Setting up the Asterisk PJSIP with Zadarma.
Information used in the example:
- 111111 - your sip-number from your personal account.
- Password - you sip-number password from the "SIP Connection" section of your personal account.
- 1234-100 - Your PBX extension number from your personal profile
- Password - your PBX extension number password from your personal profile
- 101 - the Asterisk extension number that is connected to the softphone/IP phone.
Standard setup example
Outgoing calls from extension number 101 are routed to the trunk 111111. Incoming calls are received by registration and are routed to the extension number 101.
Outgoing calls from extension number 101 are routed to the trunk 1234-100. Incoming calls are received by registration and are routed to the extension number 101.
Edit pjsip.conf
[udp-transport] type=transport protocol=udp bind=0.0.0.0 [111111] type=registration transport=udp-transport outbound_auth=111111_auth server_uri=sip:sip.zadarma.com client_uri=sip:111111@sip.zadarma.com retry_interval=60 expiration=120 contact_user=111111 [111111_auth] type=auth auth_type=userpass password=Password username=111111 [111111] type=aor contact=sip:sip.zadarma.com [111111] type=endpoint transport=udp-transport context=zadarma-in disallow=all allow=alaw allow=ulaw outbound_auth=111111_auth aors=111111 from_user=111111 direct_media=no [111111] type=identify endpoint=111111 match=sip.zadarma.com
[udp-transport] type=transport protocol=udp bind=0.0.0.0 [1234-100] type=registration transport=udp-transport outbound_auth=1234-100_auth server_uri=sip:pbx.zadarma.com client_uri=sip:1234-100@pbx.zadarma.com retry_interval=60 expiration=120 contact_user=1234-100 [1234-100_auth] type=auth auth_type=userpass password=Password username=1234-100 [1234-100] type=aor contact=sip:pbx.zadarma.com [1234-100] type=endpoint transport=udp-transport context=zadarma-in disallow=all allow=alaw allow=ulaw outbound_auth=1234-100_auth aors=1234-100 from_user=1234-100 direct_media=no [1234-100] type=identify endpoint=1234-100 match=pbx.zadarma.com
Extension number 101 to which the softphone/IP phone is connected.
[101]
type=endpoint
transport=udp-transport
context=zadarma-out
disallow=all
allow=alaw
allow=ulaw
auth=101
aors=101
[101]
type=auth
auth_type=userpass
password=101
username=101
[101]
type=aor
max_contacts=10
Incoming and outgoing routing is set in the extensions.conf file
[zadarma-in] exten => 111111,1, Dial(PJSIP/101) ; all incoming calls from the trunk 111111 are routed to the extension number 101 [zadarma-out] exten => _XXX,1,Dial(PJSIP/${EXTEN}) ; calls to 3-digit extension numbers of Asterisk exten => _XXX.,1,Dial(PJSIP/${EXTEN}@111111) ; calls to numbers with 4 digits or more via trunk 111111
[zadarma-in] exten => 1234-100,1, Dial(PJSIP/101) ; all incoming calls from the trunk 1234-100 are routed to the extension number 101 [zadarma-out] exten => _XXX,1,Dial(PJSIP/${EXTEN}) ; calls to 3-digit extension numbers of Asterisk exten => _XXX.,1,Dial(PJSIP/${EXTEN}@1234-100) ; calls to numbers with 4 digits or more via trunk 1234-100
Standard setup is complete.
If you have several active virtual phone numbers, you can "name" each number (for example manchester and London), and set the incoming routing based on this parameter. The virtual phone number "Name" can be set in the CALLERID(name) parameter.
In the following example, calls from the number named manchester are routed to the extension number 101, and calls from the number named London are routed to the extension number 102. All other calls will be declined by the Asterisk with a "Busy" tone.
[zadarma-in]
exten => _X.,1,GotoIf($["${CALLERID(name)}" = "manchester"]?2:3)
exten => _X.,2,Dial(PJSIP/101)
exten => _X.,3,GotoIf($["${CALLERID(name)}" = "london"]?4:5)
exten => _X.,4,Dial(PJSIP/102)
exten => _X.,5,Busy
Your virtual phone number that received the call will be displayed in the CALLED_DID header. It is possible to configure incoming routing based on this parameter.
In the following example, calls from the number 442045770077 are routed to extension number 101, calls from the number 442037691880 are routed to extension number 102. The Asterisk will decline all other calls with a "Busy" tone.
[zadarma-in]
exten => _X.,1,GotoIf($["${PJSIP_HEADER(read,CALLED_DID)}" = "442045770077"]?2:3)
exten => _X.,2,Dial(PJSIP/101)
exten => _X.,3,GotoIf($["${PJSIP_HEADER(read,CALLED_DID)}" = "442037691880"]?4:5)
exten => _X.,4,Dial(PJSIP/102)
exten => _X.,5,Busy
The setup is complete.
Example №2 (SIP URI)
If the Asterisk is located on a "white" IP address (not behind a router, for example in a data centre), incoming calls can be received without registration by SIP URI scheme.
Information used in the example:
- 15555555555 - Your virtual phone number connected to Zadarma.
- 2.20.190.41 - your Asterisk server IP address.
Go to your personal account, "Settings - Direct phone number" and route the calls from the virtual number to the external server (SIP URI) using this format 15555555555@2.20.190.41
Edit pjsip.conf
[15555555555]
type=aor
contact=sip:sip.zadarma.com
[15555555555]
type=endpoint
transport=udp-transport
context=zadarma-in
disallow=all
allow=alaw
allow=ulaw
aors=15555555555
direct_media=no
[15555555555]
type=identify
endpoint=15555555555
match=sip.zadarma.com
match=sipurifr.zadarma.com
match=sipurims.zadarma.com
match=sipuriny.zadarma.com
Incoming route is in the extensions.conf file
[zadarma-in]
exten => 15555555555,1, Dial(PJSIP/101)
The setup is complete.
Enabling encryption
1.In the file pjsip.conf add the following
[transport-z]
type=transport
protocol=tls
verify_server=no
bind=0.0.0.0:7065
method=tlsv1_2
2.In server_uri after the server address add :5061