How To Register Trunk Asterisk Siptrunk.com

IP PBX Configuration - Asterisk¶

Asterisk is a popular and versatile telephony software which tin be used to deploy advanced PBX systems. SIP Trunk configuration instructions below apply to the following Asterisk versions:

• Asterisk 11
• Asterisk 13

Documentation is provided for scenario where Asterisk server uses Static IP address on the public Internet and when Asterisk server is on Dynamic IP address.

Consummate set of Asterisk configuration files described below can exist establish at https://github.com/GoTrunk/asterisk-config/

Static IP address¶

Effigy eight.1 Asterisk SIP Trunk - static IP address.

1. Static IP address (a.b.c.d in our example above) of your Asterisk server will exist added to GoTrunk service IP ACL (Admission Control List) and outbound calls coming from that IP address will be accepted without requiring whatsoever farther authentication (SIP username and password). This is the most efficient way of authenticating SIP calls.

2. Inbound calls to one of Telephone Numbers on your GoTrunk account will exist sent directly to Asterisk PBX public IP accost. Since the calls volition be coming from known peer (IP address of SIP Trunking service q.10.y.z in our example above) Asterisk volition accept them without requiring any further hallmark.

To configure Asterisk server to work with GoTrunk SIP trunk using IP authentication the following changes are required:

one. Add [body] peer definition to sip.conf file:

            [body] type=peer host=european union.st.ssl7.internet    ; Europe POP ; host=amn.st.ssl7.net ; North America POP context=from-trunk                      

2. To send outbound calls to GoTrunk SIP Trunk update extensions.conf file:

            [from-internal] ; internal calls between extensions exten => 201,one,Dial(SIP/201) exten => 201,north,Hangup() exten => 202,i,Dial(SIP/202) exten => 202,n,Hangup() ; ship all other numbers to outbound trunk exten => _[0-nine].,1,NoOp(#### [from-internal] ####) exten => _[0-9].,n,Fix(CALLERID(num)=442036427900) ; fix Caller ID exten => _[0-ix].,n,Punch(SIP/+${EXTEN}@trunk)       ; send call to trunk peer defined in sip.conf exten => _[0-ix].,n,Hangup()                      

2. To accept incoming calls from Telephone Numbers assigned to your GoTrunk account update extensions.conf file:

            [from-trunk] ; <- 'from-trunk' context defined in sip.conf exten => _[0-ix].,one,NoOp(#### [from-body] ####) exten => _[0-nine].,n,Punch(SIP/201) ; ship all incoming calls from trunk to extension 201 exten => _[0-9].,n,Hangup()                      

Note: example above is provided for analogy purposes only. Your actual dialplan volition depend on specific PBX configuration.

Dynamic IP address¶

Figure 8.two Asterisk SIP Body - dynamic IP accost.

1. For outbound calls from Asterisk PBX to GoTrunk SIP Credentials (SIP username and password) authentication is used.

2. For entering calls to one of Telephone Numbers on your GoTrunk account to piece of work Asterisk PBX needs to Register with GoTrunk service (and periodically refresh registration in case IP address changes). Calls volition be sent to IP address which was sent in the most recent Asterisk Registration. Since the calls volition be coming from known peer (IP address of SIP Trunking service q.x.y.z in our instance above) Asterisk will accept them without requiring any further authentication.

To configure Asterisk server to work with GoTrunk SIP Body using SIP Credentials authentication the following changes are required:

1. Add register and [trunk] peer definition to sip.conf file:

            [full general] ; ... ; ... ; supervene upon INBOUND_SIP_USERNAME and INBOUND_SIP_PASSWORD ; with your Inbound SIP Registration credentials annals => INBOUND_SIP_USERNAME:INBOUND_SIP_PASSWORD@trunk [trunk] type=peer host=eu.st.ssl7.cyberspace  ; Europe Popular ;host=amn.st.ssl7.net ; Due north America POP context=from-trunk qualify=yes defaultuser=OUTBOUND_SIP_USERNAME   ; <- replace with your Outbound SIP Username remotesecret=OUTBOUND_SIP_PASSWORD  ; <- supplant with your Outbound SIP Countersign                      

To send outbound calls to GoTrunk SIP Trunk update extensions.conf file:

            [from-internal] ; internal calls between extensions exten => 201,1,Dial(SIP/201) exten => 201,n,Hangup() exten => 202,i,Dial(SIP/202) exten => 202,northward,Hangup() ; ship all other numbers to outbound body exten => _[0-9].,1,NoOp(#### [from-internal] ####) exten => _[0-9].,northward,Fix(CALLERID(num)=442036427900) ; set Caller ID exten => _[0-9].,n,Punch(SIP/+${EXTEN}@body)       ; send call to trunk peer divers in sip.conf exten => _[0-nine].,n,Hangup()                      

two. To accept incoming calls from Telephone Numbers assigned to your GoTrunk business relationship update extensions.conf file:

            [from-trunk] ; <- 'from-torso' context defined in sip.conf exten => _[0-ix].,1,NoOp(#### [from-body] ####) exten => _[0-ix].,n,Dial(SIP/201) ; send all incoming calls from torso to extension 201 exten => _[0-ix].,n,Hangup()                      

Note: example above is provided for illustration purposes simply. Your bodily dialplan will depend on specific PBX configuration.

Source: https://gotrunk.com/docs/ippbxconfiguration/asterisk/

Posted by: bullardwhictime.blogspot.com

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