Hello, I need some advice:
I use 2 different suppliers of trunk SIP in my infrastructure, both send me
regularly prices in a .csv format.
So I have two SQL tables that contain the prefix and the tariff.
For now, I generate a dialplan with a Perl script that allows me to select
the prefix trunk
Hi
A small question on Asterisk Manager. I use Perl Script for start a call:
my $response = $astman-sendcommand( Action = 'Originate',
Channel =
'SIP/ASTERISK/$Extension',
Exten = '200',
t...@ovm-group.com
Take a look here:
http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/
Am 16.06.2013 09:43, schrieb Olivier CALVANO:
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow 11.4.0
and now, when we want play sound, we have a errors:
-- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c,
Fermeture) in new stack
[Jun 16 07:35:06]
exists.
ipbx*CLI
2013/6/16 Matthew Jordan mjor...@digium.com
On Sun, Jun 16, 2013 at 2:43 AM, Olivier CALVANO o.calv...@gmail.comwrote:
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow 11.4.0
and now, when we want play
Hi
We want optimize my extensions file conf on asterisk 11.4.0 :
We have a big quantity of extensions, all are same design:
; Destination: Gambia Type: Fixe
exten = _00220X.,1,Set(CDR(CodeCom)=BUS-GMB)
exten = _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)})
exten =
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when i configure on the new server, the same
information in MySQL, we have a error:
[Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime:
defaault values as
if it did not find any database configuration.
Ron
On 03/06/2013 10:49 AM, Olivier CALVANO wrote:
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when i configure
to connect
database server SSI on with SSI and correct into my config file
2013/6/3 Olivier CALVANO o.calv...@gmail.com
The database schema (table) is different in Asterisk 11.4 ?
because i have configured:
cdr_mysql.conf
extconfig.conf
res_config_mysql.conf
and on the mysql server
No other idea ?
2013/6/3 Olivier CALVANO o.calv...@gmail.com
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when i configure on the new server, the same
information in MySQL, we have
that they know about.
Ron
On 03/06/2013 12:19 PM, Olivier CALVANO wrote:
No other idea ?
2013/6/3 Olivier CALVANO o.calv...@gmail.com
Hi
i have installed a new Asterisk server on Fedora. My first server use
Asterisk 1.6.x with a MySQL CDR and
realtime.
I have a small problems, when
that there is no database at all so they
just complain about the piece that they know about.
Ron
On 03/06/2013 12:19 PM, Olivier CALVANO wrote:
No other idea ?
2013/6/3 Olivier CALVANO o.calv...@gmail.com
Hi
i have installed a new Asterisk server on Fedora. My first server use
an ; (semicolon) for remarks! So he found at the
first the old remarks and tried to access my database with the false data.
Ron
On 03/06/2013 3:18 PM, Olivier CALVANO wrote:
on this server we don't have mysql.socket because he don't have mysql
server
we want access to a mysql based
RealTime: Failed to
connect database server xxx on xxx.xxx.net (err 2003). Check debug for more
info.
what is the command in asterisk for i see the SQL query ?
2013/6/4 Olivier CALVANO o.calv...@gmail.com
oh ron thanks for your help :
We have deleted all commented line, only put
?
2013/6/4 Ron Wheeler rwhee...@artifact-software.com
Well, at least you are making progress.
What is the error in the debug log?
Ron
On 03/06/2013 8:03 PM, Olivier CALVANO wrote:
grrr no in asterisk -d i have no error, but when i start normaly
asterisk i have :
[Jun 4 02:01:45
Perfect that's work ;=)
very thanks
Le 25 avril 2012 10:19, Olivier CALVANO o.calv...@gmail.com a écrit :
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't think
because it's a VMWare server and in the same server i have other
asterisk without this problems.
best regards
Olivier
Le 25 avril 2012 09:35, Olivier CALVANO o.calv...@gmail.com a écrit
, at 8:15 PM, Olivier CALVANO wrote:
Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't
think
because it's a VMWare server and in the same server i have other
asterisk without this problems.
best regards
Olivier
Le 25
and so lets authenticate
this one and here it fails and rejects the call.
BR,
Sammy.
On Tue, Apr 24, 2012 at 7:06 PM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL
Hi
i have a lot of error in the CLI of one of my Asterisk:
[Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation
not permitted
[Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8639de8
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
Le 25 avril 2012 09:34, Leandro Dardini ldard...@gmail.com a écrit :
2012/4/25 Olivier CALVANO o.calv
Hi
I have a iax configuration:
[IaxServer]
type=friend
host=172.16.1.14
port=4569
defaultuser=IaxServer
auth=md5
secret=mypassword
context=Internal
peercontext=Internal
qualify=yes
trunk=no
disallow=all
allow=alaw
i see the peer:
ipbx*CLI iax2 show peers
Name/UsernameHost
Hi Sammy,
Yes my telco have a lot of IP, i receive a call from ~20 ip ..
I can't put a subnet ?
best regards
Le 23 avril 2012 07:57, SamyGo govoi...@gmail.com a écrit :
Hi,
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
This line is telling you everything. The peer you've
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.
On my second server (phone is connected on this
Hi
No idea ?
thanks
Olivier
Le 24 avril 2012 16:06, Olivier CALVANO o.calv...@gmail.com a écrit :
Hi
i have a strange problems on my asterisk server:
I have two asterisk server.
On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same
Hi
I have a small problems with incoming call.
I have a peer actually configured for outcall :
sip.conf:
[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
Hi
can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6
Best regards
Olivier
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New to Asterisk? Join us for a live introductory
a écrit :
On Wed, Apr 18, 2012 at 05:42:18PM +0200, Olivier CALVANO wrote:
Hi
can i don't sent into the SIP invite the Session Timer ? on asterisk 1.6
Have you tried 'session-timers=refuse' ?
--
Barry
--
_
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.
http://lists.digium.com/pipermail/asterisk-dev/2006-August/022313.html
Kind Regards
Stuart Elvish
On 04/16/2012 08:16 AM, Steve Edwards wrote:
On Sun, 15 Apr 2012, Olivier CALVANO wrote:
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten =
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 1:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set variables from one asterisk ta a second.
Hi
Thanks for your help but i don't know this variable
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO o.calv...@gmail.com a écrit :
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten
){0:4}})
and post your CLI output. We need to see if the OP's suggestion is getting
to Asterisk #2.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 2:29 PM
Hi
i am search a solution for change the number called.
Sample:
I have a Linksys SPA942 connected in SIP with my server.
When this phone call a number: 043112
automatiquely change in 3343112
because my carrier want a number in international format.
It's possible ?
thanks
Olivier
--
Hi
it's possible into Asterisk 1.6.x to limit a call at 120 mn ?
after 120mn, hangup and the customer call a new time
thanks
olivier
--
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New to Asterisk?
Thanks but i read:
; The maximum number of concurrent calls you want to allow
Not limit the duration of a call ;=)
Le 2 avril 2012 16:55, Bakko asannu...@gmail.com a écrit :
Hi,
look at maxcalls parameter on the asterisk.conf file.
regards
El 02/04/2012 16:46, Olivier CALVANO escribió
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Wednesday, August 03, 2011 4:56 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Know the number of concurrent dial ?
Hi
I connected Asterisk 1.6
Hi
I connected Asterisk 1.6 has several SIP provider, Do you know a tool
to make a graph of the number of simultaneous calls incoming and
outgoing ? and know the max outgoing call in same time ?
thanks
Olivier.
--
_
--
Hi
i want add a numeric password to a call in :
User call to a number,
Asterisk answer and request: please insert your pin code
the user enter a numeric code of 4 number and #
when asterisk have the code, he start a api.
Anyone have a sample of extension.conf for this ?
thanks
Olivier
--
, Olivier CALVANO wrote:
Hi Mark
Thanks for your answer, but i am new in asterisk ;=) the context
start-audio ...
i put it into the extension.conf ?
because i have a error:
[Apr 3 20:12:08] WARNING[28078]: config.c:1154 process_text_line: No
'=' (equal sign) in line 134 of /etc/asterisk
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2)})
exten = _00339,3,Set(CLI=${CUT(cut1,,1)})
exten = _00339,4,Set(toexten=${CUT(CLI,@,1)})
exten =
an audio track on the callee's
channel (SIP/MyOperator-) before bridging the audio.
