Some friends and I want to leave WhatsApp for good and as I've been playing around with my Rpi4 for years now, I thought I'd use that to set up an XMPP-Server for us to use, to have a secure and customizable experience.
I've tried it with ejabberd and installed it from the repo. I also tried to compile the newest version, but I get an error output every time.
It works fine so far, but it differs from what I've seen online.
My problem is, that registration from outside doesn't work at all. I can register from ejabberdctl, but that's it. On the tested clients it says "server doesn't allow registration", although I have enabled everything that says anything about registration allowance.
System: Raspberry Pi 4 (4GB)
OS: TwisterOS 1.9.6 (Based on Raspbian 32bit)
Ejabberd Version: 18.12.1-2
Config files:
Ejabberd.yml https://textuploader.com/18mqx
Ejabberdctl.cfg https://textuploader.com/18mqz
When the "mod_register" line is enabled/uncommented (or any other not by default enabled mod), ejabberdctl live gives following output: [error] Cannot load /etc/ejabberd/ejabberd.yml: Syntax error on line 272 at position 2: did not find expected key
I am very thankful for any help, whether it's getting my current version to allow registration or compiling the current version which seems easier to work with, as all of the solutions I've seen on the internet won't work with my version.
Kind Regards,
Arokan
I figured it out. You won't believe it...
The prewritten spaces in
" ## mod_register: "
seem to not have been spaces but something else, though they weren't wide as tabs but exactly as spaces.
So I deleted the spaces before mod_register: and hammered the space key myself. After that, everything worked out perfectly and other modules do as well in the same way.
tl;dr: Replace the pseudo spaces with own spaces.
Related
I want to enable logging of MAC adresses as well as hostnames using Bro. I have been using Bro for a while, but I am still a bit new to it.
Version: Bro 2.5.1
From researching this a bit, I found that I can log this by enabling policy/protocols/dhcp/known-devices-and-hostnames.bro but for this, I also need to enable policy/misc/known-devices.log.
This will then log to a devices.log.
Now the problem I am experiencing is that from those files, is that this will only be logged once a day (by default).
I need this to be logged more frequently (as soon as there is the specific connection, I want that connections mac address and hostname to be logged. Is this possible?
If possible, do I need to change the defaults and where?
Or did I possibly miss something?
Try a redef of Log::default_rotation_interval (docs here) to something that suits your need. If you're running a cluster, consider the LogRotationInterval in BroControl.
I am trying to add Apache Nifi in ambari but continuously failing with error Error occured during stack advisor command invocation:
Unable to delete directory /var/run/ambari-server/stack-recommendations/1.
There is a similar thread with the same error in hortonworks community, I have tried everything mentioned in that thread but unable to fix it. My sandbox is installed in vmware workstation 12 player. I also tried to create and remove directory manually but it is failing with the error invalid argument. Created a thread for this error also on stackexchange. Please help!!!
Here is a link to Hortonworks forum thread. And it seems like sandbox is just broken:
This is due to a docker issue in this 2.5 sandbox build. It will be
fixed in next revision of the sandbox.
There are also some workarounds described (like use older HDP 2.4 or establishing own cluser based on the HDP 2.5 docker image)
Updated sandbox arrived: http://hortonworks.com/downloads
Trust me, active member of community see your posts in multiple locations. In a good, no Big Brother ways :) but cross-posting is an old as world ... Well, you got it.
Did you see a notice for this service in Ambari? Telling it's been deprecated? Same note in the github. There's a good reason for that, it's now been implemented properly by the dev team and with many more features. I.e. all the action is there now.
I think I replied a similar question, though not sure it was yours, take a look in HCC.
I'm trying to get asterisk 11.20.0 running with WebRTC (sip.js 0.72 which I believe is a fork of jssip), but I'm seeing the following (and the called party rings, but when the phone is answered the call gets hung up).
This is my setup:
What I see:
In the CLI:
[2015-11-24 01:01:53] NOTICE[43619][C-00000002]: res_rtp_asterisk.c:4441 ast_rtp_read: Unknown RTP codec 95 received from '(null)'
In Firefox:
InvalidSessionDescriptionError: Invalid description, no ice-ufrag attribute
Attachments:
SIP Dialogue (Asterisk CLI)
Webphone Log
Config Files (httpd.conf, sip.conf, rtp.conf)
Asterisk Compiled with Libuuid & Friends
What I've tried so far:
Changed webRTC implementations (tried chrome and firefox both with SIPML and SIP.JS)
Set the STUN server to null on the client side (stunServers: ['stun:null'])
Configured properly (I hope) my sip.conf and rtp.conf and httpd.conf
Made sure I have libuuid, uuid and their -devel companions and after i've recompiled asterisk.
