Hello! Thnxs for reading!
I've an IPLAN virtual PBX, that allows me to connect via zoiper or gigaset,
for instance (and it works!)
Connection parameters are:
Authentication Name: Número 11
Authentication password: 12345678
Username: 11
Display name: 11
Domain:
Marco--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
t38_udptl_ec=fec
t38_udptl_maxdatagram=400
[trunk-patton]
type=auth
auth_type=userpass
password=X
username=X
=
Thanks
Marco
--
_
-- Bandwidth and Colocation Provided by http
Hello,
I set up a new server for Asterisk with 11 cert 6 on it. I am migrating from a
previous server. I have replicated all the configurations, modules and setup
that I know of. However, when I tested an outbound call, it didn’t work.
Checking the asterisk message log yielded nothing. Any
problem.
I've already read all the information about canreinvite and directmedia
Can anybody help me?
Thanks a lot
Marco
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
the SIPML5 seems not able
to connect to the asterisk box.
Thank you and best regards,
Marco Signorini.
On 06/12/2014 03:21 AM, Steve Ng wrote:
I am using Asterisk v12.3.
As far as DTLS, I understand that applying the following Javascript
will temporarily fix for SIPML5 to Asterisk:
https
?
Below is my configuration. The sofpthone is registered as 1060.
Thanks in advance.
Marco Signorini.
pjsip.conf:
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/etc/asterisk/sslcert.pem
method=tlsv1
[1060]
type=endpoint
transport=transport-tls
context=from-internal
use_avpf=yes
so I can't
tell you if this is something true for Debian 6.06 too.
Thanks.
Marco Signorini.
On 02/26/2013 05:38 PM, Doug Lytle wrote:
I'm hoping someone can help me here.
I've purchased replacement systems for 3 aging 1.4.x installs. I'm
hoping to setup Asterisk 11, dahdi 2.6.1 and Oslec
: http://pastebin.com/ARUC0j4t
The asterisk IP : 87.248.56.101
The next hop IP : 87.248.56.100
Is it a bug? i'm already search on google, but i dont find anything.
Let me know, if you need more information.
Thanks for all
Best Regards
Marco
Ok, thanks for all
Best Regards
Marco
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Joshua Colp
Inviato: mercoledì 26 settembre 2012 19:37
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto
Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google, but I could not find anything...
Thanks for all
Best Regards
MC
http://www.connectmi.it/?utm_source=email-signatureutm_medium=emailutm_campaign=Email%2Bsignature
-boun...@lists.digium.com]
On Behalf Of Marco Colombo
Sent: Wednesday, September 26, 2012 10:33 AM
To: Asterisk-Users
Subject: [asterisk-users] Asterisk and History-Info
Hi All,
Someone can tell me if asterisk support the SIP History-Info?
If it supports, how can enable it?
I searched on Google
Thanks a lot!
Marco
Da: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Danny Nicholas
Inviato: mercoledì 26 settembre 2012 17:48
A: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Oggetto: Re: [asterisk-users] R: Asterisk
[mailto:asterisk-users-boun...@lists.digium.com] Per conto di Matthew Jordan
Inviato: giovedì 20 settembre 2012 13:42
A: Asterisk Users Mailing List - Non-Commercial Discussion
Oggetto: Re: [asterisk-users] SIP CANCEL, Reason
- Original Message -
From: Marco Colombo mcolo...@enter.it
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line Reason
Example : Reason : SIP;cause=16;text=Normal Call Clearing
I have already enable use_q850_reason=yes, but this not work.
In my dialplan I have already add : exten = _X.,n,Hangup(${HANGUPCAUSE})
, version of LibPRI
etc.
has anybody experienced these problems on BRI? Any suggestions with regards
to these warnings are welcome!
Kind regards,
Marco Mooijekind.
--
_
-- Bandwidth and Colocation Provided by http://www.api
to register 0x1ab but got back 0x4 statements.
