Hello,
SIP invites are accepted from imitator , but 'SIP 180' is not
responded back to imitator.
By inspecting the issue , we can *see* the response is generated and
sent from asterisk (via asterisk logger (sip debug )) , but while
sniffing the interface with tcpdump, we can't see 180
Maybe it is buffering issues with the kernel? Does it happen only when
there is a peak in the new calls rate? Do all 180 messages get dropped?
__Yehavi:
2009/4/19 Nir Levi n...@bezeqint.co.il
Hello,
SIP invites are accepted from imitator , but 'SIP 180' is not
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 and
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
I am having a similar issue. Asterisk does not show ringback tone and I
investigated this due to it not reading sip invite 180. (or supposedly not
receiving it).. My solution is that now I am using h323
(ver 1.4.19)
CS
From: asterisk-users-boun...@lists.digium.com
How do I know that de hardware echo canceller module on my Digium
TDM403E is recognized by Asterisk ?
After having configured /etc/zaptel.conf :
[r...@asterisk etc]# /sbin/ztcfg -vv
Zaptel Version: 1.4.12.1
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01:
VoIP-wiki.org states :
Digium resources http://www.asterisk.org/zaptel-to-dahdi
/etc/zaptel.conf Becomes /etc/dahdi/system.conf
/etc/asterisk/zapata.conf Becomes /etc/asterisk/chan_dahdi.conf
Now, what do I have installed on my system :
/etc/zaptel.conf and /etc/asterisk/chan_dahdi.conf
Will
2009/4/19 Steve Prior spr...@geekster.com
Tilghman Lesher wrote:
Emacs is a nice operating system, but it lacks a decent editor. :-P
It's also a nice religion.
:-
(I didn't this one)
Steve
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ok,
just in case check if you have /usr/share/zoneinfo/UTC
also if you still have the coredump file ... enter gdb
and do
frame 0
print p
print name
Martin
On Sun, Apr 19, 2009 at 3:31 AM, Justin Piszcz jpis...@lucidpixels.com wrote:
On Sat, 18 Apr 2009, Martin wrote:
Hi,
Your
On Sunday 19 April 2009 14:25:26 Martin wrote:
ok,
just in case check if you have /usr/share/zoneinfo/UTC
also if you still have the coredump file ... enter gdb
and do
frame 0
print p
print name
There's already a working theory and a patch:
http://bugs.digium.com/view.php?id=14932
--
How do I disable H323 when compiling Asterisk Addons?
All I want is the MySQL CDR
Michael
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On Sunday 19 April 2009 19:01:06 Michael wrote:
How do I disable H323 when compiling Asterisk Addons?
All I want is the MySQL CDR
make menuselect
--
Tilghman
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asterisk-users
Does the older 1.4.7 Asterisk Addons work with Asterisk 1.6?
The compile for 1.6x Asterisk Addons keeps failing.
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On Sunday 19 April 2009 19:29:48 Michael wrote:
Does the older 1.4.7 Asterisk Addons work with Asterisk 1.6?
The compile for 1.6x Asterisk Addons keeps failing.
No, 1.4 will not work with 1.6. Make sure you're using the same release
version of addons with Asterisk. Addons 1.6.1 will not work
I have found that Asterisk Addons requires the Asterisk includes files to be
in /usr/includes, which they are not by default in Asterisk 1.6.
This can be set at compile time -
--includedir=/usr/include
I hope this observation may be of use to someone.
Michael
Steve Underwood wrote:
I wonder how much demand there is for colour FAXing.
It's quite niche. When it's available people will have their fun with
it for a few times, and after the novelty quickly wears off they never
use it again.
I have quietly enabled color receiving support on high-volume
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label
PHONE NUMBER = the number I called.
Michael wrote:
pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in
extension 'PHONE NUMBER' in context 'phones'
[Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto:
Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label
PHONE NUMBER = the
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