Jakub Arkon Syrek wrote:
Hello, we have strange problem, till now everything was working fine,
there where no problems with dial and answer calls.
Yesterday our system crashed and we notice strange behavior.
What type of event caused the box to crash? Given the fact that you've
also
Olivier wrote:
Hi,
Though this is a bit off-topic on this list, I think this might
interest those looking to build Asterisk appliances out of mini-ITX
boards such as http://www.pcengines.ch/alix1c.htm.
If you're interested in the ALIX type boards, there is a reseller in
Australia that
Ken Williams wrote:
We're entertaining moving our intranet to Hughes satelite for our
remote locations. I'm curious if anyone with Asterisk servers has
used satellite, and if so, is the latency an issue. My understanding
is that you immediately introduce 250ms latency for travel time up
Tilghman Lesher wrote:
Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is incompatible with realtime, unless
rtcachefriends is turned on, b) automatically disallow qualify=yes if the
peer is realtime and caching is not turned on, or c)
VoIP Cyprus wrote:
Can you share with me your experiences with Asterisk 1.6? Is it stable
enough for commercial service?
No. No matter how good some people may tell you it is, 1.6 is still
beta software and software is rarely beta for no good reason. Don't
even THINK about running 1.6
Olivier wrote:
Hi,
I'm receiving this :
[Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for
peer without mailbox: 9163
I've read this :
http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html
I typed this:
asterisk -rx reload
asterisk -rx voicemail show
Olivier wrote:
Now that root cause is found, would you say that warnings or CLI
should have been different ?
Obviously, MWI subscriptions must come from SIP hardphones (at least
those supporting MWI feature).
So in this case, Received SIP subscribe for peer without mailbox:
9163 rather
Joseph wrote:
I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage
but I think this version has a problem with RFC2833 DTMF signaling and I
don't think there
will be any newer version available anytime soon on portage.
I need stable version, I'm using Asterisk
Josiah Bryan wrote:
The script design supports plugin formatting as it stands. E.g. I can
insert any formatting algorithm if anyone has any suggestions. Right
now, the formatter script just does:
#!/usr/bin/perl
use strict;
my $file = $ARGV[0];
print ~pp~\n;
print `cat $file`;
print
Giorgio Incantalupo wrote:
Hi,
why do not you simply delete them from zapata.conf and restart your PBX?
Because that simply doesn't acheive what he's wanting to achieve. On
PRI circuits you can dynamically enable and disable circuits at the
data-link level. Whether this can be achieved with
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include
this kind of line :
To: sip [EMAIL PROTECTED]
Babcock, Michael Alex wrote:
windows smart phone v 6.0 example
htc shadow
is what i have. It has wifi abilitys.
Googling for windows mobile sip yeilds a multitude of results. I'm
sure one of them will point you in the right direction.
___
--
Tilghman Lesher wrote:
Can someone suggest the best way to deal with this without resoring to a
highly repetitive/iterative dialplan?
Leif and I discussed something like this at Astricon 2008, and we came up with
this patch:
http://bugs.digium.com/view.php?id=13632
Nice! For those of
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dialplan completely breaks).
I agree completely. As it stands, the REALTIME() function is nearly
completely useless. If Asterisk had better
Tilghman Lesher wrote:
On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote:
Wesley Haut wrote:
Yell at me if you will, but I hate func_realtime - it's not very
usable nor is it change-friendly (update your database and your
dialplan completely breaks).
I agree completely
Eric Chamberlain wrote:
I should have clarified, we're only making outbound calls, not
inbound, so there is no registration.
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality you require, even if you're
only making
Eric Chamberlain wrote:
Is there a particular reason you /can't/ register? It would seem that
registration would provide the functionality you require, even if
you're
only making outbound calls.
In the case of a server like Asterisk, wouldn't sending a register
disrupt the flow of
Tim Panton wrote:
Does anybody know any free WebCall solution to let our customer call
us directly via our web site?
Any clue will be welcomed.
Yep, take a look at our offering on www.phonefromhere.com
A per-minute charge does not constitute a free solution. Please read
requests
Kev Szaszvari wrote:
Hi there
Our company is using the Linksys SPA-942 Phones, and they are pretty
useless.
They dont have any central management or provisioning, as well as a
pretty bad interface.
This is completely incorrect. Linksys SPA-942s *do* have the ability
for central
Peter Galiovsky wrote:
Does anyone have any idea what should I try next?
Either contact Digium support directly or the people you bought the G729
license from. You're more likely to get the assistance you need in a
shorter period from these people than this list.
