Displaying 20 results from an estimated 10000 matches similar to: "GUI for Asterisk"
2010 Oct 26
11
Auto provisioning from public server
Hello,
has anyone experience with auto provisioning IP-phones on different
locations through a central public provisioning server ? You use http or
https ?
Is there a danger that one uses a different MAC-address in the
provisioning link to obtain SIP username / password settings ?
Kind regards,
Jonas.
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2010 Apr 24
2
Manager events & safety
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
is it save to send manager events from a remote website (php) to an
Asterisk-server ? Is
2009 Jun 23
1
SIP 482 Loop detected
-- Executing [0473775006 at intern:1] NoOp("SIP/twinkle-088e6ea8",
"conversation to GSM") in new stack
-- Executing [0473775006 at intern:2] Dial("SIP/twinkle-088e6ea8",
"SIP/3starsnet/0473775006") in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 "Loop Detected" back from 85.119.188.3
-- Now forwarding
2011 Jan 11
2
Show voicemail in GUI
Hello list,
I have a management user interface written in php for controlling some
functions of Asterisk PBX.
I use realtime a lot.
Is there a way to easily get the details of a voicemail account and the
messages that have been left ?
In use realtime voicemail, but how to get the messages that have been
left for a certain mailbox-extension ?
Kind regards,
Jonas.
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2010 Feb 25
2
Problems installing dahdi : kernel sources
Hello list,
when installing Dahdi, the following error comes up :
You do not appear to have the sources for the 2.6.18-164.11.1.el5xen kernel installed.
make[1]: *** [modules] Error 1
The running kernel version :
-bash-3.2# uname -a
Linux vds.hosting.net 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux
-bash-3.2# ls /usr/src/kernels/
2009 Oct 25
2
help sip show on CLI : no such command
What is wrong when I can not execute any command that starts with
sip ???
> freepbx*CLI> help sip show
> No such command 'sip show'.
> freepbx*CLI> help sip
> No such command 'sip'.
IAX works fine :
> freepbx*CLI> help iax
> iax2 provision Provision an IAX device
> iax2 prune realtime Prune a cached realtime lookup
>
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 Dec 27
2
Call ends when picked up
Hello list.
My phone rings, I pick up, and the conversation is terminated. Every
time.
The setup :
Grandstream GXP2010 --> SIPproxy (Endian Firewall) --> Asterisk Server
--> ITSP
Could it be the SIP proxy of my Endian firewall ??
I have 4 accounts on the Grandstream which listen on port 5060 --> 5063.
They have a proxy defined namely my Endian firewall.
On this SIPproxy I have a
2009 Jun 24
2
Asterisk + Jabber
I want to use JabberSend in my dialplan, but I saw that my Asterisk does
not support Jabber.
Also I have nowhere a module res_jabber.so...
So I thought I'd rebuild my Asterisk. In menuselect I saw that
res_jabber was dependent of 'iksemel' and 'gnutls'.
In my yum repositories I can find a gnutls.i386, but what is this
iksemel-beast ???
There is info to find via google on
2010 May 31
6
Voicemail : mail attachment to multiple mail-addresses
Hello list,
google returns a discussion on the dev-list when I search for how to
mail a voicemail to multiple mail addresses.
Is there yet a seperator that actually works to define multiple mail
addresses ?
Kind regards,
Jonas.
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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
2011 Mar 09
6
SIPAddHeader not working
Hello list,
I notice that the dialplan method SIPAddHeader is not working :
in dialplan :
/exten => s,n,SIPAddHeader(Privacy: id)/
in SIP invite no trace of this header :
/INVITE sip:0473 at sip.domain.be SIP/2.0
Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97
From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2
To: <sip:0473 at sip.domain.be>
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
2010 Oct 18
15
SIP DNS SRV
Hello list.
When using SIP DNS SRV to define a production Asterisk server with high
priority and a backup Asterisk server with a lower priority on this
DNS-server, will this work as follow :
- production server is reachable, so registration of the IP-phone goes
to this server
- production server is unreachable, so registration goes to the backup
Asterisk server
- production server is
2010 Sep 09
5
info about application not available asterisk 1.6.2.11
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI> core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind regards,
Jonas.
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2016 Aug 16
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
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 telenet.be>> wrote:
>
> Hello
>
> using pjproject 2.5.5
> using asterisk-certified-13.8-cert1
>
>
> IIRC there were API changes in pjproject 2.5 that aren't accounted for
> in
2010 Dec 02
5
Push central phone book to phones
Hello,
I have Snom, Cisco, Grandstream & YeaLink phones.
Is there a way to push a centralized phone book to these phones ??
Kind regards,
Jonas.
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2016 Aug 11
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
On 11-08-16 18:03, Matt Fredrickson wrote:
> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at telenet.be> wrote:
>> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
>> functionality as there are certain functions deprecated/replaced. This can
>> also cause headache :-)
>>
>> I will do so if there is no other option.
2012 Sep 28
1
Disconnect calls : known reasons
Hello,
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call ending, as if one of the connected parties hung up
but that is not the case !
So what could be a bottleneck ? Any known reasons for random hangup ?
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