Our service is 100% compatible with Asterisk using either standard SIP registration or IP authentication where SIP trunks are configured as such. Asterisk is the base software behind many open-source PBX distributions, including FreePBX, Trixbox and Elastix, and is also the enabler behind many other ITSPs and commercial PABX manufacturers.
Please Note: Chan SIP is now deprecated in favor of PJSIP
https://www.asterisk.org/asterisk-21-module-removal-chan_sip/
Suitable for Asterisk versions >13, you should have the following in your pjsip.conf file:
This configuration is for authentication based on a registration with username and password. The protocol can be changed as desired. TLS and SRTP will require additional configuration.
[transport-udp] ; Configures res_pjsip transport layer interaction.
type = transport
protocol = udp
bind = 0.0.0.0
[voicehost] ; Contains information about an outbound SIP registration
type=registration
transport=transport-udp
outbound_auth=voicehost_auth
server_uri=sip:[sip-trunk-username]@[hostname]:5060
client_uri=sip:[hostname]:5060
retry_interval=20
expiration=3600
[voicehost_auth] ; Stores inbound or outbound authentication credentials for use by trunks, endpoints, registrations.
type=auth
password=[sip-trunk-password]
username=[sip-trunk-username]
[voicehost_aor] ; Stores contact information for use by endpoints.
type=aor
contact=sip:[sip-trunk-username]@[hostname]
[voicehost] ; Configures core SIP functionality related to SIP endpoints.
type=endpoint
context=voicehost-in
dtmf_mode=rfc4733
disallow=all
allow=alaw
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
timers=yes
from_user=[sip-trunk-username]
from_domain=[domain]
language=en
outbound_auth=voicehost_auth
aors=voicehost_aor
[voicehost] ; Maps a host directly to an endpoint
type=identify
endpoint=voicehost
match=[hostname]
