Example pjsip.conf for DIDWW
; pjsip.conf — trunk "didww" generated by prefixkit.com
; Reload: asterisk -rx "pjsip reload" Check: pjsip show registrations / pjsip show endpoint didww
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
[didww-auth]
type=auth
auth_type=userpass
username=username
password=PASSWORD
[didww]
type=aor
contact=sip:any.out.didww.com:5060
qualify_frequency=60
[didww]
type=endpoint
transport=transport-udp
context=from-didww
disallow=all
allow=alaw,ulaw,g729
aors=didww
outbound_auth=didww-auth
from_user=username
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733
[didww]
type=identify
endpoint=didww
match=any.out.didww.com
match=46.19.209.14
match=46.19.210.14
match=46.19.212.14
match=46.19.213.14
match=46.19.214.14
match=46.19.215.14
match=185.238.173.14
FreeSWITCH gateway
<!-- conf/sip_profiles/external/didww.xml — reload: fs_cli -x "sofia profile external rescan" -->
<include>
<gateway name="didww">
<param name="username" value="username"/>
<param name="password" value="PASSWORD"/>
<param name="realm" value="any.out.didww.com"/>
<param name="proxy" value="any.out.didww.com:5060"/>
<param name="register" value="false"/>
<param name="from-user" value="username"/>
<param name="caller-id-in-from" value="true"/>
<param name="ping" value="60"/>
</gateway>
</include>DIDWW in 3CX, FreePBX, Yeastar and other web-managed PBXs
| Trunk type | Generic SIP trunk, authentication without registration |
|---|---|
| Registrar / SIP server / Host | any.out.didww.com |
| Port | 5060 UDP |
| Auth ID / Username | the login from your account (outbound trunk username) |
| Codecs | G.711 A-law, G.711 µ-law, G.729 |
| DTMF | RFC 2833 / 4733 |
If it does not work
- Registration rejected (403) : login, password, From domain.
- No response to registration : host, port, firewall, SIP ALG.
- Inbound rejected: No matching endpoint found : provider IPs missing from identify.
- Extension not found in context : the inbound number arrives in another format.
- No joint codecs (488) : allow the provider codecs.
Check whether the provider server answers: SIP check. Read a registration or call log: SIP log analyzer. No audio: NAT test.
Other providers
Zadarma · MTT Business · Telphin · SIPNET · Mango Office · Twilio · Telnyx · VoIP.ms · Flowroute · Plivo · sipgate