Hello,
I have been using asterisk for the past decade and never had an issue with
upgrades until now. Recently, in November I upgraded from 18.14.0 to
20.0.0 and afterwards my SPA3102 can no longer register with asterisk. I
have not made any asterisk or SPA3102 configuration changes in ~1-2
Hello,
I use asterisk with an SPA3102 (latest F/W).
I have my asterisk 1.8.13.1 voicemail.conf setup as follows:
; Limit the minimum message length to 3 seconds
minsecs = 3
This works perfectly, however, when the caller hangs up before the beep (or
during it?) then I get 1 minute and 22 seconds
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Thursday, November 01, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk
Package: asterisk
Version: 1.6.2.0~dfsg~rc1-1
See below for issue:
On Wed, 21 Oct 2009, Justin Piszcz wrote:
On Tue, 20 Oct 2009, Justin Piszcz wrote:
On Tue, 20 Oct 2009, Dave Chinner wrote:
On Mon, Oct 19, 2009 at 06:18:58AM -0400, Justin Piszcz wrote:
On Mon, 19 Oct 2009, Dave
On Sat, 21 Nov 2009, Faidon Liambotis wrote:
Justin Piszcz wrote:
Found root cause-- root cause is asterisk PBX software. I use an
SPA3102.
When someone called me, they accidentally dropped the connection, I called
them back in a short period. It is during this time (and the last time
Hello,
I have regular phone service (not VoIP) with an SPA3102.
It works fine, I can dial out, incoming calls work as well, no issues.
With the regular phone service, while I am on a call, I can initiate a
3-way call via:
1. Press [FLASH]
2. Dial the number.
3. Press [FLASH]
Would this be
Hello,
I am using an SPA3102, all is working with asterisk 1.4, voice mail,
outbound calling etc, and it even passes the cid name/num to my analog
phone. However, when someone is calling me, I hear the beeps but the
caller-id information is not showing up on my phone, is this an SPA3102
Hello all,
What is the recommended way to remove spaces in the name of the caller ID?
Justin.
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On Mon, 20 Apr 2009, Justin Piszcz wrote:
Hello,
When a voice message is saved and e-mailed as a wav, the total time of the
voice mail does not show up in, e.g., windows media player, why is this?
I have only used wav49/wav:
; Use wav49 format for all voicemail messages
format=wav49
Hello,
When a voice message is saved and e-mailed as a wav, the total time of the
voice mail does not show up in, e.g., windows media player, why is this?
I have only used wav49/wav:
; Use wav49 format for all voicemail messages
format=wav49|gsm|wav
Justin.
On Sat, 18 Apr 2009, Martin wrote:
Hi,
Your backtrace doesn't make sense to me.
Do you have in main/stdtime/localtime.c
this function that way ?
struct ast_tm *ast_localtime(const struct timeval *timep, struct
ast_tm *tmp, const char *zone)
{
const struct state *sp =
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sat, 18 Apr 2009, Martin wrote:
Any other recommendations?
Here is a strace of asterisk right before it core dumps:
9374 read(24, ..., 4096) = 0
9374 close(24) = 0
9374 munmap(0x7f023a404000, 4096
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sat, 18 Apr 2009, Martin wrote:
Another person with the same issue:
http://www.freepbx.org/forum/freepbx/users/voicemail-crashes-segfault
He notes:
A call to a extension to listen MOH allways
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sat, 18 Apr 2009, Martin wrote:
Looks like someone already filed a bug report(?)
http://bugs.digium.com/file_download.php?file_id=21839type=bug
Other
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sat, 18 Apr 2009, Martin wrote:
Two items:
First, the Debian Testing version (1.4.x) works
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sun, 19 Apr 2009, Justin Piszcz wrote:
On Sat, 18 Apr 2009, Martin wrote:
I am now using 1.4
Hello,
Information:
gcc -v: gcc version 4.3.3 (Debian 4.3.3-3)
os: Debian/Testing
Pulled latest release from asterisk site, compiled, installed it.
I have a barebones configuration:
$ ls -l asterisk
extensions.conf
modules.conf
sip.conf
users.conf
voicemail.conf
You can see them here:
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