Hi,
I had the same problem, it is due to the fact that your server is
behind a nat, I solved the problem adding in my sip.conf
[general]
port = 5060
bindaddr=0.0.0.0
externip=200.121.56.70
localnet=192.168.1.0/24
context=llamadas
srvlookup=yes
Greetings
Carlos Rojas
Lima - Peru
On Mon, 21 Mar
Hi
here there are some of them
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sound%20files%20international
Carlos Rojas
Lima - Peru
On Apr 5, 2005 12:24 PM, Dov Bigio [EMAIL PROTECTED] wrote:
Hello all,
I am looking for a list of all available sound files for asterisk
Hi,
You have well formed your file zapata.conf?
Carlos Rojas
On Apr 8, 2005 9:40 PM, Drew Einhorn [EMAIL PROTECTED] wrote:
The ATA generates it's own dialtone, and waits for
the user to dial a number, before sending anything
to the * box. So one of the first examples in the
in the Brief
Hi,
My name is Carlos,
do you know that 1.0 exist rpm of asterisk for suse 9.2?
in ftp.suse.com
greetings
Carlos Rojas
Lima - Peru
www.isoimsa.com
On Fri, 11 Mar 2005 00:09:44 +0100, Aldo Bergamini [EMAIL PROTECTED] wrote:
Hi all,
sorry bothering again.
I am still stuck in compiling
Hello,
I
Check this page:
http://www.asterisk.net.au/general/1/
It's very interesting
Best Regards
Carlos Rojas
On 7/18/07, Dmytro Mishchenko [EMAIL PROTECTED] wrote:
Tim Reimers wrote:
Hi -
I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both
ports.
I need to be able
Hello,
Do you have porf forwardin for SIP protocol in your firewall?
SIP: 5060 udp
rtp 1 - 2 udp (default)
and IAX2 4569 udp
Best Regards
Carlos Rojas
On 7/28/07, Ary Junior [EMAIL PROTECTED] wrote:
Hi, Im a asterisk newbie and I've configured an asterisk server here in my
?
Thanks very much!!!
On 7/28/07, Carlos Rojas [EMAIL PROTECTED] wrote:
Hello,
Do you have porf forwardin for SIP protocol in your firewall?
SIP: 5060 udp
rtp 1 - 2 udp (default)
and IAX2 4569 udp
Best Regards
Carlos Rojas
On 7/28/07, Ary Junior [EMAIL
Hello,
In Asterisk 1.4 and zaptel 1.4,
don't work make linux26,
zaptel and asterisk works with kernel 26, and only work with
./configure
make menuselect
make
make install
Best Regards
Carlos Rojas
Lima - Peru
On 7/31/07, hugolivude [EMAIL PROTECTED] wrote:
Hi,
I'm having trouble
Hello,
I prefere, asterisk
Best Regards
On 7/31/07, Al lists [EMAIL PROTECTED] wrote:
You can use both Asterisk or AsteriskNow to have meetme (conference room)
On 7/30/07, fateme fatah [EMAIL PROTECTED] wrote:
Hi:
I want to have conference call service.You offer me use asterisk or
Hello,
Do you have install doxygen?
Best regards
On 8/13/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
MOSBAH ABDELKADER wrote:
After installing Asterisk, i have installed the docs by make progdocs.
But i don't know where to locate this documentation.
Maybe
Heloo,
I think that your error is:
zaptel.conf:
---
fxsks=1
loadzone= uk
defaultzone = uk
zapata.conf:
[channels]
language=en
context=incoming
signalling=fxs_ks
busydetect=yes
busycount=4
callprogress=no
relaxdtmf=yes
callwaiting=no
Hi,
I am trying to connect the one cellular whit my asterisk box, with a
cable usb of cellular me, to be able to call from the asterisk, someone
has proved this?
Carlos Rojas
Lima - Peru
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is connected to x100p FXO port which received incoming
cell calls and routes them to asterisk and the same port routes
outbound calls from asterisk through the cell phone to the outside.
