouni wrote:
 Hello,
 
 In the dialplan I put:
 exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

 and when I call, all goes fine except that pre_dst has always NULL value
 in cdr.

 Do you know why?
 Is something wrong I did?
 
 As far as I know, custom fields doesn't work with any database backend,
 only with CSV. There is an addon in the bug tracker but seems that it
 isn't finished.
 
 
 Regards.

Thank you for this information.

Regards
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Re: [asterisk-users] multiple phones registered for the same user

2007-02-28 Thread Bayrouni 
Eric ManxPower Wieling a écrit :
 Yuan LIU wrote:
 From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
 Date: Wed, 28 Feb 2007 10:57:43 -0600

 Ricardo Carvalho wrote:
 Can't I register multiple phones with the same user/password? That's
 what I pretend to do, not ring groups...

 Ricardo,

 Any particular reason for not using ring groups?

 No, you cannot register multiple phones with the same user/password.

 Just curious: can I register multiple phones with one user name but
 different passwords?
 
 no.
 ___


Which is relevant for asterisk (like any other client/server based
architecture), is the session.

Your phone (hard||soft) is the client.
Your PBX asterix is the server.

Your session is defined by your agent confiuration (and  configuration
data is sent in SIP protocol over TCP/IP suite protocol) .

But first there is a connection.(tcp/ip)

And on the same IP/PORT there is only one connection. If you change
username/password this is still one connection and the same connection.

username password are mostly used to authenticate and not to connect.


cheers

Bayrouni

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[asterisk-users] this i a test

2007-02-28 Thread Bayrouni 
Sorry for disturbing, but I sent some messages today and I am not seeing
them on this list.
Can sombody tell me, in case this message appear on the list.

Thank you
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[asterisk-users] read write or only read fields in cdr?

2007-02-28 Thread Bayrouni 
Hello,


I created a new field named pre_dst of type varchar(80) exactly like dst
 field in cdr table.

In the dialplan I put:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

and when I call, all goes fine except that pre_dst has always NULL value
in cdr.

Do you know why?
Is something wrong I did?


I know that original fields in cdr are only readable, but in this cas
pre_dst is one I created myself !!!

Thank you.



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[asterisk-users] asterisk CDR and mysql

2007-02-27 Thread Bayrouni 
Hello all,
I added a record named pre_dst in the cdr table.

It has the same type as dst field.

And I used this line in the dialplan:
exten = _7.,1,Set(CDR(pre_dst)=${EXTEN:1})

When I call, 70123456, (7 is only to use the provider trunk),
I have this in the CLI:
Executing Set(SIP/foo-0816a490, CDR(predst)=0123456) in new stack
-- Executing NoOp(SIP/foo-0816a490, 0123456) in new stack
-- Executing Dial(SIP/moi-0816a490, SIP/[EMAIL PROTECTED]) in new stack

But  nothing in the pre_dst field in cdr table.

Is there something wrong I did?

Thanks in advance.
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[Asterisk-Users] how to run asterisk?

2006-03-01 Thread Bayrouni 

Hello all
Can I run asterisk under normal user?
Or any other user than root?

If yes, are there any changes to do?

When I run asterisk under my username, I have some errors:
Parsing '/etc/asterisk/logger.conf': Found
Logger Warning: Unable to open log file 
'/var/log/asterisk/messages': Permission denied
Mar 1 10:41:42 ERROR[3140]: logger.c:609 init_logger: Unable 
to create event log: Permission denied.


Thank you
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[Asterisk-Users] not consistent log from asterisk

2006-02-24 Thread Bayrouni 

Hello,
I have 2 channels in iax.conf

[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes

[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xx
secret=***
auth=md5


The problem is:
When I tell FWD to call me I have this output in my asterisk
consol:


Executing Dial(IAX2/iaxfwd-outbound-3, SIP/|PHONE_1|30)
in new stack

If I comment  iaxfwd-outbound channel [iaxfwd-outbound],
then the output is correct:
Executing Dial(IAX2/192.246.69.186:4569-1,
SIP/PHONE_1|30) in new stack.

(192.246.69.186:4569 : this is from FWD)

Thank's  for any help
a+
--
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[Asterisk-Users] not consistent log from asterisk

2006-02-23 Thread Bayrouni 

Hello,
I have 2 channels in iax.conf

[iaxfwd]
type=user
callerid= Free World Dialup
inkeys=freeworlddialup
auth=rsa
context=incoming
qualify=yes

[iaxfwd-outbound]
type=peer
host=iax2.fwdnet.net
username=xx
secret=***
auth=md5


The problem is:
When I tell FWD to call me I have this output in my asterisk 
consol:



Executing Dial(IAX2/iaxfwd-outbound-3, SIP/|PHONE_1|30) 
in new stack


If I comment  iaxfwd-outbound channel [iaxfwd-outbound],
then the output is correct:
Executing Dial(IAX2/192.246.69.186:4569-1, 
SIP/PHONE_1|30) in new stack.


