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SIP provider settings

sipgate trunking SIP trunk setup

How to connect sipgate 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 serversipconnect.sipgate.de
AuthenticationRegistration with the SIP ID
LoginSIP ID as username and From user; From domain = sipconnect.sipgate.de
CodecsG.722, G.711 A-law, µ-law (GSM, iLBC)
Number formatE.164 without + and without leading zeros, e.g. 492111234567; inbound calls carry the full E.164 number in the Request-URI
Inbound IP addresses217.10.68.150
Provider guidehelp.sipgate.de

The caller ID is sent in P-Preferred-Identity: <sip:492111234567@sipconnect.sipgate.de>.

Generator with sipgate 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 sipgate

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

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

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

[sipgate]
type=aor
contact=sip:sipconnect.sipgate.de:5060
qualify_frequency=60

[sipgate]
type=endpoint
transport=transport-udp
context=from-sipgate
disallow=all
allow=g722,alaw,ulaw
aors=sipgate
outbound_auth=sipgate-auth
from_user=username
from_domain=sipconnect.sipgate.de
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733

[sipgate]
type=identify
endpoint=sipgate
match=sipconnect.sipgate.de
match=217.10.68.150

[sipgate-reg]
type=registration
transport=transport-udp
outbound_auth=sipgate-auth
server_uri=sip:sipconnect.sipgate.de:5060
client_uri=sip:username@sipconnect.sipgate.de
contact_user=username
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
endpoint=sipgate

FreeSWITCH gateway

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

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

Trunk typeGeneric SIP trunk with registration (Register Trunk)
Registrar / SIP server / Hostsipconnect.sipgate.de
Port5060 UDP
Auth ID / Usernamethe login from your account (SIP ID)
From domainsipconnect.sipgate.de
CodecsG.722, G.711 A-law, G.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 · Plivo

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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