How To Setup CHAN SIP Trunk4 min read
FreePBX is an open source IP Telephony system. If you can use home and office for communication. At first install FreePBX on Ubuntu 14.04. After installation completed then setup CHAN SIP TRUNK on your server. Browse your FreePBX server via any browser. Now you follow this step by step configure CHAN SIP TRUNK.
★ How To Setup CHAN SIP Trunk
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Step #01: Browse your FreePBX server IP and type login credential created after installation.
Click Connectivity / Trunks (Drop down position 4) .
Appear on show This page is used to manage various system trunks.
Then click +Add Trunk and choose drop down +Add SIP (chan_sip) Trunk.
Step #02: You can see three tabs such as General, Dialed Number Manipulation Rules and sip Settings.
General Settings:
Trunk Name : Any Name Here Outbound CallerID : Your VOIP Number CID Options : Allow Any CID Maximum Channerls : N/A Asterisk Trunk Dial Options : N/A Continue if Busy : Must Be No Disable Trunk : Must Be No
Step #02: Dialed Number Manipulation Rules:
These rules can manipulate the dialed number before sending it out this trunk.
If no rule applies, the number is not changed. The original dialed number is passed down from the route where some manipulation may have already occurred. If the number matches the combined values in the prefix plus the match pattern boxes, the rule will be applied and all subsequent rules ignored.
Upon a match the prefix, if defined, will be stripped. Next the prepend will inserted in front of the match pattern and the resulting number will be sent to the trunk. All fields are optional.
Rules:
X – matches any degit from 0-9
Z – matches any digit from 1-9
N – matches any digit from 2-9
Step #03: You can two tab Outgoing and Incoming.
- Outgoing Settings:
Trunk Name : Use Any Nmae
PEER Details :
host=SIP server IP [Provided From SIP provider]
username=VOIP number [Provided From SIP provider]
secret=Connected crediantials [Provided From SIP provider]
type=peer
nat=yes
insecure=very
- Incoming Settings:
USER Context : from-trunk
USER Details :
secret=Connected crediantials [Provided From SIP provider]
type=peer
context=from-trunk
nat=yes
insecure=very
quality=yes
Step #04: Register String setup example below:
UserName:Password@SIP_Server_IP_Address
Now click Submit. Then click Apply Config
Final Step: Goto your server command line ant type asterisk -rvvvvvv
FreePBX*CLI> sip reload
If you can’t see any error massage on your astersik cli then you confirm that, you asterisk server connected on SIP provider service.
If face any problem then see my YouTube video and Subscribe my channel. Please subscribe my channel for more update.
Thanks to all.
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