,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
where ${MAX_CALLS} is a variable defined by you that is the limit of
calls to be accepted
On Thu, Jun 12, 2008 at 12:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk
.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
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Is there a way to limit or set the calls per second on SIP?
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== Spawn extension (avaya, *, 3) exited non-zero on
'SIP/63.251.216.50-b7902770'
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Hi,
im trying to config ooh323 in asterisk. i compiled the one inside the
asterisk-addons im trying to connect my softphone(Xmeeting).
here's my ooh323.conf
[marq]
type=friend
context=avaya
ip=dynamic
port=1720
e164=888
username=marq
secret=marq
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Mark Quitoriano
http
Hi,
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
thanks!
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On Sat, Mar 29, 2008 at 7:26 AM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
You could save it to your asterisk voicemail directory, which is often
something like:
/var/spool/asterisk/voicemail/your_context/your_voicemailbox_number
The files used are unavail.*, busy.*, and
1.4.6 i set callerid=asreceived in my
zapata.conf
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to the carrier
is valid
exten = _9.,n(die),Hangup()
btw the number being dialled is not standard. Sometimes its 10 digits
sometimes it 7 digits and most of the time it's 6 digits.
Thanks!
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)
exten = 100,2,SetVar(foo=AEIOU:${whichVowel}:1) - sets ${foo} to
the single letter 'U'
On Wed, Mar 12, 2008 at 2:22 PM, Mark Quitoriano [EMAIL PROTECTED]
wrote:
Hi,
I need to create a simple number checking for authorizing the calls. if a
person dial 91800555121212345 where 12345
h... seems like this command doesn't do what i want.
what i wanted to do is remove the 12345 number at 1800555121212345 and send
the rest to the next cmd or operation.
On Wed, Mar 12, 2008 at 2:29 PM, Mark Quitoriano [EMAIL PROTECTED]
wrote:
ok i got the answer
Substrings${foo
for this sort of thing...
PaulH
On Wed, 2008-03-12 at 14:22 +0800, Mark Quitoriano wrote:
Hi,
I need to create a simple number checking for authorizing the calls.
if a person dial 91800555121212345 where 12345 is the authorization
code. If the authorization code is correct the call
Is it possible to include other context under globals? what i wanted to do
is create a new file which have other variables on it because i don't want
to edit every now and then the extensions.conf just for the global variable.
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http://asterisk.org.ph
Hi Olle,
On Thu, Feb 14, 2008 at 5:35 PM, Johansson Olle E [EMAIL PROTECTED] wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe a segment
On Sat, Feb 16, 2008 at 12:31 AM, Faruk Kasumovic [EMAIL PROTECTED]
wrote:
Johansson Olle E wrote:
Hi Mark!
13 feb 2008 kl. 23.42 skrev Mark Quitoriano:
Is it possilble for a single context to have multiple host=
something like this
First context is something we use to describe
Is it possilble for a single context to have multiple host= something like
this
[carrier]
host=ip address1
host=ip address2
host=ip address3
type=peer
disallow=all
allow=g729
allow=ulaw
canreinvite=no
insecure=yes
qualify=yes
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http://asterisk.org.ph
Fan the flame
Problem solved. Disabled usb serial ports and other unnecessary
hardwares built-in.
On Nov 23, 2007 2:38 PM, Mark Quitoriano [EMAIL PROTECTED] wrote:
On Nov 19, 2007 2:31 PM, Mark Quitoriano [EMAIL PROTECTED] wrote:
On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote
On Nov 19, 2007 2:31 PM, Mark Quitoriano [EMAIL PROTECTED] wrote:
On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
that's the same question i got(regarding question 1). Is it possible
for PCI compatibility issue? i need to check
On Nov 18, 2007 12:46 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message when in running asterisk with the tdm card
in.
here's
On Nov 18, 2007 10:11 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote:
On Sun, Nov 18, 2007 at 10:46:57AM +0800, Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message
hmmm... everythings is in there too...
