As I am trying to install new imc module to my kamailio sip server, I also followed the documentation of it. But unfortunately, I could not run kamailio sip server.
The error which is coming while running it is on this line:
modparam("imc","db_mysql","DBURL")
Where
DBURL = "mysql://kamailio:kamailiorw#localhost/kamailio"
The errors are like syntax error and invalid arguments
What could be the possible chances of this kind of error?
In last what can be the best imc testing software or client?
Please help.
You've set db_mysql instead of db_url while defining a module parameters. please change it to.
modparam("imc","db_url", DBURL)
Also make sure you're loading this module after db_mysql and tm module.
http://www.kamailio.org/docs/modules/3.2.x/modules_k/imc.html#idp1754096
And as it follows the architecture of IRC channels you can use many IRC chat clients to connect such as Pidgin, mIRC and Icechat for Windows, Limechat for Mac OS X, Irssi, WeeChat or XChat for Unix-like systems, ChatZilla for Mozilla browsers or any other standard IRC program.
Related
I'm trying to implement a basic audio/video-chat functionality through WebRTC by means of ejabberd in JavaScript. To do it, I'm using the Stanza library which implements the following protocols:
XEP-0166: Jingle
XEP-0176: Jingle ICE-UDP Transport Method
And from the docs I know that Ejabberd supports Jingle ICE (XEP-0176). https://docs.ejabberd.im/admin/configuration/#stun-and-turn
When I try sending a Jingle "session-initiate" stanza, as described in XEP-0166, it throws the following error:
<error code='503' type='cancel'>
<service-unavailable xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'/>
<text xml:lang='en' xmlns='urn:ietf:params:xml:ns:xmpp-stanzas'>No module is handling this query</text>
</error>
What version of ejabberd are you using?
18.1
What operating system (version) are you using?
Centos 7
How did you install ejabberd (source, package, distribution)?
Source
If needed, I can post here my installation steps and a config.
Please, advise on how to proceed further with my implementation?
are you trying to send the session-initate to a bare jid (user#host) instead of a full jid (user#host/resource)? The former will be handled by the server and it is quite unlikely that it supports accepting calls that way...
I am setting up DR.Racket for SICP. By following this guide
http://docs.racket-lang.org/sicp-manual /index.html?q=sicp#%28part._.Introduction_to_the__lang_sicp_language%29
But I am not able to download sicp package. Following is the error message I recieved after hitting install.
Resolving "sicp" via https://download.racket-lang.org/releases/6.11/catalog/
ssl-connect: connect failed (error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed)
Please explain what to do?
Apologies if I'm telling you things you already know: it appears that certificate validation failed while contacting the racket package server.
There are several things that could cause this, but I have a sneaking suspicion that you have some kind of proxy standing between you and the server that is rewriting your traffic. Is this possible?
One way to test this would be to try installing the package on a different computer, connected to a different network.
(FWIW, I just tested this installation myself, and didn't encounter this problem. So, at a minimum, it's not that the world is broken :).)
I 'm working on esp8266 module these days.
The firmware on this module is nodemcu 0.9.6 and its language is lua.
I use mqtt protocol for my project and it works well.But TLS is necessary.
I don't know how to set tls on this firmware. I 've searched but documents about TLS on nodemcu website is not enough for me.
I know that I should use this instruction at connection time to broker:
m:connect(BROKER, 8883, 1 , function ( connection ) ... end )
and to put the CA file on module i should use something like this:
enter image description here
that 's what i read from nodemcu website --> https://nodemcu.readthedocs.io/en/master/en/modules/net/#netcertverify
but the problem is when i use this on my code the module doesn't understand it and gives error at upload time and runtime.
is this problem because of the version of my firmware? Upgrading will make it? Or I am just doing something wrong on my code?
I haven't seen sample code about putting CA file on module in other project so I guess my work is wrong.
if you have worked on TLS ( on nodemcu firmware ) I would be happy by your helps.
Build a recent firmware with SSL/TLS enabled.
Flash the binary to upgrade. Make sure all flashing parameters are exactly as documented and include the init data binary.
The rest of your code seems fine.
Take a look at this question and read through comments of the answer.
You may have to get esp_init_data_default.bin from latest sdk.
I used esptool.
esptool.py --port <PORT> write_flash -fm dio -fs 32m 0x00000 <BIN_PATH_AND_FILENAME> 0x3fc000 <PATH\>esp_init_data_default.bin
Please do upvote this post if you find it is useful. Thumbs up for #MarcelStör.
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
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.