Hello,
When I get a SIP INVITE as follows:
INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
To: sip:02XX@IP:5060
Contact: sip:1053212@IP:5060
Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com
CSeq: 102 INVITE
Date: Wed,
trunk to a SIP provider Internet, the user does not have to
know...
Best regards,
Mickael
2013/6/13 Matthew J. Roth mr...@imminc.com
Mickael MONSIEUR wrote:
I have a standard Asterisk configuration:
SIP friends (phones) - Asterisk - SIP gateway to PSTN
converter
80.236.215.61
Good morning, or Good afternoon! It depends :-)
I have a standard Asterisk configuration:
SIP friends (phones)-Asterisk-SIP gateway to
PSTN converter
80.236.215.61 109.69.217.6internal IP (
10.4.0.10/255.255.255.0)
When analyzing
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back from 10.4.0.10
-- Stopped music on hold on
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello Steve,
After checking, I confirm that the call is cut precisely to 900 seconds
(15 min).
10.4.0.1
Le 7/03/13 11:12, Mickael Monsieur a écrit :
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back
Hello,
If someone has an example of configuration for Cisco AS5300 / Asterisk,
I am very interested.
Thank you,
Mickael
Le 28/12/12 00:48, Mickael MONSIEUR a écrit :
Hello,
I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem.
Sound from POTS - Asterisk does not work
Hello,
I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem.
Sound from POTS - Asterisk does not work. (In the sense Asterisk - POTS
it works!!)
The problem lies in two directions (call initiated from the Asterisk or
POTS)
I have no firewall between Asterisk and Cisco. (it's a
Thank you I'll watch. Support for Asterisk-Mysql is a bit minimal ... :-(
2011/7/1 Mickael MONSIEUR mickael.monsi...@gmail.com
Hello,
I just implement the SIP Peers with MySQL.
In the structure mySQL missing the following fields:
nat = yes
notransfer = yes
dtmfmode = rfc2833
call-limit
Hello,
I just implement the SIP Peers with MySQL.
In the structure mySQL missing the following fields:
nat = yes
notransfer = yes
dtmfmode = rfc2833
call-limit = 2
canreinvite = no
subscribecontext = blf
subscribecontext (BLF) and call-limit (Protection) are very important ...
Can you help me?
Hello,
I have Asterisk 1.6.2.9-2, the directive sendrpid does not work!
I placed this in my peer: (sip.conf)
sendrpid=yes
trustrpid=yes
or
sendrpid=yes
trustrpid=no
(and restarted Asterisk)
and the line Remote-Party-ID does not appear in my sip debug!
Please help me,
Mickael.
--
Thank you, Andrew.
So, with Asterisk 1.6, I have no alternative but to use SIPAddHeader?
2011/1/10 Andrew Latham lath...@gmail.com
On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR
mickael.monsi...@gmail.com wrote:
Hello,
I have Asterisk 1.6.2.9-2, the directive sendrpid does not work
!
2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com
Hi,
marked - noticed.
I do not know where it comes from, my CPU goes from 2% to 60-70% at a
command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU
e4...@2.40ghz
2010/11/5 Norbert Zawodsky norb...@zawodsky.at
Am
not related to MixMonitor.
Are you 100% sure that your PHP-AGi script is not looping somewhere?
You should try to understand which is the process that is taken you CPU.
On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR
mickael.monsi...@gmail.com wrote:
Hi,
After disabling MixMonitor, I realize
none ?
2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com
Hi,
Have you noticed a marked increase in CPU load when using MixMonitor?
I use PHPAgi and Asterisk 1.6.2.9-2.
Mickael.
--
_
-- Bandwidth and Colocation
Hi,
marked - noticed.
I do not know where it comes from, my CPU goes from 2% to 60-70% at a
command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU
e4...@2.40ghz
2010/11/5 Norbert Zawodsky norb...@zawodsky.at
Am 05.11.2010 10:16, schrieb Mickael MONSIEUR:
none ?
2010
Hi,
Have you noticed a marked increase in CPU load when using MixMonitor?
I use PHPAgi and Asterisk 1.6.2.9-2.
Mickael.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do so but the average is done on
time or day ...
I use Mysql CDR
Thank you,
Mickael
--
_
--
http://forums.cacti.net/viewtopic.php?p=111317
Thank you.
