Troubleshooting
From Taridium
| Line 1: | Line 1: | ||
| - | |||
== Using SIP Trunks == | == Using SIP Trunks == | ||
My calls work fine, but I don't hear a ringback tone. | My calls work fine, but I don't hear a ringback tone. | ||
| - | Go to '''Advanced Setup''' | + | Go to '''Advanced Setup''' → '''Configuration''' and check the '''Force ringback tone on trunk calls''' under '''Trunks'''. |
== CDR == | == CDR == | ||
Revision as of 22:28, 8 June 2010
Contents |
Using SIP Trunks
My calls work fine, but I don't hear a ringback tone.
Go to Advanced Setup → Configuration and check the Force ringback tone on trunk calls under Trunks.
CDR
No call ID displayed
There is no entry for call ID in the CDR viewer.
You might be using a custom version of the underlying Asterisk Addons package. Please make sure you use the latest taridium packages. E.g.:
[root@pbx ~]# yum install asterisk-addons
Installation
Ioncube Loader Permission Error
Failed loading /usr/local/php-ioncube/ioncube/ioncube_loader_lin_5.1.so: /usr/local/php-ioncube/ioncube/ioncube_loader_lin_5.1.so: cannot restore segment prot after reloc: Permission denied
Try disabling SELinux in /etc/selinux/config
SELINUX=disabled
To make this active you can either re-start your server or enter the following command:
[root@pbx ~]# setenforce 0
Ioncube File Not Permissioned
Example:
The encoded file /var/www/html/conf/license-XXXXXXXXXXXX.php is not permissioned for 192.168.213.124
Your license is locked to your network interface MAC address. Please contact Taridium licensing@taridium.com to reset your license.
Conferencing
Personal Conference Rooms Disabled in End-User Panel
The end-user interface states that personal conference rooms have been disabled. Go to Applications -> Features -> Feature Map and map the feature Personal Conference Rooms as the prefix you wish to use for the rooms. The extension format is *[selected number prefix] + [extension number]
Invalid Conference Room
Ipbx claims the conference room number is invalid. This is usually a problem with the conferencing timing source. Meetme conferencing requires a timing source, either in the form of a Zaptel compatible timing source or using the pseudo timing device ztdummy.
[root@ipbx ~]# service asterisk stop [root@ipbx ~]# modprobe ztdummy [root@ipbx ~]# cat /var/log/messages
and check for any errors, if you get an error message regarding the kernel version, you will have to re-compile the zaptel source for your kernel. Use our pre-compilation package zaptel-compile by entering
[root@ipbx ~]# yum install zaptel-compile
and try the modprobe command again. If there are no error you can start Asterisk again.
[root@ipbx ~]# service asterisk start
To configure zaptel to start the timing source at boot edit /etc/sysconfig/zaptel and uncomment the appropriate line.
