Asterisk Trunk Config

Owned by Tom Cavey
Last updated: Jul 07, 2021
2 min read

VoIP Innovations uses IP-based authentication. This means we authenticate based on the IP address you are sending calls from--not a username/password. You can add an IP address through the Back Office. Included below is are sample configurations for an Asterisk-based PBX.

Note that information in parenthesis is informational only and not included as part of the trunk.
[voipinnovations] (can be anything, only the local name of the trunk)
host=64.136.174.30
type=friend
disallow=all
allow=ulaw
allow=g729 (optional)
dtmfmode=rfc2833 (rfc2833 is recommended)
insecure=port,invite (usually not necessary, but sometimes required for inbound calls to work, depending on various other settings)
context=from-trunk (or context=from-pstn or context=default) (one of these three)
[voipinnovations] is the friendly name of the trunk.
Type: This will allow your device to both send and receive calls. Type=peer is used for outbound trunks. Type=user is used for inbound trunks. Type=friend allows both inbound and outbound to work on one trunk.
Host: This is the address that you will be sending to and receiving from.

Disallow: This will disallow all the codecs, and later allow you to set the codecs you wish to use. (You must first disallow all as required by Asterisk.)

Allow: This is where you enter the codecs you which to use. You can set priority on which codec to offer first by specifying the order of appearance. The above configuration sets ulaw as the priority.
DTMF: This is for touchtones for IVRs. RFC 2833 is recommended for VoIP Innovations' trunks.
Context: This depends on how your server is set up for calls. If you are sending a 404 SIP response on an inbound call and have ensured you have properly configured the inbound route, try changing the context to one of the three suggestions above.

pjsip.aor.conf
[TEST_PJSIP]
type=aor
qualify_frequency=60
contact=sip:64.136.174.30

pjsip.auth.conf
[TEST_PJSIP]
type=auth
auth_type=userpass
password=
username=TEST_PJSIP__NO_AUTH

pjsip.endpoint.conf
[TEST_PJSIP]
type=endpoint
transport=0.0.0.0-udp
context=from-pstn
disallow=all
allow=ulaw
aors=TEST_PJSIP
language=en
user_eq_phone=no
t38_udptl=no
t38_udptl_ec=none
fax_detect=no
trust_id_inbound=no
t38_udptl_nat=no
direct_media=no
dtmf_mode=auto

pjsip.identify.conf
[TEST_PJSIP]
type=identify
endpoint=TEST_PJSIP
match=64.136.160.0/20,209.166.128.200,209.166.128.201,209.166.154.70,209.166.154.71,192.254.151.100,192.254.151.101

 

Additional information:
VoIP Innovations will send traffic to you for DIDs in 10, 11, or 12 digits. For example, if you used DID 2052163165 with us, then we could send the DID to you in one of three ways:

  • 2052163165

  • 12052163165

  • +12052163165


The way we send traffic to you is managed in our Back Office under Endpoints > Endpoint Group Management. For more information, please see our Wiki Article on this subject. As long as your inbound route matches the way we are sending you traffic, you should be fine.


Please note that other IP addresses may be necessary to configure. Below is a list of all of the IP addresses that we use. To cover all bases, we recommend creating a SIP trunk for each of these IPs. More information regarding these IPs can be found in our Wiki Article.