prefixkit Ask an engineerAsk Support RU

SIP provider settings

Plivo Zentrunk SIP trunk setup

How to connect Plivo to your PBX: SIP server, port, login, codecs, number format and a ready pjsip.conf for Asterisk, a FreeSWITCH gateway and the fields for 3CX. The generator fills in your details.

SIP serverYour trunk’s Termination SIP Domain, like xxxxxxxxxxxxxxxx.zt.plivo.com
Port5060 UDP
AuthenticationCredentials (username and password) on the outbound trunk
CodecsG.711 µ-law
Number formatCaller ID = your Plivo number
Provider guidewww.plivo.com

Inbound calls come through a separate origination (inbound) trunk pointing at your PBX.

Generator with Plivo settings

Provider fields are filled in: add your login, password and number.

Provider

Numbers and routing

Media and NAT

The config is built in your browser; the password is not sent anywhere.

Example pjsip.conf for Plivo

; pjsip.conf — trunk "plivo" generated by prefixkit.com
; Reload: asterisk -rx "pjsip reload"   Check: pjsip show registrations / pjsip show endpoint plivo

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060

[plivo-auth]
type=auth
auth_type=userpass
username=username
password=PASSWORD

[plivo]
type=aor
contact=sip:1234567890123456.zt.plivo.com:5060
qualify_frequency=60

[plivo]
type=endpoint
transport=transport-udp
context=from-plivo
disallow=all
allow=ulaw
aors=plivo
outbound_auth=plivo-auth
from_user=username
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733

[plivo]
type=identify
endpoint=plivo
match=1234567890123456.zt.plivo.com

FreeSWITCH gateway

<!-- conf/sip_profiles/external/plivo.xml — reload: fs_cli -x "sofia profile external rescan" -->
<include>
  <gateway name="plivo">
    <param name="username" value="username"/>
    <param name="password" value="PASSWORD"/>
    <param name="realm" value="1234567890123456.zt.plivo.com"/>
    <param name="proxy" value="1234567890123456.zt.plivo.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>

Plivo in 3CX, FreePBX, Yeastar and other web-managed PBXs

Trunk typeGeneric SIP trunk, authentication without registration
Registrar / SIP server / Host1234567890123456.zt.plivo.com
Port5060 UDP
Auth ID / Usernamethe login from your account (trunk username)
CodecsG.711 µ-law
DTMFRFC 2833 / 4733

If it does not work

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 · DIDWW · VoIP.ms · Flowroute · sipgate

Settings come from the provider’s guides and may change; check your account. If something is out of date, tell us through the feedback form.

Idea?