Connect your PBX

Keep your office phone system and add an AI agent to it. Your PBX registers a SIP trunk out to Voice Logica, like it does with a phone provider. Nothing opens a port, and the PBX needs no public IP.

Once connected:

• the PBX sends the calls you choose to the AI agent: a number, an IVR option, a queue overflow, after-hours calls or a short code;
• the agent can transfer callers back to your extensions over the same trunk.

In the app

1. Go to Telephony - Trunks and click Connect your PBX.
2. Pick your phone system and the AI agent that answers the calls. Click Create connection.
3. Copy the server, port, username and password into the PBX. Save the password: it is shown only once.
4. Follow the steps for your make, which the dialog shows with your values already filled in.
5. Leave the dialog open. It changes to Connected as soon as the PBX registers, usually within a minute.

If it says Your PBX was rejected, the password or username is wrong. If it keeps Waiting for your PBX, see Troubleshooting.

From Claude or ChatGPT

With the Voice Logica connector, say “connect my Panasonic PBX to my agent”. The assistant creates the connection (connect_pbx), gives you the values, walks you through the steps one at a time, and checks the connection (pbx_query, action "check_pbx_connection") until it is live.

The values

Field on the PBX	Value
Server, registrar, proxy, domain	<your-company>.sip.voicelogica.ai (shown in the dialog)
Port	5060 over UDP (5061 for TLS)
Username and authentication ID	the generated username, for example pbx123456
Password	the generated password
Registration	on
Codecs	G.711 A-law first, then G.711 u-law
DTMF	RFC 2833 / RFC 4733 (out of band)

Steps per phone system

Menu names differ between software versions. Where a step says “check”, confirm it on your system.

Panasonic

KX-NS500 / NS700 / NS1000 and KX-NSX1000 / NSX2000.

Before you start: SIP trunk activation keys (IP trunk licence) from the Panasonic Key Management System and a DSP card. You also need a virtual SIP gateway card on the virtual shelf: V-SIPGW16 (NS) or V-SIPGW128 (NSX).

1. Web Maintenance Console → 1.Configuration → 1.Slot → virtual shelf. Put the V-SIPGW card out of service (OUS) and open Port Property.
2. [Main]: Provider Name Voice Logica, SIP Server Location Name = the server, SIP Server Port Number = 5060, SIP Service Domain = the server.
3. [Account]: User Name and Authentication ID = the username, Authentication Password = the password.
4. [Register]: Register Ability = Enable, Registrar Server Name = the server, Register Interval = 3600.
5. [Voice/FAX]: IP Codec Priority G.711A, then G.711Mu.
6. Put the card back in service (INS). Restart the PBX if you changed port numbers.

Send calls to the agent: point a DDI/DID or incoming-call destination (CO & Incoming Call → DDI/DID Table), or an ARS / trunk-group access code, at the trunk group that holds this port.

Good to know: the PBX’s own SIP extension server and the SIP trunk cannot both use local port 5060. Behind NAT, set NAT Traversal to Fixed IP with your office public IP. Changing SIP to TCP on NS deletes existing SIPGW cards, so stay on UDP unless you must change.

Grandstream UCM

1. Extensions/Trunk → VoIP Trunks → Add. Type = Register SIP Trunk.
2. Provider Name Voice Logica, Host Name = server:5060, Transport = UDP.
3. Username and Authenticate ID = the username, Password = the password.
4. Advanced: PCMA first, then PCMU; DTMF Mode = RFC4733; Heartbeat Detection on.
5. Save and Apply Changes.

Send calls to the agent: an Outbound Route (for example a short code) that uses this trunk, or an IVR option or ring group overflow pointing at it. Transfers back: an Inbound Route for this trunk with your extensions as the destination. Use attended transfer.

Yeastar P-Series

1. Extension and Trunk → Trunk → Add. Name Voice Logica, ITSP Template = General.
2. Trunk Type = Register Trunk, Transport = UDP, Hostname/IP = the server, Port = 5060, Domain = the server.
3. Username and Authentication Name = the username, Password = the password.
4. Codecs: G.711 A-law, then u-law. DTMF RFC 2833 / 4733.
5. Save and Apply.

Send calls to the agent: an Outbound Route (dial pattern or short code) using this trunk, or an IVR option or queue overflow. Transfers back: an Inbound Route for this trunk to your extensions.