On 1/21/06, Carlos Rojas
[EMAIL PROTECTED] wrote:
Hi,I
am trying to connect the one cellular whit my asterisk box
problems, someone to dyeing
some similar problem?
Carlos Rojas
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Hello,
Do your verify, the codecs, of both clients, in your sip.conf?
What codec do you use?
Best Regards
On Thu, Mar 20, 2008 at 12:13 AM, Pete Kay [EMAIL PROTECTED] wrote:
Hi,
I am sorry my questinos are too fundamental. I am new to Asterisk, and
hope to catch up as fast as I can.
Hello,
Do you redirected the rtp ports to your phone?
usually 1 - 2 defautl rtp ports
Best Regards
Carlos Rojas
On Thu, May 22, 2008 at 8:48 AM, Phibee Network Operation Center
[EMAIL PROTECTED] wrote:
I have a problem connecting a Grandstream ipphone to an asterisk
Hello,
Do you download zaptel of Redfone website?
Best Regards
On Fri, Aug 22, 2008 at 6:28 PM, Bill Michaelson [EMAIL PROTECTED] wrote:
I expected to find th module ztd-ethmf[.c...] in support of the redfone
TDMoE product in my zaptel distro (I have 1.4.11). But it's not there. I
am
Hello
And lspci -vb ??
Regards
On 5/4/07, Daniel Pittman [EMAIL PROTECTED] wrote:
Steve Edwards [EMAIL PROTECTED] writes:
I see the following on one of my new servers:
-ts10::sedwards:~$ cat /proc/interrupts
CPU0 CPU1 CPU2 CPU3
0:2979045
Hey
Look
http://www.asterisk-es.org
Best Regards
On 5/5/07, Cesar Benjamin Garcia Martinez [EMAIL PROTECTED] wrote:
Bueno, no esta en chile, yo no estoy en chile, pero hablo español, soy de
mexico, asi que en parte tienes razón, pero tb creo que deberías haber
puesto de donde eres.
Hello,
I take the example:
exten = 300,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN},30)
Best Regards
On 5/26/07, Alex Balashov [EMAIL PROTECTED] wrote:
Matt,
On Sat, 26 May 2007, Matt Darnell wrote:
exten = _3xx,1,dial(IAX2/{$EXTEN})
exten = 300,1,dial(IAX2/301)
You do not appear
Hello,
In your sip.conf you don't have the user for you provider:
[yourprovider]
username=1234
secret=sdfdsf
host=sip.yourprovider.com
type=peer
...
In yor extensions.conf
[mycontext]
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,103,Hangup
exten = 2001,1,Dial(SIP/2001,20)
exten =
Hi,
I work with
gnudialer
vicidal
Best Regards
On 7/14/07, Todd H [EMAIL PROTECTED] wrote:
I like ADM as it has a URL popup feature (open a URL with a DID or
CallerID in URL). The problem is that for each call, I tend to get 4
or 5 popups... But as the other author said, there are
Hello,
What's your zapata.conf and zaptel.conf?
On 1/20/07, Matt Brown [EMAIL PROTECTED] wrote:
Well,
I have just phoned BT today who said they can increase the CPC value
on the line - however it needs to be done at the exchange - and has
been booked for Tues.
I suppose I will know
Hello everybody
Anyone, know TDM800 of yeastar?
Anyone to test him with asterisk?
Regards
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Hello,
Do you include in your zapata.conf
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
There are any problems with hang up
Regards
On 1/29/07, Lee Jenkins [EMAIL PROTECTED] wrote:
Hi everyone,
I just installed a TDM02B and surprisingly, I had really no problems
except one.
If I
Hi, Giorgio
I'm from lima Peru, I have the same problem, and the solution was the
signaling in zapata.conf
I use :
fxsls
for
fxsks,
Regards
On 1/30/07, Giorgio Incantalupo [EMAIL PROTECTED] wrote:
Hi,
the problem was the telco line was not a pure telephone line but a mixed
one (phone +
Hello
I use trafic shapper, is very good.
