On 10/31/09 10:24, Ivan Stepaniuk wrote:
Joseph wrote:
I always had a problem with SIP and DTMF, I'm using old sipura adapters and
have one digium iaxy FXS unit which works almost perfectly, never had any
problem
with DTMF on this unit.
However, all phones connected to Sipura don't work
Vincent,
You can get hardphones from http://www.atcom.cn/
I've been using a few since long and its pretty reliable. Go for the
AT620's. They are not that costly.
Softphone with Zoiper in the free version doesnt support G723 or G729
codecs. If you need them try IAXLITE.
Just google IAXLITE, you
You can set it up in extconfig.conf:
iaxusers = mysql,asterisk,iaxusers
iaxpeers = mysql,asterisk,iaxusers
sipusers = mysql,asterisk,sipusers
sippeers = mysql,asterisk,sipusers
voicemail = mysql,asterisk,voicemail
extensions = mysql,asterisk,extensions
queues = mysql,asterisk,queues
queue_members
Hi
ok i have understand ;=)
bye
Phibee Network Operation Center a écrit :
Hi
actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
I read on the wiki:
===
Database Config
put the following in res_mysql.conf
Hi
when i use MeetMe, i have this errors:
app_meetme.c: Unable to open pseudo device
Where is the problems ?
i have too warning and error into my logs:
[Nov 1 07:26:17] WARNING[18544] res_musiconhold.c: Unable to open
pseudo channel for timing... Sound may be choppy.
[Nov 1 07:26:17]
Dear all
due to some custom requirements we are planning to use the manager
events for creating some custom reports.
I've enabled cdr_manager, then in manager.conf i've enabled
timestampevents = yes and in queue.conf eventmemberstatus = yes.
I know that these settings can generate a lot of
Hi,
I get confused about all solutions for Skype!
I want to connect as simple as possible out home * to be able to at
least answer Skype calls.
Now I use a PC USB box and a FXO, works ok both call directions but uses
a PC.
Any good and free idea ?
Thank you!
HB
What version of the IAXy are you running the ones that I have do not
have a web interface and require IAXprov to provision?
= -Original Message-
= From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
= boun...@lists.digium.com] On Behalf Of Joseph
= Sent: Saturday,
Hey has anyone gotten Exchange 2007 working with Imap Storage? I know
that Exchange 2007 didn't support the usual delegated Imap access
until Rollup 4, but it should work now right? I have it working fine
with Exchange 2003. When I do DomainName\Username\mailboxalias, I get
In 1.4.26, I'm experiencing a case where a manager- (or CLI-) originated
call where the channel is Local and the extension primarily runs an app
(e.g. Playback) immediately gets up. The extension may be rung but is
cancelled.
Anybody seen this? Know what to do about it?
On Sun, 1 Nov 2009, nik600 wrote:
due to some custom requirements we are planning to use the manager
events for creating some custom reports.
So the call-flow in the events listener will be:
1) new event detected
2) check if the event has an Uniqueid information
3) push the event into a
You need to have the dadhi_dummy driver loaded, or have a digium (or
similar) card plugged in. Meetme needs a timing source. dahdi_dummy is
used as the timing source in case you dont have a card.
sam!!
Phibee Network Operation Center wrote:
Hi
when i use MeetMe, i have this errors:
Dear all,
I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
trouble, I see on XLITE console:
Registration Error: 503 - Service unavailable.
Someone have a tutorial or a step by step description how to do that ?
Thanks in advance
--
Giancarlo Lombardo
How to tell Dial to execute Macro AFTER connecting to the channel?
Dial macro definition:
Execute the specified macro for the called channel before connecting to the
calling channel.
and I want to execute the macro after connecting (channels answers).
--
Joseph
Dear Giancarlo,
On which OS your are installing XLITE. If you are trying to connect XLITE
using Winodws XP please make a entry in your firewall. I think that would
solve your problem
On Sun, Nov 1, 2009 at 10:27 AM, giancarlo lombardo gianclomba...@gmail.com
wrote:
Dear all,
I'm trying to
I am having the same issue.
Please assist.
On Sun, Nov 1, 2009 at 1:27 PM, giancarlo lombardo
gianclomba...@gmail.comwrote:
Dear all,
I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
trouble, I see on XLITE console:
Registration Error: 503 - Service unavailable.
I
have a setup of Asterisk 1.2.28 and Digium AEX808B (8 FXOs), I connected the
PSTN lines to the Digium card, everything is working fine but the issue is that
when I make calls through the PSTN lines, some of calls get out successfully and
the other give me a different dummy long ring back tone
I have two DID numbers.
I want callers who dial 1 703 to get placed in a specific part of
IVR
I want other callers who dial 1 567 to get placed in a different
area.
How do I do this please?
___
-- Bandwidth and Colocation Provided by
Hi. When I dial a Dahdi extension using asterisk 1.6.0, and there is no
answer, the extension hangs up, but the dial status is busy instead of
no answer. How do I get this to work -- do I need to update dahdi? The
card is an X400p using its FXS module.
Thanks in advance for any ideas on this.
I can not seem to get dial status to work,
in sip.conf I have: qualify=yes
simple plan:
exten = 51,1,Dial(SIP/11,20,r)
exten = 51,n,Goto(s-${DIALSTATUS},1)
exten = s-Busy,1,Hangup()
exten = s-Answer,1,Macro(atb)
I'm dialing from exten.11 to exten.11 so I get busy signal and the channel
should
Hi,
it's quite straightforward: you can do your dialplan like this (default is the
default context answered when inbound calls happen) - remember the underscores!