Yeastar S-Series

1. Settings → PBX → Trunks → Add. Type = Register Trunk (VoIP).
2. Transport = UDP, Hostname/IP = the server, Port = 5060, Domain = the server.
3. Username and Authentication Name = the username, Password = the password.
4. Advanced: DTMF Mode = RFC2833, Qualify on. Codecs: G.711 A-law, then u-law.
5. Save and Apply.

Send calls to the agent: Settings → PBX → Call Control → Outbound Routes, or an IVR option. Transfers back: an Inbound Route for this trunk.

3CX

1. Admin console → Voice & Chat → Add Trunk. Country Generic, Provider Generic SIP Trunk.
2. Registrar / Server = the server, port 5060.
3. Type of Authentication = Register/Account based. Authentication ID = the username, Password = the password.
4. Main Trunk No: any number to identify the trunk. Save and check it shows as registered.

Send calls to the agent: an outbound rule (for example a short code) using this trunk, or a queue, IVR or no-answer destination sent to it. Transfers back: the trunk’s inbound routing to your extensions. The 3CX licence caps simultaneous calls.

FreePBX / Asterisk

1. Connectivity → Trunks → Add SIP (chan_pjsip) Trunk, name voicelogica.
2. pjsip Settings → General: Username and Auth username = the username, Secret = the password, Authentication = Outbound, Registration = Send, SIP Server = the server, Port = 5060, Context = from-pstn.
3. Advanced: From Domain = the server, DTMF Mode = RFC 4733, Qualify Frequency 60.
4. Codecs: alaw, then ulaw. Submit and Apply Config.
5. Behind NAT: set External Address and Local Networks in Settings → Asterisk SIP Settings.

Send calls to the agent: an Outbound Route with a dial pattern using this trunk, or an IVR / ring group failover. Transfers back: an Inbound Route for this trunk.

Alcatel-Lucent OXO

OmniPCX Office and OXO Connect, configured in OMC.

Before you start: an IP trunks licence (Hardware and Limits → Software Key Features), and VoIP-trunk channels set under Voice Over IP → VoIP: Parameters.

1. External Lines → List of Accesses: make the VoIP trunk channels public trunks and add them to a trunk group.
2. Gateway Parameters (from Numbering → Automatic Routing Selection): Target Domain Name = the server, Transport UDP, Registration = Requested, Registrar = the server, Port 5060, Expiration 3600, DTMF out of band (RFC 4733).
3. SIP Accounts: Login and Registered User Name = the username, Password = the password.
4. Codecs: G.711 A-law first.

Send calls to the agent: an ARS prefix whose destination is this gateway and trunk group. OXO uses port 5059 for its own SIP phones and 5060 for trunks. These steps come from a carrier guide: your business partner can confirm menu names for your release.

Avaya IP Office

Before you start: SIP Trunk Channels licences for the number of simultaneous calls.

1. Manager: right-click Line → New → SIP Line.
2. SIP Line: ITSP Domain Name = the server.
3. Transport: ITSP Proxy Address = the server, Layer 4 Protocol UDP, Send Port 5060.
4. SIP Credentials: User name and Authentication Name = the username, Password = the password, Registration required.
5. VoIP: G.711 A-law first, DTMF RFC2833. Merge the configuration.

Send calls to the agent: a short code or ARS entry dialling out on this line group. Transfers back: an Incoming Call Route for this line group.

Other

Any PBX that can register a SIP trunk to a provider (often called “SIP provider”, “ITSP” or “SIP line”): enter the values from the table above, turn registration on, and route calls to the trunk with an outbound route, short code, IVR option or overflow.

Troubleshooting

• Waiting for your PBX: nothing has arrived yet. Check the trunk is saved and enabled with registration on, the server and username are exactly as shown, and the PBX can reach the internet on UDP 5060. Turn off SIP ALG on the office router: it rewrites SIP and is the most common cause. Some PBXs need a restart (Panasonic: card OUS, then INS).
• Your PBX was rejected: the password or username is wrong. Username and authentication ID must both be the generated username. Some PBXs keep the old password until the trunk is restarted.
• Connected, but no audio: allow RTP (UDP) out from the PBX and turn off SIP ALG.
• The trunk’s SIP logs (Telephony - Trunks → the trunk) show every REGISTER and our reply.

A PBX that cannot register a SIP trunk at all can still be reached through an edge device.

Questions

Does this use my phone lines or numbers?

No. Your lines stay on the PBX. The trunk only carries the calls your PBX sends to the agent and the transfers back.

I lost the password.

Open the trunk in Telephony - Trunks, set a new password, and enter it on the PBX.

Can I connect more than one PBX?

Yes. Each connection gets its own username, so each site or PBX registers separately.