Regards
On 2/14/07, Angel Heart [EMAIL PROTECTED] wrote:
Hi,
What Network Switch you are using? I do traffic/bandwidth shapping on the
edge switch where the port the voice installed, you can configure each port
to 128Kbps or just plain Ethernet
Hello
I'd like to know too
On 4/2/07, Gustavo Felisberto [EMAIL PROTECTED] wrote:
I am in the process of buying a TDM800 card from Yeastar (
http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20CardcTypeName=1)
Any one has tested this cards? How reliable are them? I am specially
Hello,
canreinvite, don't work with all softphone or hardphone.
Regards
On Wed, Dec 3, 2008 at 12:38 PM, BERGANZ François
[EMAIL PROTECTED] wrote:
Someone have a solution for me ?
*De :* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *De la part de* BERGANZ François
*Envoyé :* mercredi
Hello
asterisk -vvvgc
Regards
On Wed, Dec 10, 2008 at 7:45 PM, Scott Berry n7...@northlc.com wrote:
Hello there,
I am reading Asterisk: The Future of Telephony Chapter four. I am using a
Ubuntu box with Asterisk precompiled at this time so I can learn. I am
finding that I am
Carlos Rojas
On Wed, Apr 1, 2009 at 10:45 AM, ludo perrot ludoper...@gmail.com wrote:
hello,
I am beginning to asterisk.
I have a sip trunk access to operator and VPN access with operator.
i booked 10 sda numbers.
IP adress asterisk : 192.168.600.1
IP adress operator : 192.168.700.50
i can
Hi,
In this link is the script Suse
http://www.leals.com/~mm/asterisk/asterisk_suse.sh
On 8/18/05, James Oakley [EMAIL PROTECTED] wrote:
On Wednesday 17 August 2005 3:04 pm, Tzafrir Cohen wrote:
On Wed, Aug 17, 2005 at 01:27:08PM +0300, [EMAIL PROTECTED] wrote:
Hi!
I'm trying to
Hi,
if only you see the page of the Apache
put the following thing
http://(The asterisk ip)/cgi-bin/vmail.cgi
On 6/20/05, Time Bandit [EMAIL PROTECTED] wrote:
On 6/20/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
snip
Is there anyway to shorten that or even give users the option to not
Hi
Someone to integrated Asterisk with pbx Panasonic KX-T336?
Regards
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greetings to all
Carlos Rojas
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Hi,The 4606 phone of Avaya it is possible that it works whith asterisk?On 5/23/06, Tom Lynn [EMAIL PROTECTED]
wrote:The 4606 is a h.323 based phone.There is no SIP image to use with this phone.
On Fri, 12 May 2006 11:11:48 -0500, you wrote:Hello all,I have asterisk working well with, Sipura, but
Hi, I have a avaya 4602, and have similar problem, in the website of avaya, in support, there are, sip firmware and h323 firmwareRegardsOn 6/6/06,
Gabriel Rosca [EMAIL PROTECTED] wrote:
Hi guys, I installed asterisk and it's working really well.
For now I`m using soft phones IAX and
does anybody have any file of configuration of phones avaya?in H323 protocol?RegardsOn 6/6/06, Mark Phillips
[EMAIL PROTECTED] wrote:Ig nore my last post. I had not seen this posting
On Tue, 2006-06-06 at 22:33 +0200, Henk wrote: Have a look at the attached link.
Hi, I solved this problem in zapata.conf file;rxgain=0.0 ;Volume RXrxgain=8.0 ;Volume RX;txgain=0.0 ;Volume TX
txgain=1.0 ;Volume TXRegardsOn 6/9/06, news.asterisk.users
[EMAIL PROTECTED] wrote:
I've been attempting to get this TDM400 card to work since March.
While it sort of works, I'm
Hi,Well, I'm working with a2billing http://www.asterisk2billing.org/, without problems.RegardsOn 6/13/06,
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,I look at voip-info for a simple billing application .I wish to calculate price to pay according to thedatas stored in cdr table
it louder when the two PSTN lines are bridged but also affects
the audio quality adversely. It's very touchy.. it also causes
horrible feedback (loud screeching when lines bridge).
