smarks
Guru
- Joined
- Jan 7, 2015
- Messages
- 116
- Reaction score
- 26
Yes I am looking that that one as well as xlog
http://kamailio.org/docs/modules/1.6.x/xlog.html
http://kamailio.org/docs/modules/1.6.x/xlog.html
Thanks for the lesson, smarks. The core show config mappings and realitme load sippeers name_voipms both work. Everything else is empty. I suspect the culprit is in creating any new organization. You get "Organization already exits" regardless of how random the data happens to be. This in turn means Elastix doesn't create dialplan for new organization. I suspect the trunk info works only because I embedded the data in sip.conf first and reloaded Asterisk.
I might add that we couldn't even get extensions to call each other until I noticed that extensions built as part of a user left half the settings out in the Asterisk DB. Once I shoved those entries in there manually (see installer for the code), calls started working at least between the extensions.
Heaven help you if you try to actually delete something. It leaves pieces all over the place in the MySQL tables.
So... things are still a bit of a mess unless I've missed something.
#!WITH_DEBUG
log_stderror=yes
xlog("The value of auth realm pseudovariable at this point is currently $au\n")
smarks: host= entry inserted where??
Realtime SIP friends
The SIP realtime objects are users and peers that are loaded in memory when needed, then deleted. This means that Asterisk currently can't handle voicemail notification and NAT keepalives for these peers. Other than that, most of the functionality works the same way for realtime friends as for the ones in static configuration.
tomodparam("auth_db", "use_domain", 1)
modparam("auth_db", "use_domain", 0)
service kamailio restart
line 91 centos=x86_64 to i686
line 317 change x86_64 to i386
line 435 change to http://li.nux.ro/download/nux/misc/el6/i386/knock-server-0.5-7.el6.nux.i686.rpm
Try run this script.
/usr/sbin/elastix-setup-kamailio-rtpproxy
Go through and check that it is setting the IP address correctly in /etc/kamailio/kamailio-mhomed-elastix.cfg and /etc/sysconfig/rtpproxy-multi
Hello,
I had same issue, I spend a lot of time for checking. The problem was that kamailio wanted authentification for incoming calls even if in Elastix 3.0 tried to change a lot of parameters for insecure etc. But as I suppose these parameters are only for asterisk not kamailio. I also tried to look at settings of kamailio, but I didnt find nothing. But that I tried to do an upgrade and after a while it started to work, but there are some bugs after upgrade, so I describe steps how to:
Login to SSH:
1. before update save to your pc file from /var/www/elastixdir/asteriskconf/elastix_pbx.conf
2. do "yum update" - takes long time to proceed
3. do "reboot"
4. After reboot you cannot connect to management WEB it says error connection to DB.
5. login to SSH again
6. find the file : /var/www/elastixdir/asteriskconf/elastix_pbx.conf
7. check it with file saved in step one - you should correct the DBPASSWORD, MGPASSWORD, maybe more.
8. you are done.
After these steps should work the SIP Trunk inbound calls without 407 Authentification required.
I'm going to try the build yourself from here: http://pbxinaflash.com/community/index.php?threads/gotcha-free-pbx-elastixmt-gui.17055/ seeing as centos 6.7 was released yesterday.
Incredible PBX 11.13-02 for Elastix 3.0
Asterisk: UP Apache: UP MySQL: UP
Postfix: UP IPtables: UP SSH: UP
LAN port: UP Fail2Ban: DN Webmin: UP
Kamailio: DN IAX Modem: DN HylaFax: UP
RAM:160MB CentOS 6.7 (i686) Disk:5.6G
Asterisk 11.13.0 Elastix MT: 3.0
Private IP: 168.235.149.68
Public IP: 168.235.149.68
System Time: Sat Nov 21 16:54:52 EST 2015
< OK >
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