If i run dahdi_tools it fails with a segmentation fault.
Any suggestions are appreciated!
Kind regards,
Marco Mooijekind.
--
_
-- Bandwidth and Colocation Provided by http://www.api
!
Thanks in advance!
Marco Mooijekind.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Hello Gord,
the line icon is solid black, which should indicate the lines are
registered.
Marco.
On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart gord...@gmail.com wrote:
Does the phone show the line as registered? The little phone icon on the
display should be solid for a registered line
Maybe local channels will do the trick? They allow you to schedule delays
between subsequent devices ringing. Not sure whether they work as queue
members.. Marco.
Op 5 dec. 2011 16:35 schreef Sammy Govind govoi...@gmail.com het
volgende:
Hi,
I dont think that 2 Queue commands would help, also
Maybe use a power supply instead of PoE, see if problem still occurs. Marco.
Op 30 nov. 2011 18:46 schreef Olivier oza_4...@yahoo.fr het volgende:
2011/11/30 Mike l...@net-wall.com
Hi Olivier,
** **
It if occurs only on the sidecar, I would imagine this is either a
defective sidecar
need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in
Italy.
My concern is about reliability of USB
Any success stories with it? Tips and tricks?
Thank you and regards,
Marco Signorini.
--
INGEGNI Tech S.r.l.
sitehttp://www.ingegnitech.com
maili...@ingegnitech.com
Gilles
shield on top of Arduino.
Thanks,
Marco Signorini
--
http://www.ethermania.com
http://www.ingegnitech.com
David - asterisk list wrote:
Asterisk as a phone system makes perfect sense in a condo. You can get
all the DID's you want and eliminate costs for the owners. You can offer
standard FXO
me gettng started with asterisk
Marco
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Hi
Did you looked at Arduino + Ethernet Shield?
Is something you can program in C or C++ to receive a simple TCP and/or
HTTP packet and turn on an external relay.
From the dialplan you can run an http query through curl and/or an
external AGI command.
Best regards,
Marco Signorini.
--
Marco
Hello
I recently heard this should be possible. Has anyone experience with this?
Thanks!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
regards,
Marco Signorini.
--
=
- http://www.ethermania.com -
- http://www.ingegnitech.com -
Jose P. Espinal wrote:
Hello list,
I'm facing a little issue with dahdi attempting to load the OSLEC echo
canceller into my current kernel.
After compiling dahdi
:
FW_FORWARD_MASQ=0/0,192.168.10.1,udp,5060,80,192.168.2.3
lets you able to forward the udp 5060 from the IP 192.168.10.1 to
192.168.2.3
You need to add all the other RTP relevant rules.
Best regards.
Marco Signorini
--
=
EtherMania di Signorini Marco
For network enthusiast
and is well known due to the fact u don't have a
precise clock source for meetme..
You need to have chan_dahdi dummie...
Hope it helps.
Marco Mouta
Enviada do dispositivo sem fios BlackBerry®
-Original Message-
From: Jeff Brower jbro...@signalogic.com
Date: Wed, 24 Feb 2010 18:25:07
of its span in /proc/dahdi file for a source: in the
description. Or even run:
strings dahdi.ko | grep source:
--
Marco Mouta
On Thu, Feb 25, 2010 at 8:15 AM, marco.mo...@gmail.com wrote:
It looks to me that u are having clock synchronism problems due to the fact
you are using Virtual Machine so
configuration was found.
Regards,
Marco.
--
http://www.ingegnitech.com
http://www.ethermania.com
Greg Woods wrote:
On Sat, 2010-01-02 at 20:25 +0100, F6HQZ wrote:
cat /proc/interrupts
Search the Digium cards drivers and look if several interfaces are using the
same IRQ number.
If yes
Hello All,
Do you guys suggest any 1800 DID Provider in the US ?
I'm having a hard time to find one.
