Skip to main content

Documentation Index

Fetch the complete documentation index at: https://docs.getsmartalex.com/llms.txt

Use this file to discover all available pages before exploring further.

Audience: network or voice engineer with admin access to a PBX or softswitch. Time: 15 minutes.
Use this guide for any of the following:
  • A PBX not covered by a named integration guide
  • A custom softswitch or SBC
  • An in-house Asterisk variant or a fork
  • A cloud PBX whose documentation isn’t vendor-specific
The SmartAlex side is identical regardless of which system you’re connecting. What varies is where to click in your own PBX admin interface.

​
What the PBX needs to support

  • SIP (any version, RFC 3261)
  • Generic SIP trunk (register-based or IP-based auth)
  • SIP REFER (RFC 3515) for transfers
  • RFC 2833 DTMF
  • G.711 u-law or A-law codecs (G.722 optional, G.729 only if licensed on both sides)
Most production-grade PBXes built in the last fifteen years support all of these.

​
Step 1: create the SIP trunk in SmartAlex

  1. Phone Numbers, then Connect Your Carrier, then SIP Trunk
  2. Pick Other from the PBX dropdown
  3. Trunk Name: label it so you recognise it later
  4. PBX SIP Domain: the hostname or IP your PBX presents (required)
  5. Authentication: Credentials (most common) or IP Whitelist (if your PBX has a static public IP)
  6. Leave codecs and transport on the defaults unless your PBX has specific requirements
  7. Create
Copy the connection details from the success screen. Leave that tab open.

​
Step 2: add the trunk on your PBX

Generic-SIP-trunk setup will look roughly the same on any PBX. Look for these menu paths or the equivalent:
PBX conceptTypical menu name
Create a new trunk”SIP Trunks”, “Voice Gateways”, “External Lines”, “Providers”
Registrar / Outbound Proxy”Registrar Domain”, “SIP Server”, “Host”
Credentials”Auth Username”, “SIP Auth ID”, “Display Name”
Codecs”Codec Priority”, “Voice Codec Selection”
Transport”Protocol”, “Transport Layer”
DIDs”DID Numbers”, “Inbound Routes”, “Telephone Numbers”
Enter these values:
SettingValue
Registrar / Serversip.voice.getsmartalex.com
Outbound Proxysip.voice.getsmartalex.com
Port5060 for UDP/TCP, 5061 for TLS
TransportUDP (default), TCP, or TLS
Auth Username(from SmartAlex)
Auth Password(from SmartAlex)
Auth realmleave blank unless your PBX requires one
Codec priorityG.711 u-law (PCMU) then G.711 A-law (PCMA), then G.722 if supported
DTMFRFC 2833

​
Step 3: confirm registration

Your PBX should show the trunk as Registered, Active, or Up within 30 seconds. If not, see Troubleshooting.

​
Step 4: route inbound DIDs to the trunk

Create an inbound rule (or equivalent): when this DID rings, send the call out via the SmartAlex trunk. Your PBX’s “external routing” or “forward” mechanism should handle this.

​
Step 5: add extensions in SmartAlex

Settings, then PBX. Click through to the trunk and add your extensions: 
extension,display_name,owner,department,aliases
101,Sales,Sarah Jones,Sales,sarah|sales team
102,Support,John Smith,Support,support|tech
Use CSV for bulk. Agent Studio, then Telephony tab, then pick the trunk in the dropdown. Save.

​
Step 7: test

Call the DID from an external phone. AI should answer. Ask for a transfer by name. Extension should ring within 3 seconds.

​
Common compatibility notes

SmartAlex’s outbound headers are standard RFC 3261. If your PBX rejects based on User-Agent, contact support with the PBX’s specific requirements.
G.729 is a licensed codec. If your PBX requires it, we can enable it on the trunk. Contact support. Note: G.729 adds ~10–20 ms latency compared to G.711 and has slightly lower MOS.
A handful of very old or locked-down PBXes reject REFER. If that’s the case, transfers won’t work directly. Contact support for alternative transfer models.
Most generic SIP provider templates default this off. Leave it that way.

​
If your PBX is exotic

Some platforms (certain SBCs, very old Asterisk forks, or vertical-specific systems like hospitality PBXes) have quirks. Capture a SIP trace from your PBX during a failing call and email it to support with:
  • PBX vendor and version
  • The exact error or symptom
  • Trace file (PCAP or text-format SIP log)
Most issues resolve within 24 hours.

​
Next steps

Call Routing & Transfers

Transfer mechanics.

Network Requirements

Firewall and NAT.

Troubleshooting

Error reference.