Why make a brand new?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> moona ather
> Sent: Sunday, May 07, 2006 11:36 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] asterisk management interface
>
> Hi,
> I have to make a
Hi,
I have to make a web-based management interface of configuring asterisk
i wanted to know if it is as simple as reading the .conf files and searching
for the required section in the file and adding users etc. or there are
other steps involved too?? As I have seen many such built codes on
I've been wondering about how to do what you're describing as well! Anyone
know how it can be done?
Regards,
Bevan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe
Sent: Monday, 8 May 2006 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-
I was just looking through the Wiki for some info on how to retain the
original caller's callerid when make transfers to internal extensions, and I
came across the parameter below:
useincomingcalleridonzaptransfer=yes
There is nothing in the zapata.conf file from vers. 1.2.7 so I am wondering
if
On Mon, May 08, 2006 at 12:40:55AM +0200, Francesco Peeters wrote:
> Hi,
>
> I would like to execute a command on a different system using ssh.
>
> When I execute the command from the CLI on the asterisk machine, it
> works fine (I set up RSA keys on both sides)
>
> When I execute the same comm
Does anyone have any suggestions as how to enter a CDR Account code
during a call?
I know it can be done in the extension logic before the answering the
call, but I wanted to optionally enter an account code on certain calls
without prompting on every call before or after the call?
_
I use chanspy to listen to agents for QA. It works great for listening
to random conversations. * works to switch between calls.
I cannot however specify the agent I want to listen to. I dial the
agent's extension and press # but get connected to some random
conversation, not the one I dial
Francesco Peeters schreef:
Hi,
I would like to execute a command on a different system using ssh.
When I execute the command from the CLI on the asterisk machine, it
works fine (I set up RSA keys on both sides)
When I execute the same command from System() inside the dialplan, the
log shows
Hi,
I would like to execute a command on a different system using ssh.
When I execute the command from the CLI on the asterisk machine, it
works fine (I set up RSA keys on both sides)
When I execute the same command from System() inside the dialplan, the
log shows it is being executed, and a
At 04:33 PM 5/6/2006, you wrote:
All I need is a way to uppercase a string, which from everything
I've read so far isn't in the code. Then again, I could just use
all uppercase for my SIP/IAX device names even if it *does* look ugly. ;)
What if you just prefix all names with the number one?
Bob's Leaky News Service schreef:
Simple as that please email me direct. [EMAIL PROTECTED]
Also looking for a U.S. DID provider as well as orig provider.
FWD (FreeWorld Dialup) will allow Toll Free outbound...
So will some of the Finarea services (check the wiki for their various
services. A
I have 20 or so users with analog cordless phones connected via a 24port FXS box (vegastream). The vegastream supports voicemail indication via a studder tone, which is great but I have some users asking for a more positive (or proactive) indication of voicemail.
I had a couple ideas;
1.If an
Simple as that please email me direct. [EMAIL PROTECTED]
Also looking for a U.S. DID provider as well as orig provider.
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> >> Since when do these use IAX? He asked for IAX hardphones... If I
> >am mistaken
> >> let me know since I am looking for good reliable SNOM-like IAX phones as
> >> well! :-)
> >
> >I'm sorry if i recommend some foolish (i've just joined the maillist)
> >but have you tried PA168 chip based hard
Well, to tell the truth, the phones, what available in Hungary, is 90%
working. The other 10% is sometimes bad as you get out off the box, sometimes
it's noisy, echoing, crappy sound, rebooting, etc.
Is i asked so many folks on Cebit (who resells this phone) most of them, told
me, there are two k
I'd rather shoot myself in the head! other day we had a site that flashed the PA168 chipset phones with new firmware and they all ended up with the same MAC address!! I thought that shouldn't happen normally ...And talk about nasty cheap effects, sidetone, distortion and the list goes on.
RobOn 07/
At 12:55 PM 5/7/2006, you wrote:
Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter
osszedobalt bytejaira:"
>
> Since when do these use IAX? He asked for IAX hardphones... If I
am mistaken
> let me know since I am looking for good reliable SNOM-like IAX phones as
> well! :-)
I'm sorry
Keyboardot ragadtam, hogy va'laszoljak Koopmann, Jan-Peter osszedobalt
bytejaira:"
>
> Since when do these use IAX? He asked for IAX hardphones... If I am mistaken
> let me know since I am looking for good reliable SNOM-like IAX phones as
> well! :-)
I'm sorry if i recommend some foolish (i've j
On 7 May 2006, at 16:16, Aaron Daniel wrote:
On Sun, 7 May 2006, Tofik Suleymanov wrote:
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
On Sun, 7 May 2006 19:58:26 +0500, "Farhad Ibragimov" <[EMAIL PROTECTED]> wrote:
> Thanks
>
Try reading this URL (spanish language):
http://www.ecualug.org/?q=2006/02/28/comos/asterisk_1_2_4_agregando_soporte_para_el_protocolo_h_323
With the page instructions I can call from and to H.323 to
This extremely useful dialplan requires the standard Asterisk sounds,
plus the additional ones in the asterisk-sounds package.
Scott.
[haiku]
exten => s,1,Playback(privacy-please-dial)
exten => s,n,Playback(letters/a)
exten => s,n,Playback(high)
exten => s,n,Playback(letters/q)
exten => s,n,
On Sun, May 07, 2006 at 08:44:41AM -0400, Doug Lytle wrote:
> Wilson Pickett wrote:
> >>> No, you have to kill the op_server app and restart it
> >>This is incorrect. You can just send it the HUP (Hangup) signal and it
> >>will reload it's configuration files.
> >
> >Isn't that what HUP does? :)
>
On Sun, 7 May 2006, Tofik Suleymanov wrote:
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
Tofik Suleymanov
I'll pipe in on this one. We
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Sunday, May 07, 2006 7:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] H323 to SIP
You could begin with:
http://www.voip-info.org
You could begin with:
http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
http://www.voip-info.org/wiki/view/Asterisk+H323+channels
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels
and much more.
You need to install chan_h323 module and configure as well as
I dont have practice to work with Asterisk but I see that is a great soft.
If you have any idea or some config files can you help me
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alberto
Sagredo
Sent: Sunday, May 07, 2006 7:34 PM
To: Asterisk Users Ma
You could make a H323 to SIP transport. Before to do that, you need to
have installed and working both chan protocolos on Asterisk.
aFarhad Ibragimov escribió:
Hi all
I have installed station which support only H323 protocol. I want to
install SIP telephone. Is it possible to call SIP teleph
Hi all
I have installed station which support only H323
protocol. I want to install SIP telephone. Is it possible to call SIP telephone
throught my station
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Asterisk-Users ma
Turns out there is a fault due to the new way that asterisk installs It isnt feasible to fix it until the asterisk installation method has settled down a bit.
Does anyone know a time scale on this?
Dan Journo
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Wilson Pickett wrote:
> No, you have to kill the op_server app and restart it
This is incorrect. You can just send it the HUP (Hangup) signal and it
will reload it's configuration files.
Isn't that what HUP does? :)
No,
HUP sends the Hang UP signal, causing an application to reload/re-read
On Tuesday 02 May 2006 11:08, richard Coco wrote:
> Hi all,
>
> i have an Asterisk box with an Eicon 4BRI with
> chan_capi-cm and every thing works fine. We now plan
> to install a new Asterisk using a Dialogic BRI/2VFD.
> Is the Dialogic card supported and can i use
> chan_capi-cm? Has anyone mana
>
> Small java applications seem to be quite common among many phones. I
> tend to stay away from such phones. I'd first like phone vendors to
get
> their acts together and provide me decent standard interfaces.
Currently
> I need an expensive cable just for the pleasure of connecting my
mobile
>
On Sunday, May 07, 2006 10:07 AM Steve Totaro wrote:
> Oooops, sorry its late.
Obviously. :-)
> My favorites in order, Polycom, Snom, Cisco.
Since when do these use IAX? He asked for IAX hardphones... If I am mistaken
let me know since I am looking for good reliable SNOM-like IAX phones as
we
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA256
Obelix wrote:
> Does Asterisk have voice prompts for the following.
>
> 1. The number you dialled is not available. Please try again later.
>
> 2. The number you dialled is not recognised
Take a look at the following URLs for a good list of the so
Hi!
I'm using freepbx, with * 1.2.6, everything is working nice, except fax
handling.
the incoming faxes got received:
May 6 23:24:39 DEBUG[12505] app_rxfax.c:
==
May 6 23:24:39 DEBUG[12505] app_rxfax.c: Pages transfe
Hello list,
We are going to build and deploy telephony-system for approx ~1000 users
with ASTERISK as main PBX.I was googling through this mailing list and
found a lot of useful information.Some of my questions solved ,some
other are still on agenda.Below I will formulate short questions that
Hello folks,
firstly, thank you for your useful and fast answers !
Is there anybody using D-Link SIP phones ?
Are D-Link SIP phones ok to install in production environment ?
Give your comments please.
Tofik Suleymanov
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Does Asterisk have voice prompts for the following.
1. The number you dialled is not available. Please try again later.
2. The number you dialled is not recognised
/Obelix
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Asterisk-Users
Tofik Suleymanov wrote:
Steve Totaro wrote:
Give idefisk a try. It works very well for me, its free, and does
not crash all the time like Cubix (formerly Firefly).
Hello Steve,
As far as i know 'idefisk' is a softphone, but i need a hardware phone.
Thank you for reply.
Tofik Suleymano
Tofik Suleymanov wrote:
Steve Totaro wrote:
Give idefisk a try. It works very well for me, its free, and does
not crash all the time like Cubix (formerly Firefly).
Hello Steve,
As far as i know 'idefisk' is a softphone, but i need a hardware phone.
Thank you for reply.
Tofik Suleymanov
Wilson Pickett wrote:
anyone using hardware IAX phones with asterisk ?
I've googled on this issue and found several hardware phones which
support IAX protocol, but before paying money I'd like to know more
about what people experiencing with them.
I have had three of them for neary two years.
> No, you have to kill the op_server app and restart it
This is incorrect. You can just send it the HUP (Hangup) signal and it
will reload it's configuration files.
Isn't that what HUP does? :)
___
--Bandwidth and Colocation provided by Easynews.com -
On Sun, May 07, 2006 at 12:21:53AM -0400, Roger Gulbranson wrote:
> On Sat, 2006-05-06 at 19:43 -0400, Steve Totaro wrote:
> > Roger Gulbranson wrote:
> > > On Sat, 2006-05-06 at 07:42 -0400, Steve Totaro wrote:
> > >
> > >> I have a TDM4xxp card with no modules. My question is, will this card
anyone using hardware IAX phones with asterisk ?
I've googled on this issue and found several hardware phones which
support IAX protocol, but before paying money I'd like to know more
about what people experiencing with them.
I have had three of them for neary two years. Here's an "executive rev
He asked about hard phones not soft phones.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Steve Totaro
> Sent: Sunday, May 07, 2006 12:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] asterisk
Steve Totaro wrote:
Give idefisk a try. It works very well for me, its free, and does not
crash all the time like Cubix (formerly Firefly).
Hello Steve,
As far as i know 'idefisk' is a softphone, but i need a hardware phone.
Thank you for reply.
Tofik Suleymanov
_
Give idefisk a try. It works very well for me, its free, and does not
crash all the time like Cubix (formerly Firefly).
Tofik Suleymanov wrote:
Hello folks,
anyone using hardware IAX phones with asterisk ?
I've googled on this issue and found several hardware phones which
support IAX protoc
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