On 31 July 2010 15:28, Leif Madsen leif.mad...@asteriskdocs.org wrote:
On 7/30/2010 5:49 AM, Lenz Emilitri wrote:
QueueMetrics is actually free (as in beer) for very small call centers and
individual hackers.
Oh really! I didn't know that! Very nice.
What is considered a small call centre?
On 19 July 2010 00:35, Anthony Messina amess...@messinet.com wrote:
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote:
Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones,
how can one receive distinctive ring tones for INTERNAL calls ONLY?
Using Aastra 4801 CT
I'm looking for a good Linux Softphone that
a has Consultation Transfer built in, I know you can do this by
dialling what ever is in features.conf but this is not ideal.
b has the ability to handle more than 2 lines eg calls at a time.
c Works with Asterisk.
d Has a feature where someone can dial
the returned list using
cut searching for the device.
So.
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
or
2. Is there some way of creating such a function.
Thanks in advanced
Peter Childs
On 11 July 2010 14:19, Paul Belanger paul.belan...@polybeacon.com wrote:
On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote:
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
*CLI core show function QUEUE_MEMBER
No function by that name
On 11 July 2010 15:56, Paul Belanger paul.belan...@polybeacon.com wrote:
On Sun, Jul 11, 2010 at 9:28 AM, Peter Childs pchi...@bcs.org wrote:
No function by that name registered.
also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21
The function is in 1.6.2. Best you could do
On 17 May 2010 08:40, Lenz Emilitri lenz.lo...@gmail.com wrote:
Use Addmember and removemeber instead :)
l.
Hmm I'm getting that kind of.
From What I can work out.
Agents have been deprecated and are going to be removed.
The replacement, is some complex dialplan using Local Channels which
I've been trying to get the hang of Agents and Queues and I must say
its a little unclear as to how things work.
So maybe someone has some better idea
From what I can work out an Agent is meant to be nothing more than a
virtual device that can be moved around physical devices... by login
On 30 March 2010 02:04, Mark Phillips g7...@g7ltt.com wrote:
They say confession is good for the soul. Perhaps they are offering a
phone in confessional service?
Unfortunately the business of the church often flies in the face of
the business of the Church.
I think you'll find a lot of
On 11 March 2010 21:09, Matt Riddell li...@venturevoip.com wrote:
On 9/03/10 9:13 PM, Peter Childs wrote:
Also is there some way to get the starting end to auto pickup, (or at
least hit for this to happen (I'm using SIP if that helps))
When you make an originate request it works like
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote:
8 mar 2010 kl. 11.13 skrev Peter Childs:
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life is rarely that simple, and this does not really answer
On 9 March 2010 07:58, Peter Childs pchi...@bcs.org wrote:
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote:
8 mar 2010 kl. 11.13 skrev Peter Childs:
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote:
At an Asterisk CLI use the command manager show commands.
Life is rarely that simple, and this does not really answer the question.
Oh and Channel can mean different things in different contexts
ie
Channel in a PlayDTMF
Is there a list of input's / out puts from the management API together
with there parameters, there meanings and which are required and what
they do/mean.
Its just all the docs I've found seam to be rather sketchy and
gathered by trial and error, not really up to what I would call a
protocol
On 16 January 2010 06:04, Sean Brady sbr...@gtfservices.com wrote:
Looking at all the docs I can find Asterisks looks like it should be
able to do the job and a whole lot more.
This is for a small call centre so ideally we want all the features of
an average call centre, ACD, Call Recording,
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
purposes
Anyway I can make calls fine (if only to the testing line and other
sipgate lines
quickly.
Peter
option for your endpoints?
y.
Peter Childs schrieb:
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash
I've managed to get a basic system set up. and can now take and make
sip calls over the sip trunk I've got from sipgate.co.uk for testing
purposes
2010/1/26 Peter Childs pchi...@bcs.org:
2010/1/26 Yves Arikoglu yves...@gmx.de:
do you use the
qualify=yes
No, If I do it does not work at all.
