Hi all,
I have 2 question.
I have a call center queue with 5 agent; the following are the configuration
files:
*queue.conf*
[name_of_queue]
musicclass = default
announce = queue-name_of_queue
strategy = ringall
servicelevel = 60
context = callcenter
timeout = 60
retry = 5
wrapuptime=15
Hi at all !
I've a problem and I don't know how to solve it.
My configuration is the following:
ISDN LINE --- PATTON (SIP) --- ASTERISK
in asterisk my sip.conf for sip patton account is the following:
[general]
port=5060
bindaddr=0.0.0.0
context=default
language=it
limitonpeers=yes
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
Hi all,
I've a problem: I update my asterisk to version 1.4.25, and the attended
transfer doesn't work.
A call B, B press *2 and voice announce to digit internal and select
internal of C. CORRECT
A hear music on hold and B talks with C. CORRECT
If B press *0, the call return
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
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
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
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
Hi all,
I try to install FOP. It's very nice.
In documentation I red that from my dial plan I can launch a popup window
with UserEvent() application.
I try to follow FOP documentation but I can't popup anything. My structure
is:
- server 1: Asterisk system
- server 2: FOP system
- client
On client
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
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-server-1.4.32-1.i386.rpm
Regards
David.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Marco Sambo
*Sent:* Thursday, 23 April 2009 7:29 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Asterisk
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
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
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Hi Enrico,
I do that by modifying logger.conf
[logfiles]
logpro = notice,warning,error,debug,verbose
and modifying asterisk.conf
[directories]
astetcdir = /etc/asterisk
astmoddir = /usr/lib/asterisk/modules
astvarlibdir = /var/lib/asterisk
astdatadir = /var/lib/asterisk
astagidir =
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
Hi,
I've a problem: I can't configure DAHDI with ech canceller OSLEC.
I have Asterisk 1.4.24 and DAHDI 2.1.0.2. I compiled also OSLEC.
But when in /etc/dahdi/systems.conf I insert value echocanceller=oslec,1-4,
command dahdi_cfg - give me an error about oslec.
Someone can help me?
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 ;)
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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
Hi Ira,
for Aastra phones I have done this application to resolve busy/xfer
transfer:
extensions.conf
===
exten = _1X,1,GotoIf($[${SIPPEER(${EXTEN}|curcalls)}1]?free:busy)
exten = _1X,n(free),Dial(SIP/${EXTEN},,tTr)
exten = _1X,n,Hangup()
exten
, 16 Mar 2009, Marco Sambo wrote:
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone
busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find
it
busy.
When SIP phone make a call and I
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
Hi,
I have a question. How can I configure my sip.conf to make a SIP phone busy
on incoming and outcoming calls? I explain my problem.
When SIP phone receive a call and then I try to call that phone, I find it
busy.
When SIP phone make a call and I try to call that phone, I find it avaible
and it
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