Author: reveal

Lecture and Lab: Use sample config files to build a working PBX

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CONFIG FILES:

• • sip.conf – SIP peers for extensions and carriers

• • extensions.conf – Phone system dialplan for call routing

• • voicemail.conf – Voicemail users

• • followme.conf – Route unaswered calls to a user’s mobile phone

ASTERISK CONFIG FILES

• Leave rest as default
• This config is for SIP only PBX
• No GUI interface for end users
• All configuration is done using text config files

CREATE CUSTOM RECORDINGS

• Make a custom recording for your auto attendant greetings
• Configure voicemail accounts
• Record name for voicemail account, option 3
• Record unavailable greeting for voicemail, option 1

CUSTOMIZED DIALPLAN

• Edit extensions.conf to route your inbound phone numbers
• Edit extensions.conf to set outbound caller ID numbers
• Edit followme.conf to dial alternate phone number if not answered

vi /etc/asterisk/voicemail.conf

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• Scroll down to bottom of file

Should see example imapvm same sequence sort of

• • Create new name in example:

;[internal-vm]

[internal-vm]

101 => 123456,John Doe,john-email@gmail.com

102 => 123456,Jane Doe,jane-email@gmail.com

What it means:

101 => 123456,John Doe,john-email@gmail.com ** 101 is exten 123456 is password, name, email

• What we just did was create extension 101 and 102 with password 123456 and name and email for voicemail, we will add this context to sip.conf and enable voicemail

Edit /etc/asterisk/extensions.conf

You’ll create your dialplan here to making outbound calls with your SIP CARRIER

voipms has a sample sip.conf and extensions.conf you can use and replace with your sub_account username, here is what outbound looks like:

[voipms-outbound]

exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)

exten => _1NXXNXXXXXX,n,Hangup()

exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)

exten => _NXXNXXXXXX,n,Hangup()

exten => _011.,1,Dial(SIP/${EXTEN}@voipms)

exten => _011.,n,Hangup()

exten => _00.,1,Dial(SIP/${EXTEN}@voipms)

exten => _00.,n,Hangup()

Important to know that you need this also in your dialplan:

[mycontext]

; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop

include => voipms-inbound

include => voipms-outbound

Edit /etc/asterisk/extensions.conf

go to bottom add the following under [voipms-inbound]

exten => 2242179435,1,Answer

exten => 2242179435,n,NoOp(Inbound-Call)

exten => 2242179435,n,Dial(SIP/101,24)

exten => 2242179435,n,Voicemail(101@internal-vm)

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You will need to find a SIP provider and signup, I used voip.ms and dumped $25 into it for testing..

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It has support for Asterisk

It will give you the sample configuration for Asterisk sip.conf

edit /etc/asterisk/sip.conf

find the line marked [general]

under there insert your registration tag you got from the sample and replace with needed information:

register => 220450_USERNAME:PASSWORD..@chicago.voip.ms:5060

Go to bottom need to insert the SIP CARRIER information do that before EXTENSIONS line

; ## SIP CARRIERS PEERS ###

[voipms]

canreinvite=no

context=mycontext

host=chicago.voip.ms ;(one of our multiple servers, you can choose the one closer to your location)

secret=PASSWORD.. ;your password

type=peer

username=220450_USERNAME ;(Replace with your 6 digit Main SIP Account User ID or Sub Account )

disallow=all

allow=ulaw

fromuser=220450_USERNAME;(Replace with your 6 digit Main SIP Account User ID or Sub Account username)

trustrpid=yes

sendrpid=yes

insecure=invite

nat=yes

edit /etc/asterisk/extensions.conf

** ADD THE FOLLOWING TO THE BOTTOM **

[mycontext]

; Make sure to include inbound prior to outbound because the _NXXNXXXXXX handler will match the incoming call and create a loop

include => voipms-inbound

include => voipms-outbound

[voipms-outbound]

exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)

exten => _1NXXNXXXXXX,n,Hangup()

exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)

exten => _NXXNXXXXXX,n,Hangup()

exten => _011.,1,Dial(SIP/${EXTEN}@voipms)

exten => _011.,n,Hangup()

exten => _00.,1,Dial(SIP/${EXTEN}@voipms)

exten => _00.,n,Hangup()

; inbound context example for your DID numbers, do not add the number 1 in front

[voipms-inbound]

exten => 7863643011,1,Answer() ;your DID

edit /etc/asterisk/sip.conf

go to our extensions we setup 101 and 102 need to replace context=internal to context=mycontext

go to asterisk -rvvv

reload

sip show peers

sip show registry

You can dial 4443 from your SIP device to start the echo test

Need to make changes for this to work you will see some errors

[Aug 30 14:59:40] NOTICE[2099][C-00000005]: chan_sip.c:25503 handle_request_invite: Call from ‘101’ (172.16.24.173:58413) to extension ‘4443’ rejected because extension not found in context ‘mycontext’.

edit /etc/asterisk/extensions.conf

add the following

exten => 4443,1,Answer

exten => 4443,n,NoOp(Echo-Test-4443)

exten => 4443,n,Dial(4443@voipms)

open asterisk -rvvv

reload

dial again, get new error

then open /etc/asterisk/extensions.conf

edit the following

Aug 30 15:02:10] WARNING[2223][C-00000006]: app_dial.c:2330 dial_exec_full: Dial argument takes format (technology/resource)

== Spawn extension (mycontext, 4443, 3) exited non-zero on ‘SIP/101-00000006’

exten => 4443,n,Dial(4443@voipms) ** change to exten => 4443,n,Dial(SIP/4443@voipms) **

asterisk -rvvv

reload

dial and it’s a success!

Configuration for a Linksys router settings

• Single Port Forward: SIP port 5060 UDP to Server IP address
• Port Forward Range: Media ports 10,000-20,000 UDP to Server IP address
• Remote SSH Terminal: SSH port 22 TCP to Server IP address

Disable ALG Setting

Basic troubleshooting issues you might experience in Asterisk

— Executing [103@internal:1] Answer(“SIP/101-00000003”, “”) in new stack

— Executing [103@internal:2] NoOp(“SIP/101-00000003”, “Internal-phone-call”) in new stack

— Executing [103@internal:3] Dial(“SIP/101-00000003”, “SIP/103,12,r”) in new stack

1. [Aug 30 11:43:24] WARNING[2128][C-00000002]: chan_sip.c:6071 create_addr: Purely numeric hostname (103), and not a peer–rejecting!
2. [Aug 30 11:43:24] WARNING[2128][C-00000002]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 – Subscriber absent)

== Everyone is busy/congested at this time (1:0/0/1)

— Executing [103@internal:4] VoiceMail(“SIP/101-00000003”, “103@internal-vm”) in new stack

3. [Aug 30 11:43:24] WARNING[2128][C-00000002]: app_voicemail.c:6321 leave_voicemail: No entry in voicemail config file for ‘103’

1 This error occurred because the extension/user was not defined in sip.conf

2 This error occurred because the phone does not exist in sip.conf
1 This error occurs because a voicemail is not defined in voicemail.confView mailbox users, voicemail show users ** lists users from /etc/asterisk/voicemail.conf with voicemail accounts**