there is a time synchronization problem with your system, please tell your system administrator

there is a time synchronization problem with your system, please tell your system administrator

Vicidial is having issue with time synchronization when agent is login phone is ringing and giving error as above .

In my case server time and agent machine time was showing  correct.

Vicidial works on meetme application on asterisk and  dahdi

1- check meet me installed and working

  • go to your installation source folder mostly  /usr/src/
  • go to asterisk installation folder  cd asterisk-13.38.1/  ” you choose your folder”
  • run “make menuselect”
  • check under application  > Extended > app_meetme  should look like below

Continue reading there is a time synchronization problem with your system, please tell your system administrator

How to configure Reliance Jio SIP trunk on asterisk

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 .

Continue reading How to configure Reliance Jio SIP trunk on asterisk

Protect your Asterisk PBX server from Black listed IP address

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.

Continue reading Protect your Asterisk PBX server from Black listed IP address

Extension information saved in Elastix

Extension information saved in Elastix

cat /etc/asterisk/sip_additional.conf

 

[root@pbx asterisk]# cat /etc/asterisk/sip_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make    ;
; custom modifications, details at: http://freepbx.org/configuration_files       ;
;--------------------------------------------------------------------------------;
;


[2235]
deny=0.0.0.0/0.0.0.0
secret=P@ssw0rd
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
nat=yes
port=5060
qualify=yes
qualifyfreq=3600
transport=udp
avpf=no
icesupport=no
dtlsenable=no
dtlsverify=no
dtlssetup=actpass
encryption=no
callgroup=
pickupgroup=
dial=SIP/2235
mailbox=2235@device
permit=0.0.0.0/0.0.0.0
callerid=Gkhan <2235>
callcounter=yes
faxdetect=no

 

How to configure Cisco Phone 7911 with Elastix

How to configure Cisco Phone 7911 with Elastix

Cisco Phone 7911 looks for  term06.default.loads  file to upgrade in to SIP firmware.

Download the SIP firmware from cisco or google it and get the firmware files

Configure your TFTP server and upgrade SIP firmware.

For SIP configuration this phone support XML configuration file . SEPMACADD.cnf.xml

Continue reading How to configure Cisco Phone 7911 with Elastix