seem to happen if the external
call hangs up, or if the call is answered by the reception phone (first call in
the queue).
Thanks again,
Xavier
-Original Message-
From: Xavier Singer - EcuTek
Sent: 11 July 2013 12:02
To: 'asterisk-users@lists.digium.com'
Subject: IPcort
?
Thanks!
Xavier
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Yes but what about the conference number ?
On 08/27/2010 11:58 AM, Doug Lytle wrote:
Xavier wrote:
Hi All,
Is there a way to use the dynamic feature of the meetme application
(D) and to set an option to configure the minimum length of the
numbers for the conference and the associated pin
Hi All,
Is there a way to use the dynamic feature of the meetme application (D)
and to set an option to configure the minimum length of the numbers for
the conference and the associated pin.
In my case, I'd like them to be at least four digits.
Thanks in advance !
--
___
I totally agree with the barge mode but for future evolution, what
about if there is more than 3 people ?
On 07/14/2010 04:36 PM, Russell Bryant wrote:
- Original Message -
On 07/12/2010 05:36 PM, Xavier wrote:
I've got a question about chanspy and meetme.
I'd like to tr
No one have, at least, an idea ?
On 07/12/2010 05:36 PM, Xavier wrote:
Hi guys,
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the
spied one -> t
Hi guys,
I've got a question about chanspy and meetme.
I'd like to transfer all the persons involved in a chanspy (the guy
spying, the guy that is spied and the guy that is speaking to the spied
one -> total: 3) in a conference room.
Is there a way to do it quickly without especially knowing e
Hi, I want
to interconnect a Alcatel OmniPCX PBX with SIP support with an Asterisk PBX. My
intention is Alcatel PBX manage all external calls and analog extensions and
Asterisk
manage all the SIP users (because I have to pay for every SIP license in
Alcatel PBX and I can’t edit configuration
Has Asterisk any protection against brute force attack for SIP authentication?
Something like a maximum login attempt limit
Thanks
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To UNSUBSC
Have you set the realm in the sip settings in the mobile? Default one is
"asterisk" . It's important too, defining Registration to "Always on", because
if not, it doesn't enable the wifi connection. Finally, don't enable
compression and security
--- El mié, 28/10/09, bilal ghayyad escribió:
Did you publish it somewhere ?
On 10/05/2009 09:19 PM, Danny Nicholas wrote:
There are plenty of good products out there, but I use my own
PERL/Apache/AMI interface for this
*From:* asterisk-users-boun...@lists.digium.
at 3:33 PM, Steve Underwood wrote:
> Klaus Darilion wrote:
> > Xavier Cardil schrieb:
> >
> >> Hi, I've managed to get HYLAFAX>T38MODEM->
> >> ASTERISK>CISCOAS5400 working, but when they are negotiating asterisk
> >> drops
--
NOT ACCEPTABLE HERE<
ACK>
BYE<---
200 OK---->
Thanks for your help.
On Thu, Jul 16, 2009 at 6:17 PM, Kevin P. Fleming wrote:
> Xavier Cardil wrote:
> > Hi
Hi, I've managed to get HYLAFAX>T38MODEM->ASTERISK>CISCOAS5400
working, but when they are negotiating asterisk drops a message telling
"Unknown RTP codec 96 received from gateway" Do somebody know how to fix it
?
Thank you !
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISC
Hello, I heard that since asterisk 1.6.0.6, now you can send faxes with t.38
through asterisk to a PST gateway that supports t.38 too. Is that true ? If
so, what elements you need to make it work beside asterisk and the PSTN
trunk ?
Thanks all.-
___
--
You can handle 600 SIP sessions and about 400 calls doing transcoding (
passing RTP )
On Mon, Jul 6, 2009 at 1:26 PM, Steve Totaro wrote:
> It can make 9977.39 Bogocalls of course!
>
> On Mon, Jul 6, 2009 at 5:17 AM, abdelkader
> wrote:
> > Hello,
> >
> > This is the configuration of my server g
[ OK ]
/usr/sbin/usermod[ OK ]
/usr/sbin/vipw [ OK ]
/usr/sbin/unhide-linux26 [ Warning ]
On Wed, Jul 1, 2009 at 1:42 PM, Bruce Ferrell wrote:
>
>
> Xavi
; > On 1 Jul 2009, at 09:54, Xavier Cardil wrote:
> > > udp0 0 0.0.0.0:2727
> > > 0.0.0.0:* 4989/asterisk
> > > udp0 0 0.0.0.0:9001
> > > 0.0.0.0:* 26354/udp-sender
&
-- Forwarded message --
From: Xavier Cardil
Date: Wed, Jul 1, 2009 at 10:51 AM
Subject: Unknown udp ports listening experts calling !
To: asterisk-users-requ...@lists.digium.com
Hello, last days we run under an very heavy issue with one audio stream
getting mixed with our RTP
No, it doesn't sound like any of our IVR applications. We have 2 two Cisco
AS5400 PSTN to SIP gateways, so yes we do outbound / inbound calls.
Thank you.
On Mon, Jun 29, 2009 at 12:52 PM, Steve Howes wrote:
>
> On 29 Jun 2009, at 11:00, Xavier Cardil wrote:
>
> > I meant t
pdump -r to read it back and clean
> it out a bit more
>
> Cheers Duncan
>
> Xavier Cardil wrote:
> > Hi, do somebody knows how to sniff RTP and
can you hear? (BTW locution
> is a rather uncommonly used word in english).
