Displaying 20 results from an estimated 20000 matches similar to: "Manage users & administrator for the asterisk-gui"
2009 May 22
1
VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
Don't be afraid about the info that I'm going to post in this mail, but
I want you to give as much info as possible. Also I want to show you
what I've tried.
What do I want
When a voicemail-message is left via the Voicemail()-application, I want
the .wav-file send to my mail-address as an attachment.
My mail-setup
I'm not using sendmail as MTA. I have msmtp as MTA and mutt as
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2009 Sep 07
2
features.conf : feature map ==> getting feature to work
Hi there,
I need some help with a 'custom' feature.
I have following feature defined in features.conf :
[applicationmap]
opnemencallee =>
#3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m
In my dialplan :
[from-HostAst]
exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee)
exten => s,n,Dial(SIP/grandstream,30)
I want the callee to be able to press #3 to be able
2009 May 08
2
Not receiving voicemail message in mailbox
It should be as simple as editing voicemail.conf :
; Voicemail Configuration
;
[general]
; Formats for writing Voicemail. Note that when using IMAP storage for
; voicemail, only the first format specified will be used.
format=wav49|wav|gsm
; Who the e-mail notification should appear to come from
serveremail=asterisk-voicemail
; Should the email contain the voicemail as an attachment
attach=yes
;
2009 Aug 29
1
GoToIfTime : how to define sep 25th till oct 10th ?
Hi list,
quick question :
With GoToIfTime, how to define a period of holiday that starts at the
end of the month and ends at the beginning of the next month ??
Like September 25th till October 10th when incoming calls need to go to
the voicemail...
Greetingz,
Jonas.
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2010 Mar 01
2
Is answer() necessary ?
Hello list,
is it necessary to properly answer() an incoming call ?
I don't want to answer a call because the caller has to pay even if the
attached SIP-phones do not answer the phone call. Because I answer() the
incoming call, the caller has to pay for 60 seconds of 'ringtone'.
On the other hand, sometimes an incoming call is send to a macro where
the caller is given the
2009 Oct 21
3
Searching on how to keep local calls... local
Hi list.
Does anyone know how to keep calls between 2 local SIP-phones on the
local private network when the 2 local IP-phones are registered to an
online public Asterisk-server ??
What network-element / router do I need to install to prevent the
RTP-traffic from flowing via the internet ?
Config :
Asterisk --internet-- > router/firewall --> connected local IP-phones
Internal call :
2009 Oct 14
2
FXS to SIP gateway
Hello list !
I don't have the money to test out all the products and reading the
manuals is not always that enlightening...
Does someone here know a good gateway-product that lets analogue
telephones communicate with an Asterisk-server.
I have found the Grandstream GXW-400x to be able to add SIP-accounts to
analogue telephone devices that are connected to the FXS-ports. Moreover
this
2009 Aug 31
0
asterisk-users Digest, Vol 61, Issue 85
Topic 6: RE:unable to execute command
hi there
i tried to execute the command as suggest like
exten => 1987,1,Playback(posix-restarting)
exten => 1987,2,wait(1)
exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py)
exten=> 1987,4,Hangup
it still doesn't work,now it does it give unable to execute command but
it doesn't reach the system command it just
2010 Mar 01
1
rtcachefriends & qualify
[Mar 1 14:54:07] WARNING[15290]: chan_sip.c:17669 build_peer: Qualify
is incompatible with dynamic uncached realtime. Please either turn
rtcachefriends on or turn qualify off on peer 'gerrie'
Am I correct that when I turn on rtcachefriends in sip.conf,
database-changes in my MySQL-DB will not be reflected untill a reload ??
Am I correct that when I turn off qualify in my realtime
2009 Aug 18
2
You do not appear to have the sources for the 2.6.20-prep kernel installed
I want to install Dahdi and Dahdi-tools on a CentOS 5.3 Xen host and I
receive the following error :
"You do not appear to have the sources for the 2.6.20-prep kernel
installed."
I have installed :
- kernel-headers-2.6.18-128.4.1.el5.x86_64
- kernel-devel-2.6.18-128.4.1.el5.x86_64
- kernel-xen-devel-2.6.18-128.4.1.el5.x86_64
bash-3.2# uname -r
2.6.20-prep
bash-3.2# ls -l
2009 Aug 30
2
MySQL syntax error : I really don't see where...
Hi list,
I'm stuck for the moment @ the following :
My Query (in a macro) :
exten => s,n,MYSQL(Query resultid ${connid} SELECT\ vakantie_set\
vakantie_data1\ vakantie_data2\ FROM\ AstDB\ where\
SIPACCOUNT="${ARG1}")
Asterisk CLI :
[Aug 30 14:07:42] -- Executing [s at macro-vakantie:2]
MYSQL("IAX2/zoiper-9238", "Query resultid 1 SELECT vakantie_set
2009 Jun 24
3
GUI for Asterisk
I wonder if there is a GUI that does not change the underlying hand-made
configuration ?!
What I'm looking for actually is a GUI for adding a new SIP-client +
voicemail, so that a company does not have to call me when they hired a
new employee.
I don't want a GUI that over-writes my hand-made SIP-configuration, and
my hand-made dialplan.
Jonas.
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2009 May 12
2
Hangup()-command does not hang up the line
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the following :
- Asterisk tells me that the conversation was hung up (the log files
tell me the command was executed)
- On my cell phone I hear
2009 Apr 26
1
file.c:655 ast_openstream_full: File /tmp/winkel-gesloten.alaw does not exist in any format
part of extensions.conf:
exten => 11,1,Answer()
exten => 11,n,NoOp(CallerID : ${CALLERID(all)})
exten => 11,n,Playback(/tmp/welkom-tcs.alaw)
exten => 11,n,GoToIfTime(09:00-17:59|mon-fri|*|*?open,s,1)
; wordt doorgerouteerd naar context open, maar indien gesloten :
exten => 11,n,NoOp(Oproep tijdens winkel gesloten)
exten => 11,n,Playback(/tmp/winkel-gesloten.alaw)
exten =>
2016 Aug 12
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-16 09:25, ????? ?????? wrote:
>
> Try delete nat from 770000wrtc settings ice should do the same
>
>
> On Aug 11, 2016 10:00 PM, "Jonas
2016 Aug 17
4
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 17:45, George Joseph wrote:
>
>
> On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens
> <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote:
>
> On 16-08-16 04:38, George Joseph wrote:
>>
>>
>> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens
>> <jonas.kellens at telenet.be <mailto:jonas.kellens at
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2009 Jul 06
2
SIP registry fails during night
Every morning I check my SIP registry to the SIP-provider. And I must
conclude that during the night somewhere registry has failed.
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
85.119.188.3:5060 092779077 105 Failed Sun,
05 Jul 2009 23:11:40
asterisk*CLI> sip reload
[Jul 6 10:30:43]
2009 Sep 17
1
I'm not getting the ability to leave a voicemail-message
I'm having a little problem with voicemail. Actually I'm not getting the
ability to leave a voicemail-message.
This is part of the dialplan :
> exten => s,n(voicemail),PlayBack(/var/lib/asterisk/sounds/voicemail/${ARG1})
> exten => s,n,NoOp(${ARG1}@boxes)
> exten => s,n,Voicemail(${ARG1}@boxes)
> exten => s,n,Hangup()
> exten => s,n,MacroExit
This is the