Thanks,
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
help me
Thanks to all for your help
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE
, but I believe this is only the small beginning….
Looking forward to hearing from you guys ;)
Cheers,
--
Marco Mouta
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15
, but I believe this is only the small beginning….
Looking forward to hearing from you guys ;)
Cheers,
--
Marco Mouta
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15
... the call to NUMBERTOCALL on
acc1 continue to ring until the called answer, but the call is out.
Someone can help me ?!?!?
Thanks to all
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15
Just done it ... and all works fine.
Thanks all.
Marco
2009/7/24 Administrator TOOTAI ad...@tootai.net
Marco Sambo a écrit :
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
[...]
Marco,
attented transfer are broken
to A. CORRECT
if B hangup, .. also the call hangup
Someone can help me??? Please!
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit
and to
http://lists.digium.com/pipermail/asterisk-dev/2009-March/037003.html
Best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Chris YM wrote:
hello:
I wan to use the test tools-patgen and pattest for pri cards.
according to Tzafrir Cohen
Hi all,
I want to try to use a USB phone with Ekiga under Linux (Debian Lenny). It
works: I can receive and make calls. But some buttons of USB phone don't
work properly. In particular, button *, #, and hangup have wrong key
mapping.
Someone have tried a USB phone
Thamks all
Marco
,
Marco Signorini.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
be forcing interrupts from some cards
to use a single line/IRQ.
Thank you for your complete description on how PCI IRQ subsystem works.
It's probably the best explanation I've found since years.
My warm compliments, you've my best appreciation.
Regards,
Marco Signorini
Hi,
I try Noojee Click and Outcall, and for my context they work fine. Some
times ago I tried SanpANumber, but it was bought by Digium and substitute
with ADA.
Bye
Marco
2009/6/15 Stefanov, Milen milen.stefa...@compuware.com
Hello guys,
Is there a decent click-to-dial CTI which works
a suggestion, or real scenario similar to this that
could help me??
Thanks again,
Marco
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada em: terça-feira, 2 de junho de 2009 14:22
Para
with AstDB.
My doubt is if, only a SIP Proxy would be able to trigger SIP Response 181,
or if it would be possible with an Asterisk Server.
Thanks,
Marco Cordeiro
mhcorde...@gmail.com
___
-- Bandwidth and Colocation Provided by http
Thanks Philipp,
Sorry about my ignorance, but what would be IIRC Asterisk Trunk? Where could
I find info about it?
Thanks again,
Marco
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada
Hi Philipp,
So, what you are saying is that SIP trunks between 2 Asteriks might be able
to handle SIP Response 181 ?
Marco
-Mensagem original-
De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Em nome de Philipp Kempgen
Enviada em: terça-feira
Hi,
I do this by creating a directory waitingtransfer with only 1 file (the
audio message, the name isn't important, so you can change it everytime you
want) and then add new musiconhold class with specific waitingtransfer
directory. In your extensions.conf you change the musiconhold class to
Hi,
in Asterisk 1.4 to limit the simoultaneous calls I use the following
parameters:
[general]
...
limitonpeers=yes
notifyringing=yes
[phone]
...
host=dynamic
username=phone
call-limit=2
So I can receive and make max 2 calls simoultaneous.
Fo me that's work fine.
2009/5/29 Yuri
I set a variable CalledID to ${EXTEN} before dial it. So in h extension I
can use ${CalledID}.
2009/5/26 Thomas Kenyon dig...@sanguinarius.co.uk
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my
notebook using with asterisk. But I have some problem with firewall and
port.
Someone knows which ports I should open on my firewall??? I can't connect
...
Thanks all.
