If you can use curl, and can do some text parsing and know regular
expressions, you may be able to use this free CNAM service:
http://www.numberguru.com/ and integrate into your system. This one appears
to have a more complete database. When I tried my number, I have gotten my
full name, but
That's how analog lines work. Asterisk do not know when the called is picked
up so it goes straight to the context execution. You may want to try
setting callprogress=yes and answeronpolarity=yes on your chan dahdi conf
file as a work around, or switch to PRI
On Sun, May 16, 2010 at 3:38 PM,
Why don't you copy the files to your asterisk box and play them from there?
-- Sent from my Android device
On Apr 16, 2010 5:03 PM, Edwin Quijada listas_quij...@hotmail.com wrote:
Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with ...
I did.
I have asked a GSM operator in my country if he can route a number or a
short code to my asterisk server via SIP (since they dont give DIDs in my
country) the operator said they do not support SIP, they have no way of
converting GSM calls to SIP to then send them to me. I would like to know
what
You might be able to do it using FastAGI
http://www.voip-info.org/wiki/view/Asterisk+FastAGI
On Mon, Mar 8, 2010 at 4:33 AM, Pham Quy qu...@vega.com.vn wrote:
hi all,
We going to implement a music service which enable user to playback a
song by dialing to a service number.
The problem is
This may help you:
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
On Wed, Mar 3, 2010 at 11:20 PM, Pham Quy qu...@vega.com.vn wrote:
Hi all,
It maybe not clear that what i'm going to do.
What i want to do is that enable user to call to a number then a
background
I would love to hear some inputs on Aastra and Snom IP phones.
On Wed, Feb 10, 2010 at 4:36 PM, Jeff LaCoursiere j...@jeff.net wrote:
On Wed, 10 Feb 2010, Tim Nelson wrote:
- Gordon Henderson
gordon+aster...@drogon.netgordon%2baster...@drogon.net
wrote:
If not using PoE I'd
.
This way the gateway does not have to register, and I can keep the settings
that passes the right caller id. Another way would be to have asterisk read
another field for the caller id, because the number of the caller is
somewhere on the sip invite.
2009/11/27 Massimo Nuvoli mass...@archivio.it
Pascal
Hi,
I am experiencing a weird issue with my MV-372.
Mobile1 Mobile2 are both registered to my asterisk server, I am able to
use them for outgoing call with no problem, but when I call the sims in my
gateway, they are routed to the right context/extension/priority, but as
soon as I hangup, the
What are you trying to achieve here? The h extension is for when the
channel hangs up. And if the caller hangs up how will he leave you a
voicemail?
Sent from my iPod
On Nov 11, 2009, at 7:20 AM, Anahi Ludueña a_ludu...@hotmail.com
wrote:
Hi people, just a question:
Is it possible to
Lol
Sent from my iPod
On Oct 27, 2009, at 6:59 AM, Alex Balashov abalas...@evaristesys.com
wrote:
aster...@opensourcesolution.in wrote:
installing asterisk
I am intrigued by your ideas and would like to subscribe to your
quarterly newsletter, as well as attend your biannual leadership
pfsense
On Tue, Oct 13, 2009 at 12:12 PM, Doug Lytle supp...@drdos.info wrote:
David Wathen wrote:
Hi,
My customer has a outdated firewall that is also presenting a NAT
nightmare for getting the Asterisk server reachable from the internet.
What firewalls work good with VOIP? I
I believe the administrator can see what is on your screen with screen with
those screen sharing stuff, this makes it harder a lil bit, and
www.boratproxy.com becomes useless in that case.
On Wed, Sep 16, 2009 at 1:28 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
On Wed, Sep 16,
On Wed, Sep 16, 2009 at 2:37 PM, Zoa zoach...@securax.org wrote:
What if i send my twin brother to take the exam instead of me... ?
z
If you think you cannot pass the test yourself, your twin wont be able to
pass it neither, he can be even worst than you
lol
For example if it was Alex to reply to that msg, i would feel bad for
this guy, because Alex would make him feel like if he cannot do this
by himself or use google to find that answer by himself, he does not
belong to that list. He would never give him a chance and try to help
him.
Sent
You have to fix the dependency issues, which means install the stuff
you are missing that cdrmysql depends on so u can recompile it.
Sent from my iPod
On Aug 30, 2009, at 11:18 AM, Cyprus VoIP voi...@gmail.com wrote:
Thanks. I found out that the module didn't load:
[Aug 30 20:35:59]
On Tue, Aug 25, 2009 at 1:29 PM, Alex Balashov abalas...@evaristesys.comwrote:
Sanjoy Rath wrote:
I would prefer to use AMI. Let me start looking into AMI. I would like
to include functionalities like upload numbers to call from an
interface, i want reports back numbers called, setup call
On Tue, Aug 25, 2009 at 2:52 PM, Alex Balashov abalas...@evaristesys.comwrote:
With enough spiritual commitment, anything can be done; you certainly
*can* do it this way. You can write a fairly sophisticated dialer in
Bash, too.
The issue is whether it is methodologically correct and
http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu
On Fri, Aug 21, 2009 at 5:21 PM, trebaum treb...@telepaths.org wrote:
Start here, http://www.asterisk.org/support
~T
On Aug 21, 2009, at 1:22 PM, aster...@opensourcesolution.in
aster...@opensourcesolution.in
wrote:
I have a DID from IPKall that is forwarded to my Asterisk box and I have
done no special configuration for SIP URI:
On IPKall you put the IP address of your asterisk box or the hostname then
is Sip phone number you put the did number they gave you and thats it.
On your asterisk box you set your
Why not record a ring tone, and playback the file? with $agi-streamfile???
On Tue, Aug 18, 2009 at 11:38 AM, Barton Fisher bhfis...@icpage.com wrote:
I'm going blind searching - maybe you know?
During the execution of a script I want to play fake ring to caller. Both
of these examples
Lol but he has a good point and makes a lot of sense. Never thought about
that strategy...
On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon
dig...@sanguinarius.co.ukwrote:
Michael Graves wrote:
Pricing is a very legitimate way to minimise support effort. It winnows
down the market size to
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
http://store.digium.com/productview.php?product_code=1SFA0001
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon
Did you get CDRTool to work with Asterisk or Areski's CDR Stats?
On Fri, Aug 14, 2009 at 10:20 AM, harry R rhm.noa...@gmail.com wrote:
Hi
I just solve my problem today. Just a package on redhat that I need
install.
H.
___
-- Bandwidth and
Where you able to compile DAHDI in a virtual environment? How about skype
for asterisk? Has anyone tried that in a virtual environment? Seems like
to register the license, digium tool is looking for a connection on eth0,
and in a virtual environment I see the name as vnet0 or vnet1. At least
On linux you can use Sox. Google it and resd the documentation to see
how you can convert files from the command line. On windows you can
use Switch by NCH Software. Download the trial then you can pay a
small fee if you want to keep it.
Sent from my iPod
On Aug 2, 2009, at 10:30 AM,
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Sent from my iPod
On Aug 2, 2009, at 3:49 AM, Thomas Kenyon dig...@sanguinarius.co.uk
wrote:
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could
Well I think thats what the problem was, I dont have it named as eth0. So
if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant
it just scan you nic handles? Can someone point me to where I can change
the NIC name in the source file or something???
On Sun, Aug 2, 2009
This question has been asked thousands of time on this list. You mat want
to search the archive, but to sum it all, there is no limit as far as calls
on an Asterisk server. It all depends on your server's specs, and how it is
setup. A celeron processor with 256Mb Ram could handle a fews calls
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning ttbrown...@gmail.com wrote:
Nice job. It worked right away for me with my 10 channel trial license.
