Hi Thorsten
Thanks very much, at this point my preference is rfc2833 but I will try some
other options.
The system is generating audible tones (that I can hear), although I think the
audio is generated by the last sip device in the network so if thats so I don't
have any control of it.
Freepbx really needs its own list but it doesn't seem to have one
But - if you have mysql setup and records being logged then the reports should
show you usage on a daily, weekly, monthly level. Make sure you built asterisk
with cdrs logged into mysql - its in the addons
Cheers Duncan
On
I think there is a new version of Outcall, the pop up was pretty good, but the
dialout wasn't ideal in Win 7 , and I believe thats fixed now with good
integration with 2007 and 2010
http://code.google.com/p/outcall/
You can buy commercial options from Biocom - who make Outcall
You often don't get cdrs or at least useful ones unless you run the call files
through a Local channel
You maybe already doing this
Can you check the Master.csv and see if it also is recorded incorrectly there.
Is this just an issue with mysql cdrs or something else. In my setups which use
The Lumenvox works fine in my limited use, easy to setup, good dictionary
options but it always depends on your circumstance.
http://www.lumenvox.com/partners/digium/Asterisk.aspx
Most of it is being really careful in planning the customer experience. The
technology is secondary to the
So in broad terms
You need to know when the queue is empty, and when there is voicemail (in a
generic queue mailbox presumably) and also that you haven't already delivered
the voicemail, and probably that when you deliver the mail its been
successfully been heard and actioned.
Are you also
You can include the label of the context in the custom area instead of
including a different context
i.e. [ext-queues](+)
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Not sure if it affects the order of processing or if that matters
Cheers Duncan
On
No its a split
FreePBX is still the same, V3 is still the same, this is a fork from some guys
who had got involved (or maybe paid some money)
Cheers Duncan
On 4/08/2010, at 2:56 AM, Tzafrir Cohen wrote:
On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote:
Have a look at the call files examples of voipinfo
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Its not too hard to do what you want
Cheers Duncan
On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote:
I want to use Asterisk as a general message delivery system here.
That is, I
Hi Frank
I have found Freepbx on top of Asterisk a good solution for the church I look
after and the rest of my customers, the callcentre functions you need are built
in it and if they have someone technical then they can expand what they are
doing
It has both queues and ring groups (which
On 1/03/2010, at 2:41 PM, Peter Serwe wrote:
I checked the firewall, iptables -L showed no rules whatsoever. No other
traffic has indicated it was blocked, iptables was set in allow all
everywhere mode.
I went ahead and turned it off, still don't have RTP. No audio either
direction
The other way on Debian/Ubuntu is just to test the existence of the dir and
create it if needed
If you add this to the /etc/init.d/asterisk near the start you should be fine
if ! [ -d /var/run/asterisk ] ; then
mkdir /var/run/asterisk
chown $AST_USER.$AST_GROUP /var/run/asterisk
Having looked at the outputs into PMS they are very simple stop start records.
Line by line text that can easily be recreated. They have about 4-5 fields,
origin number, destination, time of call, duration, or similar things
Usually they go out via a serial port or TCP port expecting a
Hi there
I have a client who has an AA50 from DIgium. I am really challenged getting any
support as the client doesn't have any of the original registration or
subscription info, someone did the install and left without any records. I
thought okay we can ask Digium, but you can't get help
AM, Kevin P. Fleming wrote:
Duncan Turnbull wrote:
I did get one response which was to email customer services and eventually
found an email address for them but that seems to have fallen on deaf ears.
Perhaps my expectations are too high but it was an email a week ago and no
response
Freepbx comes with setup of ring groups and queues with different hunt
strategies
Also it has Flash Operator Panel which gives you the state of the system
in real time graphical format
No money - just a small bit of installation time and learning how to use it
Cheers Duncan
Ken D'Ambrosio
Usually that message comes up because the caller is anonymous and
freepbx doesn't like anonymous calls by default.
There is an option to accept anonymous calls, or set the incoming trunk
to accept calls from the specific IP address
Of course it could be something else
Cheers Duncan
Ben
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a
custom module to add auto configuring endpoints for linksys (but i cna't
find it again right now)
Trixbox had too much stuff whereas the source install of just what you
want is nice and clean
Cheers Duncan
Jeff
Generally with FreePBX the ring options are set in the General Options -
you can set the Dial options which are normally tr, but I guess that
isn't working for you.