Marco
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
*Sent:* Tuesday, May 26, 2009 11:21 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] SIP over VPN
Hi all,
I have a question. I have a VPN and I want to use a SIP softphone on my
notebook using
,
Marco Wind
dipfees.com
Ph: 646-736-7816
Tf: 888-780-0253
F : (347) 626-2242
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http
FXO channels shuld have FXS signalling, and FXS channels shuld have FXO
signalling, so:
# FXO channels are 1,2,3
fxsks=1,2,3
# FXS channel is 4
fxoks=4
sorry why do you think that, I have 3 fxs + 1 fxo (my understanding is
that a attached fxs presents internally as a fxo
I have a
I connect to FOP panel, but I don't see any popup.
Someone can help me to configure FOP popups and in the use of UserEvent()
application?
Thanks all
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Hi,
someone has installed on an Asterisk box (not Trixbox) with Debian Linux,
the HUDlite Server?
Can someone help me in retrieve and install packages???
Thanks all
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Well, I see the rpm but my Asterisk box has Debian Linux, and I'm a little
afraid to use alien package to transform rpm to deb. Has HUDlite Server
source?? Like in tar.gz??
2009/4/23 David Klaverstyn d...@klaverstyn.com.au
Hi Marco,
Try this:
http://yum.trixbox.org/centos/4/RPMS/hudlite
Lee Howard wrote:
Marco wrote:
I've IAXModem and asterisk Asterisk 1.4.24 running on the same machine.
They are linked together through localhost. I've turned qualify on for the
iax peer. Periodically I've this message:
[Apr 20 23:47:46] NOTICE[4641]: chan_iax2.c:9049
Hi,
I have the same problem: sometimes my Asterisk box crash (or similar) and in
asterisk log doesn't appear nothing. Also into syslog.
I don't understand what is it
Marco
2009/4/21 Adrien Lemoine alemo...@legos.fr
Hi all,
I experienced for a second time the crash of asterisk
of this?
Thank you and best regards,
Marco Signorini.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi all,
I have a question: how can I see hints of a remote Asterisk in IAX2 trunk??
I want to set BLF on my phones to look state of other phones also in other
Asterisk server.
Someone have any idea or solution?
I use Asterisk 1.4.24.
Thanks all
Marco
Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use
SIP trunks.
Can you help me?
2009/4/16 Philipp Kempgen philipp.kemp...@amooma.de
Marco Sambo schrieb:
I have a question: how can I see hints of a remote Asterisk in IAX2
trunk??
I want to set BLF on my phones
So thanks, but in Asterisk 1.4.24 is not present in any way??
Any mystique solution??
Marco
2009/4/16 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Thursday 16 April 2009 07:08:49 Marco Sambo wrote:
Well, I use SIP phones and IAX2 trunk. For the moment I don't want to use
SIP
Hi, excuse me, but I see in your code that you configure DAHDI with OSLEC.
How do you do? Which version you have installed?
Thank you.
Marco
2009/4/16 Giovanni Magallanes gmagalla...@gmail.com
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a
call but only
Hi all,
someone has used the voice recognition software named Sphinx??? Can he tell
me how to use and its performance???
Thanks
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
/asterisk/agi-bin
astspooldir = /var/spool/asterisk
astrundir = /var/run/asterisk
astlogdir = /var/log/asterisk
[options]
verbose = 3
and so I find into /var/log/asterisk the logpro file with the output of CLI
(verbose) and notice, warning, error, debug message of Asterisk.
Ciao
Marco
2009/4/7
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton dfullertaster...@shorelinecontainer.com
Marco Sambo wrote:
Mhmm. Thaht's strange!
modinfo oslec
--
modinfo: could not find module oslec
and
modinfo dahdi_echocan_oslec
One thing!
I saw that I use kernel 2.6.26 in my asterisk machine. I should use kernel
2.6.28 or newer to use oslec with DAHDI???
2009/4/1 Marco Sambo derwid...@gmail.com
But I don't have also echo
modinfo echo
modinfo: could not find module echo
2009/4/1 Dave Fullerton
?
