Displaying 20 results from an estimated 300 matches similar to: "1.4 segfaulting when manager client is connected"
2006 Jun 06
1
Asterisk 1.2.7.1 bad file descriptor
Hi all,
could someone tell me what this does mean "bad file descriptor" when
trying to start asterisk. It goes till the CLI command and then die with
this message. Below an strace output from asterisk -vvvvvvvvvc
It's on debian Sarge kernel 2.6.7 with packages from debian VoIP team.
The server was running fine till now with this version.
Thanks
2004 Dec 30
1
More * weirdness
Well I am about to reserve a small padded room so I can bounce off the
walls without inflicting tooo much damage... Nothing is making sense at
this point. I tried several releases last night before settling on the
latest CVS (seemed to work the best). Asterisk was running GREAT for the
first few hours. Now since around 10AM EST SIP can't register and incoming
calls are rejected with "all
2017 Apr 29
2
Can't compile asterisk-certified-11.6-cert16 on Ubuntu 16
All;
I'm trying to install certified asterisk 11.6 cert16 on a Ubuntu 16
server. However, when I try to compile it, I'm getting hundreds and hundreds
of errors. Here is a sample of the output.
make[1]: Leaving directory
'/usr/src/asterisk-certified-11.6-cert16/menuselect'
[LD] aelparse.o aelbison.o pbx_ael.o hashtab.o lock.o ael_main.o
ast_expr2f.o ast_expr2.o
2007 Apr 03
0
Faxing issues
I have spandsp, rxfax and asterisk-1.4.2 installed and whenever a fax call
comes in we get this. This isn't good. Any ideas?
[New Thread -1215390800 (LWP 8504)]
-- Accepting call from 'DELETED' to '539' on channel 0/1, span 1
-- Executing [539@telco-incoming:1] Set("Zap/1-1", "DIALEDNUM=539") in
new stack
-- Executing [539@telco-incoming:2]
2009 Feb 19
1
NUT 2.4.1 crashes on FreeBSD - additional info
Hi Volker,
I forward your request to the user list since I don't currently have much
time to process it.
quickly testing 2.4.1, I wasn't able to reproduce it.
a question: was it working with the exact same context/config with 2.2.2?
cheers,
Arnaud
--
Linux / Unix Expert R&D - Eaton - http://www.eaton.com/mgeops
Network UPS Tools (NUT) Project Leader -
2011 Mar 07
1
[1.8.3] Error compiling Asterisk: __sync_fetch_and_add
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting this
error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o
2011 Mar 07
1
Error compiling Asterisk 1.8.3 on Sun SPARC x64 w/Debian Squeeze
Hello all,
Figured I'd repost this with an edited subject line, to attract attention of
people with Debian On Sparc experience. Apologies in advance if this kind of
thing is frowned upon :)
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
2005 Jun 21
0
chan_unicall and /dev/zap/channel
Hello again :-(
I have a problem with chan_unicall. If I have two simultaneous incoming or
outgoing calls, they sound broken because cpu load goes to 99%. Also with one
call, the cpu load goes to 99%. Seems like device /dev/zap/channel is busy
after 5 or 10 seconds , and chan_unicall does not write to this.
strace with asterisk-1.0.7, zaptel-1.0.7, kernel-2.6.10
================================
2020 Aug 27
0
[klibc:master] alpha: Fix definitions of _NSIG and struct sigaction
Commit-ID: 1cd11aaed1dece773c6b1ce2e99a0fe98b51321e
Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=1cd11aaed1dece773c6b1ce2e99a0fe98b51321e
Author: Ben Hutchings <ben at decadent.org.uk>
AuthorDate: Thu, 27 Aug 2020 01:58:19 +0100
Committer: Ben Hutchings <ben at decadent.org.uk>
CommitDate: Thu, 27 Aug 2020 03:51:11 +0100
[klibc] alpha: Fix definitions of
2010 Jan 11
0
[PATCH] Fix arm signals
Following example from usr/include/arch/i386/klibc/archsignal.h:
The in-kernel headers for arm still have libc5
crap in them. Reconsider using <asm/signal.h>
when/if it gets cleaned up; for now, duplicate
the definitions here.