I've upgraded zapata and libpri too.
JD
Carlos Rojas wrote:
Hi,
I solved this problem
Hi, Kris,
It is possible to add mysql serber to astlinux?
Carlos R.
On 12/22/05, Kristian Kielhofner [EMAIL PROTECTED] wrote:
John Reynolds wrote: Kris, has this module been added to Astlinux? will it be?
John R.John, It's in -testing.You should have gotten it when you pulled -testingdown for
Greetings from Lima Peru
Carlos Rojas
On 12/23/05, Guillermo Salas M [EMAIL PROTECTED] wrote:
Xmas is tomorrow at my country.. Merry Xmast to all :)Greetings from Ecuador - South America ;)
On Fri, 2005-12-23 at 19:20 -0500, tracinet wrote: Nothing wrong at all - this is the Merry Christmas
Instal subversion package, in your linux to be abale to use svn.
Regards
Carlos Rojas
On 12/20/05, Douglas Garstang [EMAIL PROTECTED] wrote:
No idea on this. Can't find it on digium's web site. How do we download the latest source for Asterisk? Looks like they switched to SVN from CVS? Never
Hello,
I'm working with supermicro servers, for the irq problems with Dell, any people have problems
Regards
On 10/30/06, Paul Hales [EMAIL PROTECTED] wrote:
How many analog lines are you looking at? Hundreds?PaulHOn Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote:
We have a number of clients
Hello,LookX-web litehttp://www.asterisk-es.org/modules/mydownloads/visit.php?cid=6lid=12Regards
On 11/1/06, Vladimir Montealegre Estailes [EMAIL PROTECTED] wrote:
Hello list partners
you know about a softphone made in java attachable
in a web page?
GNU!
Thaks in
advance!
Visita
Hello,
The X100P, don't support reverse polarity, I have same problem, then I
bougth a TDM.
Regards
On 11/25/06, txus [EMAIL PROTECTED] wrote:
Hi, I mean that the server finish the action
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I'm
Hello,
Anyone saw asterisknow, ?
Regards
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iftop
On 12/12/06, Mochamad Susantok [EMAIL PROTECTED] wrote:
Dear all,
Are there anyone have ben to use some tool or method to measure latency
and packet loss for VoIP packet ?
-
This email was sent using Student EEPIS-Webmail.
Hello everybody
Anyone know a good carrier of voip for international calls?
Regards
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a2billing
Is very good
On 12/19/06, Giedrius Augys [EMAIL PROTECTED] wrote:
2006/12/19, C F [EMAIL PROTECTED]:
Can anyone recommend a call accounting solution with rating for post
paid billing that works well with asterisk using the account code or
any other info from the CDR?
I don't
Hello everybody
HAPPY and Merry Christmas to all.
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Hello
Do no forget the rtp ports 1 to 2
Regards
On 1/4/07, Facundo Barrera - GMail [EMAIL PROTECTED] wrote:
007/1/4, Bob Chiodini [EMAIL PROTECTED]:
Facundo Barrera - GMail wrote:
Hi list:
This is my first post and first off all i want to wish a good
year for everone!
Hello,
What's your sip.conf and extensions.conf?
Regards
On 1/12/07, kevin bergner [EMAIL PROTECTED] wrote:
i am having a problem where the phones are registered and can make
outgoing calls but all incoming calls go directly to voicemail and do not
ring any of the phones
any ideas?
--
Hola
Que distribucion usas?
On 1/14/07, Jugleni Jr [EMAIL PROTECTED] wrote:
*Olá pessoal,
alguém pode me ajudar, como posso solucionar isto?
toda hora que preciso parar um iniciar um serviço me mostra isto, e tenho
alguns serviços que não esta rodando corretamente, pelo que eu estudei é
Hello everybody
Anyone know a software for callcenter, with statistics and reports and work
with asterisk?