Thanks
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
wrapper
depends:dahdi
vermagic: 2.6.26-1-486 mod_unload modversions 486
2009/3/31 Tzafrir Cohen tzafrir.co...@xorcom.com
On Tue, Mar 31, 2009 at 05:02:36PM +0200, Marco Sambo wrote:
Hi,
I've a problem: I can't configure DAHDI with ech canceller OSLEC.
I have Asterisk
I have to try Skip2PBX, integrated into my Asterisk machine, but it seem
more invasive than Gizmo5 opensky. Doesn't it?
Marco
2009/3/26 Grygoriy Dobrovolskyy megaho...@gmail.com
skip2pbx is the best i tryed, but nasty price ;)
___
-- Bandwidth
Well,
anyone knows a good Skype vs SIP channel or program or something else to
integrate it into an Asterisk machine, to call normal skype users and not
and receive normal skype calls?
I red that Digium and Skype are working to integrate a chan_skype. Anyone
can tell me about?
Bye
Marco
2009/3
=documentation
accountcode=sip10
callerid=sip10 10
call-limit=2
dial=SIP/10
canreinvite=no
And this resolve for me problems for busy and for xfer Aastra button.
Marco
2009/3/17 Ira i...@extrasensory.com
At 01:29 AM 3/17/2009, you wrote:
But there is another little problem. On Aastra phone
It's so uncommon for me fxs and fxo cards and based on the reference
of sip.conf files and accounts i totally missed last paragraph where
it was mentioned only analogue lines and fxs (phone).
my appologies.
E1 and BRIs and sip trunks have been overloading my last month of work.
cheers,
--
Marco
?
Thanks all
Marco
2009/3/16 Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
On Mon, 16 Mar 2009, Olivier wrote:
2009/3/16 Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net
gordon%2baster...@drogon.net gordon%252baster...@drogon.net
On Mon
Ok, I read it.
Thank u. For busy on SIP I use also the Asterisk peer function SIPPEER with
field CURCALLS.
2009/3/17 Philipp Kempgen philipp.kemp...@amooma.de
Marco Sambo schrieb:
Anyone know how to use busy-level parameter or some other useful
parameters?
call-limit=2
busy-level=1
understand why I find it avaible when it makes
an outgoing call.
Thanks all
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
,Dial(SIP/phone,10)
exten = s,2,Voicemail(line)
exten = s,3,Hangup
hope it helps.
don't forget to asterisk reload on cli.
Looking forward to hearing from you.
cheers
--
Marco Mouta
On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal vad...@gmail.com wrote:
Hi I looked at few emails related
Thank you, Doug, for precious information.
Best regards,
Marco Signorini.
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Doug Lytle wrote:
Main fax server:
Mandriva 2008.1
Kernel 2.6.24.5 (Compiled for source)
(1) Intel(R) Xeon(TM) CPU 2.80GHz
Digium TE110P (23
you to All People answered me on this subject.
Analyzing your answers, seems that fax handling is still today
problematic with IAXModem and Hylafax... or I'm wrong?
What about other solutions?
Thank you and best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http
Thanks Doug and Lee,
your testimonials are changing my opinion :-)
Can you provide some details about your setup? What PRI solution are you
using? And what version of Asterisk, IAXModem, SpanDSP?
Thank you and best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http
.
Thank you for writing SpanDSP and best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Steve Underwood wrote:
Marco Signorini wrote:
Thank you to All People answered me on this subject.
Analyzing your answers, seems that fax handling
, but I would
like to know if someone has experience on this and could share their
opinion, tricks and/or statistical results on failure/success rate when
faxing. I think that this could be useful to other people have to realize
a system like that one depicted.
Thank you in advance.
Marco Signorini
file with a wave editor (Audacity).
I had better results if the maximum level is near half to the full
dynamic. Then switch to T38, if you need it.
Hope this helps you.
Best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Joseph wrote:
On 03/04
Joseph wrote:
On 03/04/09 15:44, Marco Signorini wrote:
Hi Joseph.
