On Tue, 16 Dec 2008 17:55:00 -0500, Drew Gibson d...@oanda.com
wrote:
http://www.theregister.co.uk/2008/12/16/infonetics_enterprise_telephony_numbers_q3_2008/
... whilst...
http://www.theregister.co.uk/2008/12/15/adsl_voip/
G
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Hi,
At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point
to multi-point mode.
Here (France), most small PBXes are connected to ISDN through BRI trunks in
PtmP (don't know why but it seems the general case).
So this NT-PtmP function would be very helpful to easily slide an
Hi,
I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would
you please contact me?
Thanks,
Pete
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If you are connecting to BRI lines then you should be TE - not NT.
You can run as TE ptp or ptmp with mISDN (not sure about DAHDI yet - not tried
the new release).
HTH
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
Hello Andrew,
2008/12/17 Andrew Thomas a...@datavox.co.uk
If you are connecting to BRI lines then you should be TE - not NT.
Yes of course, you're right.
I was mostly referring to this :
ISDN --BRI asterisk -BRI- legacy PBX
Then, in this case, as legacy PBX has a set of
I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with no
problems. The ones I used for testing were the Avaya IP Office, Siemens
Hi-Path/Hi-Com and various old Panasonics.
All I had to do was to turn on the 100ohm termination on my S0 ports (set as NT
on the B410P of
Where are you actually doing the diverting? In Asterisk at the telco
exchange?
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith
-- Sent: 17 December 2008
Chris Bagnall schrieb:
Also seem to be getting some errors writing CDRs to a postgresql database.
What errors precisely?
How can you tell?
I'm using the schema for pgsql from voip-info.org, which, again, has worked
fine logging 1.2 and 1.4. Have there been any schema changes in 1.6 one
Jerry Geis schrieb:
This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038
apparently has been fixed.
I dont see anything on the page saying what released version of asterisk
this is in.
How can I tell that?
It (svnbot) says:
U branches/1.4/main/dial.c
No, ${exten} is the final destination number
myphone calls 123456, which is diverted to 22334455 would givc an
${exten} of 22334455, but I wanted to know the 123456.
Julian
Andrew Thomas wrote:
Isn't that the ${exten} number? In other words, the number called.
--
Hello,
I want that after client and queue member call would be established, cmd
queue runs some 'procedures' . So I am using cmd Queue option 'gosub'. This
is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
Answer();
2008/12/17 Artifex Maximus artife...@gmail.com
On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote:
2008/12/17 Artifex Maximus artife...@gmail.com
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls,
Hello may situation is the next:
Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite
^
|
ip phone (cisco)
Asterisk and de cisco phone are in the same LAN. I want to make a
call between the x-lite and the
So what is the middle name that causes problems? atre you sure you don't
have strange characters in it, like spaces, nonprintables, weird
encodings, etc?
l.
In data Wed, 17 Dec 2008 00:35:04 +0100, Eve Ellen Cole
ec...@mail.plymouth.edu ha scritto:
I’ve got an interesting problem and
Giedrius Augys wrote:
Hello,
I want that after client and queue member call would be established,
cmd queue runs some 'procedures' . So I am using cmd Queue option
'gosub'. This is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
OR
Q: What is the most annoying thing in email?
Q: What is the most annoying thing in email?
A: Top-posting.
Q: What is the most annoying thing in email?
A: Top-posting.
Q: Why is top-posting such a bad thing?
Q: What is the most annoying thing in email?
A: Top-posting.
Q: Why is top-posting
-- Where are you actually doing the diverting? In Asterisk at the
telco
-- exchange?
...or at...
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Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Not when you take the time to properly trim your reply it's not.
BK
I would say the 'norm' in the UK is TE-ptp and NT-ptp or NT-ptmp (depends what
is on the end of the port(s)).
If using NT-ptmp, then a 100ohm resistor is usually needed in the circuit
somewhere - aka ISDN balun - (unless the card has this facility - like the
B410P has).
