Displaying 20 results from an estimated 6000 matches similar to: "Realtime call-limit"
2007 Mar 30
3
Multi-Level Queue
I am trying to setup a queue in a very specific way and I can't quite
figure it out. I'm sure someone else has done this.
I want calls to come into a queue and do a ringall on a number of phones
(let's say 3). So ring them for 20 seconds or so. If there is no
answer, I want it to ring a second set of phones for 20 seconds. If no
answer, then go back to the first set of phones.
2007 Apr 11
2
SIP INFO message
I've got a very strange problem and I can't figure it out. I have a
Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I
see callerID name, but it is not getting to * via SIP. I am running *
1.4.2 and the latest Cisco IOS for my router. Here is what is happening:
A call comes into the gateway. It sends a SIP INVITE to * with
"pending" as the callerID
2007 Mar 18
2
camp on off-line phone
When phone A registers, I want phone B to ring, when picked up, it should
call phone A and connect the phones.
Translated: When GF in Mexico powers up laptop where soft iax-phone
registers automatically, I want to talk to her asap :-)
How to?
Leif
2007 Nov 28
2
cvs or svn
Hi All;
Which is better (to have more stable or release
versions) of zaptel, libpri and asterisk: to use cvs
or svn?
In case of using cvs, why I need to type:
export
CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot
In other words: what is the use of pserver, anoncvs,
... with cvs checkout?
Note: How can I know all the variables needed for cvs
checkout so I might need to do
2007 Jul 01
2
the-asterisk-book.com online (unstable version)
Hi,
this is to inform everybody that the translation of my new book
(unstable version) is online at http://www.the-asterisk-book.com
The book is a GNU FDL project. So everybody who wants to participate
is welcome to do so. Also, everybody who needs material for his own
work, feel free to take it as long as the new material will become
GNU FDL too.
I am glad that Stephen Bosch (who you
2007 Dec 03
2
Hoteling
I'm sure this has been discussed many times, but I have a question about
hoteling.
My understanding would be this:
A phone sitting on a desk. A user hits 9000 and it asks what extension
you'd like to become. You type "1001" and then it asks for your
password. You type 1234, and it says you're "logged in". You now are
accepting calls at your phone and you're
2009 Jun 04
3
PHP/AGI/SetVar Issue
Is there a limitation to the number of variables you can set from a PHP agi
script? I have a simple example and I can't get it to let me set more than
1. I am pretty sure I am just missing something, but I've searched all over
an can't find the answer. Here is the extensions.conf part:
exten => _XXXXXXXXXX,1,AGI,diallocal.agi exten =>
_XXXXXXXXXX,n,NoOp(${ISLOCALCONTEXT})
2007 May 02
2
Large dial plans and variables
I have a large dial plan here with over 3000 lines, and several dozen
macros. As it grew, it became apparent that there was some problems.
1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc,
if that macro calls another macro, and passes arguments like this as
well, you lose the original values.
2. When the macro's 'return' some value, it has to set a channel
2007 Feb 26
2
Ex-Girlfriend syntax and RealTime Extensions
As seen in the following URL:
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I
also tested some time ago with an old release of Asterisk, RealTime
Extensions didn't support the Ex-Girlfriend syntax.
Is it already working in recent 1.4 or 1.2.15 releases?
Is there any other way that I can use to do the same thing but only
using contexts, for example? If yes, please
2007 Feb 21
1
Monitoring which users are online in realtime
Hi all,
Is there a way to keep track in Asterisk of which phones are online in
realtime using some MySQL DB table for exemple, much like "sip show
peers" does in the CLI?
Regards,
Ricardo.
2007 Mar 08
2
Hinting and Realtime
hello all,
My problem if i have my extensions and sipusers in a realtime database
it is not possible to use BLF or hinting.
i see only idle or unavailable status but if the phone is ringing or in
use i can't see it.
Is there a fix or any workaround? Version is Release 1.4.1
regards rene
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2007 Feb 21
2
How does Asterisk use SIP info command
What Asterisk command I can use to send a SIP INFO command? Thanks for
pointers.
Yuan Liu
2007 Oct 14
4
ResponseTimeOut() and t extension
Hi List;
Can someone advise me why in the below context, it
does not run the Background step? Once I dial 1000,
then it hangup and give congestion signal? If I
comment the ResponseTimeOut, then it run the
Background but it does not wait till caller enter the
digits, once the sound file finish, then it hangup
(congestion signal), also in all the situation, it
does not go for the t extension, why?
2009 Jan 27
2
Module res_odbc is not loading
Hi,
I have remove the comment defor res_odbc.so and res_config_odbc.so in my
modules.conf, but the module is still not loading
when I do:
module show like odbc
I have o module returned
anybody knows why?
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2007 Mar 07
2
queue information in mySQL
Hi,
is it possible to have the information stored in
/var/log/asterisk/queue_log
realtime in mySQL?
thanks
2009 Jan 07
5
recommendation for German sound files
Hi!
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon).
thanks
klaus
2007 Dec 19
5
Using * in extension name
I am trying to setup an extension of *7XXX that will allow me to dial
*7 and then any extension and use the Pickup application to pickup a
ringing phone. Ideally it will also check if the phone is ringing
somehow and then either dial the extension or pick it up if it is
ringing. But I can't get that far. If I use *7268 specially it works
fine, but as soon as I introduce any wild
2009 Jan 21
1
SIP realtime status...
Since 1.4.22 realtime status for sip peers seems to be broken. If I do
a "sip show peers" from the CLI I get this:
2001/2001 192.168.2.234 D 5060 UNKNOWN
Cached RT
It is arbitrary which peers will say OK and which will say UNKNOWN and
it changes over time. This is a problem with an application like the
Asternic Flash panel because it uses the peer
2007 Nov 27
4
Snom phones, blinking lights and call pickup
Hi!
I have the following questions/problems with * 1.4.
We have several Snom phones (320 and 360). Hints are configured in
extensions.conf (core show hints shows the correct values). My Snom phone
is registered to some numbers (validated by using sip show
subscriptions). I see the lights blinking if someone calls the subscribed
number and steady lights if the call is established.
So far, so
2007 Nov 14
1
"Whats New at Digium the Asterisk Company" -- Junk?
Is the "Whats New at Digium the Asterisk Company" message I got from
digium at en25.com really from Digium?
If so I suggest to send it from digium.com and not to use those
shady Eloqua redirect URLs.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk?