So far - the 4-port ISDN HFC chipset cards from Junghanns.net works well
and is half the price of a 4-port Eicon card.
On Mon, 2005-11-14 at 10:07 +, David Waugh wrote:
Hi Lee,
I use a Diva Server card here with Asterisk using Chan_capi.
The basic BRI card has one BRI port. They also
I can not see that its illegal to have Asterisk in India. The TDM400P
card should work fine - but it may not be approved to be interconnected
to the phone system. (This never stopped me doing similar things).
I'm assuming that its possible to connect a 2-wire phone to the Indian
phone system - ie
Couldn't find anything on the lists or in Wiki..
Customer wants to be able to dial complete SIP URL's... from his SNOM
phone.
ie - He dials on his phone [EMAIL PROTECTED] (which is more
difficult than a Number - but not undo-able)
How do I configure my extensions.conf to handle this sort of
On Fri, 2005-10-28 at 14:26 +0200, Tomasz Chmielewski wrote:
So the idea is to put a SIM card inside the Asterisk box, equipped with
a special card, a card which would be a mobile phone really.
Does anyone have an idea if such cards exist, and if so, if they work
with Asterisk?
You can
I'm using a SNOM 360 with Ver 4.3 software.
Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff +
Head)
I've used the wiki info to set up some lines to monitor some internal
extensions.
When the extension is rung - the lamp comes on, when the call is
answered, the lamp goes
I'm also using SNOM320/360 phones. Ideally - set up one button to toggle
the Agent Status (in/out == On/Off) ???
Kinda make sense if app_devstate (or similar) made it into mainstrean
Asterisk - so line indication lamps could be used at will.
The SNOM320 is so ideal for Call Centres (the Headset
On Mon, 2005-08-01 at 18:31 -0400, Joseph wrote:
Hall, Eric M. wrote:
Looking for a good web app that will show agents that are login to
queue. I tried the operator panel but I'm unable to get the LED to
change color per the doco I have.. It works well for everything else but
no luck on
I want to authenticate a user before he is able to use the phone. I also
want to set his privilege as to where he is allowed to call to...
Preferably, the password should be their VoiceMail password, (every
extension (or is that user?) can have voicemail defined - even if its
not in use?)
I see the GXP2000 has a headset socket. Are their any compatible
headsets for it. How does the functionality change?
What else would people suggest for a Call-Centre?
Would like Headset, Call Details - etc...
The call centre answers the phone according to which number is called..
--
. .
On Sat, 2005-07-16 at 16:47 +0200, Zoltan Szecsei wrote:
Hi,
Can anyone please point me in a direction as to how to set up these 2
pci cards with AAH 1.1?
Rather load [EMAIL PROTECTED] 1.3 - fixes other problems
I have (am still) googling left, right center - but haven't found a
My 2c worth...
For the beginner, AAH is great. The PC that you install on will be
totally reformatted / fdisk-ed (assuming single drive - etc).
With AAH 1.3 - the installation goes to sleep and sort of finishes
when its Syncing with a Time Server. A reboot at this point seems to do
no harm.
As
On Fri, 2005-07-15 at 04:17 -0700, /dev/null wrote:
I'm trying to build a justification case to get the firm I work for to
start working with Asterisk more. How could I build this case?
The argument I'm raising is that people need phones. PBX systems are
too expensive for fewer options and
On Fri, 2005-07-15 at 17:36 +0300, [EMAIL PROTECTED] wrote:
I was wondering which would be the best GUI to use for Asterisk management?
astGUIclient or AMP?
I'd use AMP - mainly because [EMAIL PROTECTED] uses it - so the user base
and knowledge base should be bigger...
--
. . ___. .__
On Wed, 2005-06-08 at 23:39 +0100, David John Walsh wrote:
Angus
Jumping in with both feet
a BT socket with a capacitor in is commonly refered to as a Master
socket, and are very cheap even without wholesale. It gets its name
from being the socket that BT installed into the house for
On Fri, 2005-06-03 at 06:28 -0700, Nardis Dome wrote:
--- Brett, Gary [EMAIL PROTECTED] wrote:
Is the Eicon that much better ?
sorry, i have only experience with Eicon... maybe
someone else is able to give a feedback...
I'm using Junghanns 4 port card. There is also an 8 port card.
I have a Grandstream GSX-2000 with ..
Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3
I tried to do an HTTP update from the Grand Stream web site...
After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
On Thu, 2005-05-12 at 12:40 -0600, Tim Pushor wrote:
I had trouble calling people who were using FWD/SIP from my FWD/IAX
account. I switched back to using SIP and could call SIP users, but not
IAX users. I've since de-registered myself for the IAX *beta* and can
now talk to everyone again.