-
[default]
exten = _1703,1,Goto(place-IVR,s,1)
exten = _1567 ,1,Goto(place-other,s,1)
[place-IVR]
exten =
Thank you.
I am trying it shortly. This is a lot of fun. I am trying to find
places where I can get customers with IVR or anything relating to
Asterisk. Any ideas?
Cheers
Tom
On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote:
Hi,
it's quite straightforward: you can do your dialplan
Where is everyone located? I am in Virginia, USA
On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote:
Hi,
it's quite straightforward: you can do your dialplan like this (default is
the default context answered when inbound calls happen) - remember the
underscores! -
[default]
Does anyone know of a low price SIP termination service to Nepal? For
VoIP calling card solutIon
On 11/1/09, Aggio Alberto alberto.ag...@loquendo.com wrote:
Hi,
it's quite straightforward: you can do your dialplan like this (default is
the default context answered when inbound calls happen) -
On 11/01/09 11:04, Alexander Lopez wrote:
What version of the IAXy are you running the ones that I have do not
have a web interface and require IAXprov to provision?
[snip]
The one I have is using standard Linux command line for provisioning but one
time I run onto a web-page that offered iaxy
Is this going to work:
[default]
include = stdexten
include = big10-IVR
include = cleveland-IVR
exten = _17035745353,1,Goto(big10-IVR,s,1)
exten = _15672528431,1,Goto(cleveland-IVR,s,1)
[big10-IVR]
exten = s,1,Answer()
exten = s,n,Background(dir-welcome)
;exten = s,n,WaitExten(1)
;exten =
Joseph wrote:
On 10/30/09 12:55, Vincent wrote:
Hello
Since SIP/RTP is a pain to use with road warriors who need to connect
from any location over the Internet, I'd like to get them some IAX
phones instead.
For those of you using this protocol instead of SIP, what would you
I want to match specific contexts to menus.
If users dial a number (example: 1703444) then start with context
big10-IVR
If users dial a number (example: 1567444) then start with context
cleveland-IVR
It is not working. I have played with the include statements and am close
but no cigar.
Try removing the include statements from the default context and see
what happens. Also double check to make sure calls are sent to the
default context.
Peter
On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
I want to match specific contexts to menus.
If users dial a number (example:
How do I check
On 11/1/09, Peter peter.johans...@omnitor.se wrote:
Try removing the include statements from the default context and see
what happens. Also double check to make sure calls are sent to the
default context.
Peter
On Nov 2, 2009, at 3:40 AM, Thomas Person wrote:
I want to
Check the channel driver configuration file, or fire up CLI with max
verbosity and monitor its output while calling the dialplan
extensions. CLI is like a good friend that tells you whats going on
and if there are any errors in you configuration.
Peter
On Nov 2, 2009, at 4:39 AM, Thomas
ok
thanks
On Sun, Nov 1, 2009 at 11:16 PM, Peter peter.johans...@omnitor.se wrote:
Check the channel driver configuration file, or fire up CLI with max
verbosity and monitor its output while calling the dialplan
extensions. CLI is like a good friend that tells you whats going on
and if
As I see here, you do not have to include the big10 context inside the
default context, as you have an extension defined to reach that context
and its extention is start extension.
If the cleveland-IVR is based on the start extension too, the same
applies.
Besides that, it would
On 11/01/09 13:01, John Novack wrote:
[snip]
Isn't the IAXy a discontinued product?
The web page is a 3rd party solution to provision the IAXy, if my memory
isn't completely shot, and isn't built into the IAXy
John Novack
It looks like it is discontinued. When did they discontinued it? It
Hi Juan,
I have this:
[default]
;include = stdexten
include = big10-IVR
include = cleveland-IVR
exten = _1703XXX,1,Goto(big10-IVR,s,1)
exten = _1567XXX,1,Goto(cleveland-IVR,s,1)
You recommend I have this:
[default]
exten = _1703XXX,1,Goto(big10-IVR,s,1)
exten =
Try running your asterisk service with the -vvvc option or connect to it
via the -r option, and then try making a call that would cause it to
land in the default context, you will see the way asterisk traverses the
dial plan, this will give you good debug info.
sam!!
Thomas Perron wrote:
Hi
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and another is only
have local IP address
now on public IP there are some 7 DID forwarded , now i want to forward 3
DID out of 7 DID to
local machine we called server B , I know there are DIal , and Switch
statement in
Dhaval,
Why is your name capitalised?
DHAVAL INDRODIYA wrote:
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and another is
only have local IP address
now on public IP there are some 7 DID forwarded , now i want to forward
3 DID out of 7 DID to
local machine
Ok, now pretty much everything is up 'n running, however when I try to send an
ANSWER (or any) command to *, it replies with
org.asteriskjava.manager.response.ManagerError Permission Denied. In
manager.conf for the *-java client, I have
read =
is this answer of my question?
i cant understand?
regards
Dhaval
On Mon, Nov 2, 2009 at 12:25 PM, Alex Balashov abalas...@evaristesys.comwrote:
Dhaval,
Why is your name capitalised?
DHAVAL INDRODIYA wrote:
hello all,
i have 2 asterisk boxes on that 1 have public IP Address and
In a manner of speaking.
DHAVAL INDRODIYA wrote:
is this answer of my question?
i cant understand?
regards
Dhaval
On Mon, Nov 2, 2009 at 12:25 PM, Alex Balashov
abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote:
Dhaval,
Why is your name capitalised?
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