Emmanuel Pascal Bruno wrote:
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the
other party can hear me, but I cannot hear anything the
Hi guys,
I'm about to embark on a small (undoubtedly to get much larger) project
to write a set of scripts to handle provisioning of phones - Snom to
begin with, possibly with others (most likely Polycom and Linksys) to
follow later. Since I want this script to handle *all* aspects of phone
Emmanuel Pascal Bruno wrote:
I have turned off firewall on the linux box, I have turned off
firewall on the router I still have the same problem :-(
Disabling firewalls is almost certainly going to ensure the problem
persists. You need to ensure that all SIP and RTP ports are
port-forwarded
Hales wrote:
It should ignore the keywords, but you will get lots of errors in the CLI.
My guess is that if you put it all in a DB (and use realtime) you can
probably do whatever you want.
PaulH
Rob Hillis wrote:
Hi guys,
I'm about to embark on a small (undoubtedly to get much
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rob Hillis wrote:
Unfortunately RealTime isn't going to be an option - it's another level
of configuration I want to avoid, but more importantly since I'm
planning on being able to run these scripts on an Astlinux
Steve Anness wrote:
Good Day,
I have been tasked with fixing the time on our asterisk server. I am
having a hard time finding documentation to tell my what asterisk uses
to get its time information to push to phones (or a better question,
where does the SPA-962 get its time information)?
While I can't speak for the Linksys SPA-921, I /can/ comment on the
Grandstream GXP-2000.
We're running half a dozen of these at the moment, primarily for
testing. I can confirm that the LCD display /does/ display both caller
name and number - assuming of course that both are presented.
I put such a request for enhancement in sometime, and as is seeming to
be frustratingly common for CounterPath, it was completely ignored.
Were it not for the Plantronics CS-50 headsets that we bought that have
support in a /very/ limited number of softphones, I'd be dumping EyeBeam
/and/
Oded Arbel wrote:
And you don't need physical access to the system to get to the one and
only real console. OTOH, if you do have physical access, you have full
control of Asterisk, as you may inject custom dialplan.
I wasn't aware that running asterisk -r on a physical tty has any
There are several possibilities, however the one that works across just
about every channel type is soft hangup channel
Yuan LIU wrote:
Can I disconnect an arbitrary call using a console command? I
remember reading something but can't find any more.
Uh maybe not... :) stop now will cause Asterisk to drop *all* calls
and exit immediately. Kinda the equivalent of nuking a small city to
kill one person.
Paul Hales wrote:
Stop now?
PaulH
On Sat, 2007-02-03 at 21:47 -0800, Yuan LIU wrote:
Can I disconnect an arbitrary call using a
Smartass... :)
Trixbox works off FreePBX which, while not as tightly integrated into
Asterisk, is currently far more mature and easy to use.
Note the use of the word currently. :) I wouldn't be too surprised
if FreePBX made the move to the Asteri/s/kNow framework. Removes the
big ugly
Try changing the shell for the asterisk user to /bin/false. This should
disallow anything passed through the ! command since it runs the command
via the shell for the asterisk user.
jeremij jerome wrote:
Hi,
I configured Asterisk to run as asterisk user, but I see that a user
can anyway
Hi James,
The only solution I've managed to find so far is to set the wrap-up time
to 5 seconds and tell the operators that if they need more time, they
need to put themselves on pause. See PauseQueueMember and
UnpauseQueueMember.
If someone has a better solution, I'd be most pleased to
I guess the obvious question would be whether the callingcard context
is included into the context that the call is coming from. That's the
usual reason for a failure like this.
[EMAIL PROTECTED] wrote:
I have followed all the install note for A2billing and have everything
installed and
Only when the chicken is provided with sufficient stimulation.
Salvatore Giudice wrote:
I think it's a small, feather covered appendage.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Jenkins
Sent: Monday, April 09, 2007 9:21 AM
To: Asterisk
In the case of our Sangoma card, the echo cancellation module
constituted approximately half the price of the card, so yes you should
find it considerably cheaper than the Digium card.
Just be aware of the extra fiddling around having to install the Sangoma
drivers in addition to the Zaptel
Louis-David Mitterrand wrote:
On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote:
Your monitoring app is not sending valid IAX2 packets to the server. If
it was sending a true IAX2 POKE, it would be a valid packet and wouldn't
generate this warning.
Could asterisk at
Tzafrir Cohen wrote:
On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote:
I'd take this warning seriously. It means that your monitoring app isn't
monitoring what you think it is.
