Date: Mon, 23 Jun 2008 08:00:08 -0400
From: David Backeberg [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with
ringing sound
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL
2008/12/10 [EMAIL PROTECTED] [EMAIL PROTECTED]
This should be sufficient to get it to work from zoiper to zoiper.
http://asteriskguru.org/tutorials/zoiper2zoiperfaxt38.html
If you would still experience any issues, please send us a packet
capture + a description of the setup.
Fine !
Hi,
I am interested in building my own DIY IP hardphone to connect to Asterisk
for my personal usage. Does anyone know of any good reference design in
guidance me on how to build one? I am capable of building it from even raw
material or circuit design if I can get some info on how to start.
On Wed, 10 Dec 2008, Gideon Hack wrote:
Hi Gordon,
DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the DIDs
that you require. And they can forward to IAX if that is preferable to you.
Thanks.
I was actually hoping I'd find a Swedish company, but I'll pass this and
mark morreny wrote:
Hi,
I am interested in building my own DIY IP hardphone to connect to
Asterisk for my personal usage. Does anyone know of any good
reference design in guidance me on how to build one? I am capable of
building it from even raw material or circuit design if I can get
Hi Aldo,
sorry not having posted it. ::))
You have to set canreinvite=no.
Giorgio Incantalupo
Aldo Alexander Leyva Alvarado wrote:
What parameter???
2008/1/17 gincantalupo [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
Hi Olle,
that was a phone misconfigurationa parameter
Hi Gordon,
Take a look at http://www.cellip.com/
//Peter
Gordon Henderson wrote:
On Wed, 10 Dec 2008, Gideon Hack wrote:
Hi Gordon,
DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has the
DIDs that you require. And they can forward to IAX if that is preferable to
Have a nice day,
Scott Berry
E-mail: [EMAIL PROTECTED]
I am studying out of the book Asterisk: The Future of Telephony on
Chapter 4, and right now for practicing using the built in Debian
version of Asterisk for Ubuntu. I am however having some problem where
I cannot do asterisk -r
Scott Berry wrote:
Have a nice day,
Scott Berry
E-mail: [EMAIL PROTECTED]
I am studying out of the book Asterisk: The Future of Telephony on
Chapter 4, and right now for practicing using the built in Debian
version of Asterisk for Ubuntu. I am however having some problem where
I
You've checked that another asterisk is running (ps -ef|grep asterisk)?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott Berry
Sent: Wednesday, December 10, 2008 9:06 AM
To: Asterisk Users
Subject: [asterisk-users] a problem on Ubuntu with Asterisk
On Wed, Dec 10, 2008 at 5:59 AM, mark morreny [EMAIL PROTECTED] wrote:
Hi,
I am interested in building my own DIY IP hardphone to connect to Asterisk
for my personal usage. Does anyone know of any good reference design in
guidance me on how to build one? I am capable of building it from
On Wed, Dec 10, 2008 at 09:05:34AM -0600, Scott Berry wrote:
Have a nice day,
Scott Berry
E-mail: [EMAIL PROTECTED]
I am studying out of the book Asterisk: The Future of Telephony on
Chapter 4, and right now for practicing using the built in Debian
version of Asterisk for
On Wed, 10 Dec 2008, Peter Lindquist wrote:
Hi Gordon,
Take a look at http://www.cellip.com/
Ah! Thanks! I'll pass it on.
Gordon
//Peter
Gordon Henderson wrote:
On Wed, 10 Dec 2008, Gideon Hack wrote:
Hi Gordon,
DID World Wide (see http://www.didww.com/virtual_numbers/Sweden) has
Have you do just asterisk before try to reconect to cli??
Try asterisk - to see if is crashing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo
Gonzalez
Sent: miércoles, 10 de diciembre de 2008 01:12 p.m.