${CALLERIDNUM})
exten = s,4,SetCIDNum(87${CALLERIDNUM})
exten = s,5,Goto(default,s,1)
When Executing the above - and I presume incoming Caller Info looks like
the name is Mark Elkins and the Number is 638936...
The purpose is to prefix the number (only the number) with 87.
Sometimes, incoming CallerID data
On Fri, 2005-05-06 at 14:29 +0200, Eugenio De Vena wrote:
Hi there,
will someone suggest me a good and * combatible isdn card ( 1 , 2 , 4 , 8
channels ).
I am currently working with but can not stand their complete lack of
support.
In all fairness to Junghanns, my current release Asterisk
I see that there seems to be a 'callto' URL/URI for dialling a phone
number... ie - on my web site's Contact Page - I have added the
code...
a href=callto:+27128070590+27 12 807-0590/a
There should be some generic way for Mozilla (firefox - etc) to somehow
turn a click on such a link into
On Wed, 2005-03-02 at 07:23 -1000, Jean-Denis Girard wrote:
Is anyone using mozPhone?
If so any feedback you can provide?
Yes. For what I'm doing with it work. Could be improved.
Thanks for your feedback. MozPhone could obviously be improved in many
ways, what would be your
Hi list, I'm getting the message...
Apr 4 15:13:09 WARNING[1069]: chan_zap.c:7512 zt_pri_error: PRI:
received SETUP message for call that is not a new call, wicked!!!
This is running Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k.
These messages happen when someone calls from the Telco on a BRI line...
On Thu, 2005-03-24 at 10:50 +0100, Marc SCHAEFER wrote:
On Wed, Mar 23, 2005 at 06:55:28PM +0200, Mark Elkins wrote:
Last time I tried - there were a few problems...
I had a few random crashes, higher delays and echo with the EICON. I
replaced it now with an HFC. The EICON on isdn4linux
On Wed, 2005-03-23 at 17:18 +0100, Tomasz Chmielewski wrote:
I just wanted to let you know that it's possible to use Eicon DIVA PCI
2.01 ISDN cards (not server divas) with asterisk.
Last time I tried - there were a few problems...
1 - Outbound DTMF - never made it... ie You can not interact
On Tue, 2005-03-08 at 14:16 -0500, John Fullington wrote:
I set up a monitoring system that calls my techs when a problem occurs on
one of our networks, everything works fine unless asterisk calls a cell
phone in which case the tech can not respond using dtmf. It works fine if
the tech call
manufactured device called
Digi-Cell - frmaritz (at) global.co.za is the email address on the box
it came in
Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit???
On Fri, 25 Feb 2005, Mark Elkins wrote:
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote:
Did you have
... :-)
-Herman
On Mon, 2005-02-28 at 17:21, Mark Elkins wrote:
On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote:
Hello Mark , C. All , Is this device available for sale
in the US ? All the digging I've only found outside US
mentions of sales . Any help appreciated
On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote:
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10)
and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are
configured in TE mode and connected to the PSTN; the other 8 are in NT mode
and connected to
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote:
Did you have to make any changes to use the premicell, or was it as simple
as an outgoing landline call?
I am looking into doing this as you can get deals where calls between chosen
numbers are free :-)
Absolutely no changes at all I
On Sat, 2005-02-12 at 12:20 +, JunkMail wrote:
For the single card I was using with isdntool for initialization,
wich
works fine but has no support for two cards.
Can anyone tell me exactly how to initialize the ISDN system manually
???
It all starts with modprobe -v hisax
On Tue, 2005-02-08 at 06:27 -0600, Rich Adamson wrote:
Looking for some advanced thoughts relative to exten number assignments.
We're in the planning stage for rolling out asterisk at multiple small
US telco/isp operations. Their typical voip customer has had their
pstn line for a long
On Mon, 2005-01-31 at 15:30 +0500, [EMAIL PROTECTED] wrote:
Hello,
I need some clarification on TDM400P.
The TDM400P card by itself has no use. You purchase a mix of FXS and FXO
daughter cards (they are coloured Red and Green) which pug into four
available positions on the card. That
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a
call with '*8' - the call will drop after about 20 or so seconds. Is
this a general problem with Asterisk 1.0.2?
As this is the latest release that it appears Klaus-Peter Junghanns has
for public consumption - is there
Just added a new Grandstream BT102 to my network. Its running new
firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to
(SIP) register
Gripe 1: The New Firmware does NOT show the current version of all the
firmware. You have to ask the phone manually with its menu button.
On Sat, 2005-01-15 at 09:09 -0500, Doug Lytle wrote:
Mike Dent wrote:
Whilst on the subject of BT's, do the callers and called buttons function?
they dont seem to do anything on mine?
Yes, but the hand set needs to be off hook.
To add to Doug's reply...
---for people you have called---
1 -
Curiosity got hold of me. I opened up my BT-10 (and it still works
afterwards..)
Under the keyboard (buttons) are four red LED's that appear to run in
parallel (they all flash at the same time when you put the power on).
These are used to light up the keyboard.
The Display LED (blue in my case)
On Tue, 2005-01-04 at 15:34 +0100, Erik Versaevel wrote:
Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The caller was +886288097680 - am I getting the wrong
ClID because
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with myself again - that is - I
On Thu, 2004-11-25 at 13:24 -0600, Carmi Weinzweig wrote:
Again, note that I am not asking to display trunk status, just
extension status, and to allow a user to place a call on hold on one
phone and pick it up on another (that has that shared extension).
From another posting today (SNOM
On Fri, 2004-11-26 at 09:05 +0100, hhandresen wrote:
OT:
http://www.grandstream.com/BETATEST/
(as someone else on this list stated)
I've not seen any problems with it yet
Sequence is, you have a call, push Flash, dial new extension - speak,
push transfer - and you're out of the loop.
But
http://www.grandstream.com/VON_Fall_2004_Product_Announce.pdf talks
about the new GXP-2000 - the replacement for the planned BT-102D (which
I was waiting for)
Anyone seen one yet?
Anyone care to say anything about it - price, performance - etc...
...or should I look elsewhere...
Been
Setting up a Gatekeeper can be a pain. After looking at Speed Dial /
New Context from Wed, 3 Nov 2004 18:24:31, I added the following bits
into 'extensions.conf'.
Maybe useful to others..
In my incoming default profile - I have...
; Calls from the H323 Extentions
exten =
On Wed, 2004-10-27 at 08:15 -0500, Eric Wieling wrote:
George Gardiner wrote:
I would be grateful for any pointers in the right direction. In short, I get
CallerID to display on Xten and a SipTone II; but have failed miserably to get my
BudgeTone 101 to display anything other than the
On Wed, 2004-07-21 at 16:55, Massimo De Nadal wrote:
I'm using a Cologne chip card in my Asterisk box with zapHFC drivers
(bristuff-0.0.2). The system works well, but this way I'm not able to run
newer version of Asterisk.
Do you think it's better to use i4l support and newer version of
On Sun, 2004-07-18 at 23:52, Bruce Komito wrote:
Following a(n apparently) failed attempt to upgrade a BT102, the phone is
now brain-dead. Although it still has enough smarts to get a dhcp address
and try to download the firmware and config, it never gets past the blue
screen, nor will it
On Tue, 2004-07-06 at 11:29, Martin Bene wrote:
The bristuff distribution comes with a install.sh script
(./install.sh)
which downloads, compiles the required software on your system.
If you want to do it manually, look into download.sh to see the exact
cvs checkout options which
On Wed, 2004-06-23 at 17:12, Sathya wrote:
Hi,
Greatly appreciate if some one help me with the application read().
I have added a feature to reload asterisk from a phone...
it uses 'read' to get a 3 digit password
I was using '#' to end the sequence until I realised I could specify the
On Tue, 2004-06-15 at 17:44, Mark Elkins wrote:
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote:
This is an issues with DTMF clamping, you need to use
chan_capi to get DTMF
working correctly.
That's the last thing I wanted to hear :-(
The jist of this is that i4l does not allow
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote:
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jason Williams
Sent: Tuesday, 15 June 2004 6:55 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Outgoing DTMF when using BRI
i4l
On Sat, 2004-06-12 at 17:47, Mark Elkins wrote:
I have a 'day' and a 'night' mode. In the day mode, I play a
'background' message which is interruptable by the pushing of a DTMF key
- ie - all is normal.
Let me try again...
If I mix background announcements with SayUnixTime - then my IVR
menu
I have a 'day' and a 'night' mode. In the day mode, I play a
'background' message which is interruptable by the pushing of a DTMF key
- ie - all is normal.
In night mode - I decided to get smarter...
I play two backgrounds with a 'sayunixtime' in between and now DTMF does
nothing - the menu
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys -
On Mon, 2004-05-10 at 14:53, Philipp von Klitzing wrote:
Hi there,
could anyone drop a short line on what cidinternalcontexts exactly does
in voicemail.conf? The Wiki explanation isn't sufficient - at least not
for me... :-
From my understanding..
I have defined my internal extentions
On Mon, 2004-05-10 at 05:54, Simon Brown wrote:
I have just installed one of the new TDM400 cards with an FXS and an FXO
module into my * server.
I also checked out the latest cvs head.