HTH
Andy
2008/12/17 Mark Michelson mmichel...@digium.com
Giedrius Augys wrote:
Hello,
I want that after client and queue member call would be established,
cmd queue runs some 'procedures' . So I am using cmd Queue option
'gosub'. This is my example of ael :
context QUEUE {
_X. = {
Scott,
I had the same problem when I downloaded
http://downloads.digium.com/pub/asterisk/asterisk-1.6-current.tar.gz This
downloaded asterisk-1.6.0.2.tar.gz To fix the problem I downloaded
http://downloads.digium.com/pub/asterisk/asterisk-1.6.0.3-rc1.tar.gz and I
was able to compile without any
I have the following setup: DS3 - Cisco AS5400 - H.323 (ooh323) -
Asterisk
Inbound calls work great but outbound calls fail. So to check and
make sure we have outbound calling ability on the DS3 we setup a Cisco
Call Manager Express and it can make outbound calls both local and
long distance
Isn't that the ${exten} number? In other words, the number called.
-- -Original Message-
-- From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
-- boun...@lists.digium.com] On Behalf Of Tony Mountifield
-- Sent: 17 December 2008 10:17
-- To:
In article 49483005.8030...@dotr.com,
Julian Lyndon-Smith aster...@dotr.com wrote:
I have a couple of numbers that are diverted to a number that is
conected to an isdn30 card, running asterisk 1.4.
eg.
123456 = 22334455
654321 = 22334455
What I would like to know is the number of the
2008/12/17 Andrew Thomas a...@datavox.co.uk
I have piggy backed a few PBX's off the back of a B410P (4 x BRI) card with
no problems. The ones I used for testing were the Avaya IP Office, Siemens
Hi-Path/Hi-Com and various old Panasonics.
All I had to do was to turn on the 100ohm termination
It is precisely relevant to this issue. All subroutines, whether they're
called macros or not, in AEL (in 1.6) are Gosub routines. So to invoke that
subroutine, you need to call out with Gosub, not with Macro. So it probably
should be along the lines of: Gosub(outbound,s,1
Hi,
I've read README file in agx-ast-addons-1.4.17.5.tar.bz2
It says Install libTiff =3.8 and 4.0
Should you really use this agx-ast-addons to get app_rxfax and app-_txfax
running with latest 1.4.22 ?
If positive, should you take this libtiff warning into account ?
If positive, where can you
This bug report http://bugs.digium.com/print_bug_page.php?bug_id=12038
apparently has been fixed.
I dont see anything on the page saying what released version of asterisk
this is in.
How can I tell that?
jerry
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On Wed, Dec 17, 2008 at 11:52 AM, Olivier oza-4...@myamail.com wrote:
2008/12/17 Artifex Maximus artife...@gmail.com
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls, handling Agents
2. Asterisk as external IVR on extension
http://shop.ebay.com/items/_W0QQ_nkwZdigiumQQ_armrsZ1QQ_fromZR40QQ_mdoZ
2008/12/17 Pete Kay pete...@gmail.com
Hi,
I am looking to buy 2 used 1 or 4 ports E1 Cards. If you have one, would
you please contact me?
Thanks,
Pete
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The original poster is looking for RDNIS as the number initially dialed,
i.e. the DNIS recognized by the first PSTN switch handling the call. The
call may have been diverted to a different number, e.g. unconditionally
forwarded to a call center (answering service, for us older types). If there
are
2008/12/17 Barry L. Kline blkl...@attglobal.net
Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Not when you take the time
On Thu, Dec 18, 2008 at 12:46 AM, Silvia Menendez silvia.menen...@gmail.com
wrote:
Hello may situation is the next:
Asterisk -- NAT1 (router)--- internet -- NAT2 (router) -- x-lite
^
|
ip phone (cisco)
If the diversion takes place in asterisk then the dialplan can set a
variable before it diverts and then use that variable at the destination of
the diversion.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Julian Lyndon-Smith aster...@dotr.com
Reply-To:
2008/12/17 Artifex Maximus artife...@gmail.com
Hi all!
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls, handling Agents
2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931)
PRI
The incoming calling
Hi,
I figured out, that app_pppd suffered from
overruns under high out traffic.
(ping -s 600 destip was already high in this context.)
I've just made a quick and dirty hack to fix it.
If interested, just download the original package
by Sirrix (as mentioned on VoIP-Wiki) and the replace
their
On Wed, Dec 17, 2008 at 07:03:15PM -0200, David fire wrote:
2008/12/17 Barry L. Kline blkl...@attglobal.net
Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either...
Hi all!
Is anyone using the $subject setup?
What I would like to do the following setup:
1. OXE is setup for receiving calls, handling Agents
2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) PRI
The incoming calling route:
1. OXE handles incoming calls, answer
2.
