Displaying 20 results from an estimated 10000 matches similar to: "Advice on AMI and SIP (was: SIP)"
2008 Jan 18
0
SIP
Hello,
For each incoming or outgoing call, sip hardphones I'm using, turn BLF on
and off like this:
the first call (after leaving idle status) turns 1st BLF on,
the second one turns 2nd BLF and so on,
when a call is hanged, its BLF is turn off.
My first question is : do you think such behaviour is general ?
My 2nd question is : using AMI, how can I tell for a given extension :
1. the number
2008 Dec 22
1
AMI and ExtensionState command returning bogus 'status' number
Hello List,
I have been working on a PHP application in order to build a BLF style
script.
Until now everything is going Ok but something a little (in my oppinion)
strange is going on with the 'ExtensionState' command;
The problem is that it does not returns the 'Status' as it's suposed to,
mentioned in the A.T.F.O.T book - version 2.,
where it sais something like:
2008 Aug 01
1
Comparing origination from CLI and from AMI
Hi,
Using FOP, I've met a situation which makes me ask this simple question :
Are both A and B commands bellow equivalent ?
A. CLI:
originate SIP/9122 application dial Local/9123 at local
B. AMI/FOP:
192.168.64.5 -> Action: Originate
192.168.64.5 -> Channel: SIP/9122
192.168.64.5 -> Async: True
192.168.64.5 -> Callerid: 9122 Guest2 <9122>
192.168.64.5 -> Exten: 9123
2009 Jul 20
0
No subject
<snip>
Replaces: pickup-9582-c0a80101-d-4 at 192.168.101.102
<snip>
This Replaces header refers to RFC3891 which is not yet supported in
Asterisk (see http://www.voip-info.org/wiki/view/Asterisk+SLA)
This INVITE fails with :
<snip>
chan_sip.c: Trying to pick up 7792 at subs
<snip>
app_directed_pickup.c: No target channel found for 7792.
If I'm dialing *87792 instead
2009 May 19
5
OT: SIP hardphone with multi-color BLF
Hi,
Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors
(or more) ?
This could be very useful to support extended presence, for instance.
Regards
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2005 Feb 02
0
ExtensionState problems using Manager.conf API
This is my first attempt to write software of any sort. What I am
trying to is to use a .php page to query asterisk Manager and get the
ExtensionState for each particular extension. Then when it has the
answer it outputs an XML file for use as the directory on a Cisco 7960
phone. What I am thinking is that when the user hits the directory
button to veiw the directory that is at this URL it will
2008 Jun 28
0
AMI extenstion state
Hi,
I would like to get the status of asterisk extension with my php program.
*My program as follows,*
<html>
<!--<meta http-equiv="refresh" content="1" />-->
<?php
$fp = fsockopen("xxx.xxx.xxx.xxx", 5038, $errno, $errstr, 30);
if (!$fp)
{
echo "$errstr ($errno)<br />\n";
}
else
{
$out = "Action: Login\r\n";
$out
2007 Mar 20
2
Which parameters of a live Asterisk server would you monitor ?
Hi,
Let's say you have an Asterisk server running.
Which parameters would you check to improve service continuity ?
I was thinking of :
- telco lines status (make sure every is up)
- registered hardphones
- config files backup (compare live and saved configuration files, if files
differ, notifies the administration team)
- systems variables (disk and CPU)
- log files (trigger an alarm for
2007 Jan 02
2
802.1x support in wired sip hardphones ?
Hi,
Is anyone aware of a wired sip hardphone supporting 802.1x authentication ?
I've been told some Avaya and Alcatel ip phones supported 802.1x.
As 802.1x is widely used with wireless hardphones, I'm wondering whether or
not, 802.1x could also be valuable for wired environments.
Regards
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2011 Jun 10
2
AMI question
Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X
Thanks,
jerry
2009 Dec 20
1
What changed in Directed PickUp between 1.6.1 and 1.6.2 ?
Hi,
I'm banging my head over this.
Usually, I'm using a SIP hardphone feature called "Call Pickup Starcode" to
enhance BLF with Directed Call Pickup :
basically, SIP hardphone (here a Thomson ST2030S) is configured to send an
INVITE message whenever a BLF is pressed while blinking.
The INVITE is build with the extension number (attached to the BLF that was
blinking and pressed)
2009 Apr 14
0
Gxp 2000 softkey question
I have a function *1 that starts and stops recording in a call. I use a
function so I can use MixMonitor.
It works well, however I would like to make it a little more integrated for
my users. We have GXP 200 hardphones. So far I've been able to configure a
softkey using the speeddial option to dial *1 during a call. I also have
setup another key to monitor the status of recording (on/off)
2008 Oct 04
0
2 stage dialing and 484 address incomplete [SOLVED]
Replying to myself, I've just read in 1.6.1 announcement that a new
Incomplete dialplan application is the one that provides what I'm looking
for ...
2008/10/3 Olivier <oza-4h07 at myamail.com>
> Hi,
>
> If my memory serves me right, there was thread (in dev mailing list ?)
> explaining how we could implement 2 stages dialing with SIP endpoints:
> user dials 1234
2007 Jun 12
4
Gigabit SIP Phones
Hello,
Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone.
Did I miss something ?
Regards
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2017 Nov 23
2
How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?
Hi,
1. How do you then, synced then unread message presence with custom device
status ? From an external program ? When a user leaves VoiceMailMan
application ? Using externnotify ?
2. What is MWI:101 at default expression for (see [2] ?
Cheers
[2]
https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box
2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at
2017 Nov 20
2
How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?
Hello,
I'm trying to supervise an existing Voicemail box with a BLF button on
Debian's asterisk 13.14.1 system.
I mostly found this [1] document.
I added in a context a line like:
exten = *7000,hint,MWI:31 at default
With "core show hints", I can read this:
*7000 at subs : MWI:31 at default State:Unavailable
Presence:not_set Watchers 1
My questions
2008 Feb 05
6
External MWI question for Asterisk
Hey there. I've been working on a project to integrate Asterisk with
Exchange Unified Messaging via sipX using large parts borrowed from:
http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html
... and everything works surprisingly well. The one problem I have is MWI,
or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so
I've been looking into
2014 Jan 17
1
Which multi-platform client-side Python library implements AMI ?
Hi,
Though ARI exists these days, is there any Python library implementing AMI
that fulfil the following requirements :
- client-side (may be installed on bax in which Asterisk is not running)
- multi-platform (Linux, Windows)
- take care of filtering various events to simplify operations such as
login/logout, origination,
- include some examples.
Regards
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2007 Mar 26
9
Multi-registration ?
Hello,
1. Is it possible to install several SIP softphones on the same PC, have
them registered to the same Asterisk server and attribute to each softphone
a specific extension, ringtones or call forwarding rules ?
2. Is possible to do the same with SIP hardphones ?
Regards
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2007 Apr 01
5
Best Hardphone (Subjective?)
After working with the Grandstream GXP 2000 series phones, I have
decided that I am quite unhappy with their problems, both voice quality,
volume, features and others. For their price now, there are plenty of
phones to choose from as well.
So subjectively what would be the best Hardphone for a small/medium
business with multiple line support, BLF, etc.
Are the Cisco 7960 the best of the