Category: Elastix

How to configure BSNL SIP Trunk on Asterisk

How to configure BSNL SIP Trunk on Asterisk or Issabel

BSNL now provides SIP Trunk with DIDs  in genreal they call it SIP PRI  and asking you to buy channels as they were selling copper  PRI links .Now they are providing same service with fiber optice cables on FTTH connections.

They are giving fiber link as same they are providing FTTH  internet services  with ONT and providing PRI service through SIP Protocal over the VPN  using OPEN VPN or SoftEther .

Depend on your convinent they are asking which VPN service you will use . Here I am using OPENVPN. But before implementing SIP PRI from BSNL as I was thinking it  can be  use  over OPENVPN from any internet network .

but they have played the game that there OPENVPN server can be accessed by only there network they have provided and sip server can be access only over VPN connection .I dont know  why they have used OPENVPN  if you are using IP based security and SIP authentication and SIP registration . they have just made it complicated if your Applicance does not support OPENVPN then you have to used additional device or router which support OPENVPN.then you can reach there SIP Proxy server.

Now below are the breif  steps to configure BSNL SIP trunk

BSNL SIP PRI
BSNL SIP PRI

 

Continue reading How to configure BSNL SIP Trunk on Asterisk

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

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

How to configure Cisco Phone 7960 or 7940 with Elastix

How to configure Cisco Phone 7960 or 7940 with Elastix

Cisco Phone 7960 or 7940 comes default with SCCP firmware you need to change the firmware to SIP .

After the SIP firmware update you need to configure these phone to work with Elastix .

Cisco Phone 7960 or 7940 does not support XML configuration file it support cnf  configuration file.

If my Phone MAC address is 0014A9713F98 then my configuration file name should be SIP0014A9713F98.cnf and place it in to Elastix server /tftpboot/ folder

 

I am giving simple example to configure one Extension . You can change according to your requirement.

Continue reading How to configure Cisco Phone 7960 or 7940 with Elastix

How to Configure Cisco Phone 7912G with Elastix

How to Configure Cisco Phone 7912G with Elastix

Cisco Phone 7912G comes with SCCP firmware and you have to convert it to SIP .

The easy way to convert this phone in to SIP .

1- Download the SIP firmware from Cisco or  google it and get it.

SIP Firmware Version: 8.0001
Filename: CP7912080001SIP060412A.tar

2- Extract the file and upload to Elastix   server /tftpboot/    folder

CP7912080001SIP060412A.sbin     and    gkdefault.cfg

Continue reading How to Configure Cisco Phone 7912G with Elastix

How to troubleshoot CallerID issues on a FXO

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.

Requirements

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

Troubleshooting Steps

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.
Continue reading How to troubleshoot CallerID issues on a FXO