PBX & contact center platforms

1. Create a Generic SIP Trunk in 3CX with IP-based authentication, not SIP registration.
2. Point it at the platform SIP FQDN, set the trunk transport to TCP (UDP fragments large INVITEs), and restrict codecs to PCMA/PCMU with RFC2833 DTMF — disable Opus and G.722.
3. Set the trunk’s From: Display-Name and Remote-Party-ID to OriginatorCallerID so the assistant sees the real caller, not 3CX’s own system number.
4. Configure inbound in two parts: an outbound rule with a distinct prefix (for example 999) routed to the trunk, and an inbound rule that dials 999<number> back through that same outbound rule.
5. Assign the DID on the platform side and test.

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FreePBX (GUI)

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Raw config (pjsip.conf)

[platform]
type=endpoint
context=from-platform
disallow=all
allow=ulaw
allow=alaw
aors=platform
auth=platform-auth
outbound_auth=platform-auth
from_domain=<platform-sip-fqdn>
from_user=<did-in-e164-without-plus>
direct_media=no
rewrite_contact=yes

[platform]
type=aor
contact=sip:<platform-sip-fqdn>

[platform-auth]
type=auth
auth_type=userpass
username=<username>
password=<password>

[platform]
type=identify
endpoint=platform
match=<platform-sip-fqdn>

1. Create the SIP line as host/IP-based, pointed at the platform SIP FQDN, and assign the number.
2. Enable CLIP No Screening so the real caller ID passes through instead of the trunk’s own number.
3. For time-based routing to the assistant, use a line-prefix dial string such as **<line-digit>*<number>.
4. If audio is one-way or missing, set Behind NAT? yes on both the Starface server and the line.

1. Open WMS → Trunks and click + below the SIP trunk table.
2. Set Address or Host Name to the platform SIP FQDN, without a sip: prefix, and leave Enable registration off.
3. Enter the platform trunk’s SIP username as Auth Login and its password as Password. Leave From user empty: Wildix forces that value into the From header, so the assistant would see it instead of the caller’s number.
4. Under Advanced, set Transport to TCP. For encrypted calls, use TLS with SDES-SRTP instead, and enable Secure trunking on the platform trunk too.
5. In Audio codecs, keep only alaw and ulaw — Wildix enables g729 by default, so remove it. Set DTMF mode to rfc2833.
6. Enable Show original caller number so the assistant sees the real caller, not your Wildix number. Enable Support Refer and Hold so the assistant can cold-transfer calls back to Wildix users.
7. Add a Wildix dialplan rule that calls the exact +E.164 DID through this trunk — for example after hours from a timetable, as overflow from a group, or as an IVR option.

1. Create the trunk with inbound (the FQDN Genesys Cloud generates) and outbound (pointed at the platform SIP FQDN) configured as separate legs.
2. Open the non-standard ports at your firewall — TCP/UDP 32681 and TLS 32682 — instead of the usual 5060/5061. A firewall that only opens the standard ports fails silently.
3. Order codecs PCMU then PCMA, and remove G.722.
4. Restrict SIP Access Control to only the platform’s current signaling addresses.

1. Open a support ticket with Five9 — there’s no self-service path.
2. Set codecs to G.711 only (no Opus/G.722).
3. Configure human transfers with SIP REFER — Five9 doesn’t honor SIP 302 redirects.
4. Target a fully-qualified URI for every REFER (sip:queue@sbc-{region}.five9.com); a bare extension is silently dropped.