I always want to know when I get malformed protocol packets in. It is
always bad news, mostly either
Tzafrir Cohen wrote:
On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote:
Maybe it's me, but I think that warning should be regarding a problem
I can fix. Malformed network content does not neceserily fall under that
definition. notice?
Absolutely it does. Warnings
Russell Bryant wrote:
On Nov 8, 2008, at 1:30 PM, Atis Lezdins wrote:
Asterisk offers very much the same flexibility. You can disable
specific log levels (for example warnings) in logger.conf or you can
log everything to syslog, where filter out this specific message.
Of course,
Alex Balashov wrote:
The solution for the problem of an IAX client is a SIP client.
That's not a particularly good solution if you have a NAT between your
client and Asterisk. IAX is still *much* easier to get working through
a firewall.
___
--
Carlos Chavez wrote:
I have a new customer that wants to upgrade their Asterisk installation
from 1.2.27 to 1.4.22. They use FreePBX for administration. Since
there are many syntax and command changes from those versions of
Asterisk, is there an easy way to convert the FreePBX
Philipp Kempgen wrote:
SendImage() in 1.4:
---cut---
SendImage(filename): Sends an image on a channel.
If the channel supports image transport but the image send
fails, the channel will be hung up. Otherwise, the dialplan
continues execution.
The option string may contain the following
David fire wrote:
hi
i need an open source callcenter manager system like queuemetrics but
opensource any one know any?
i prefer to search before start a new one
You'll be pushing to find something even close to QueueMetrics' quality
available in open source. The closest I'm aware of is
dubravko caric wrote:
Hi all,
I'm testing Linksys SPA922 phone and I have strange issue. when call
is finished on the phone I see CallEnded and normal silence for cca.
5 seconds and then I get fast busy for cca. 20 sec. So, this isn't
automatic hangup as on other phones I have tried
Michael wrote:
My experience with Grandstream is that are one of the better 'cheap' ones,
but
cheap non the less.
I am yet to run into a worse IP phone than the Grandstreams - although
having said that, I should say that I've always steered clear of most of
the Chinese no-name brand phones.
Michael wrote:
I bought it. The SPA962 went on ebay within 3 months of me buying it.
I have a few grandstream 286's I like to use for traveling and placing
in remote areas of an installation.
3 months... that long?
Again I'm surprised. I've had no problems at all with the Linksys
forums - sigma wrote:
having deployed a fair amount of phones I have the following observation
(and these observations are worth what you paid for them :-) )
1. Linksys 942, my preferred mainstream desk phone, a bit more expensive
than the Polycom IP330. Be careful as there are two SKUs
Michael wrote:
Change it to the following:
exten = _10,1,Dial(SIP/10,10)
exten =_10,n,Background(vm-nobodyavail)
exten = _11,1,Dial(SIP/11,5)
exten =_11,n,Background(vm-nobodyavail)
The only time I am aware of that you can leave out the prefix underscore is
for exten = s and exten = i
No,
Administrator TOOTAI wrote:
[MyPeer]
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142
permit=yyy.yyy.yyy.yyy/255.255.255.255
On incoming calls, when the peer address is the one terminating with
.139 everything is OK.
If I
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
Which line of code is generating this log entry?
[Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing
[91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack
...because this appears to be where your problem lies.
joek...@gmail.com wrote:
Hi all,
I have a connect between a siemens hipath
Fabio Mosti wrote:
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to
another asterisk (zap).
client-asterisk (Spandsp)-asterisk (zap)-fax
To quote the
Duncan Turnbull wrote:
Hi All
I am looking at a replacement for a hotel PBX which requires at least 60
analogue extensions.
I tend to use Sangoma equipment but haven't tried this many analogue
extensions before. I am interested in anyone's experience of which
server platform literally
Asterisk wrote:
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
I've never seen that before, does that definitely work in 1.4.x? If so,
cool...
About two and a half years ago, I upgraded a small call centre from
corded handsets to X-Lite with Plantronics CS60 USB headsets.
X-Lite lasted about two or three months before we ditched it in favour
of Eyebeam. X-Lite disables too many features to be useful. With the
Plantronics headset,
Yes. Grandstreams suck.
Oguzhan Kayhan wrote:
Hello,
I configured both asterisk and grandstream 2000 accourding to howtos on
the web..
And everything seems working fin.
But if i reload asterisk grandstream stops working with BLF.
I need to restart the phone to enable BLF again.
Any
Rilawich Ango wrote:
My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
That would be my bet. No, Asterisk can't do anything to remove EM
noise. That's up to
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing an FXS port integrated into a small footprint computer.