On 11/19/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Nov 19, 2007 at 12:30:01AM +0800, Mark Quitoriano wrote:
On Nov 18, 2007 10:11 PM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Sun, Nov 18, 2007 at 06:46:18AM +0200, Tzafrir Cohen wrote
On Nov 19, 2007 12:10 PM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
that's the same question i got(regarding question 1). Is it possible
for PCI compatibility issue? i need to check for the motherboard specs
to post later :)
Hopefully someone will have someone
Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5
im having an error message when in running asterisk with the tdm card
in.
here's the error from the console of asterisk:
[Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get
span status: Inappropriate ioctl for
=24
# Global data
loadzone= us
defaultzone = us
On Nov 18, 2007 11:36 AM, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
You have 18 channels defined in zaptel.conf, but 24 channels configured
in zapata.conf
Mark Quitoriano wrote:
Hi i have a tdm2400p and installed asterisk
Hi im using centos 5 what is the prerequisite to be installed before
compilling asterisk 1.4?
Thanks!
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On 10/3/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Tuesday 02 October 2007 16:55:52 Brian West wrote:
On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote:
anyway still if there's a hack for meetme to work with g729 codec
this won't be an issue. So is there a hack or patch that i can
On 10/2/07, Brian West [EMAIL PROTECTED] wrote:
Ok Let me chime in on this one.
If you can use ulaw/alaw because you'll end up with tandem encoding which
will make the conference sound worse to some people.
All audio coming in will get transcoded to signed linear and pushed down
into zaptel
but is there a way to use g729 codec in meetme?
On 10/2/07, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote:
In my experience, and theoretically by design, it doesn't matter what
codec you are using when you call a meetme conference.
Moj
On 9/27/07, Tilghman Lesher [EMAIL PROTECTED] wrote:
On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote:
Some company asked me to do audits with there asterisk boxes. Is there a
standard that i should be following in auditing? anyway can give me a
start
what to do with asterisk
Hi,
is there a way to use g729 in meetme?
Thanks!
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Hi,
Some company asked me to do audits with there asterisk boxes. Is there a
standard that i should be following in auditing? anyway can give me a start
what to do with asterisk audits?
thanks!
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Fan the flame...
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with
Telia, I'm not sure if I want/need a Digital or an
Analog card... And 'how big'...
if they said they can do PRI you'll be needing a Digium card. but if they
can use VOIP go with VOIP to save up from buying the hardware.
HTH
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Fan the flame
ok tnx guys
On 8/25/07, Jean-Michel Hiver [EMAIL PROTECTED] wrote:
Le Fri, 24 Aug 2007 20:50:05 +0400, Mark Quitoriano
[EMAIL PROTECTED] a écrit:
What is a good softswitch that is also open source rather than asterisk?
You may want to check out freeswitch
to be passing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Friday, 24 August 2007 1:11 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk as a softswitch
Mark Quitoriano wrote
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12 WARNING[15466]: pbx.c:1720 pbx_extension_helper: No
application 'MeetMe' for extension (sample, 65000, 1)
i recompiled
On 8/24/07, ram [EMAIL PROTECTED] wrote:
On 8/23/07, Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi,
im using asterisk-1.2.24 and zaptel-1.2.20, im having a problem running
meetme conference,
when i try to call meetme i get this from the asterisk console
Aug 24 00:14:12 WARNING
Can i use asterisk as a softswitch?
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got it working... looks like the tar file is corrupted or something
redownload it again and installed it.
Thanks!
On 8/15/07, Anthony Francis [EMAIL PROTECTED] wrote:
looks broken, is there an apps dir in the source directory?
Mark Quitoriano wrote:
is there a new way to install asterisk
is there a new way to install asterisk? im using centos 4.5 and trying to
install asterisk. when i do make clean and make install i get this error.
# make clean
--snip--
make[1]: Leaving directory `/usr/src/asterisk-1.2.24/apps'
make: *** codecs: No such file or directory. Stop.
make: ***
Hi,
is there a way to limit an account to do simultaneous calls in sip and iax?