Regards
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Hello,
Only copy the configuration files, extensions.conf, sip.conf, iax.conf
,
Best regards
On Nov 27, 2007 1:27 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
If I have a running Asterisk on one machine and I need
to have another Asterisk on another machine, can I
copy
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Best Regards
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Hello,
Remember, that linux has problems with irq and pci cards of digium, do you
have 3 digium card, and don't have any problems ?
Best Regards
On Jan 5, 2008 11:01 PM, Eric S López [EMAIL PROTECTED] wrote:
Gres,
Me, as an asterisk and linux newbie installed redhat 4 (without the gui)
Hello everybody
Anyone, to know a gateway that works with nextel simm cards?
I'm looking for them, in internet, but I did'n look.
Best regards
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To
Hello,
Your smtp server is on?
Best regards
Carlos Rojas
On Mon, Jul 6, 2009 at 7:30 PM, Steve Anness steve.ann...@gmail.com wrote:
Today I discovered that voicemail attachments are not working on our
latest asterisk server (version 1.4.24.1). I have two other asterisk
servers that I
Hello everybody
I have an asterisk with an integration of alcatel pbx, by sip trunk, all
calls are fine, but tha calls calls that originate from a analog line,
the recipient is not listening, and that if they hear the call originates,
the lines are E1 in alcatel pbx.
When a asteris user call to
Hello,
In your sip.conf
You need
host=sip.xxx.com
or IP
don't work with dynamic
Regards
On Wed, Aug 12, 2009 at 8:27 AM, harry R rhm.noa...@gmail.com wrote:
Dear all,
I want to setup the incoming calls, that don't use authentication in
sip.conf file.
My configurations as
Hello,
I never use externhost
y use \
externip=public ip
And work fine
Regards
On Fri, Aug 14, 2009 at 9:18 AM, Ott Rose sixfourimp...@hotmail.com wrote:
how do i troubleshoot no ring tone. It was working and all i added was the
lines below now it doesn't ring.
Edit sip_nat.conf for
Hello
One question
In sip.con or sip_additionals.conf, in freepbx, the context of your client
do you put
nat = yes
externip =
You put your public ip.
Are you sure that?
Regards
On Fri, Aug 14, 2009 at 10:43 AM, Ott Rose sixfourimp...@hotmail.comwrote:
i changed it and still didn't
Hello,
I use Authenticate command in dialplan.
Regards
Carlos Rojas
On Wed, Aug 19, 2009 at 6:33 AM, James Mutuku listmut...@gmail.com wrote:
Hellos,
I have astersist 1.2 working with freepbx. I want to tie pin codes to
extensions(users). How do I do this?
--
Best Regards,
James
Hello,
You need configure a queue, with agents for that.
Regards.
On Thu, Aug 20, 2009 at 11:22 AM, kaustuva...@bbsr.syscomes.com wrote:
I have tried a lot like as
exten = 123,1,Dial(SIP/114SIP/113SIP/115)
and all the channels are dialing and if i answered any 3 of one, all the
Hello,
I use cri
http://www.tikalnetworks.com/voip/index.php?cid=38
Best regards
On Thu, Jun 24, 2010 at 3:22 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Hello Bruce,
This module is not reliable on FreePBX?
You know if there is a open source web-voicemail for Asterisk?
Best
Hello
Do you set your callerid in the context outgoing?
[outgoing]
exten = _X.,1,Set(CALLERID(num)=4663000)
exten = _X.,n,Dial(..
On Mon, May 9, 2011 at 4:45 AM, mahesh katta maheshka...@flexydial.comwrote:
Sir ,
this is not working
On Mon, May 9, 2011 at 1:52 PM, A J Stiles
Can you send the logs in cli console for help you?
Regards
On Tue, May 17, 2011 at 9:16 AM, virendra ban hati virbh...@gmail.comwrote:
hi list,
please help me how to know how many calls are on hold.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
Hello,
I use no-ip service, is similar than dyndns.com
Best Regards
asterisk-l...@puzzled.xs4all.nl wrote:
On 09/07/2011 02:17 AM, A Dunor wrote:
Hello list, I am a beginner at asterisk. I want to access my asterisk
box from my laptop, on a different network (mobile hotspot). The
asterisk
Hello, every body
Anyone set up, the sla sharing line appearances, in asterisk, I'm setting,
tha but, don't, work, I change the sla.conf, extensions.conf, and sip.cfg,
but don't work fine.
Any one, could setup, tha?
Regards
Carlos Rojas
Hello
Did you use callerid(num) in your dial plan?
On Dec 16, 2011 7:38 PM, Kaushal Shriyan kaushalshri...@gmail.com wrote:
Hi,
I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel
with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI
Card on the
Hello everybody
I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine?
I've been find any information and saw heatbeat + cysnc2 and heartbeat +
rdbd, any one has worked any these aplications fine?
Best regards
--
Hello,
Do you saw this solution?
http://linuxnotes.us/
Regards
On Sun, Dec 18, 2011 at 12:26 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I have asterisk 1.6.2.20 installed at production server, I have 2 SIP voip
trunk for making outgoing and DID for incoming to server.
My
Hello
It is possible but how do you have the dialplan ?
In your dial plan you can do that
Regards
On Dec 20, 2011 2:40 PM, Matt mhop...@gmail.com wrote:
Hi,
Has anyone here any experiencing with linking an Asterisk PBX to a
GOIP GSM to SIP Gateway? We've got inbound calls from the GSM
Hello
I use fail2ban, and works fine,
Regards
On Tue, Dec 27, 2011 at 1:54 AM, virendra bhati virbh...@gmail.com wrote:
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account
Hello,
Do you use monitor?, because in asterisk 1.4 to new versions, It's use
mixmonitor, in asterisk 1.2 had this mistake.
Regards
On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards
asterisk@sedwards.comwrote:
Un-top-posting, snarky comments inline...
On Wed, 28 Dec 2011, Faraj Khasib
Hello,
Do you set up, your logrotate in /etc/asterisk ?
Do you test that your fail2ban work fine?
Regards
On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I happened to be in the cli tonight as some (208.122.57.58) initiated a
simple attack - just trying to make long
Hello,
Your blackberry sip client, works in your wifi network? or by blackberry
internet?
do you set nat=yes if your phone, register by internet?
What is your sip.conf?
Regards
On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I have a softphone I'm trying on a
Hello
Asterisk only says that the iax2 channel don't work maybe you look the
iax.conf. you trunk. Is iax I think
Regards
On Dec 29, 2011 6:49 AM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hello all,
I attempted to make a couple of outbound calls this morning and always got
the busy tone. I
Hello
Do you use hard phone or softphone?
In many ip phones you can change the ring tones or use w option in Dial
command
Regards
On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote:
i could add r option in dial command. this will generate a ringtone
during connection. could i
Hi everybody
I have been presenting a periodic problem, do not know if anyone listed has
happened something similar,I'm using the asterisk, asterisk-1.6.2.13, in
different locations works well, but every so often fails, hangs on Asterisk
server or simply asterisk, SIP requirements do not answer,
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Carlos Rojas
*Sent:* Saturday, 14 January 2012 3:37 p.m.
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] asterisk problem sip
Hi everybody
I have been presenting a periodic problem, do not know
Hello everybody
someone in this list, has installed asterisk, in a virtual server like
proxmox? I'm thinking install some asterisk servers in a machine dell
xeon 64 processor, but I'm not sure, about virtual Server software.
I heard, about proxmox, but I don't know if works fine.
Regards
Hello
Are you using a amd server?
Sometimes openvox doesn't work fine with amd processor
Regards
On Mar 1, 2012 2:07 PM, Dave Platt dpl...@radagast.org wrote:
5. Placing ferrite cores on the phone cables.
Do either of the phone lines in question have DSL on them?
If so, a ferrite core
Hello
http://www.voip-info.org/wiki/view/Asterisk
I prefer asterisk under linux sistem works better.
