Asterisk configuration guide

⚠ These are placeholder values — set the real ones in the editor (Global tab) and they update on every guide at once.

1. Define the trunk in pjsip.conf

These examples use the PJSIP driver, which is the supported channel driver from Asterisk 12 onwards. Replace YOUR_USERNAME and YOUR_PASSWORD with the credentials from your welcome email.

;--- Transport -------------------------------------------------------------
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0:5060
; If the PBX is behind NAT, uncomment and set these:
; external_media_address = YOUR.PUBLIC.IP
; external_signaling_address = YOUR.PUBLIC.IP
; local_net = 192.168.0.0/16

;--- Registration ----------------------------------------------------------
[sipandvoip-reg]
type = registration
transport = transport-udp
outbound_auth = sipandvoip-auth
server_uri = sip:sip.sipandvoip.com.cy
client_uri = sip:YOUR_USERNAME@sip.sipandvoip.com.cy
contact_user = YOUR_USERNAME
retry_interval = 60
forbidden_retry_interval = 600
expiration = 3600
line = yes
endpoint = SipandVoip

;--- Authentication --------------------------------------------------------
[sipandvoip-auth]
type = auth
auth_type = userpass
username = YOUR_USERNAME
password = YOUR_PASSWORD

;--- Address of record -----------------------------------------------------
[SipandVoip]
type = aor
contact = sip:sip.sipandvoip.com.cy:5060
qualify_frequency = 60

;--- Endpoint --------------------------------------------------------------
[SipandVoip]
type = endpoint
transport = transport-udp
context = from-sipandvoip
disallow = all
allow = alaw,g729,opus
outbound_auth = sipandvoip-auth
aors = SipandVoip
from_user = YOUR_USERNAME
from_domain = sip.sipandvoip.com.cy
dtmf_mode = rfc4733
direct_media = no
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
send_rpid = yes

;--- Identify inbound traffic ----------------------------------------------
[SipandVoip]
type = identify
endpoint = SipandVoip
match = sip.sipandvoip.com.cy

2. Add the dialplan in extensions.conf

[from-sipandvoip]
; Inbound: route your DID to an extension, queue or IVR
exten => _X.,1,NoOp(Inbound from ${CALLERID(num)} to ${EXTEN})
 same =>    n,Set(CALLERID(name)=${CALLERID(num)})
 same =>    n,Goto(internal,100,1)   ; <-- change to your destination

[internal]
; Outbound: strip the trunk prefix and send in E.164
exten => _9.,1,NoOp(Outbound to ${EXTEN:1})
 same =>    n,Set(CALLERID(num)=YOUR_DID)
 same =>    n,Dial(PJSIP/${EXTEN:1}@sipandvoip,60)
 same =>    n,Hangup()

3. Reload and verify

asterisk -rx "pjsip reload"
asterisk -rx "pjsip show registrations"
asterisk -rx "pjsip show endpoint SipandVoip"

You are looking for the registration in state Registered. If it shows Rejected, check the credentials; if it never leaves Unregistered, the packets are not reaching us — check the firewall.

4. Test

1. Place an outbound call to a mobile you can hear ring. Confirm two-way audio.
2. Call your DID from that mobile and confirm it lands where the dialplan sends it.
3. Check the presented caller ID on the mobile matches the DID you set.

Turn on pjsip set logger on while testing — it gives you the full SIP trace, which is the first thing our support team will ask for.

If it does not register

• 401 / 403 on REGISTER — username or password is wrong, or the device is presenting a different username in the auth header than in the From header. Check both.
• No response at all — a firewall is dropping SIP. Allow outbound UDP 5060 (or TCP/TLS 5061) and the RTP range to our network, and make sure any SIP ALG on the router is disabled. SIP ALG causes more problems than it solves.
• Registers, but one-way or no audio — almost always NAT. Set the external address on your PBX, forward or open the RTP port range, and confirm the device is advertising its public IP in SDP.
• Calls drop after ~30 seconds — SIP session timers or ACK not reaching us. Enable session timers and re-check NAT traversal settings.
• Outbound rejected with 404 — number format. Send E.164 with the leading + or full international format without a trunk prefix.

Still stuck? Send us the SIP trace and we will look at it against our side of the call.