SIPNET SIP trunk setup

Example pjsip.conf for SIPNET

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

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

[sipnet-auth]
type=auth
auth_type=userpass
username=0012345678
password=PASSWORD

[sipnet]
type=aor
contact=sip:sipnet.ru:5060
qualify_frequency=60

[sipnet]
type=endpoint
transport=transport-udp
context=from-sipnet
disallow=all
allow=alaw,ulaw,g729
aors=sipnet
outbound_auth=sipnet-auth
from_user=0012345678
from_domain=sipnet.ru
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
dtmf_mode=rfc4733

[sipnet]
type=identify
endpoint=sipnet
match=sipnet.ru

[sipnet-reg]
type=registration
transport=transport-udp
outbound_auth=sipnet-auth
server_uri=sip:sipnet.ru:5060
client_uri=sip:0012345678@sipnet.ru
contact_user=0012345678
retry_interval=60
forbidden_retry_interval=600
expiration=3600
line=yes
endpoint=sipnet

FreeSWITCH gateway

<!-- conf/sip_profiles/external/sipnet.xml — reload: fs_cli -x "sofia profile external rescan" -->
<include>
  <gateway name="sipnet">
    <param name="username" value="0012345678"/>
    <param name="password" value="PASSWORD"/>
    <param name="realm" value="sipnet.ru"/>
    <param name="proxy" value="sipnet.ru:5060"/>
    <param name="register" value="true"/>
    <param name="expire-seconds" value="3600"/>
    <param name="retry-seconds" value="30"/>
    <param name="from-user" value="0012345678"/>
    <param name="from-domain" value="sipnet.ru"/>
    <param name="caller-id-in-from" value="true"/>
    <param name="ping" value="60"/>
  </gateway>
</include>

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

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 · Mango Office · Twilio · Telnyx · DIDWW · VoIP.ms · Flowroute · Plivo · sipgate