I've spent some time tuning the SPA3102 FXS line input and output gain
and I think that this is an important variable.
Let's try to record incoming and outgoing fax tones with asterisk on SIP
channel (disabling the fax
/entry/138/ for
the E1.
Thank you and best regards,
Marco Signorini
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE
,
Marco Signorini
===
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Tiago Durante wrote:
On Thu, Feb 26, 2009 at 12:52 PM, Paulo Santos paulo.r.san...@sapo.pt wrote:
Marco Signorini wrote:
Hi Tiago.
I've it working on PHP 5.2.6 but only after having modified
streamer, or SIP compliant
actuators.
I have a (very) little experience on electronic projects. Is there
something I can do to help starting a similar project?
Thank you and best regards.
Marco Signorini
Tzafrir Cohen wrote:
Hi folks
A common wisdom here is that one should use a proper
Jon Pounder wrote:
Marco Signorini wrote:
It's a dream!
It's since years that I'm thinking to have an open hardware project
targeted to a SIP application.
there is already a project called openmoko - join it and buy some hardware.
The phone is large and clunky - the idea
Hi Tiago.
I've it working on PHP 5.2.6 but only after having modified the php.ini
default configuration keys:
zend.ze1_compatibility_mode = Off
short_open_tag = Off
setting together to On and restarting apache forces PHP5 to behave like
PHP 4.x version.
regards,
Marco Signorini
try to set in your zapata.conf
overlapdial=yes
then in your asterisk cli
reload chan_zap.so
--
Marco Mouta
On Fri, Feb 13, 2009 at 9:21 AM, joek...@gmail.com wrote:
Default FreePBX context,
[from-pstn]
include = from-pstn-custom ; create this context
in Italy.
Any suggestion is really welcomed.
Thank you very much.
Best regards,
Marco Signorini
http://www.ingegnitech.com
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
, except, if I
remember well, some little problem on reload (but stopping and starting
again works fine).
Best regards,
Marco Signorini.
==
INGEGNI Tech S.r.l.
http://www.ingegnitech.com
Anyone by chance got an Init script for /etc/init.d/dahdi on a SLES 10 box
that'll work
Yes.
That's the correct way to do it. Placing # as a rule in callnum forces
the Portech to use the number defined in the SIP INVITE packet.
Bye.
Marco.
Marco Signorini
INGEGNI Tech S.r.l.
http://www.ingegnitech.com http://www.ingegnitechcom/
Pascal Bruno wrote:
Sorry
to specify the asterisk raw IP address in the Portech.
Best regards,
Marco Signorini.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
Pascal Bruno wrote:
Thanks for your reply!
Can you tell me what you have in your Portech configuration settings
(Mobile to Lan Settings; Sip Proxy settings etc...) My sip.conf file
is pretty similar to yours but still cant register.
On Fri, Jan 16, 2009 at 3:47 AM, Marco Signorini
one zap channel was present and I was
not able to test it on these type of situations.
Thank you and bye
Marco Signorini
Joseph L. Casale wrote:
I spent some time to understand what's missing in the OSLEC patch for
dahdi... I can confirm the same problem you reported some days ago and I
need
on how to do it.
Thank you and best regards.
Marco Signorini.
Joseph L. Casale wrote:
Yesterday I pulled in the latest svn of Dahdi and added the files
from a recent kernel in the drivers/staging/echo structure and modified
the Kbuild file so it would compile without error. I insmod'ed the module
,
Marco
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
the svn revision 5366 into my temporary folder
/home/marco/Install/dahdi-linux
2. Taken the linux-2.6.27 kernel sources baseline and placed in my
temporary folder /home/marco/install/linux-2.6.27
3. Taken the Linux kernel patch-2.6.28-rc6.gz, unzipped and applied to
the baseline kernel 2.6.27
1 - 100 of 523 matches
Mail list logo