To: [EMAIL PROTECTED]; Asterisk
A bit of clarification on my previous answer that may help with some of
these replies:
Asterisk is theoretically launched at start up or some other time with
/usr/sbin/asterisk
-g causes core dump
-vvv causes verbosity level to be set
When you try to run your command, it should always include
On Wed, Dec 10, 2008 at 01:33:39PM -0200, Sebastian wrote:
Have you do just asterisk before try to reconect to cli??
Try asterisk - to see if is crashing.
This runs asterisk as root (instead of as the user asterisk). In the
worst case it might write some files as root (e.g. the log
Scott,
Login as root user and start asterisk by typing asterisk and then give
command asterisk -r
Amit Mehta
Cell: +91 9898340962
On Wed, Dec 10, 2008 at 8:35 PM, Scott Berry [EMAIL PROTECTED] wrote:
Have a nice day,
Scott Berry
E-mail: [EMAIL PROTECTED]
I am studying out of the book
if you are using MYSQL.. why don't you query your DB directly from Asterisk ?
the following example is something i use with my servers
[ivr1-cont]exten = 7700,1,Answer
exten = 7700,n,MYSQL(Connect connid 127.0.0.1 root rootpass TarekDB)exten =
7700,n,MYSQL(Query resultid_2 ${connid} SELECT
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer, busy, etc., it is not a
problem, because dialplan continues after the Dial() application, but
when the call is
Use the h extension and execute DeadAGI.
On Wed, 2008-12-10 at 18:21 +0100, Martin Tirsel wrote:
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer,
use deadagi on the h extension maybe?
Cheers
Geraint
2008/12/10 Martin Tirsel [EMAIL PROTECTED]
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer, busy, etc.,
2008/12/10 Martin Tirsel [EMAIL PROTECTED]
Hello,
I am googling for a while but google seems to be broken today, no
solution yet :D I have a PHP script which needs to be started after
Dial() has ended. If there is no answer, busy, etc., it is not a
problem, because dialplan continues after
Steve Murphy wrote:
Just to be pedantic, how would src_cid be different from the clid field
that cdr's have now?
and the same with src_exten vs. src;
A simple example might help to let this sink into my brain.
murf
The main thing is that the originating number shouldn't be linked to
On Wednesday 10 December 2008 01:27:17 Olivier wrote:
When listening to the time and date a voicemail message was received, you
can hear french sentences like :
message reçu à vingt-et-un heure
(message received at twenty and one hour)
It should be (in whatever french flavour):
message reçu
Hi All,
Got some problem with asterisk video, i'm testing an eyebeam and a
grandstream video phone.
call from grandstream to eyebeam works ok, video shows up.
but calls from eyebeam to grandstream there's no video, but audio is ok.
Regards,
Nhadie
Stefan I tried this and now I get this:
-- ast_get_srv: SRV lookup for '_sip._udp.grimlock.vtnoc.net' mapped
to host galvatron.vtnoc.net, port 5060
-- ast_get_srv: SRV lookup for '_sip._udp.megatron.vtnoc.net' mapped
to host grimlock.vtnoc.net, port 5060
-- ast_get_srv: SRV lookup
Hello,
When I execute parkandannounce() in the dialplan, is the extension that the
call is parked to stored in a variable? I would like to send it to an AGI
script but can't seem to figure out where the 'announcer' gets its information.
Thanks
Dave
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Gibbons
Sent: Wednesday, December 10, 2008 1:02 PM
To: 'Asterisk Users Mailing List -
I tried really quickly the other day to send an image to these phones
from the dialplan like this:
exten = 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro)
or
exten = 2821,n,SendImage(asterisk-intro)
It didn't work for me.
Should this work? Is anyone else using this with Polycom
snip
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Danny Nicholas
According to lists.digium.com/pipermail/asterisk-dev/2006-March/019516.html
the value is stored in ${PARKEDAT}
/snip
*grin*
I guess I deserved that.
Thanks for checking.
