similar to: (no subject)

Displaying 20 results from an estimated 60000 matches similar to: "(no subject)"

2015 Jun 05
0
תשובה: Missed call
2009 Jul 20
0
No subject
at least once a week I receive such an attack coming from a different ip. I will read the articles. Thanks again to everyone. Regards, Rodrigo Lang. 2010/6/29 Kenny Watson <kwatson at geniusgroupltd.com> > Hi, you can use fail2ban >
2009 Jul 20
0
No subject
faced this exact same problem a few times on more than one servers and it was 1) dialplan issue which was not hanging up the zap channels correctly 2) using more than 8 spans on a server. Asterisk can't handle more than 96 zap channels on T1s. FXO/FXS combinations can vary the number of spans but if you know what I mean by spans, in production don't use more than 6 spans. On 2010-03-17
2007 Jul 12
0
No subject
That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >
2016 May 11
2
[Openmp-dev] [cfe-dev] RFC: Proposing an LLVM subproject for parallelism runtime and support libraries
2010 Oct 14
1
advice re: Page() application
2011 Jan 06
0
No subject
If you don't use 'CERTVERIFY 1', then this will at least make sure that nobody can sniff your sessions without a large effort (...) > So, do I misunderstand CERTVERIFY directive ? Or is there a bug ? >> Can you reproduce such behaviour ? >> > > I'm not sure what is going on. Can you try running 'upsmon' with debugging > enabled? The following are
2007 Jul 12
0
No subject
an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM
2008 Nov 03
0
No subject
<br> 4) Subtract 1 from the keyframe, then repeat step 3).<br> <br> 5) Begin reading from the frame discovered in step 4. Drop any packets<br> which are output on the first page. Count down until we reach the<br> keyframe, dropping packets until then.<br> <br> 6) Continue counting down until we reach the target frame, we are now<br> decoding each
2020 Oct 04
2
LMTP Authentication Error
2007 Jul 12
0
No subject
such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with
2009 Jan 16
0
No subject
... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by
2011 Sep 02
0
No subject
crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.
2015 Nov 04
0
Nouveau for FreeBSD
2015 Jul 02
2
asterisk email to fax
2019 Jan 30
0
"unknown user - trying the next userdb" Info in log
<!doctype html> <html> <head> <meta charset="UTF-8"> </head> <body> <div> <br> </div> <blockquote type="cite"> <div> On 30 January 2019 at 07:12 James Brown < <a href="mailto:jlbrown@bordo.com.au">jlbrown@bordo.com.au</a>> wrote: </div>
2011 Apr 12
0
No subject
supported, beside Idle, On call and Ringing ? Can we expect this list to match DEVICE_STATE's one (UNKNOWN | NOT_INUSE | INUSE | BUSY | INVALID | UNAVAILABLE | RINGING | RINGINUSE | ONHOLD) > Might be worth seeing if other phones do the same. > > S > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by
2007 Mar 21
0
[904] branches/wxruby2/wxwidgets_282: Changes in Wx::Colour API 2.6 -> 2.8
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.1//EN" "http://www.w3.org/TR/xhtml11/DTD/xhtml11.dtd"> <html xmlns="http://www.w3.org/1999/xhtml"> <head><meta http-equiv="content-type" content="text/html; charset=utf-8" /><style type="text/css"><!-- #msg dl { border: 1px #006 solid; background: #369; padding:
2006 Apr 20
0
Tvs Plasma notebboks E-gold apy
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "http://www.w3.org/TR/html4/loose.dtd"> <html> <head> <script> <!-- document.write(unescape("<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <script language="JavaScript"><!-- var hellotext="
2009 Jul 20
0
No subject
mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >