how to configure sip trunk between cme and issabel or how to configure sip trunk between cme and elastix. Some time we require sip communication between Cisco CME and Asterisk…
How to reset FreePBX admin password If you forget the freepbx admin password and you have root ssh access you can reset the admin password . 1- Access your login…
FreePBX firewall commands fwconsole firewall disable fwconsole firewall stop fwconsole firewall start fwconsole firewall restart fwconsole firewall trust www.google.com fwconsole firewall untrust www.google.com fwconsole firewall untrust 192.168.0.1/24 fwconsole firewall list…
You do not have permission to be here: |6666|GOOD|
above error comes due to by mistaken you have change the value under user properties
API Only User to 1
it should be as below
you have to change it from database
SSH and login to your server
Now change it from database as below
How to configure Reliance Jio SIP trunk on asterisk
Reliance Jio provide SIP E1 trunks with DIDs . For old PBX they give you SIP to E1 converter which give you RJ45 connector to connect your E1 port.
Jio put a fiber cable and terminate on ONT from there they provide a network CAT 6 cable to put in to your network switch or directly to your IP PBX network interface not E1 interface .
Reliance jio provide SIP E1 trunk on there own IP address which will be configured on your asterisk server or Grandstream UCM or any IP IPBX .
How to configure GoIP GSM SIP VOIP Gateway with ISSABEL PBX
1- Login in to GoIP
GoIP default IP address is 192.168.8.1 default user name is admin password is admin .
Step 1- Create trunk in the ISSABEL PBX to configure GoIP
Protect your Asterisk PBX server from Black listed IP address
VoIPBL is a distributed VoIP blacklist that is aimed to protects against VoIP Fraud and minimizing abuse for network that have publicly accessible PBX’s.
For more details http://www.voipbl.org/
For Asterisk PBX you need to install Fail2ban. This is the only required dependency needed to run VoIP Blacklist on your server.
How to troubleshoot CallerID issues on a FXO
The Digium® X100M FXO module and X400M FXO module are daughter cards that allow Digium® analog cards to terminate analog telephone lines (POTS).
This document is intended to be a brief description of toublehsooting steps that Digium’s customers can take to resolve Caller ID issues on their Asterisk Server.
At least one analog line.
A regular analog phone with Caller ID display feature
Fully configured Asterisk server with the latest version of DAHDI and Asterisk
A Caller ID issue could be caused by several factors. This section explores the possibility that the issue could be located at the telco or by noise on the lines.
Connect an analog phone to the demarcation point. In telephony, the demarcation point is the point at which the telephone company network ends and connects with the wiring at the customer premises.
Place a call — using another line or mobile phone — from the PSTN to the analog phone connected at the demarcation point and check if the phone displays Caller ID. If the Caller ID information is not being shown by the phone, please contact your telco and ask them how to enable this feature.
Otherwise, please disconnect the phone and plug it to the line that is directly connected or closest to your Asterisk server and repeat the test call.
How to use another sip port on Elastix or Asterisk
I was trying to configure Elastix PBX SIP client but the SIP port UDP 5060 was blocked from the ISP .
I don’t wanted to change server SIP port 5060 . I google the internet and get the trick and tested and
its worked for me .
Hope it will work for others also
How to configure Elastix Virtual FAX
Elastix is having inbuilt FAX services as HylaFAX . which provide a facility to send and receive fax through web or fax client software or email .
To configure Fax in Elastix you have to configure Virtual Fax , IAX extension, Email services ,Fax Client List.
1- Login in to the Elastix Server Click on Fax > New Virtual Fax
2- Fill all the information