(192.246.69.186:4569 : this is from FWD)

Thank's  for any help
a+
--
Bayrouni
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[Asterisk-Users] how to tape letters in xlite

2006-02-21 Thread Bayrouni 

Hello all,
How to tape letters in xlite softphone, when using the 
Directory application (or generally when is needed).


Thank you.
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[Asterisk-Users] test

2006-02-21 Thread Bayrouni 


--
Bayrouni
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[Asterisk-Users] asterisk to FWD

2006-02-07 Thread Bayrouni 

Hello all,
Here is my problem,

I try to place a call to FWD (free world dialup) trough my asterisk PBX.

my config is as follow:

extensions.conf

[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD

iax.conf

[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr

register = xx:[EMAIL PROTECTED]

[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes

when I call the 613 number (echo FWD service), I have this
message from my PBX:
 Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new 
stack

-- Called iaxfwd-outbound/613
Feb  7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest: 
Auto-congesting call due to slow response

-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
  == Everyone is busy/congested at this time (1:0/1/0)

Please, how can I resolve this problem?

Thank you very much


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[Asterisk-Users] test

2006-02-07 Thread Bayrouni 


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Re: [Asterisk-Users] asterisk to FWD

2006-02-07 Thread Bayrouni 

Mark Phillips wrote:
One problem I can see is that you're not using the keys that come with 
asterisk.


Mine (which works!) looks like this

iax.conf

register = user:[EMAIL PROTECTED]

[iaxfwd]
type=peer
context=from-fwd
permit=65.39.205.0/24
auth=rsa
host=iax2.fwdnet.net
inkeys=freeworlddialup
disallow=all
allow=ulaw
qualify=yes

extensions.conf

; Calls to FWD
exten = _393.,1,Set(CALLERID=37720)
exten = _393.,2,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:3}|20)
exten = _393.,3,Congestion

[from-fwd]
exten = 37720,1,SetCallerID(393${CALLERIDNUM})
exten = 37720,2,Dial(SIP/2208,20)
exten = 37720,3,Voicemail,u2208
exten = 37720,4,Hangup
exten = 37720,103,Voicemail,b2208
exten = 37720,104,Hangup

Try this and see how it goes.

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com


Bayrouni wrote:


Hello all,
Here is my problem,

I try to place a call to FWD (free world dialup) trough my asterisk PBX.

my config is as follow:

extensions.conf

[internal]
exten = 613,1,Dial(IAX2/iaxfwd-outbound/613)(service echo de FWD)
exten = xx,1,Dial(IAX2/iaxfwd-outbound/xx) mon numero FWD
exten = yy,1,Dial(IAX2/iaxfwd-outbound/yy) celui d'un ami FWD

iax.conf

[general]
context=default
bandwidth=low
disallow=lpc10
jitterbuffer=no
forcejitterbuffer=no
tos=lowdelay
autokill=yes
allow=ulaw
language=fr

register = xx:[EMAIL PROTECTED]

[iaxfwd-outbound]
type=peer
username=xx
host=fwd.pulver.com
secret=mon_passwd_FWD
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
allow=g726
nat=yes

when I call the 613 number (echo FWD service), I have this
message from my PBX:
 Executing Dial(SIP/xlite-9f55, IAX2/iaxfwd-outbound/613) in new 
stack

-- Called iaxfwd-outbound/613
Feb  7 09:38:17 NOTICE[2744]: chan_iax2.c:2821 auto_congest: 
Auto-congesting call due to slow response

-- IAX2/iaxfwd-outbound-1 is circuit-busy
-- Hungup 'IAX2/iaxfwd-outbound-1'
  == Everyone is busy/congested at this time (1:0/1/0)

Please, how can I resolve this problem?

Thank you very much



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Thank you,
yes,
there was problems with some keys.
the secret was incorrect and host was too incorrect.
Thanks
a +
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Re: [Asterisk-Users] MP3player Problem

2006-02-07 Thread Bayrouni 

office wrote:

Hi,
 
 
i use in my extensions.conf a testline for an internal test : 
 
exten = 10,1,MP3Player(/var/lib/asterisk/mohmp3/fpm-calm-river.mp3)
 
 
When i call 10, Asterisk answer and i see in the CLI, that MP3player 
works without problems - but i can't hear the sound at the phone ?  
 
Where is the Problem ?
 
 
Walter
 

Hello,
you are missing mpg123,Install it and moh will work

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[Asterisk-Users] xlite and letters

2006-02-07 Thread Bayrouni 

Hello

How to use letters with xlite?
Thank you very much
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