I've found if I set defaultexpiry to 30 it works fine. and was infact
working for 30 seconds every two minutes before, It looks like
Ok this has Probably been put to bed several time but never mind.
Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
installed with OpenPBX, Asterix etc by hand)
I've got a new server to run Asterix on and want to get it working
quickly and yet be configurable in the future with out
This is currently still at a proof of concept stage.
After being mis-sold a Alcatel phone system, that does None of the
things we asked for (Ok it takes calls but that's about it) We are
looking at alternatives to try and bring some of the features we
previously had on our old Analogue STS
2010/1/12 James Mutuku listmut...@gmail.com:
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a
I can use Google just as well as the next guy, and if you'd bothered
to look at the results you could see they were extremely
.) but it has its benefits, such as not being restricted by a
particular GUI or management system, and being able to customise things
a bit more.
Peter.
Rob
On Tue, 2010-01-12 at 10:55 +, Peter Childs wrote:
This is currently still at a proof of concept stage.
After being mis-sold
On Tue, 2006-07-11 at 12:29 +1000, MBIT Technologies wrote:
Hi Guys
I am just looking for a bit of help here. I am trying to integrate the
2 of these together via a E1 link. The link has no signalling and is
basically a dumb 2 meg link.
I would have thought that you would have _some_ type
, and have a 'Asterisk' T-Server
then I think you might be out of luck as they no longer have third parties
develop t-servers.
As a media gateway, pre-treatment, etc it should be usable.
Cheers,
Peter
--
Peter Childs
NEC Business Solutions Ltd
Ph:61-8-8301-4908 Mb:61-4-0819-7693
IM
The couple of AS5300's that we have seem to work fine.
We are using them for E1-SIP with SER and Asterisk. We also use
PC/Digium for E1-SIP. With far end echo issues I found that adjusting
the echo tail on the cisco 'fixed' things straight away, with the
Digium. Well... Um Perhaps I
you can use a softphone, or
we usually just throw a 'call' description file into
/var/spool/asterisk/outgoing that plays music on hold for a few minutes...
Cheers,
Peter
--
Peter Childs
NEC Business Solutions Ltd
Ph:61-8-8301-4908 Mb:61-4-0819-7693
IM: pjcinaus (yahoo)
-Original Message
Called [EMAIL PROTECTED] ??
I assume it actually didn't dial 'myhomeno' but your number?
Are you fully prefixing your number (ie 10 digits)... Does ringing other
numbers (a engine test number etc..) work?
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Have you loaded the zaptel drivers yet? Until you load the wcfxo module you
will see nothing in /proc/interrupts.
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of 163
Sent: Friday, 5 August 2005 12:35 PM
To:
--
Peter Childs
NEC Business Solutions Ltd
Ph:61-8-8301-4908 Mb:61-4-0819-7693
IM: pjcinaus (yahoo)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jochen Witte
Sent: Wednesday, 27 July 2005 10:48 PM
To: asterisk-users@lists.digium.com
Subject
What is your dmesg output when you fire up the card.
There were some problems with TE410P and the intel chipset used in the
DL380 G4's.
You need firmware at least 'TE410P version c01a010b'
Contact Digium and RMA if you have older firmware (basically the symptom
will be everything
is ok,
: Eric Bishop [mailto:[EMAIL PROTECTED]
Sent: Saturday, 12 March 2005 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Peter Childs
Subject: Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq
DL380 G4 server]
How did you go?
On Tue, 8 Mar 2005 11:28:59 +1030, Peter
Bishop [mailto:[EMAIL PROTECTED]
Sent: Saturday, 12 March 2005 11:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Peter Childs
Subject: Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq
DL380 G4 server]
How did you go?
On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs
, 8 Feb 2005 11:13:24 +1030
From: Peter Childs [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
CC: [EMAIL PROTECTED]
RMA your non
Sort of. I worked sort of ok, but I found I really just thought it sucked,
and carrying a wifi phone and mobile together just didn't impress me at all!
I had some issues with WEP, but I was trying to run adhoc so it may not have
been a problem with the device but my wifi...
Good luck.