Regards
On Sun, Jun 17, 2012 at 12:28 PM, Jim Schultz jimschultz...@gmail.comwrote:
Greetings,
I am interested in learning more ablout Asterisk. Is there a recommended
link for getting
Hello
Is your server behind nat? This problems sounds me nat problems.
Regards
On Thu, Jul 12, 2012 at 7:53 AM, Roland o/d Akker aster...@rolandow.com wrote:
I have this very specific problem with two dect sets. Problem that I have is
one-way audio, in this very rare situation.
I am
Hello
a2billing works fine
Regards
On Mon, Jul 16, 2012 at 1:47 PM, Goke M Aruna gok...@gmail.com wrote:
hi all,
Can someone give me information on any open source asterisk calling card
solution?
I have laid my hands on astpp, astcc, asterisk-prepaid-0.3.1, agi-ccard.agi
without luck.
I
Hello
You will need to do, something like
[outbound]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten = s,n,Playback(message-when-machine)
exten = s,n,Hangup
exten = s,n(humn),WaitForSilence(500)
exten =
Hi
Have you seen thirdlane?
Thirdlane has a multitenant version.
Regards
On Aug 11, 2012 11:11 AM, Carlos Alvarez car...@televolve.com wrote:
On Sat, Aug 11, 2012 at 3:16 AM, Kannan vasdevelo...@gmail.com wrote:
I am planning a multi-tenant VoIP services system with Asterisk, using
Hello
I think you must change
type = peer
insecure=invite,port
qualify=yes ; for monitor the ip
Regards
On Fri, Aug 17, 2012 at 2:11 PM, Selecstine Bucci Anukwu
buchal...@gmail.com wrote:
Hello Everyone,
We are trying to integrate a hosted soft-switch to an Asterisks server and
the error
Hello
Check voicemail.conf
maxmsg = 100
And change it.
On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi
dionisi.dan...@gmail.com wrote:
I'm sorry, I haven't been clear.
I do not have to check the inbox on Asterisk, but I have to check the free
space on a particular mailbox of Exchange
Hello
In indications.com are the tones for several countries
On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote:
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
Hi
Ok, I think vpn is good way, but , you can use tls that uses certificates,
and srtp for media encriptatio, in sip protocol.
Regards
On Sep 29, 2012 12:59 PM, Chris Nighswonger cnighswon...@foundations.edu
wrote:
On Sat, Sep 29, 2012 at 12:41 PM, Carlos Rojas crt.ro...@gmail.com
wrote
Hello
You should be modify the volume in the file, there are several
software for that, like wavepad .
Regards
On Mon, Oct 1, 2012 at 2:52 PM, Danny Nicholas da...@debsinc.com wrote:
AFAIK, there is still not a MOH volume control. What I did was to take my
moh wav files and run them
Hello
Yes, has a berckeley database, wirh function blackllist
Regards
On Oct 9, 2012 12:51 AM, Joseph syscon...@gmail.com wrote:
Can someone refresh my memory how blocking incoming call works based on
caller ID in Asterisk 1.8?
If I remember correctly in asterisk 1.4 it was possible to block
Hi
You will need change the names for your extensions
101-company_a
102-company_a
ETC
On Wed, Oct 31, 2012 at 2:23 PM, Darin Iv adari...@gmail.com wrote:
Is it possible to bul multitenant system using some third party opensouce
application My design is like this.
Company A:
Context
Hello
In SIP.find you can to use
Deny=0.0.0.0/0.0.0.0
Permit=192.168.1.25/255.255.255
Regards
On Nov 19, 2012 7:12 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi;
How I can make my configuration to allow the sip phones only from specific
IP addresses range (for example from 192.168.10.1 -
Maybe,
You can do that, with queues, and ringall strategy.
On Wed, Dec 5, 2012 at 4:53 PM, Leandro Dardini ldard...@gmail.com wrote:
You can dial all the extensions at once, putting all them in the dial
string, separated by . There is no other method.
Leandro
2012/12/5 Paolo De Michele
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