On 7/16/06, Erick Perez <[EMAIL PROTECTED]> wrote:
Hi,
I would like to know what kind of tests should I make in order to
document tone/configuration settings for analog cards and E1 cards
specifically for my country (Panama).
For example: Australia, Venezuela, etc, they have been documented and
Hi,
I would like to know what kind of tests should I make in order to
document tone/configuration settings for analog cards and E1 cards
specifically for my country (Panama).
For example: Australia, Venezuela, etc, they have been documented and
included in the zaptel config.
Thanks,
--
---
My mistake.
Adding the "\" character using ascii codes in a badly configured
console+keyboard, led to the insertion of a "dot" and that was the
unprintable char.
Note to console users (CENTOS?) vi and nano did not showed the char, I
then used MC (midnightcommander) and it showed the "dot".
See y
Hi,
Don't know if it helps, but in AU you can tell the telco to place all calls
on 2 ISDN's at the same time.
That way you could have 2 ISDN lines on 2 ISDN cards (or Spans) and all
calls would be presented on both ISDN services.
That way each asterisk server can decide what call it takes.
Than
- Mike Fedyk <[EMAIL PROTECTED]> wrote:
> When will Digium include the octasic on the TDM2400P? And maybe the
> TDM400P?
The TDM400P will never have an echo canceller module, there is no way to add
one to the board without a redesign.
The newer board(s) will likely have Octasic module opti
Hi folks,
I've been having a go at getting DUNDI working this evening to enable
users to register to any Asterisk box and to look them up from another.
The DUNDI part works just great (very impressed), as does the subsequent
joining of calls between the two servers but I'm struggling with
reg
On 15 Jul 2006, at 19:26, voiplist wrote:
On 7/15/06, Tim Panton <[EMAIL PROTECTED]> wrote:
On 15 Jul 2006, at 17:24, voiplist wrote:
> After further testing, here is what we found..
>
> The account code was actually right after all, what made us
think it
> was incorrect was the fact that
Stephen,
I would check with your polycom reseller. They should have the files
you are looking for and you know they will be at least from a
creditable source. In terms of setting up your phones for ftp
provisioning, you will need to edit the files that you obtain from the
reseller, and edit t
Hi,
While compiling zaptel 1.2.7 I get the error:
zaptel.c:384 error: stray '\194' in program
It looks like an unprintable character, this source is straight from
the .tar.gz release.
Since im compiling in a Centos 4.3 Xeon, the only thing I added was:
CFLAGS+=$(shell if uname -r | grep -q 2.6.9
Douglas Garstang wrote:
I have dialled into a Queue, and an agent has answered the call with
AgentcallbackLogin().
The agent hits #1, to transfer the call. Asterisk responds with 'Transfer',
followed by dial tone.
As soon as I enter a digit, Asterisk responds with 'I am sorry. That is not a
v
On 7/15/06, Tim Panton <[EMAIL PROTECTED]> wrote:
On 15 Jul 2006, at 17:24, voiplist wrote:
> After further testing, here is what we found..
>
> The account code was actually right after all, what made us think it
> was incorrect was the fact that * was reporting the wrong channel for
> the cal
Hi Erick,
The instructions for the SVN installation are on the Asterisk's website:
http://www.asterisk.org/download
Instruction from Asterisk's web:
===
SVN repository
Subversion is the best way to keep on the bleeding edge of source releases.
If yo
On 15 Jul 2006, at 17:24, voiplist wrote:
After further testing, here is what we found..
The account code was actually right after all, what made us think it
was incorrect was the fact that * was reporting the wrong channel for
the call.
Test 2:
We turned off the softphones mentioned abo
Yes, I initially used FreePBX to set up the system since I was new to
Asterisk, but now I feel more comfortable editing the conf files myself.
I'm not sure which conf file would be relevant to the issue of each
phone being directly mapped to a specific trunk, but here goes. I
included the pa
After further testing, here is what we found..
The account code was actually right after all, what made us think it
was incorrect was the fact that * was reporting the wrong channel for
the call.
For example we may have a call from "iaxuser1" which comes in with the
correct account code but it s
If you turn verbose on under the remote console for asterisk does it
show any information that phones are trying to register or anything for
that matter? Another thing you may want to verify is that the phones
are communicating with the server if you aren't seeing anything on
verbose. I am ass
> -Original Message-
> From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED]
> Sent: Friday, July 14, 2006 10:52 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How do you harden an Asterisk install?
>
> shadowym a écrit :
>
> >
> >I remem
Matt, What do you mean the 1.2 svn branch?
Where are the download instructions and installation procedure?
I always download tar.gz (that means the official release) but i
always question what do I do to keep my installation with the latest
bug fixes.
Thanks,
On 7/15/06, Matt Riddell (NZ) <[EMA
Salve Tzafrir, *!
I found Gustavo's mail on the vserver ml and it is IMHO
not the primary problem with a vserver - but I know to
less about ISDN to give him the right tip.
(without vserver)
Does this belongs to digium drivers or to ISDN in general?
Aren't there ISDN cards, where two diffrent ap
Same here. We have over 20 T1s in 4 locations and the one location
we've always had problems with is because of Verizon's bad lines. One
big contributor to this is that outside the building, Verizon's wiring
terminal is covered not by a metal or plastic hard cover, but a
garbage bag. I kid you not
Hi, had a few ask for this so thought may
be of interest to the list.
This is actually for the following setup:
Telstra ISDN30 <-> Asterisk <->
BP250 PABX
The ISDN10, 20, 30’s are all the
same physical link, but you may need to change the bchan and dchan settings
Thanks Warren
I've updated the wiki.
Regards
Paul
"Warren (mailing lists)"
<[EMAIL PROTECTED]>
Sent by: [EMAIL PROTECTED]
14/07/2006 17:14
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion
To
Asterisk Users Mailing List - Non-Commercial
Discussion
On Sat, Jul 15, 2006 at 01:18:02PM +0200, Sergio Sicari wrote:
> Hi, The link in this mail is my blog about asterisk and voip technology:
>
> http://skalog.blogspot.com
Unless you read Italian, don't bother following that link.
--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755
Hi, The link in this mail is my blog about asterisk and voip technology: http://skalog.blogspot.com Thanks! Chiacchiera con i tuoi amici in tempo reale! http://it.yahoo.com/mail_it/foot/*http://it.messenger.yahoo.com ___
--Bandwidth and Colocation pro
Hmm - I have had 2 bad PRI installs out of about 20, and both times it
was faulty wiring from the Telco.
But getting them to fix it can be a real struggle!
Paul Hales
Technical Manager
www.asteriskit.com.au
On Sat, 2006-07-15 at 12:23 +1000, James Sturges wrote:
> Have had L O T S of trouble
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Julian Varanini wrote:
> What is the best way to update from 1.2.9 to 1.2.10?
If it was downloaded from SVN then you can just type make update in the
directory.
If it was a .tar.gz download then you will need to reinstall. I would
recommend using th
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Wolfgang Falk wrote:
> Hello together,
>
>
>
> is it a fact that the manager api does not provide the "hold" and
> "unhold" commands?
There is no manager command by that name nor is there any asterisk
application by that name.
You could maybe tr
Hello together,
is it a fact that the manager api does not provide
the „hold” and “unhold” commands?
Best regards
Wolfgang
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thanks and I am sorry I just noticed the thread containing some compiling problem and after the intensive googling and searching for answers on maillists, i got somewhat outrageous which led to my direct reply to this thread.
well now I compiled app_rxfax and app_txfax successfully, but still no
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