The SIP files you could edit would have custom in their name, otherwise
your changes will be overwritten when you reload freepbx
I am using the beta and its pretty good for remote access for clients
It would help if they had some discount structure for volume
Cheers Duncan
Pascal Bruno wrote:
Not sure if anybody noticed, but it seems like Skype For Asterisk is out.
$66 per channels, pretty pricey
If you use a Local channel to dial it then it will fall under the same rules
Channel: Local/numbertod...@the-context-you-want
This gets a CDR produced, it does pay to check everything works the same
but it should be fine
Cheers Duncan
David Gibbons wrote:
Context: is what the call is dumped
If you create a peer definition and put the host address in it and the
context you want it to go to you should be fine
Cheers Duncan
David Klaverstyn wrote:
Hi All,
I never saw a reply to this question. Is anyone able to assist?
Regards
David.
*From:*
Trixbox I think uses FreePBX
FreePbx has an option for each extension to set it to record all calls.
It will record the extension in the file name and you can view it
through the recordings app if you want a web view.
There are all stored in a common dir /var/spool/asterisk/monitor - you
can
For Linux use tcpdump on the host you are after
tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0
where 5060 is your SIP port and 1-16000 are your rtp ranges
-s0 means snap length of 0 so capture all the packet rather than cutting
off at a point
And refine it by adding the
Yip the VoiceBlue SIP units are very good but a bit pricey
Gordon Henderson wrote:
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters connected to
a VoIP
Usually this is a routing error with openvpn setup and asterisk thinking
it needs to route someway other than the vpn. If the originating packets
have an external ip address asterisk might send them back out another route
Have a look using tcpdump on the server to see where the returned
Not too hard to do,
you can have a script generate a list of call files which automatically
ring the callers in the list and play a message
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
Cheers Duncan
Christopher Stamper wrote:
Right now, my organization is using a
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 fits and copes well with
you abstract that layer which is cleaner.
If we need to have one E1 then having more for the Astribanks sounds fine.
Cheers Duncan
Rob Hillis wrote:
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
I like the discussion, I doubt it will end.
I prefer top posting because I reply to all my customers that way, my
mail client isn't that smart and I think technology should meet the
needs rather than force you to adopt work arounds.
I can fully understand though others preferring it, but I
on the initial screen that show 'upgrading' and MAC
address and the process not continued.
Thanks.
--
Salvatore.
- Original Message -
From: Duncan Turnbull [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
phone but is displays always and
only 'upgrading' and MAC address and I cann't access the phone
configuration.
Thanks.
--
Salvatore.
- Original Message -
From: Duncan Turnbull [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
Hi Salvatore
You need to look at the logs of the tftp server, not the phone.
Hopefully you can see the ip address of the phone asking for files
If there is nothing at all being requested from the tftp server then you
probably want to reset the phone to defaults again.
Usually it stalls when
Are you sure you have set the 7960 to SIP?
By default they use SCCP, so you need to go through the process of
changing them over, which ideally would just be done with the edits you
have already in the load files but generally means going back to an
early version of the SIP code then working
Its a good question
I have lots of disk space so leave it high, I would rather have the
detail if I need it
It probably would seem sensible to revisit stable systems after a year
and lower the verbosity, but then since I can afford the space I am not
too fussed.
Cheers Duncan
Olivier wrote:
Its not so hard if the APs are purely just converting ethernet to
wireless. If there is any authing on the AP then it would be tougher.
And a centralised DHCP issuer is important i.e. just one address range
across all APs so when moving APs there is no dhcp change, no auth
change, just a
Hi All
I have an AA50 without inbound DDIs but each line has a separate number
so based on analogue port it can be routed to different people. The
challenge with this method is it appears to only allow the dial plan to
use 1 outbound route so if all the analogue ports are split into
You can use TDMoE to get an E1 running but its really designed to
replicate an E1 end to end
Its a standard and there is equipment out there that does it, e.g. from
RAD and a few others. I didn't have any joy using the Asterisk code to
get it going but it should in theory work. Its completely
I had an issue where I put a comma in the prepend digits string pn
call plans and then the call plan menu would no longer load.
It parses the menu from the text file so I used the file editor to
clear the offending line and my menu came back. Not sure if thats your
issue but I was surprised
Are you using ubuntu?
