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SIP provider settings

VoIP.ms SIP trunk setup

How to connect VoIP.ms 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 serverThe VoIP.ms server closest to you, e.g. atlanta.voip.ms (the list is in the customer portal)
AuthenticationRegistration, or IP authentication on a sub-account
LoginMain account 6-digit ID (123456) or sub-account (123456_sub); From user = the same
CodecsG.711 µ-law (G.729 optional)
Number formatNorth American 1+10 digits; 011 for international
Provider guidewiki.voip.ms

The provider guide sets rtp_symmetric, rewrite_contact and send_rpid on the endpoint: keep them when behind NAT.

Put the inbound context before the outbound one in the dialplan to avoid loops.

Generator with VoIP.ms 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 VoIP.ms

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

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

[voipms-auth]
type=auth
auth_type=userpass
username=123456_sub
password=PASSWORD

[voipms]
type=aor
contact=sip:atlanta.voip.ms:5060
qualify_frequency=60

[voipms]
type=endpoint
transport=transport-udp
context=from-voipms
disallow=all
allow=ulaw
aors=voipms
outbound_auth=voipms-auth
from_user=123456_sub
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733

[voipms]
type=identify
endpoint=voipms
match=atlanta.voip.ms

[voipms-reg]
type=registration
transport=transport-udp
outbound_auth=voipms-auth
server_uri=sip:atlanta.voip.ms:5060
client_uri=sip:123456_sub@atlanta.voip.ms
contact_user=123456_sub
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
endpoint=voipms

FreeSWITCH gateway

<!-- conf/sip_profiles/external/voipms.xml — reload: fs_cli -x "sofia profile external rescan" -->
<include>
  <gateway name="voipms">
    <param name="username" value="123456_sub"/>
    <param name="password" value="PASSWORD"/>
    <param name="realm" value="atlanta.voip.ms"/>
    <param name="proxy" value="atlanta.voip.ms:5060"/>
    <param name="register" value="true"/>
    <param name="expire-seconds" value="3600"/>
    <param name="retry-seconds" value="30"/>
    <param name="from-user" value="123456_sub"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="ping" value="60"/>
  </gateway>
</include>

VoIP.ms in 3CX, FreePBX, Yeastar and other web-managed PBXs

Trunk typeGeneric SIP trunk with registration (Register Trunk)
Registrar / SIP server / Hostatlanta.voip.ms
Port5060 UDP
Auth ID / Usernamethe login from your account (123456_sub)
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 · Flowroute · Plivo · 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.

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