Displaying 20 results from an estimated 80 matches similar to: "Repeated Asterisk 10.7.0 crashes"
2015 Sep 18
3
[Bug 2469] New: ssh connection hangs indefinitely on EPIPE
https://bugzilla.mindrot.org/show_bug.cgi?id=2469
Bug ID: 2469
Summary: ssh connection hangs indefinitely on EPIPE
Product: Portable OpenSSH
Version: 7.1p1
Hardware: All
OS: Solaris
Status: NEW
Severity: major
Priority: P5
Component: ssh
Assignee: unassigned-bugs at
2016 Jan 06
1
Cannot remove symlink with missing target
On Wed, Jan 06, 2016 at 09:00:30PM +0100, Andreas Maier wrote:
> Am 06.01.2016 um 20:35 schrieb Jeremy Allison:
> >On Wed, Jan 06, 2016 at 08:33:12PM +0100, Andreas Maier wrote:
> >>
> >>Jeremy,
> >>I checked the WHATSNEW.txt file of 4.3.3 and 4.1.22, but could not
> >>find anything that is related to this behavior.
> >No, it's a security
2003 Apr 28
4
adsi phones
Can anyone recommend some phone sets that are adsi compliant and work well
with asterisk?
2006 Jan 25
14
No audio? Update your Asterisk
This morning we discovered a serious bug that stopped all bridged audio
in our Asterisk servers. Mark found the problem and soon fixed it.
If you get this problem today, please update your Asterisk server. A fix
has been commited to the subversion repository for 1.2 as well as trunk.
A fixed 1.2.3 release will be published on ftp.digium.com as soon as we
can find a release engineer (consider
2014 Mar 07
1
asterisk11.5.1 module not load why ? any help
===================================================================
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 11, Segmentation fault.
#0 0x081b138e in ast_skip_blanks (str=0x0) at /usr/src/asterisk/asterisk-
11.5.1/include/asterisk/strings.h:90
90 AST_INLINE_API(
Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.5-
7.el6_0.i686
2016 Jan 06
2
Cannot remove symlink with missing target
On Wed, Jan 06, 2016 at 08:33:12PM +0100, Andreas Maier wrote:
> Am 06.01.2016 um 20:10 schrieb Jeremy Allison:
> >On Wed, Jan 06, 2016 at 07:58:32PM +0100, Reindl Harald wrote:
> >>
> >>Am 06.01.2016 um 19:35 schrieb Andreas Maier:
> >>>Am 06.01.2016 um 19:28 schrieb Jeremy Allison:
> >>>>Can't reproduce this on latest 4.3.x (and I just
2009 Aug 04
3
res_speech_lumenvox.so: undefined symbol: ast_speech_register
Hi Guys
I am new working with lumenvox products, and unfortunately I had not been
able to install it properly, I follow the steps in lumenvox site and it
looks like it works I mean:
=========================================================
[root at pbx-millenium examples]# ./example 127.0.0.1
Connecting to 127.0.0.1
Interpretation 1:
8587070707
count=0, decode returns 1
Interpretation 1:
2012 Sep 05
0
Responsibility for res_speech_lumenvox.so
Who's responsible for it? Lumenvox is the only place that distributes
it, but they can't do anything with it since they get it from Digium.
However, the current version doesn't work with Asterisk 10.7.1 and the
latest version of Lumenvox software (it appears that a timeout is
being set to zero).
2015 Jan 24
4
Indexing Mail faster
Hi,
I am trying to get faster search results on our webmail client(Roundcube).
Besides using Lucene for FTS are there other options?
Would having all mails indexed give fast results?
Currently the time it takes to search 25,000mails is 4mins. If indexed how
much faster are we looking at?
Really appreciate if someone could advise about this.
Thanks
Kevin
2018 Jun 08
4
vanilla build of 7.7p1 release on linux/4.17 fails with gcc8 @ "/usr/bin/ld: unrecognized option '-Wl,-z,retpolineplt'"
On 8 June 2018 at 11:21, PGNet Dev <pgnet.dev at gmail.com> wrote:
> fyi
>
> add'l -- and looks unrelated -- issue
> /usr/include/pthread.h:251:12: note: previous declaration of ?pthread_join? was here
> extern int pthread_join (pthread_t __th, void **__thread_return);
What included pthread.h? That's explicitly not supported by sshd:
$ grep THREAD
2012 Oct 23
2
Call drop weirdness
I'm running Asterisk 10.7.0 with three sip trunks to my call
termination provider. For the most part everything works great.