What I've read:
http://forums.asterisk.org/viewtopic.php?p=201702
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
http://jssip.net/documentation/misc/interoperability/asterisk/
http://sipjs.com/guides/server-configuration/asterisk/
https://kunjans.wordpress.com/2015/01/09/web-sip-client-sipml5-with-asterisk-13-on-centos-6-6/
http://forums.digium.com/viewtopic.php?f=1&t=89798
Please, if you can, give me a hand. I'm about to smash my box with a sledge hammer.
Faced same issue and followed instruction in http://forums.digium.com/viewtopic.php?f=1&t=90167 realise that:
This issue is caused because you asterisk don't have ICE support, you can solve that by installing the uuid/libuuid and uuid-devel/libuuid-devel packages on your system. Then recompile asterisk(be sure to rerun the configure script before the make command).
I did recompile my Astersik 11.16.0 with patch for ECDH support and fallback to prime256v1 https://issues.asterisk.org/jira/browse/ASTERISK-25265 and looks like lost uuid support at that time. Reverting back to non-patched version (with uuid support, use to be compiled before) resolved my issue with "no ice-ufrag attribute" error in Firefox console and calls are going well now from WebRTC client SIPML5 based to asterisk, but not in opposit direction
How to connect to MySql using Connector/c++ over SSL?
I read about mysql_ssl_set(): seems to work only with Connector/c.
I have openSSL enabled and working for require SSL users on MySql. And I can connect remotely to MySql using Connector/c++, but cannot see how to modify parameters of sql::Connection to reference certs for SSL.
Excellent tutorial from Giri Mandalika at MySql DevZone and docs on Connector/c++ don't mention SSL, which surprised me because SSL is well documented for Connector/c...or did I miss something obvious?
I realize sql::Driver::Connect() is overloaded, with the less-commonly discussed method taking one parameter sql::ConnectOptionsMap &options. Following that trail, I found just one discussion on point: Ulf Wendel/Thomas Pollak question-and-solution here (http://forums.mysql.com/read.php?167,367519,369534#msg-369534).
Did not work for me: when I incorporated it and tried to build (VC2008, WIN7) the build had warnings and failed.
I traced the exact causes as:
sql::ConnectPropertyVal generates locally defined symbol warning; and
std::map causes fatal error LNK1120: 1 unresolved externals.
With so many users out there of Connector/c++, I hoped others in the community might also be interested in how to use SSL with Connector/c++. Appreciate any ideas or help.
Spoke to Giri Mandalika (on the original development team), who mentioned that SSL support was originally planned but not completed for Connector/c++.
In case others who want SSL find this question, I worked around the problem only by using Connector/c, the underlying C api which Connector/C++ was designed to wrap.
As more than six months have elapsed without any community member coming up with a specific solution to use SSL with Connector/c++ (other than the overload solution referenced in my question), I guess I have to conclude that it just isn't possible, and mark my own question as "Answered" by me.
I have tried for days now trying to find the right version of red5phone, but to no avail. Also, I need red5 and asterisk to be on different servers. I have followed all instructions described in the various tutorials on web, but nothing helps. I have downloaded 2 different versions of red5phone: sip1, sip_47 from the red5phone google code site, but none of them worked!
When I use:
a) sip1 - shows the correct parameters being passed on the red5 server console, but connection is stuck up with the console displaying the following error:
[NioProcessor-1] ERROR o.r.server.service.ServiceInvoker - Method login with parameters [<sip user>, <sip user>,<sip user>,<sip user pwd>, <asterisk server i/p>, <asterisk server i/p>] not found in org.red5.server.webapp.sip.Application#2d0c94a7
b) sip47 - when I type in the values in the flex interface and check the red5 server console, I see all parameters correct(the ones I passed) except for asterisk server ip that i m trying to connect. Instead, it shows 127.0.0.1 by default and completely ignores the passed ip and regsitration fails.
I am using:
red5 server version: 0.9.1, centos: 4.8(final), red5phone used: sip_47 (tried sip1 as well)
As a desperate measure I tried debugging the source code for red5phone (java and flex files) myself. but when I try to create the environment on my local system, I get several compile errors for missing java packages such as javax.media, org.slf4j, org.red5, etc. really confused and desperate for some guidance. Any tips highly appreciated.
Sunil, I'm also new to this, I would try red5-voicebridge installed in your red5 server.
red5-voicebridge Please let me know if you got it to work.