Paul Hales wrote:
I would love to see the agent login/logout stuff working - but that's
just me.
I'd like to see the damn web interface become usable.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Loic Didelot wrote:
Hi,
I am trying to connect a doorbell to a Xorcom device. And the setup is
quite simple. But when I push the doorbell all I see on the asterisk cli
is:
-- Starting simple switch on 'Zap/11-1'
[Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough
digits
Michael wrote:
On Fri, 03 Apr 2009 12:32:03 you wrote:
Like:
exten =
5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
0 5,20)
That is what I am currently doing - though is there a cleaner way?
The only cleaner way is to define the group in [globals]
Sebastian Milioto wrote:
2. What E1 card should I buy for Asterisk? Is the physical interface
(conectors) E1 identical as T1?
The connectors are identical, however the protocol isn't. However, just
about all the T1 cards I'm aware of support E1 as well - usually
selected by a jumper on the
Michael wrote:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label
PHONE NUMBER = the
Benny Amorsen wrote:
Michael mich...@networkstuff.co.nz writes:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER'
Kurian Thayil wrote:
On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation
Darrick Hartman wrote:
Rob Hillis wrote:
Daily? No. However, after implementing a weekly restart of Asterisk,
I've found the instance of lockups and CPU utilisation spikes have
decreased significantly.
Unless you're using some unstable modules, there really should be no
need
Darrick Hartman wrote:
If I were to do things again, I'd be running Astlinux on a net 5501 with
an integrated hard drive (for voicemail/IVR and so on) Only time I've
ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's
time to upgrade Astlinux.
That's what we like to
Sebastian wrote:
Anyone thought about something like outgoing queues?
Many people have. I know QueueMetrics has methods for this kind of
thing, and I'm fairly sure that Vicidial does as well.
___
-- Bandwidth and Colocation Provided by
carl Lougher wrote:
Ok cheers.
Any idea when 1.6 goes stable for prod?
Theoretically it already has, however as was the case with 1.4, I
suggest you tread very carefully when it comes to migrating to 1.6.
___
-- Bandwidth and Colocation Provided by
Louis-David Mitterrand wrote:
Hi,
Is anyone here using OrderlyStats with asterisk in a call center
setting? If so what what is your experience with it? Is that software
really free for asterisk users?
Or is there a better option out there?
The short answer is OrderlyStats isn't really free
Sounds like you're looking at the wrong variable. You should be looking
at CALLERID(num).
peace keeper wrote:
Hi there,
I am using the Asterisk as the PBX, and need to know the caller ID for
the incoming call,
but when I show the caller Id, it gives the Zaptel channel that
recieves the
The clue in the log is no authority found. Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which
Christian Stredicke wrote:
Check out the snom 300 or the snom 820...
Good lord... talk about two extremes... :) The Snom 300 is pretty good,
but the 320 is much better and costs around a *third* of what the Snom
820 does.
Stick with the older model snoms. So far I've seen nothing about
Alex Samad wrote:
I have been looking at a snom 300, which seems okay. the display goes a
bit haywire occasionally - not sure why yet.
Are the 320 worth the extra money ?
IMO yes, though it really depends on what you want from the phone.
The Snom 320s handle transfers considerably better
Jeff LaCoursiere wrote:
We are still talking about a $175 phone. How about the Polycom IP 320?
$85 at 888voipstore. Can't go wrong with Polycom for voice quality.
True, Polycom's are brilliant for voice quality, but unlike the Snom, a
Polycom /will/ reboot on the drop of a hat /and/ take
Cary Fitch wrote:
We have a bunch of SNOM 360’s we are not using. I agree they are not
intuitive to the user. They work ok in general. I would part with 15
or so at an attractive price, one or more,
I like the Grandstream 2000 series. Easy to use, easy to set up, good
web page.
Dan Pilcheck wrote:
The call will go over the server fine, but when the Call Center server
answer, the CLI returns:
NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt
from 10.0.10.20, request '2...@2xxx' does not exist
What context are the phones in the extension range
jonas kellens wrote:
Do you understand what is happening ?
I don't understand what this sentence means :
SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
Pretty simple really. Your SIP trunk 3starsnet is making progress with
the call and Asterisk is passing that
Bart Coninckx wrote:
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten =
Yes, although not for connecting to the PSTN - I've used one for
connecting to a legacy NEC PABX.
Voltage isn't the issue - the difference is in the impedance. Australia
uses complex impedance (220+820Ohm resistors with a 120nF capacitor)
whereas the US uses a straight resistor.