Barry L. Kline wrote:
Bill Andersen wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Not when you take the time to properly trim your reply
On 12/17/2008 Eric ManxPower Wieling wrote:
You're not going to be able to make people stop top posting and I'm
not
going to be able to make people stop using wrong or misleading terms
like SIP Trunk. If you try all you are going to do is piss people
off
and stress yourself out.
Why
You can do a translation rule on the outbound peer, like
voice translation-rule 10
rule 1 /.*/ /\0/ type any national plan any isdn
voice translation-profile SET_TypePlan
translate calling 10 {or} translate called 10 (whatweven you want to
change)
and in the DS3 trunk if you have a
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF
THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M
@#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT
MSN IS FOR.
-Original Message-
From:
On Wed, Dec 17, 2008 at 8:39 PM, Bill Andersen ander...@mwdental.comwrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
Posting either way can be
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
I am strongly against anyone posting anything with their bottom.
later,
PaulH
Paul Hales wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
I am strongly against anyone posting anything with their bottom.
How about
On Wed, 17 Dec 2008, Danny Nicholas wrote:
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF
THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M
@#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT
MSN IS FOR.
Spoken like a true
On Tuesday 16 December 2008 14:51:47 Barton Fisher wrote:
- Original Message -
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 16, 2008 10:40 AM
Subject: Re:
RE Kushner List Account wrote:
Paul Hales wrote:
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
I am strongly against anyone posting
Steve Edwards wrote:
On Wed, 17 Dec 2008, Danny Nicholas wrote:
OSUR GONNA BE ABLE TO MAKE PEOPLE STOP POSTING. IF DIGIUM GETS ENOUGH OF
THESE STUPID HITS, THEY WILL CUT THIS OFF. I KNOW I'M SHOUTING, I'M
@#$###$# TIRED OF INTERRUPTING IMPORTANT WORK TO READ NOTHING. THAT'S WHAT
MSN
Exactly! but sadly these variables don't seem to exists as far as I can
tell
The point is that you're the first person to make this request. If nobody
had
the idea to do it before you, that is precisely the reason it never got
done.
Now that it has been requested, it is in queue for
On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote:
To me top posting is like people talking about SIP Trunks. There is
no such thing as a SIP Trunk. There are SIP connections, peers,
friends, etc. The term is simply a marketing buzzword to make people
that don't know much about
you are soamming my mail box whit this useless discution
the solution is doble posting (top and bottom)
2008/12/17 Andrew Kohlsmith (lists) akli...@mixdown.ca
On December 17, 2008 05:03:00 pm Eric ManxPower Wieling wrote:
To me top posting is like people talking about SIP Trunks. There is
Everyone read this top down for your IVR wav file.
Press 9 for the company directory
Press 8 for the billing department
Press 1 for technical support
Press 0 for the operator
Next let us know who calls into your PBX complaining that your menu is
whacked. Now discussing PBX related issues, that is
On December 17, 2008 06:59:19 pm David fire wrote:
you are soamming my mail box whit this useless discution
the solution is doble posting (top and bottom)
It's a public mailing list. If you're having trouble managing it, you may
want to try a digest version, or perhaps a moderated list.
-A.
On Wed, 17 Dec 2008, Darryl Dunkin wrote:
Steve Edwards wrote:
Top posting. Bottom posting. Honestly, if you can't use an effing
scrollbar, please tell me so I can take you out back and beat you to
death with something heavy. The .5 seconds it takes to scroll from one
end of a message to
I want to take series of user entered (via phone keypad) options/numeric entry
fields and use these with an AGI script. I have looked through voip-info and
I can't find any Asterisk functions specifically for this.
Any guidance please?
Michael
___
Has anyone seen this before, and know what is happening?
u...@host:~/asterisk/agx-ast-addons# ./build.sh
-- Configuring done
-- Generating done
-- Build files have been written to: /root/asterisk/agx-ast-addons
[ 11%] Building C object CMakeFiles/app_devstate.dir/app_devstate.o
Linking C shared
I am getting the following error during AsteriskNow installation I am
using the following AsteriskNOW-1.5.0-beta1-i386-1of1.iso
Here is the error I could piece together as I don't have access to the
screen:
EIP: [c041041c] powernow8k_init
Kernel panic -
On Wed, 17 Dec 2008, Michael wrote:
I want to take series of user entered (via phone keypad) options/numeric
entry fields and use these with an AGI script. I have looked through
voip-info and I can't find any Asterisk functions specifically for this.
Try show agi (1.2) or agi show (1.4) at
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