Asterisk 1.4.26
Just a little clarification for people refering to Asterisk as a PBX
and not an Answering Machine:
In fact, Asterisk is neither a PBX nor an Answering Machine. Asterisk
is a Telephony Toolkit. You can choose to use it as a PBX or an
Answering Machine or both or even in some case as a
That's right, they say it is a PBX because it is mostly used as such, but
it is more than just a PBX. Some people use it as a VoiceMail tool or to
handle just conference, some use it to add functionalities to other legacy
PBX systems. Calling cards applications for example, a plain PBX wont be
If you have asterisk addons installed you can use the mysql
applications to make queries. I find it to be very easy if you know
how to do select and insert queries and understand the basic mechanism
of the dialplan. Other than that, you may want to hire someone to do it.
Sent from my iPod
It seems like a few people including me DID understand what Dhaval meant, or
maybe some people used they common sense and their intelligence to
understand what somebody who's english is not the primary language wanted to
say and put some effort to guide or help someone in the community getting to
You can use the extension h
exten = h,1,app()
exten = h,n,app()
.
On Mon, Apr 20, 2009 at 10:31 AM, Michael mich...@networkstuff.co.nzwrote:
What is the syntax to progress with a dial plan after hangup please?
Michael
___
-- Bandwidth and
I believe she is refering to how she's going to send you your incoming calls
(on your DIDs) for example:
10 digits: 972-453-2345
7 digits: 453-2345
4 digits: 2345
so you know how to expect your incoming calls and configure your
extensions.conf accordingly.
On Fri, Mar 27, 2009 at 10:06 AM,
You may want to check this link
http://www.geodatasource.com/cities-free.html
it may help you
On Sun, Mar 22, 2009 at 8:14 PM, Cary Fitch ca...@usawide.net wrote:
I don’t, but it out to be “out there”. We needed a list of all (valid)
bank routing numbers for a check writing program and a
-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
*Sent:* Thursday, March 19, 2009 4:42 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
I dont want
or
whirlpool to see what they say.
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
*Sent:* Friday, March 20, 2009 9:39 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
*Sent:* Wednesday, March 18, 2009 6:24 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] T1 problem (call using a .call file)
This has to be a bug
Callerid: SIP/YYY
MaxRetries: 1
RetryTime: 5
WaitTime: 60
--
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Pascal Bruno
*Sent:* Thursday, March 19, 2009 9:22 AM
*To:* Asterisk Users Mailing
as
callerid.
On Thu, Mar 19, 2009 at 4:23 PM, Pascal Bruno tipas...@gmail.com wrote:
Here is what I get from the console with the call file:
-- Attempting call on DAHDI/g1/1201XXX for s...@fortest:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127
-users] T1 problem (call using a .call file)
Pascal Bruno wrote:
Also very strange, when in my call file I change the callerid line to
SIP/whatever like Danny said, the call go through, but I dont want
that, because when I do so, it is displaying the main number on my T1
account as caller
get the PROGRESS with cause code 127
On Wed, Mar 18, 2009 at 10:47 AM, David Backeberg dbackeb...@gmail.comwrote:
On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno tipas...@gmail.com wrote:
I have a weird problem with call using my T1 card. I can make calls fine
using my analog and IP phones
This has to be a bug, because I dont know what else to try here
On Wed, Mar 18, 2009 at 11:16 AM, Pascal Bruno tipas...@gmail.com wrote:
Nope, I always dial 1 + 10 digits for all my numbers. It works on all
numbers when I am using my phone (Analogue or IP) but when I do it using a
.call
Do you have an extension set for 246463 in your extensions.conf?
On Mon, Mar 16, 2009 at 1:54 PM, Bayardo Sanchez
bayardo.sanc...@gmail.comwrote:
i create inbound number but i calling and send this error:
[Mar 16 11:41:12] NOTICE[30847]: chan_sip.c:14383 handle_request_invite:
Call
Your sip.conf should look like this
sip.conf
[procall]
type=peer
username=XX
secret=XX
context=default
and extensions.conf
[default]
exten = 246463,1,Dial(SIP/8003)
you must also have a sip user for 8003 in your sip.conf like
[8003]
type=friend
username=XX
secret=XX
context=outgoing
And
I have a weird problem with call using my T1 card. I can make calls fine
using my analog and IP phones, but when I try to initiate a call using a
.call file, I get the following error
-- Attempting call on DAHDI/g1/1XX for s...@test:1 (Retry 1)
-- Requested transfer capability: 0x00
check ipkall.com
On Sat, Mar 14, 2009 at 12:46 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello
please ho to get a free did number for my asterisk ?
also, is it pocible to assign it to a group of extentions ?
thanks!
___
-- Bandwidth and
I have the same situation. My scenario is weird:
I have a DID with IPkall that points to my asterisk server, and I have it
play a message with Playback() after about 20 seconds call drops and give
me the same message you get: no reply to our critical packet
BUT
I have a DID from Vitelity, and
:
3 feb 2009 kl. 04.33 skrev Ex Vito:
On Tue, Jan 27, 2009 at 10:21 PM, Pascal Bruno tipas...@gmail.com
wrote:
Is it possible to retrieve the DIALSTATUS variable when placing
call through
a call file. This variable is set when using the Dial()
application from
the dialplan, but I am
Hello,
Is it possible to retrieve the DIALSTATUS variable when placing call through
a call file. This variable is set when using the Dial() application from
the dialplan, but I am using a call file for my current application and need
to get the dialstatus.
Thank you.
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
___
-- Bandwidth and Colocation Provided by
Actually I installed them after, so do you recommend I recompile asterisk?If
I do so, I wont loose my current configuration files right?
On Tue, Jan 27, 2009 at 6:27 PM, Philipp Kempgen
philipp.kemp...@amooma.dewrote:
Pascal Bruno schrieb:
I have remove the comment defor res_odbc.so
I have having a hard time setting the cdr with cdr_odbc. Below is all my
conf file related to it, let me know what I am doing wrong. Thank you.
*cdr.conf*
[general]
enable=yes
[csv]
usegmtime=yes; log date/time in GMT. Default is no
loguniqueid=yes ; log uniqueid. Default is no
by that field.
David
2009/1/24 Tilghman Lesher tilgh...@mail.jeffandtilghman.com
On Friday 23 January 2009 18:22:16 Pascal Bruno wrote:
Is it possible to log just the outgoing calls using cdr_odbc into a
custom
mysql database table?
my table will look like
Is it possible to log just the outgoing calls using cdr_odbc into a custom
mysql database table?
my table will look like this:
| call_status |
|-- --|
| · id |
| · destination |
| · status |
||
I just need to store the
Dear List,
I have just installed a Digium TDM808 (8 fxo port) on an Asterisk 1.6.3.
When I try making a call with a .call file, the call goes straight to the
dialplan and start executing the dialplan even before the called party has
pick up. Anybody knows why by any chance?
Any help would be
Is there any way of going around this??? Any tricks, configuration hacks??
On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote:
On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
I have just installed a Digium TDM808 (8 fxo port) on an Asterisk
1.6.3. When I try
...
The only real solution is to go digital...
d
2009/1/21 Pascal Bruno tipas...@gmail.com
Is there any way of going around this??? Any tricks, configuration hacks??
On Tue, Jan 20, 2009 at 4:39 PM, Jared Smith jsm...@digium.com wrote:
On Tue, 2009-01-20 at 15:30 -0500, Pascal Bruno wrote:
I
Is it possible for asterisk to send sms through a GSM gateway, tor example
the Portech MV-37X?
If yes, any examples of configurations would be really apreciated.
On Tue, Oct 14, 2008 at 11:13 PM, Steve Totaro
stot...@totarotechnologies.com wrote:
The most flexible way but will require a bit
:
Emmanuel Pascal Bruno wrote:
Has anyone been able to configure portech's mv-378 gateway with
asterisk?
I did the configuration as per the manual but it does not work.
My server sees the portech gateway, but when the gateway is trying to
register to my server it fails. It says peer
Thank you!, I will try that in a few hours and let you know what happens.