Alex Samad
Alex Samad wrote:
Voltage isn't the issue - the difference is in the impedance. Australia
I get this in my dmesg when I load up the rdm410 modules
[1083334.103487] Freed a Wildcard
[1083336.171371] ALAW override parameter detected. Device will be
operating in ALAW
[1083338.040522]
800ms is horrendous lag for a VoIP connection. If I were you, I'd be
investing some time in finding out why the lag is so great. Even if I
do a ping to a UK address, I'm getting pings of no more than 300ms from
Australia. Unless you've got multiple satellite connections in the path
(in which
Low bandwidth is another possibility, but I'd have though that any
connection slow enough to generate that much latency wouldn't be usable
for VoIP in the first place.
Ishfaq Malik wrote:
Cheers Rob, I was thinking it was due to a low bandwidth connection at
the other end but from what you're
Patrick wrote:
I believe this is called camp on. Found some examples on voip-info.org
but they assume that you do not hangup the originating phone. Anyone
have an idea how to implement this feature as described above?
It might be referred to different things in different countries. In
I'm currently starting development of an add-on to a program designed to
be used in a call-centre type environment that will interface very
closely with Asterisk - quite possibly to the point that the add-on
itself will be a softphone as well.
In order to test this application properly, I
Mojo Jojo wrote:
Anyone using one of these as a QOS device in an Asterisk environment?
If so, does it work well?
No, I don't use one of these myself. However...
Do you know what exactly it prioritizes? SIP only? IAX?
...during my recent DCE course, this product (or one extremely similar
to
Lan wrote:
Forgive me that I don't understand what you are really mean?
He means that this is the list for Asterisk Support - [EMAIL PROTECTED] is a
completely different product. (Asterisk is simply the PBX software,
where as [EMAIL PROTECTED] is a complete distro that just happens to include
Steve Underwood wrote:
We have 2 GXP-2000 dead during automatic
firmware upgrade. Devices now send out only one ARP packet for default
gateway resolution during boot and nothing more!
We've contact Grandstream support, but they cannot help. Now we want to
send devices to Grandstream
Trixbox isn't a GUI - it's a complete Linux distribution. The GUI to
Asterisk that comes with Trixbox is FreePBX (http://freepbx.org/)
Zeeshan Zakaria wrote:
Trixbox
www.trixbox.org
signature.asc
Description: OpenPGP digital signature
Hi guys!
I'm having one or two issues with queues hosted by an Asterisk machine
where the clients are on a legacy PABX - at least for the interim. I
fully expect most of these issues to be non-resolvable, but thought I'd
at least ask to find out if there is some way of working around the
issues.
Tzafrir Cohen wrote:
Note, however, that only one program can listen on the same port. So if
you want both to e.g., listen on IAX, one, at least, has to listen on a
custom port.
...or you need to run two IP addresses on the machine,
and configure each Asterisk installation to use the
Hi guys!
We have a call centre that has been moved across from an old Ericsson
MD110 PABX to an Asterisk server with those in the call centre using
X-Lite as their softphone.
I'm trying to get Agent presence configured so that X-Lite gives the
operators a visual indicator of their status -
:
You could put together a web page that talks to the Asterisk Manager.
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 26, 2006 11:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Agent presence
Hi guys
on the phone...
-Original Message-
From: Rob Hillis [mailto:[EMAIL PROTECTED]
Sent: Wednesday, December 27, 2006 8:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Agent presence
Not quite the solution I was looking for - I was wanting
Hi all,
I'm having a hell of a time trying to get a SIP trunk working between
Asterisk and a Boscom/Claro/IP Gear Robocom. (the company that owns the
things has changed names at least twice since we've had them!)
The units are the backbone of our legacy PABX network and with the
I take it you aren't using chan_agent to send calls to agents. You
might want to check voip-info.org for a few examples on how to implement
this - this should give you the desired results.
Felipe Neuwald wrote:
Hi Folks,
I implemented an Asterisk 1.2.10 on a Debian GNU/Linux, and I have
The only possible way I can think of achieving this would be to mangle
the incoming caller ID to include the extension that the call came
from. Given that Asterisk's voicemail boxes are separate to extensions,
I can't see another solution.
Benjamin Jacob wrote:
Hello All,
I am using ODBC
One of the biggest barriers to upgrading are the number of little
gotchas in syntax changes that can make an upgrade from 1.2 to 1.4
quite painful. After the pain I went through upgrading to 1.4, I've
always been recommending to people to think twice about upgrading if 1.2
does what they require.
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