Dave
Two things - Polycom phones require specific images sizes (480x180 pixels I
thinkg) and the sip.cfg has to allow presentation of images.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce
Sent: Wednesday, December 10, 2008 1:25 PM
To: Asterisk
On Thu, 11 Dec 2008, Nhadie wrote:
Hi All,
Got some problem with asterisk video, i'm testing an eyebeam and a
grandstream video phone.
call from grandstream to eyebeam works ok, video shows up.
but calls from eyebeam to grandstream there's no video, but audio is ok.
It's a Video Codec
Is anyone using fixed Park buttons (some of the ones on the left side
of the screen) on a Polycom phone? Here's what I mean:
- Call is received and parked, by the user pressing an unlit park
button (e.g. 701) and the call is parked there.
- The call can be picked up at any other extension by
Use the docs, Luke.
dev-1*CLI core show application parkandannounce
dev-1*CLI
-= Info about application 'ParkAndAnnounce' =-
[Synopsis]
Park and Announce
[Description]
ParkAndAnnounce(announce:template,timeout,dial[,return_context]):
Park a call into the parkinglot and announce the call
Asterisk Project Security Advisory - AST-2008-012
++
| Product| Asterisk|
The Asterisk.org development team has released Asterisk version 1.2.30.4.
This release is available for immediate download from
http://downloads.digium.com/.
This update for Asterisk includes a security fix for chan_iax2. Please see
the associated security advisory for more details:
Hi,
The way I understand this
(http://www.voip-info.org/wiki-Asterisk+G.729+Licensing) is that a call from
a G729 enabled phone to Asterisk, then to a ulaw SIP provider would take one
license, and would show up as 1/1 (one encoder and one decoder).
So, in short, if all my calls were
Hi Folks,
Had a quick search through the archives for softphones and cannot see any
recommended ones.
My question is what recommended free softphones are out there that can be
used with Asterix? I don't really know how many are out there. Is anyone
currently using a softphone with Asterix and if
X-lite from CounterPath work with Asterisk. No g729 support on the free
version. If u plan to use ulaw will work perfectly.
Best regards,
Chris Hariga
--Original Message--
From: Georgecooldude
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing
Dear all, I've read that it's possible to set up Asterisk 1.4 with video using
H.263 and H.264 video codecs.
Now I'm using just SIP in order to have voice over IP.
My question are:
1) Do both SIP video and voice work OK simultaneously in Asterisk 1.4 ???
2) What is the best SIP video+voice
Hi Folks,
Had a quick search through the archives for softphones and cannot see any
recommended ones.
My question is what recommended free softphones are out there that can be
used with Asterix? I don't really know how many are out there. Is anyone
currently using a softphone with
So, in short, if all my calls were from outside to a G729 enabled phone and
vice versa, I would reach the limit at 30/30, NOT 15/15.
If you had 30 licenses, yes the limit would be when you needed either
30 decoders or 30 encoders. i.e. 1/30 would max you out.
-M+
Thank you for the sanity check!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Darnell
Sent: Wednesday, December 10, 2008 22:06
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 licenses
So, in short,
Hi,
I've tried that also, i remove h263+ and leave only h263. i prioritize
h263 on grandstream aslo. does it work bothways on your side? for mine
it works if i call from grandstream to eyebeam, but eyebeam to
grandstream does not. Thanks!
Regards
Nhadie
Gordon Henderson wrote:
On Thu, 11
I want to build my own calling card system on Asterisk.
I looked at this page -
http://www.voipinfo.org/wiki/view/CallingCard+Applications
and it has listed some applications that I thought could help speed up the
development process though the link down the bottom doesn't work.
Does anyone
Hi,
Since we're moving from a legacy PABX that has been serving one
of our customers for more than 15 years, we'd like this process to
require no human habits change among the users.
Software: Asterisk 1.4.22
Hardware: Polycom phones (mainly 430/601)
Here are the problems:
We did configure call
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