Signed-off-by: Jon Ringle <jon at ringle.org>
---
usr/include/arch/arm/klibc/archsignal.h | 112 ++++++++++++++++++++++++++++++-
1 files changed, 110
2006 Nov 01
0
Fwd: Benachrichtung zum +ANw-bermittlungsstatus (Fehlgeschlagen)
Can someone get this guy off the lists?
---------- Forwarded message ----------
From: postmaster@prebit.net <postmaster@prebit.net>
Date: Nov 1, 2006 3:22 PM
Subject: Benachrichtung zum
=?unicode-1-1-utf-7?Q?+ANw-bermittlungsstatus (Fehlgeschlagen)?=
To: joakimsen@gmail.com
Dies ist eine automatisch erstellte Benachrichtigung +APw-ber den
Zustellstatus.
+ANw-bermittlung an folgende
2007 Jan 03
3
Asterisk Core Dump in app_queue - Anyone seen?
Anyone seen this? It ocurred on a 'reload app_queue.so' command.
Asterisk version is 1.2.9.1.
Tried again, but it was not immediately reproducable.
Doug.
(gdb) bt
#0 reload_queues () at app_queue.c:3339
#1 0xb778a7a8 in reload () at app_queue.c:4012
#2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257
#3 0x08092b3f in handle_reload (fd=33,
2006 Oct 31
2
Opinions on the best wholesale origination/term providers
I've been losing patience with my current provider, a small company
called Sellvoip. Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service. So I'm shopping.
I am interested in the opinions of others on the providers they
work with.
Here are my criteria, roughly in order
a) Decent quality, low latency.
In
2006 Dec 05
2
SIP firmware for Siemens Optipoint 410 Economy?
I have not seen anybody on the web to have found this so I thought
I would check here. Anybody got this firmware? I've found
firmware for the 400, but it doesn't seem to load in the 410.
2009 Nov 06
1
Function Value Not Being Overwritten?
Part of my problem is that I am in the middle of starting to use functions, so this was unexpected behavior for me.? Maybe there is a work-around because I like declaring some of the variables prior to using them.??
I have two separate files:
# File: Test.R
dog<-function(x)
{
??? templeton<-NULL
??? source("Cat.R")
??? print(templeton)
??? print(x)
???
??? print(bobby)
}
#
2006 Nov 09
2
Powering SNOM 200 phones?
Ok, not exactly an Asterisk problem, but...
I picked up some SNOM 200 phones because SNOM's have been recommended for use
with Asterisk and they have line buttons that can subscribe to presence.
However, they don't appear to power up when connected to my Negear FS108P,
which is an 802.3af Power-over-ethernet capable hub. I am pretty sure
these are the SNOM 200b, in that the ethernet
2008 Jun 23
2
Correlation Help
Hi,
I have recently been using the R program and encountered a recurring problem. I have been trying calculate the correlation of a 16 column table. Everytime I type in cor(test), where test is data that I uploaded into R using the read.table function, I get an error:
Error in cor(test) : missing observations in cov/cor
In addition: Warning message:
In cor(test) : NAs introduced by coercion
2008 Jun 25
2
T and P Statistics
How do you calculate T and P statistics (T- test) in R?
Is there a package out there that can do these calculations?
Best,
Michael Tong
Futures Associate
Quantitative Research Services
Franklin Templeton Investments, Inc.
600 Fifth Ave
New York, NY 10020
(212) 632-4254
mtong@templeton.com
Notice: All email and instant messages (including attachments) sent to
or from Franklin Templeton
2007 Feb 23
3
Sellvoip configuration....Please Help!!!!
hi guy, i have a problem, i have an sellvoip account and i want
configure asterisk for outbound calls.
this is my sip.conf
register => XXXXX0000000000:PassWord@70.42.34.200 ; this is one of the
sellvoip server
[sellvoip_out]
type=friend
secret=PassWord
username=XXXXXX0000000000
host=70.42.34.200
dtmfmode=rfc2833
context=testing
disallow=all
allow=ulaw
extensions.conf
this is a semplified
2006 Oct 31
2
Asterisk both behind a NAT and outside at the same time
I've read a lot of the descriptions of handling NAT with Asterisk,
and the use of both the nat and canreinvite flags. I am very
familiar with Sip and NAT but have not seen an answer to the following
question.
My Asterisk server runs on a machine with two ethernets. One is
an external net, with exposed IP addresses. The other is an internal
net with natted IP addresses. Thus the server