Displaying 20 results from an estimated 11000 matches similar to: "Push central phone book to phones"
2011 May 12
8
Light indicator managed by Asterisk
Hello,
is there some way to make Asterisk light up a certain light on an IP-phone ?
Like MWI, the message waiting indicator can light up if there is voicemail.
Could this light, or even other lights (like BLF-buttons) be used to
give a visual notification to the user ?
For example : if a certain value is set in the Mysql-DB and Asterisk
reads out this value, can Asterisk react upon it inside
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
2011 Feb 18
2
DTMF and Snom
Hello list,
I'm having some troubles with DTMF tones. When pressing numbers on a
Snom phone, the DTMF-signal takes too long.
I have the following in sip.conf :
dtmfmode = rfc2833
which works well for Grandstream, Yealink and Cisco phones. But not for
Snom.
Snom support tells me I should use SIP info.
Is it possible to have something like this :
dtmfmode = rfc2833, info
??
Because
2013 Feb 04
1
CallerID external call after Attended Transfer
Hello,
using Asterisk 1.8.12.2
case :
I call with my cellphone to our public telephone number
Our receptionist answers the incoming call and does an attended transfer
to my colleague ( A )
Colleague answers and the receptionist tells him that I am on the other
side.
Receptionist transfers the call and I am connected to my colleague ( B )
My question is about the CallerID that the
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 Oct 28
3
SIP client floods port 5060 and gets blocked
Hello,
Is there any reason why an IP-phone would pounder on port 5060 ? My
firewall blocks the public IP because it thinks the remote IP is port
scanning on port 5060.
I think the phone is just registering but for some reason it does this
repeatedly in a very short time.
Oct 28 09:01:48 astserver kernel: Firewall: *UDP_IN Blocked* IN=eth0
OUT= MAC=00:00:00:00:00:00:00:00:00:00:00:00:00:00
2011 Apr 13
11
Realtime SIP & peer status
Hello,
I'm using SIP realtime with MySQL DB.
Is it possible to get the status of the SIP peer (free / calling) from
this realtime DB ?
If not, is there another way to obtain the call state of a SIP peer ?
Kind regards,
Jonas.
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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
2010 Apr 27
2
Record call without caller interference
Hello list,
can a conversation be recorded without the caller or callee having to
press some combination that is defined in features.conf ??
Like in queues.conf you have the ability to record a conversation with
MixMonitor when the caller is connected to an agent/member of the queue.
Can this auto-recording also be implied on normal Dial(something) ?? So
that when the call is picked up (and
2012 May 08
4
Asterisk 1.8 Transfer CallerID
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B sees the number of colleague A on his
screen while talking to the external calling number.
I expect here that colleague B would see the external calling number on
the screen
2013 Oct 16
1
Use Asterisk Realtime Extensions with Switch-statement and include-statement
Hello,
Is it possible to use the switch => statement in extensions.conf
(http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to
point to a database and in the database use the include-statement ?
In extconfig.conf I would have :
extensions => mysql,asterisk,extensions_table
In extensions.conf I would then have :
[includecontext]
switch => Realtime/includecontext at
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 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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2010 Jul 06
2
ARA : Realtime or not ?
Hello list,
what is the use of realtime SIP peers when you always need to reload the
sip configuration as if you were just putting your SIP peers in sip.conf ??
My SIP peers are now defined in a mysql-DB and when I add a mailbox in
the field 'mailbox', the change is not active untill a do a "sip reload"
or a "module reload chan_sip.so".
Doing a "sip
2014 Nov 04
1
queue log realtime mysql
Hello,
I have 5 Asterisk servers all using mysql realtime to store queue log
information.
There is 1 out of 5 servers which stores the data in 4 columns : 'data1'
--> 'data 5'.
All other servers store data in 1 column 'data' with the data seperated
by pipe.
I see no difference in my configuration of extconfig.conf and
logger.conf. Maybe a hidden default value ?
2010 Aug 28
4
Asterisk does not translate from wav to alaw
Hello list,
I have a file to be played in wav-format.
I thought Asterisk would automatically take the wav-file and translate
it to the codec used, but I see this :
[Aug 28 11:16:29] WARNING[2705]: file.c:664 ast_openstream_full: File
/var/lib/asterisk/sounds/vprompts/*zip-code.wav* does not exist in any
format
[Aug 28 11:16:29] WARNING[2705]: file.c:991 ast_streamfile: Unable to
open
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 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
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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