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On 3/30/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Mark Quitoriano [EMAIL PROTECTED]
Date: Thu, 29 Mar 2007 23:05:57 +0800
Hi,
what could be the maximum simultaneous calls can asterisk do? i read
about
the asterisk business edition review[1] and it can only handle 120
simultaneous calls? i'm
Hi,
what could be the maximum simultaneous calls can asterisk do? i read about
the asterisk business edition review[1] and it can only handle 120
simultaneous calls? i'm using 1.2.x branch of asterisk and i use more or
less 90 simultaneous calls.
[1]
On 3/16/07, Trevor Peirce [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote:
Hi i have an asterisk pbx with E1 port connected to another PBX. Im
trying to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get
Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying
to send the DNID/DNIS to the PBX here's my dialplan
exten = 888111,1,Dial(ZAP/g2)
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or
ZAP/g1 the PBX get the number 1.
Hi im having this message in my console and dmesg.
rtc: lost some interrupts at 1024Hz
im not sure what this is.
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positionOct 24 16:39:19 WARNING[3684]: format_wav.c:247 update_header: Unable to find our position
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, something like this:exten = s,1,Answerexten = s,n,Wait(2)
exten = s,n,Congestionexten = s,n,HangupI suppose you could really just use Hangup instead, but this seems to work for me.Alex
On 10/12/06,
Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi list,i noticed from the cli my asterisk box is accepting
Hi Sylvain,How about in iax.conf? i tried allowguest=no in iax.conf but it didn't work i haven't tried it in sip we'll try that.On 10/13/06, Sylvain ZUCCA
[EMAIL PROTECTED] wrote:Try allowguest=no in your
sip.conf
2006/10/13, Mark Quitoriano [EMAIL PROTECTED]:
well what i wanted to do is let
Hi list,i noticed from the cli my asterisk box is accepting unauthenticated calls how can i prevent this?CLI:-- Accepting UNAUTHENTICATED call from 192.168.0.2:
requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (g729|ulaw|alaw), priority = mine
by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
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Hi i tried this setup what should my zapata.conf setup be?
On 4/8/06, Infobox Peru
[EMAIL PROTECTED] wrote:
Your zaptel is wrong...
it must be:
zaptel.conf:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31On 4/7/06, JP Carballo
[EMAIL PROTECTED] wrote:
Mark Quitoriano wrote: Hi Guys, Im
ok i'll try that. tnx!On 4/8/06, Infobox Peru [EMAIL PROTECTED] wrote:
Your zaptel is wrong...
it must be:
zaptel.conf:
span=1,1,0,ccs,hdb3
dchan=16
bchan=1-15,17-31On 4/7/06, JP Carballo
[EMAIL PROTECTED] wrote:
Mark Quitoriano wrote: Hi Guys, Im configuring my asterisk box as a voip gateway
Hi Guys,What model can i use for an xseries 346 server, i think the pci slot is 64-bit? Im just going to use it for asterisk timing so the cheapest will be the best.
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Hi Guys,Im configuring my asterisk box as a voip gateway. I have TE110P which is connected on my PBX and i will be using voip for my outgoing.Here's my configzaptel.conf:span=1,1,0,ccs,hdb3
fxoks=1-32 zapata.conf:context=defaultsignalling=fxs_ksgroup=1channel =1-32-- Regards,Mark Quitoriano
ok got it tnx guys!On 3/26/06, Dovid Bender [EMAIL PROTECTED] wrote:
Yes,There are issues witht he latest kernal release. Search the list archives. Dovid
Mark Quitoriano [EMAIL PROTECTED] wrote:
Hi Guys,Im having a problem compiling zaptel
1.2.4 on CentOS 4.3, anyone encountered
: 'fcstab' defined but not usedmake[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2make[1]: Leaving directory `/usr/src/kernels/2.6.9-
34.EL-i686'make: *** [linux26] Error 2-- Regards,Mark Quitoriano, CCNAFan the flame...http://www.spreadfirefox.com/?q
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) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you
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) -abefhkmnptuvxBCHP or -o optionmake: *** [.depend] Error 2 Any ideas of what the problem might be. Thank you
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then 3rd and 4th. can i implement something that if a call
get it 4 phones will ring simultaneously until someone picked up?