On Fri, Jan 16, 2009 at 11:01 AM, Marco Signorini marcota...@libero.itwrote:
Pascal Bruno wrote:
Thanks for your reply!
Can you tell me what you have in your Portech configuration settings
(Mobile to Lan Settings
Marco,
The configs work fine for me. I can receive calls with no problem. Now,
were you able to dial using the sim card? I cant figure out how I can do it
since asterisk doesnt have a channel to place call through the portech
gateway.
On Fri, Jan 16, 2009 at 12:04 PM, Pascal Bruno tipas
help would be appreciated.
Thanks
On Sat, Jan 17, 2009 at 12:39 AM, Pascal Bruno tipas...@gmail.com wrote:
Marco,
The configs work fine for me. I can receive calls with no problem. Now,
were you able to dial using the sim card? I cant figure out how I can do it
since asterisk doesnt
Sorry for bothering you, but I got it, I just had to put # in callnum!
On Sat, Jan 17, 2009 at 1:44 AM, Pascal Bruno tipas...@gmail.com wrote:
I want to dial out using the sim card. What I did, I have used the SIP
channel ex:
Channel: SIP/thenum...@mv378
It shows the called is being
Has anyone been able to configure portech's mv-378 gateway with asterisk?
I did the configuration as per the manual but it does not work.
My server sees the portech gateway, but when the gateway is trying to
register to my server it fails. It says peer is not suppose to register.
The gateway
The latest Nokia phones come with a SIP client and I like them.
On Wed, Nov 5, 2008 at 10:56 PM, Pedram M [EMAIL PROTECTED] wrote:
Any recommendations on good wireless SIP phones?
Thanks,
Pedram
___
-- Bandwidth and Colocation Provided by
I have turned off firewall on the linux box, I have turned off firewall on
the router I still have the same problem :-(
On Sat, Nov 1, 2008 at 1:22 PM, Emmanuel Pascal Bruno [EMAIL PROTECTED]wrote:
Oh ok, I knew it was something like that. I have tried many different
settings on my router
I have tried that too with no results
On Sun, Nov 2, 2008 at 1:30 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Emmanuel Pascal Bruno wrote:
I have turned off firewall on the linux box, I have turned off
firewall on the router I still have the same problem :-(
Disabling firewalls is almost
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then
Oh ok, I knew it was something like that. I have tried many different
settings on my router. I'll dig into it some more.
Thanks
On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Emmanuel Pascal Bruno wrote:
I have a DID from IPKall.com which is forwarded to my asterisk
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then
Anybody knows how to get a Coupon Code for the discount on the Asterisk
training classes??? I am interested on taking that upcoming Asterisk
Advance course, and 3K is kinda steep and considering I am still a college
student paying this training out of my pocket, every bit helps.
asking for free training, so I don't see why you are saying for that matter
you think people with experience should get the dCAP. It doesn't make any
sense to me.
On Wed, Sep 17, 2008 at 10:42 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, Sep 17, 2008 at 2:37 PM, Pascal Bruno [EMAIL
/asterisk/asterisk-1.4.21.2.tar.gz
tar -zxvf asterisk-1.4.21.2.tar.gz
cd asterisk-1.4.21.2
./configure
make menuselect (You don't have to select anything)
make
make install
make samples
Pascal Bruno wrote:
I am about to install Asterisk on a Fedora 9 box, but i see with yum
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Pascal Bruno
*Sent:* Friday, September 12, 2008 9:14 AM
*To:* [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Asterisk and Fedora 9
Ok very good, how
subject" VALUE="Immediate rek- SR.SAP Consultant-8+yrs exp --Need Locals to - ILL">
Reply via email to
- Date --
[asterisk-users] Asterisk and Fedora 9
Pascal Bruno
Re: [asterisk-users] Asterisk and Fedora 9
Anthony Messina
Re: [asterisk-users] Asterisk and Fedora 9
Pascal Bruno
Re: [asterisk-users] Asterisk and Fedora 9
Pascal Bruno
Re: [asterisk-users] Asterisk and Fedora 9
A
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