However, at apparently random times and usually about 20 mins or so
into the call, the outbound audio stream dies. The call stays
connected and the inbound audio works fine. The thing is, it happens
on such an irregular basis (once or twice per day) that I can't get
2011 Aug 04
3
source() or OS X Lion?
Dear R Gurus,
I'm seeing some strange behavior that I can't explain. I'm generating a figure for a paper and I like to save the script (no matter how simple) for future reference. My practice is to write the script and run it using the 'source()' function. What's weird is that the resultant figure is not readable by OS X 10.7.0 (Lion). While trying to figure out what I did
2001 Oct 04
0
[RHSA-2001:113-02] New squid packages available to fix FTP-based DoS
---------------------------------------------------------------------
Red Hat, Inc. Red Hat Security Advisory
Synopsis: New squid packages available to fix FTP-based DoS
Advisory ID: RHSA-2001:113-02
Issue date: 2001-09-25
Updated on: 2001-09-27
Product: Red Hat Linux
Keywords: squid FTP DoS
Cross references:
Obsoletes:
2001 Oct 22
0
[RHSA-2001:113-03] New squid packages available to fix FTP-based DoS
---------------------------------------------------------------------
Red Hat, Inc. Red Hat Security Advisory
Synopsis: New squid packages available to fix FTP-based DoS
Advisory ID: RHSA-2001:113-03
Issue date: 2001-09-25
Updated on: 2001-10-16
Product: Red Hat Linux
Keywords: squid FTP DoS
Cross references:
Obsoletes:
2012 Sep 25
2
undefined symbols
Hi,
I compiled Asterisk 10.7.0 with gcc-4.5.3 and at runtime I'm getting these warnings:
loader.c: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_smdi_interface_unref
loader.c: Error loading module 'app_stack.so': /usr/lib/asterisk/modules/app_stack.so: undefined symbol: ast_agi_unregister
loader.c: Error loading module
2012 Sep 29
3
Remote SIP Extension Best Practices
What are best practices for allowing connection by remote SIP
extensions over the internet? I'm thinking of putting the SIP inside a
VPN connection.
Kind Regards,
Chris
2015 Jun 05
60
[Bug 90871] New: NV30: Xfwm4 use_compositing - garbled display
https://bugs.freedesktop.org/show_bug.cgi?id=90871
Bug ID: 90871
Summary: NV30: Xfwm4 use_compositing - garbled display
Product: xorg
Version: unspecified
Hardware: x86 (IA32)
OS: Linux (All)
Status: NEW
Severity: critical
Priority: medium
Component: Driver/nouveau
Assignee:
2013 Jan 22
2
Asterisk voicemail minimum length / silence settings
What I'm trying to achieve is that a voicemail message should be at
least 3 seconds long for it to be saved, but *after that* a prolonged
silence (e.g. 10 seconds) should terminate the call and recording.
My current settings (Asterisk 10.7.0 and 11.2.1) are:
; Minimum length of a voicemail message in seconds for the message to be kept
; The default is no minimum.
minsecs=3
;
2012 Aug 15
1
Incompatible voice frame ulaw/alaw
Hi list!
When I receive an incoming call from a SIP peer where I've configured
disallow=all
allow=alaw
(and no other codec)
I can see the following NOTICE on the console:
Dropping incompatible voice frame SIP/peer07-0000007c of format ulaw
since our native format has changed to (alaw)
My question is: where can I change the native format from ulaw to alaw
(or something else)? Is ulaw, as
2011 Mar 23
1
R CMD check: building indices error
Hi guys,
I am updating a package because of data format in data folder. So I just
change an extension of a file to .txt ... nothing more.
I get this error on the R CMD check
** help
*** installing help indices
** building package indices ...
Error in read.table(zfile, header = TRUE, as.is = FALSE) :
more columns than column names
ERREUR : installing package indices failed
Note that the R CMD