2010/9/23 Faisal Hanif fai...@vopium.com
use CACTI
On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote:
Hello,
I want to graphically display the number of calls per minute to an
extension.
The programs I have found it possible to do
Hi,
My Asterisk is not running on a virtual machine, and Debian does not have an
X Server.
I have no value with Kernel Timing enabled. Do you think it may be bound for
the proper functioning of chan_local? I have no problem with the Dial
(SIP/XX), but only with the Dial (Local/XX) :-(
Do you
2.6.30-2-686 (Debian)
2010/7/21 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote:
Hi,
My Asterisk is not running on a virtual machine, and Debian does not have
an
X Server.
I have no value with Kernel Timing enabled. Do you
Nobody uses chan_local
2010/7/16 Mickael Monsieur mickael.monsi...@gmail.com
Hello
I just coding a AGI script for billing.
- For external calls, I pass the call directly on a trunk. I do :
Dial(trunk1/extension) - OK !
- For internal calls (shortcode, others users ...) I am
Hello
I just coding a AGI script for billing.
- For external calls, I pass the call directly on a trunk. I do :
Dial(trunk1/extension) - OK !
- For internal calls (shortcode, others users ...) I am
Dial(Local/extens...@context/n)
The problem is that through chan_local.so, I sound as
are already in trouble. It's about time someone
come up with a better moduel.
On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Hello,
I look ARI (Asterisk Recording Interface)
the publisher site is closed...
http://www.littlejohnconsulting.com/ari
Thank you
Hello,
I look ARI (Asterisk Recording Interface)
the publisher site is closed...
http://www.littlejohnconsulting.com/ari
Thank you,
Mickael
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Hello,
The MeetMe application refuses MusicOnHold personalized and skip always in
the default!
Have you any idea how to fix this?
-- Executing [028883...@default:1] Set(SIP/109.10.214.1-0002,
CHANNEL(language)=fr) in new stack
-- Executing [028883...@default:2]
Steve Edwards asterisk@sedwards.com
On Fri, 11 Jun 2010, Mickael Monsieur wrote:
Is it possible to connect two callers without going through a conference
(meetme) ?
0) A better subject may attract the interest of someone with relevant
experience. Contacting means nothing.
1) More
because... I use it! But I do not use MeetMe with!
What is the importance of providing binary packets if the conference (MeetMe
app) is impossible without compiling ??
2010/6/12 Tzafrir Cohen tzafrir.co...@xorcom.com
On Fri, Jun 11, 2010 at 04:39:46PM +0200, Mickael Monsieur wrote:
What
What is the interest to supply binary of Asterisk, under debian for example,
while to use MeetMe it is necessary to COMPILE Asterisk ??? :-))
Mickael.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Hello,
Is it possible to connect two *callers* without going through a conference
(meetme) ?
Example:
06:50pm - User 1 call extension 600 and musiconhold / parked call ..
06:51pm - User 2 call extension 600 and connect to User 1.
Thank you in advance,
Mickael.
--
DeadAGI is deprecated in Asterisk 1.6.x !
2010/4/9 Danny Nicholas da...@debsinc.com
Do the call in a context and have the context run the script as a
DeadAGI.
[call_and_do]
- exten = s,1,Dial…
- exten = h,1,Deadagi(…)
--
*From:*
Hello !
I want to call a line and play a sound from the callee before putting it
in connection with the caller. Is this possible?
Example:
Dial(SIP/11, m) // Ring or Music...
if(call==ANSWERED) Play(announce) // Play 'announce' to the called
// To connect caller and called ?
Best
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Mickael MONSIEUR
*Sent:* 26 April 2010 11:22
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* t...@zilok.com
*Subject:* [asterisk-users] play a sound from the callee before
putting it in connection.
Hello
Hi all,
I am running an AGI script in a command dial, or call a SIP trunk.
I want to execute after 10 minutes a voice message (stream file) on the
channel to warn the person that the call is about to end. How to do that?
Thank you,
Mickael.
--
minutes allowed duration of the call and
after 10 minutes it'll play message You have one minute.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-04-07 9:39 AM, Mickael MONSIEUR mickael.monsi...@gmail.com
mailto:mickael.monsi...@gmail.com wrote:
Hi all,
I am running
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