I am using 7940 phones.
Now I have some strange problems:
1. When in the VM menus, key presses do not
On Mon, 2004-05-10 at 09:04, Mark Elkins wrote:
On Mon, 2004-05-10 at 05:54, Simon Brown wrote:
I have just installed one of the new TDM400 cards with an FXS and an FXO
module into my * server.
I also checked out the latest cvs head.
I am using 7940 phones.
Now I have some strange
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the
On Sun, 2004-05-09 at 14:33, Mark Elkins wrote:
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation
Some CVS upgrade in the last day or two has broken the recognition of
DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting
the error...
*CLI -- Executing VoiceMailMain(SIP/phone1-e0dd, ) in new stack
-- Playing 'vm-login' (language 'en')
**Here I push a button**
May 9
On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote:
Mark,
Could you please add a SIP debug message with the SIP INFO?
I've done a debug with a working asterisk (V1.0) and the non-working
asterisk. The trace is attached. :-)(debug - ascii text)
When you say SIP INFO - what else are you
On Sun, 2004-05-09 at 21:39, brian k. west wrote:
What firmware you have on that BT101? And yes gnupg or what ever you use to
sign your message did produce the attachemnt on this last one too.
OK the gnuPG is off.. :-(
Product Model:BT100
Software Version: Program--1.0.4.63
I'm sure I saw a posting about someone updating the CVS with a more
richly featured voicemail system. What happened? Am I wrong?
Can't seem to find anything on this...
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/
On Fri, 2004-05-07 at 23:00, Mark Spencer wrote:
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
If you'll just be patient for a little
On Sat, 2004-05-08 at 20:43, Ryan Courtnage wrote:
On 8-May-04, at 12:09 PM, Paul Tyreman wrote:
I have a problem with my Grandstream phone. I have set it up to use
DTMFMODE=info and I am able to transfer calls that have been made from
that
phone, but I am unable to transfer calls made
After much reading and fiddling - I have the gnugk GateKeeper running
and can make calls from the H323 phone to the sip phone. Voice works
bi-directionally..
Calling from SIP to H323 gives me a problem...
Both gnuGK and Asterisk are on the same box. Someone said this was OK.
Others said No. I
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned the mailing list after buying
the keys
Seems like
On Fri, 2004-05-07 at 22:27, Billy Huddleston wrote:
SO, do you have a IDE CDROM?
Sorry - I should have said I do have an IDE CDROM -
with a mounted CD
(Yes)
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/
On Wed, 2004-05-05 at 04:02, Kevin wrote:
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is
On Sun, 2004-05-02 at 22:07, Kevin Walsh wrote:
Jon Lawrence [EMAIL PROTECTED] wrote:
I emailed sales at digium asking whether the new module supported
international (ie non bellcore) cli. The answer was yes, ...
The Digium shop (http://store.yahoo.com/asteriskpbx/newitd1pofxo.html)
says
On Mon, 2004-05-03 at 00:11, David J Carter wrote:
Mark J Elkins wrote
Um - Digium wants you to buy their hardware - but there is a CLID
issue.. would it not make more financial sense to insert a dumb ISDN
card (or two), and upgrade your PSTN to ISDN??? Would this not assist
Digium in
On Wed, 2004-04-21 at 01:03, Fran Boon wrote:
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Pattern
I have some Grandstream BT101 SIP phones. Work great (so far).
I have some Planet VIP-101T H323 phones... how do I make them
look/feel/act like a SIP phone
I can dial to them from both Trunk + SIP's
(ie - I've added 'oh323' libraries)
What config do I add so that if I dial the * IP -
I've got two patterns I want to match on making an outgoing call...
(one day - to do Least Cost Routing for Cell/Mobile calls)
Firstly - I prefer '0' rather than '9' to get an outside line...
Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084)
or its just another number to
-- Executing DateTime(SIP/phone1-07ff, ) in new stack
-- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en')
-- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en')
-- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en')
This works - the pathname
On Sun, 2004-04-18 at 11:26, Richard wrote:
Hi,
I noticed some issues with how grandstream handles
stun test. GS is running version 1.0.4.50.
The latest release of software for Grandstream (dunno if its the same
for all phone??? - but for Product Model: BT100)
is:
Software Version:
On Sat, 2004-04-17 at 15:58, Chris Orme wrote:
My dialplan is for the outgoing SIP call is:
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten =
What I've got...
Software:
Linux: Slackware 9.1
Asterisk: out of CVS - so its new.
isdn4k-utils: to test the ISDN Card
Hardware:
PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM
1 x ISDN BRI Card - DIVA EICON (Installed + working)
2 x Grandstream (Barbie?) BT100 SIP
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