Cheers,
From
http://www.supermicro.com/products/system/1U/6014/SYS-6014P-8R.cfm
you can see the board has the IntelR E7520 chipset.
I would suggest you note this to Digium when purchasing your TE410p, as
several people have had issues with this chipset in servers (see HP
DL380-G4), and Digium have a
@lists.digium.com
Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4
server
In article [EMAIL PROTECTED],
Peter Childs [EMAIL PROTECTED] wrote:
Contact Digium Support. They have been very helpful with this issue
(mention your using the G4 server with the Intel E7520 Chipset
Digium support are trailing some new firmware with the TE410P for machines
with
the Intel E75xx Chipsets that are having issues (such as the DL380 G4).
I believe they are confident they have resolved the issue that prevents the
cards working, but you may need to specifically mention that you
Contact Digium Support. They have been very helpful with this issue
(mention your using the G4 server with the Intel E7520 Chipset..)
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Erick Perez
Sent: Saturday, 5 February 2005 5:51 AM
Personally I would for the time being steer clear of anything with a
Intel E7520 Chipset (newish) such as the HP DL380 G4 etc... if you
are using a TE410P card (ie 3.3v).
Just my 2 cents. But you can always give it a go :)
Cheers,
Peter
-Original Message-
From: [EMAIL
Sounds like you don't ever get Layer 1 up.
I'd check your cabling (pins 1,2 and 4,5)
Should span=1,1,0,ccs,hdb3,crc4 be the way to go (aren't you getting
clocking
from your carrier...)
Good luck.
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
The the 'common' factor here appers to be the Intel E7520 Chipset.
I have a NEC 120Rg-2 here with this chipset with the same problem.
This chipset exists in the HP DL380 G4 Server, and the machine
mentioned below.
Someone else mentioned the same issue on a new Dual Xeon EM64T
capable
I have one sitting around here, only the SV7000T module though (not the
SV7000S sip handset server), and a few handsets and media gateways.
The SV7000 is basically just the CPU module removed from their IPX
hybrid tdm/voip models for customers that want a 'pure ip' solution.
The software
My communications was as a PH-D handler and DRS server for the ip
telephones. I'm sure the CCIS/IP would be a _whole_ different
kettle of fish, and I'm sure a NEC SIP-MG for the SV7000 would
be around (perhaps only in the NEC/J marketplace, but soon to
others...)
My thoughts of
http://sunflowerhead.com/software/yac/index.html
You only need to run the client..
exten = s,4,System(/bin/echo -e '@CALL${CALLERIDNAME} ~${CALLERIDNUM}' |
nc -q 0 -w 1 pjcm400 10629 )
Does the trick for me... and YAC has a nice caller history log etc (and
I do like those nice windows
'SIP trunking' is something I've heard before mainly from traditional
switch
vendors that are having trouble with SIP as they are used to 'stations' and
'trunks'.
You should find asterisk very capable as a 'toolkit' box, as it can
'register'
like a traditional SIP client with a SIP
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] sip trunking works?
Are you talking about the 187 page SIP tutorial? What couple of pages are
you referring to?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter
I may have missed something here but couldn't you just do this with a
bit of bash/perl/etc using 'externnotify=' option in voicemail.conf file?
I do this to set MWI via OAI (CTI) on a NEC switch without having to
'integrate' heavily. If you just need those bits you could probably just
What type of existing PABX do you have (Make and Model)
What interfaces can you use to connect to your PABX, ie
analog tie lines, E1/ISDN, anything else?
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of P J
Sent: Friday, 17 September
For what its worth I think you'll find the Philips IV2000 voip switch is
actually a NEC IVS2000. It does support IP telephony, and I have read that
H323 trunking is possible (only _read_ it though).Handsets are NOT
anything you would call 'standard'.I have no idea about the IS3090
I have the same hardware (x2)
/etc/zaptel.conf file
fxsks=1-2
loadzone=au
defaultzone=au
/etc/asterisk/zapata.conf file
[channels]
language=en
context=inbound
group=1
musiconhold=default
; need these much shorter than defaults
flash=90
signalling=fxs_ks
threewaycalling=yes
transfer=yes
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