> ## 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.
# Generic SIP
> Connect any SIP-compliant phone system to SmartAlex. Use this guide when your platform isn't one of the named integrations.
**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), OR a SIP extension you can hand to us if you'd rather we register on your side (see Native Extension Mode below)
* 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.
Two directions are supported. **Outbound trunk** (default below): your PBX registers to SmartAlex. **Native Extension Mode** (register-direction): SmartAlex registers as an extension on your PBX. Pick the one your PBX makes easier to provision, they have the same call-handling behaviour. See [Native Extension Mode](/telephony/native-extension-mode) for the alternative path.
## 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 concept | Typical 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:
| Setting | Value |
| ------------------ | ------------------------------------------------------------------- |
| Registrar / Server | `sip.voice.getsmartalex.com` |
| Outbound Proxy | `sip.voice.getsmartalex.com` |
| Port | 5060 for UDP/TCP, 5061 for TLS |
| Transport | UDP (default), TCP, or TLS |
| Auth Username | (from SmartAlex) |
| Auth Password | (from SmartAlex) |
| Auth realm | leave blank unless your PBX requires one |
| Codec priority | G.711 u-law (PCMU) then G.711 A-law (PCMA), then G.722 if supported |
| DTMF | RFC 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.
## Step 6: link the agent to the trunk
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
Transfer mechanics.
Firewall and NAT.
Error reference.