Before you start
Sign in to the Sipflex portal and open the trunk you want to connect. Keep the profile, phone number or username and account-specific credentials visible while you configure your system.
- A fixed public IPv4 address saved in Sipflex
- PJSIP enabled
- SIP and RTP restricted to 52.56.188.70/32
- A restricted inbound dialplan context
Use the current Sipflex endpoints.
The SIP registrar is sip.sipflex.co.uk and the IAX2 host is iax.sipflex.co.uk. Usernames and passwords remain account-specific and are shown in your portal.
Configure the connection
- Select IP peering and save the public IP of your Asterisk system in Sipflex.
- Back up pjsip.conf and extensions.conf.
- Create a transport, AOR and endpoint without registration authentication.
- Create an identify section matching only 52.56.188.70/32.
- Add restricted inbound and outbound dialplan contexts.
- Reload PJSIP and verify the endpoint and identify match before testing calls.
; /etc/asterisk/pjsip.conf
[sipflex-udp]
type=transport
protocol=udp
bind=0.0.0.0
[sipflex-aor]
type=aor
contact=sip:sip.sipflex.co.uk:5060
[sipflex-endpoint]
type=endpoint
transport=sipflex-udp
context=from-sipflex
disallow=all
allow=alaw
allow=ulaw
dtmf_mode=rfc4733
aors=sipflex-aor
direct_media=no
[sipflex-identify]
type=identify
endpoint=sipflex-endpoint
match=52.56.188.70/32
; /etc/asterisk/extensions.conf
[from-sipflex]
exten => _X.,1,NoOp(Incoming Sipflex peer call to ${EXTEN})
same => n,Dial(PJSIP/your-destination)
same => n,Hangup()
[outbound-sipflex]
exten => _X.,1,Dial(PJSIP/${EXTEN}@sipflex-endpoint)
same => n,Hangup()Firewall requirement
Accept provider SIP and RTP only from 52.56.188.70/32. See the generic peering guide for the complete port table.
Generate a customised configuration
Verify the service
- Confirm the trunk is registered or that the correct public peer IP is saved in the Sipflex portal.
- Place an outbound test call and check the caller ID presented.
- Call the Sipflex number from another network and confirm it reaches the expected inbound context or route.
- Check two-way audio and RFC4733/RFC2833 DTMF.
Support scope
We can check Sipflex credentials, registration state and provider-side call routing. Your PBX, device, firewall, dial plan and application remain your responsibility.