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anyone tried using asterisk on AMD64? how's the performance is better than p4?-- Regards,Mark Quitoriano, CCNAFan the flame...
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Asterisk
Not Exist back from 147.135.20.128
On 11/21/05, Mark Quitoriano [EMAIL PROTECTED] wrote:
Im having a problem configuring my broadvoice account on my asterisk. im using asterisk from cvs v1-0 here's my dialplan.
exten = _1NXXNXX,1,dial(SIP/[EMAIL PROTECTED],30)
exten = _1NXXNXX,2
@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:
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tnx!On 11/16/05, Gerard Dupont III [EMAIL PROTECTED] wrote:
This has come up a bunch of times on this list..Take a look at http://www.voip-info.org/wiki-Asterisk+sound+filesHope that helps-Gerard
Mark Quitoriano wrote: how can you play .gsm files what program can you use both in windows and linux
On 11/15/05, Mark Quitoriano [EMAIL PROTECTED]
wrote:
hmmm... so there's no way i can prioritize g.729 before ulaw and alaw? i just want to do is use all
g.729 first if all
g.729 already used now thats the time ulaw will be used.
On 11/16/05, trixter aka Bret McDanel
[EMAIL PROTECTED] wrote
great! tnx matt.On 11/16/05, Matt Riddell [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote: you mean the way you setup asterisk 1.2 dialplan is different with 1.0.9?Yes, you can read the upgrade.txt file inside the RC2 distribution forinformation on the required changes.
--Cheers,Matt
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hmmm... so there's no way i can prioritize g.729 before ulaw and alaw?
i just want to do is use all g.729 first if all g.729 already used now
thats the time ulaw will be used.On 11/16/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote:
On Wed, 2005-11-16 at 01:20 +0800, Mark Quitoriano wrote
What do they mean by pass thru?
http://www.voip-info.org/wiki/view/Asterisk+G.729+pass-thru
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great! i'll try it later tnx!On 11/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Monday 14 November 2005 07:20, Mark Quitoriano wrote: Hi, is there a howto to install
g.729 codec on asterisk?http://www.digium.com/downloads/ftp/asterisk/g729
it to 1.2?If in three or four days you go to upgrade your version of 1.0.9, it will beupgraded to 1.2.So, why not do it now, that way you won't end up creating a dialplan only for
it to not work in a couple of days.
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hey guys,
if i get the asterisk from CVS like cvs checkout -r v1-0 zaptel libpri
asterisk asterisk-addons asterisk-sounds do i get a stable one?
On 11/11/05, Mark Quitoriano [EMAIL PROTECTED] wrote:
Great! tnx matt!On 11/11/05, Matt Florell
[EMAIL PROTECTED] wrote:
It's CVS v1-0. Digium has said
Hi,
is there a howto to install g.729 codec on asterisk?
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but that's already 1.2? is it advisable to upgrade my current version
1.0.9 to 1.2 already? any big changes to be done to my current setup to
upgrade it to 1.2?On 11/14/05, Matt Riddell [EMAIL PROTECTED] wrote:
Mark Quitoriano wrote: hey guys, if i get the asterisk from CVS like cvs checkout -r v1
cards from
Digium run much better on it than they do on 1.0.9.If you want it now, just checkout from CVS like this:cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-soundsMATT---
On 11/10/05, Mark Quitoriano [EMAIL PROTECTED] wrote: in the Changelog on http://ftp.digium.com/pub
result in a crash. -- general-- Added man pages for astgenkey, autosupport, and safe_asterisk
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do i need to install libpri? my only setup is Digium TDM400P with 2 fxo port.
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards,Mark
Hi list, i'm having a problem with asterisk+pstn termination, i just
bought a TDM400p and connect my phone line(bellsouth) now when im using
the pstn through asterisk there's an echo, i don't know if this is
already have been resolved. If it does please point me to the
instruction how to resolve
they're using versions 1.0.9 for all(e.g. asterisk, zaptel).
[1]http://www.voip-info.org/wiki/index.php?page_id=2403
On 10/27/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
[EMAIL PROTECTED] wrote on 10/26/2005
05:09:30 PM:
On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote:
Hi list, i'm
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