similar to: What is bootstrap.sh for ? Possible bug in 11.3.0 ?

Hello list, I have explored https://github.com/networkupstools/nut repository and found that Riello UPS added into list of supported UPS. But current package for most distribution it's nut-2.6.5 which doesn't have Riello's models. Do we need to wait next upcoming nut release to get start with Riello? I tried to install from source but stuck: # git clone

Hi, <p>the following steps are necessary to successfully compile Icecast2 (as of today) on OpenBSD. This workaround is based on Karl Heyes' findings that _XOPEN_SOURCE is the root of all evil, at least on OBSD: 1. "autoconf version problems" OpenBSD 3.2: change all "autoconf" in autogen.sh to "autoconf-new" OpenBSD 3.3: $ export AUTOCONF_VERSION=2.52 2.

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

The Asterisk Development Team has announced the release of Asterisk 11.3.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello, I'm facing a rights issue on with Asterisk 11.3.0 running on CentOS release 5.8. Asterisk process is running with asterisk since it is define in asterisk.conf as following: runuser = asterisk rungroup = asterisk You can see asterisk proccess here: ps aux |egrep 'python|asterisk' root 11581 0.0 0.1 65940 600 ? S Apr17 0:00 /bin/sh /usr/sbin/safe_asterisk

Anyone trying openssl 3 against openssh? -- Member - Liberal International This is doctor@@nl2k.ab.ca Ici doctor@@nl2k.ab.ca Yahweh, Queen & country!Never Satan President Republic!Beware AntiChrist rising! https://www.empire.kred/ROOTNK?t=94a1f39b Put more trust in nobility of character than in an oath. -Solon

> > from source with no problem on Linux, but on my OBSD system the configure > bombs > > while checking for a function in libxslt... > > If you look at your config.log, you'll find the actual error to be > > ld: -lpthread: no match > > If you want threads on OpenBSD, -pthread is the way to go (it's a > wrapper saying "Do whatever you have to,

Dear All, I'm trying to connect to Asterisk CDR database using PHPMyAdmin but unfortunately all my trials and searches failed. So I'd be more than grateful if someone helped me with right steps to do this. Kindly note that I'm working on a remore server that I can connect to as a root using *ssh.* Asterisk Version: 11.3.0 MySQL Version: mysql-server.x86_64 0:5.1.69-1.el6_4

Building on Mac OS X doesn't work. It fails with this error: configure: error: No 16 bit type found on this platform! I've posted the config.log here: https://gist.github.com/1878663 I think something is going wrong with how the configure script is detecting the platform. Thanks, aaron -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

*I'm trying to build an application that provides statistics of calls*>* and call recording. Someone told me this could be done out of band*>* with a SPAN (?) port that would replicate SIP and media packets to a*>* separate NIC without having to actually pass the real-calls thru*>* asterisk. It was explained that this SPAN port would in the SBC*>* would replicate data

Hello, I spent a bit of time to update my Kamailio-Asterisk realtime tutorial to latest stable versions in both sides. The tutorial is available at: - http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I tried to use default names for asterisk database tables, where the structure was not changed, and different names for those that are a bit customized, in order to

Hello, I am beginning to front my Asterisk cluster with OpenSIPS/Kamailio and so far my biggest issue is the complete lack of quick-start-like documentation for either. Is there any place I can get a very simple HA configuration (telling me where the config files are, for starters, is a good thing) for OpenSIPS or Kamailio with the following features: (a) Support an arbitrarily large number of

Hi all Have recently watched Matt Jordan's session on Kamailio World 2014 On slides 26-29 of his presentation (http://www.kamailio.org/events/2014-KamailioWorld/day1/09-Matt.Jordan-Asterisk12-And-PJSIP.pdf) he speaks about a (completely new, for me at least) approach to build scalable telephony systems, using N instances of Kamailio and N instances of Asterisk Are there any

Hi Steve, Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. On Wed, 20 May 2020 at 18:37, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Wed, 20 May 2020, David Cunningham wrote: > > > We have an Asterisk

I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration (MySQL database) so that kamailio authenticates and then forwards the registration to asterisk on localhost. The setup calls for asterisk to be

Arch = x86_64 OS = CentOS-6.3 (FreePBX) Asterisk = 11.3.0 (FreePBX) Hylafax+ = 5.5.3 (epel) I am installing an Asterisk server with the intent of having it act as our fax server. The Asterisk system was installed using the current FreePBX distro which is reputedly based on CentOS-6.3. I have installed Hylafax+ from epel and I have built and packaged IAXModem-1.2.0 for CentOS-6 using mock and

I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with "this is a known

I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asterisk runs on localhost and only listens on SIP/RTP at 127.0.0.1 . Therefore, all of the SIP traffic appears to

I am experimenting with Asterisk having downloaded and installed the FreePBX i386 CentOS-6.3 based distro and updated it. The current package level on this system is: asterisk11-11.3.0-49_centos6 freepbx-2.11.0beta2-112 I am using twinkle-1.4.2-7.el6 as a softphone testing tool. There is no firewall on the asterisk host and SELinux is disabled on it. Fail2Ban is installed but I have made no