On 04/03/11 12:01, Olivier CALVANO wrote:
Hi
i use this into my extension :
exten = _00339,1,Set(foo=${SIP_HEADER(To)})
exten = _00339,2,Set(cut1=${CUT(foo,:,2
= _003318364,n,ExecIf($[${toexten} =
81169]?Dial(SIP/204,180,rt):Noop(${toexten}))
exten = _003318364,n,ExecIf($[${EXTEN} =
003318364]?Dial(SIP/203,180,rt):Noop(${toexten}))
On Thu, Mar 24, 2011 at 11:13 AM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
Anyone know
Hi
I request your help because i don't have actually a solution at my problems.
I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers:
003318364 (official number)
081169 (Nddi Number)
When i receive a call on the 081169, he don't
can edit sip.conf and in peer
entry, try to add,
context=(desired_context for peer)
and then into context write a dial-plan for given number and route a call or
whatever you want to do.
On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO o.calv...@gmail.com
wrote:
Hi
I request your help
Hi
Anyone know a solution at my problems ?
Thanks
Olivier
2011/3/23 Olivier CALVANO o.calv...@gmail.com:
Hi
I request your help because i don't have actually a solution at my problems.
I have a Asterisk Server in 1.6
Connected at a SIP Provider
This provider supply me 2 numbers
Hi
I have in a SIP invite of a incoming call:
INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From:
Hi
i want use the API on my asterisk 1.6, but i have a small problems :
In extension, i start it :
exten = _X.,3,AGI(My-Script.agi)
The perl agi file are started without problems
but i want get into this script a lot of variable:
Type (SIP or IAX)
src (from cdr)
but that's don't work:
Hi
I have two Asterisk Server:
The first server A, all phone are connected
The Second server B only route call to a lot of SIP supplier
the server A sent:
; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR
exten = _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW)
Hi
i don't see a answer at my question
Bye
Jerome
2010/11/9 Olivier CALVANO o.calv...@gmail.com:
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
in extensions.conf:
exten = 0532xx,1,Answer
exten = 0532xx,2,MusicOnHold(Sound_1)
exten = 0532xx,3,Dial(SIP/ACCOUNT001,180,t)
exten =
2010/11/24 Sherwood McGowan sherwood.mcgo...@gmail.com:
On Wed, Nov 24, 2010 at 4:30 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Hi
i have a small problems on Asterisk 1.6 with the MusiconOld :
musiconhold.conf:
[Sound_1]
mode=quietmp3
directory=/var/lib/asterisk/moh/Sound_1
Hi
In Asterisk 1.6/realtime Mysql, we can't put a username/password in a
Dial Command ?:
'Dial', 'SIP/Username:passw...@mypeer/${EXTEN},180,r'
Thanks
Olivier
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Anyone have a AudioCodes with Asterisk ???
2010/9/18 Olivier CALVANO o.calv...@gmail.com:
Hi
i have buy a Audiocode Median 2000 VoIP Gateway and connect it on :
1 E1 30 channels
1 Lan Port
Anyone use this equipements with asterisk ? because i am search a
config sample
Hi
i have buy a Audiocode Median 2000 VoIP Gateway and connect it on :
1 E1 30 channels
1 Lan Port
Anyone use this equipements with asterisk ? because i am search a
config sample for AudioCode and for Asterisk (i am new in VoIP).
I want that all calls arrives on the AudioCode are sent
Sorry i don't really understand your message ;=) my english are bad.
I am search a sample of configuration of the audiocode.
2010/9/18 Paul Belanger paul.belan...@polybeacon.com:
On Sat, Sep 18, 2010 at 6:46 AM, Olivier CALVANO o.calv...@gmail.com wrote:
Anyone use this equipements
Hi
I have a big problems on my Asterisk systems :
I have one Asterisk Server with static IP (no nat) and
6 Linksys SPA941.
All SPA are after a router with NAT:
* SPA-1 and SPA-2 are on the same network,
we have a pat 5060 = SPA-1 and 5061= SPA-2 on the internet router
*
Hi
i want change my asterisk server. Actually, Asterisk work's on a IBM
Server with a internal digium E1 card.
For High availability, i don't want now use internal E1 card.
In my new asterisk systems, i have two server and two E1 not in the same site.
I am search a hardware gateway, if possible
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