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Connect Asterisk to your CallGlobal line

Updated 2026-10-30 · Prepared by Call Global Communications Inc

Short answer: Define the provider endpoint, authentication, address-of-record and registration objects in your Asterisk PJSIP configuration. Keep secrets outside public configuration examples and restrict which extensions can use the route.

Asterisk handles the phone interface or routing environment; CallGlobal supplies the active line and its account credentials. Keep those responsibilities separate when setting up and troubleshooting.

Prepare Asterisk for the line

Define the provider endpoint, authentication, address-of-record and registration objects in your Asterisk PJSIP configuration. Keep secrets outside public configuration examples and restrict which extensions can use the route.

Values and checks before use

  1. Open the active number in your CallGlobal dashboard and obtain its current SIP server, username and password.
  2. Place each value in the corresponding account field; do not reuse the website login as the SIP password.
  3. Use a supported transport and audio configuration, then check whether the client reports a registered account.
  4. Inspect microphone access and audio routing separately if registration works but calls have no sound.

Supported connection details

SettingReference
SIP signalingUDP/TCP 5060; TLS 5061
Audio codecsG.711 PCMU and PCMA
Keypad eventsRFC 2833 or SIP INFO
AuthenticationSeparate username and password supplied for each line

Connection field reference 1

SettingValue
SIP server / registrar / domain<SIP_SERVER_FROM_DASHBOARD>
Transport and portUDP or TCP 5060; TLS 5061
Username<SIP_USERNAME> — copy the credential assigned to this line in your account
Password<SIP_PASSWORD> (from the same page)
AuthenticationRegistration authenticates through Digest using the assigned username and password
CodecsG.711 µ-law (PCMU) and G.711 A-law (PCMA)
DTMFRFC 2833 (RTP events) or SIP INFO

Configuration reference 2

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
; behind NAT also set:
; external_media_address=<PUBLIC_IP>
; external_signaling_address=<PUBLIC_IP>
; local_net=192.168.0.0/16

[callglobal]
type=registration
transport=transport-udp
outbound_auth=callglobal-auth
server_uri=sip:<SIP_SERVER_FROM_DASHBOARD>:5060
client_uri=sip:<SIP_USERNAME>@<SIP_SERVER_FROM_DASHBOARD>
contact_user=<YOUR_NUMBER>
retry_interval=60

[callglobal-auth]
type=auth
auth_type=userpass
username=<SIP_USERNAME>
password=<SIP_PASSWORD>

[callglobal]
type=aor
contact=sip:<SIP_SERVER_FROM_DASHBOARD>:5060
qualify_frequency=60

[callglobal]
type=endpoint
transport=transport-udp
context=from-callglobal
disallow=all
allow=ulaw,alaw
dtmf_mode=rfc4733
outbound_auth=callglobal-auth
aors=callglobal
from_user=<SIP_USERNAME>
from_domain=<SIP_SERVER_FROM_DASHBOARD>
callerid=<YOUR_NUMBER>
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes

[callglobal]
type=identify
endpoint=callglobal
match=<SIP_SERVER_FROM_DASHBOARD>

Configuration reference 3

[from-callglobal]
; inbound: DID = your CallGlobal number
exten => <YOUR_NUMBER>,1,NoOp(Inbound call on CallGlobal DID)
 same => n,Dial(PJSIP/1001,30)
 same => n,Hangup()
; catch other DID formats (with or without +)
exten => _X.,1,Goto(from-callglobal,<YOUR_NUMBER>,1)

[from-internal]
; outbound: dial +<country code><number>
exten => _+X.,1,Set(CALLERID(num)=<YOUR_NUMBER>)
 same => n,Dial(PJSIP/${EXTEN}@callglobal,60)
 same => n,Hangup()

Keep the connection secure

Restrict access to the SIP password and to any PBX administration panel. Redact credentials before sharing logs, and keep the number renewed so the account remains usable.

CallGlobal does not supply campaign automation, a hosted PBX or predictive dialing. Configure application-specific behavior in the software you operate and follow the rules governing each call.

Frequently asked questions

Which credentials does Asterisk need?

Use the SIP values attached to the active number. These are separate from the email and password used to sign in to the CallGlobal website.

What if the account registers but audio fails?

Check microphone permissions, the selected audio device, codec compatibility and firewall or NAT handling. Registration and audio use different parts of the connection.

Can the browser be used instead?

Yes. The dashboard includes a web dialer, so a separate SIP client is optional for ordinary browser calling.

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