>
> Steve
>
>
> On 29 Jun 2009, at 10:07, Xavier Cardil wrote:
>
> > Hi, we are experiencing a problem that is very strange, only on SOME
> > calls, a locution jumps in to the RTP stream and both p
Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster
debugging ?
Thanks.
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Hi, we are experiencing a problem that is very strange, only on SOME calls,
a locution jumps in to the RTP stream and both persons between the phones
can hear it. It is looped and it does not stop till hang up. Do you have any
clue about what could be happening ?
Thank you !
__
Hi we have set up two asterisk machines where we do all the managing of SIP
calls. Now, sometimes we call and we get an underlying sound that is a
locution from a customer. What could make this to happen ? Is very strange
to us, but maybe we are missing something... some configuration, or anything
evboard-R2_01/target/cris-axis-linux-gnu/bin
[EMAIL PROTECTED]:/home/francois/foxboard# find . -iname 'target'
./devboard-R2_01/os/linux-2.6-tag--devboard-R2_01/include/config/ip/nf/target
./devboard-R2_01/target
Can someone explain me what I should change in Asterisk Makefile,
could forward each account for each user, but
it wouldn't be practice. So I hope there is a solution for that.
Thanks a lot I wait some feedback.
--
Francois-Xavier Bas
RSS-Global Technologies Ltd.
Bachemer Strasse 266
50935 Cologne
Germany
phone: +49221 297-6491
email: [EMAIL PROTECTED
number.Are tehre some companies that provide this kind
of service instead of suscribing to many official numbers?
--
Francois-Xavier Bas
RSS-Global Technologies Ltd.
Bachemer Strasse 266
50935 Cologne
Germany
phone: +49221 297-6491
email: [EMAIL PROTECTED]
url:www.rss-global.com
begin:vcard
Hi All,
anyone in Spain is using a ONO PRI? In that case are you experiencing any
problems with asterisk
and ONO? Wich are your zaptel parameters?
Thanks
Xavier Gil
__
LLama Gratis a cualquier PC del Mundo.
Llamadas a fijos y
Here is the debuging information when trying to call out
__
LLama Gratis a cualquier PC del Mundo.
Llamadas a fijos y móviles desde 1 céntimo por minuto.
http://es.voice.yahoo.com
error.log
Description: 2522182428-error.log
Hi All,
the * is working rigth for incoming calls and internal calls, but when trying
to call out we got
hanged up. The hangup reason is AST_CAUSE_INVALID_IE_CONTENTS
I've been searching in the mailing list archive as I thing that some thing
similar happens to
someone else but did not find.
We
>On 16 Jan 2006, at 11:05, Xavier Gil wrote:
>
>>
>> We run a asterisk 1.2.1 on a HP Proliant ML310, PIV 3Ghz 2 Mb L2
>> cache, 1 Gb Ram. We have a TE210P
>> digium card configured for E1.
>>
>> This pbx has been running for almost a moth before giv
orward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
channel=>1-15,17-31
callerid=asreceived
thanks in advance,
Xavier Gil Estarellas.
We can call out almost every number. But when calling to numbers with automatic
attenders the
asterisk returns a NO ANSWER as dial status, like the number doesn't exist. Can
any one help as?
We have no idea about was it's happening.
We are runnig an Asterisk 1.2 with a TE210p digium card.
thx
Hi all,
when calling to a queue that has no agents logged in we expect to hang up, here
is the
extensions.conf queue configuration.
exten=> 2020,1,Answer
exten=> 2020,2,Ringing
exten=> 2020,3,Wait(2)
exten=> 2020,4,Queue(gestoria)
exten=> 2020,5,Hangup
But althougth there isn't any agent it let
Hi All,
I have to install an Asterisk PBX into an office with a legacy PBX, an Ericsson
BussinessPhone
250. I wonder which is the best way to comunicate the new asterisk PBX and the
Ericsson's.
My frist aproach is to connect with a Digium TE411P the Asterisk server with
the Ericsson (3
E1's), a
Well, there was a problem with my Fedora installation.
Thanks to everyone.
Xavier Olivella i Rigol
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> May be bison is too new there was an issue a few months ago go back to
1.35
WOW!!!
Downgrading... sounds interesting.
Thanks!
Xavier Olivella i Rigol
e-mail: [EMAIL PROTECTED]
Sirt S.L. www.sirt
> Sounds like you need to update your version of bison or install bison
Hum... I thought it, but nope.
> I'm running 1.35
I'm running 1.875-5
:-OOO
********
Xavier Olivella i Rigol
e-mail: [EMAIL PROTECTED]
Sirt S
t_expr.c] broken pipe
Just this, nothing else.
Zaptel and Libpri installed without any problem and system is Fedora core 2
with kernel 2.6.5-1.358 and bison installed.
Thanks a lot in advance!
****
Xavier Olivella i Rigol
e-mail: [EMAIL PROTECTED]
S
stuff that everybody talk about on this forum. In other words, can I
link the free "T2" card of the Bosch to a Linux box with an E100P
interface and make it a secondary PBX with Asterisk ? One which will
be aware of VoIP ?
Any hint will be very much appreciated ...
Xavier
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