Usually I have to edit the Makefile in the else section of Global
variable declaration based on architecture
# ASTVARRUNDIR=$(localstatedir)/run
ASTVARRUNDIR=$(localstatedir)/run/asterisk
This seems to do it
Cheers Duncan
on 07/09/08 04:53 Cyril SCETBON said the
Try some of the shell scripts in the asteriskcookbook recipe heap
http://asteriskcookbook.com/wiki/index.php/RecipeHeap
Specifically
http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention
Cheers Duncan
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL
I had a similar issue in 1.2 after transfer and we were using SIP only
but an upgrade cured it
We are now on 1.4.18 still without issues
Cheers Duncan
Rilawich Ango wrote:
Hi all,
Recently, I experienced one way audio after call transfer.
incalling call (PSTN) A -- GXP2000 thro' zap
Hi Femi
We have about 50 Cisco 7960s on one site off Asterisk 1.4.18
Its all SIP and it doesn't stress a P3 system much at all.
I am not sure what phones you are using - the 7960s are not hard to configure,
a bit of process to convert from the Cisco Skinny to
SIP (using SIP v8.6) but
Hi All
For the scenario of a single asterisk server that needs to serve clients
on the net, as well as local office clients, I would be very interested
in people's views of the best method to handle security to prevent net
based attacks while still allowing the client access.
Some of the
We build and maintain 7 Asterisk boxes for our customers, I have
recently moved 3 to 1.4. I also use iaxmodem and on the last one 1.4.14
I was getting iax thread errors - which was reported as a bug in much
earlier versions but apparently fixed. When 1.4.15 came out (two days
later) it solved
The freepbx system has a primary number option in its ring group dialing
which if selected as a ring strategy means it won't ring any further if
the primary number is engaged. This is useful in follow me setups.
I haven't dug into how its implemented but it works for ring groups and
follow me
Hi Helen
Sounds good, I think Troy will need me to setup the notification list to
the winners though so it might pay to send me those details directly
Should be better rugby this weekend for one of us ;-)
Cheers Duncan
on 10/09/07 14:20 Paul Hales said the following:
We have used a quite a
I am yet to use 2.3 but have 2.2 on 8 ubuntu based installations with Asterisk
1.2.18 or greater
FreePbx is really useful as an interface to all the config files, stats etc,
its also really great if your customers need some
control
The documentation has recently been updated and there is a lot
This message arrived today 18 July NZ time
Full headers below but most of my mail is like this - the offending bit seems
to be: INXS.digium.internal which took 4 days to
deliver it
Cheers Duncan
Return-path: [EMAIL PROTECTED]
Envelope-to: [EMAIL PROTECTED]
Delivery-date: Wed, 18 Jul 2007
I thought initially it was a pretty poor generalization about postgrey and our
capabilities until I realized that this was sent a
few weeks ago when this probably wasn't an as obvious issue. But it clearly is
an issue now.
I have checked my mail servers for failures, implicitly greylisting is
If you look through the Trixbox without Tears by Ben Sharif - google for it,
it's a good read for things you can do for asterisk
Ch 31 has this below
I would search the tribox and sugar forums for more info - really its just
using click to dial from sugar, and potentially CID
lookup - I
I doubt it's the PRI itself
SHDSL isn't part of the internet per se, its just an access technology.
SHDSL is just synchronous DSL which can be used to deliver E1s over.
ISDN PRI's are delivered in a 2Mbit/sec G703/G704 frame and will give you lots
of alarms if they are having any issues
It
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no
baby sitting, I receive about 20 and it requires no baby
sitting
Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and
hylafax lists for much bigger examples
Cheers Duncan
-Original
I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/)
Its built to use voip or gsm and is about the same
price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's
okay and they say WPA is only a firmware load away ;-)
, and it has a browser to login
I have the same challenge and issue, the server dies shortly after being fired
up, although I am using Asterisk 1.2
Even with strace its very trying to work out whether the messages are errors or
importance or just run of the mill
All advice and options appreciated
Cheers Duncan
I have this happening with a Cisco 7960 - I can't see what the difference is, I
have asterisk 1.2.13 and a number of 7960s which
happily work, as well as some 7961s which also work.
However one 7960 doesn't, although it dials quite happily but that's probably
due to dtmf being put into SIP
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