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Difficulty: Easy · ~10 min

Grandstream UCM SIP Trunk Setup

Connect your Grandstream UCM6200, UCM6300, or UCM6500 series IP PBX to AIVO Connect for reliable SIP trunking. This guide covers the UCM web admin panel configuration.

Prerequisites

  • Grandstream UCM6200, UCM6300, or UCM6500 series with latest firmware
  • An active AIVO Connect SIP trunking account with credentials
  • Admin access to the UCM web admin panel (typically at https://<UCM-IP>:8089)
  • Your AIVO Connect SIP username and password (from the Connect dashboard)

AIVO Connect SIP Settings

SIP Configuration Reference

SIP Server: sip.aivo.bz

Port: 5060 (UDP/TCP) or 5061 (TLS)

Codecs: G.722, G.711u, G.711a, Opus

DTMF: RFC 2833

Registration: Not required (IP auth) / Required (credential auth)

Auth Methods: Credential (username/password) or IP-based

1

Navigate to VoIP Trunks

Log in to the Grandstream UCM web admin panel. In the left sidebar, navigate to Extension/Trunk > VoIP Trunks.

Click Add SIP Trunk to create a new VoIP trunk.

2

Configure SIP Trunk Settings

In the trunk configuration form, select Register SIP Trunk as the type (for credential auth) or Peer SIP Trunk (for IP auth). Enter the following settings:

Provider Name: AIVO-Connect

Hostname/IP: sip.aivo.bz

Transport: UDP (or TLS for encrypted signaling)

Port: 5060 (or 5061 for TLS)

Keep Alive Interval: 60 seconds

3

Configure Authentication

Set up authentication based on your trunk type:

Register SIP Trunk (Credential Auth)

SIP User ID: (your AIVO Connect SIP username)

Authenticate ID: (your AIVO Connect SIP username)

Authenticate Password: (your AIVO Connect SIP password)

The UCM will register with the SIP server upon saving and applying the configuration.

Peer SIP Trunk (IP Auth)

Leave SIP User ID and password fields empty. In the AIVO Connect dashboard, add your UCM's public IP address under SIP Trunking > IP Authentication. No registration is necessary.

4

Set Codec and DTMF Preferences

Scroll to the Codec Preference section. Move the following codecs to the Selected column in this order:

1. G.722 (wideband, recommended for HD voice)

2. PCMU (G.711u) (narrowband, universal compatibility)

3. PCMA (G.711a) (narrowband, common internationally)

4. Opus (wideband, if supported by your firmware version)

Remove any other codecs from the Selected list (e.g., GSM, iLBC, G.729) to avoid codec mismatch issues.

Set DTMF Mode to RFC 2833. This is usually found in the trunk's advanced settings section.

5

Create Outbound Route

Navigate to Extension/Trunk > Outbound Routes and click Add.

Configure the outbound route to use the new trunk:

Calling Rule Name: AIVO-Outbound

Pattern: _X. (matches any number starting with a digit)

Use Trunk: AIVO-Connect

Privilege Level: International (to allow all call types)

You can create multiple outbound routes with different patterns to control which calls go through AIVO Connect. For example, local calls only, or international calls with a specific prefix.

For inbound calls, go to Extension/Trunk > Inbound Routes to configure how incoming calls on your AIVO Connect DID numbers are routed to extensions, ring groups, or IVR menus.

6

Save and Apply Changes

Click Save on both the trunk and outbound route configurations.

An Apply Changes notification will appear at the top of the UCM admin panel. Click it to reload the dial plan and activate your new settings.

Navigate back to Extension/Trunk > VoIP Trunks and check the status column. A Register trunk should show Registered (green icon). A Peer trunk will show as Available.

Test Your Connection

After applying the configuration, verify everything works correctly:

  1. Check trunk status — In the UCM admin panel, go to System Status > PBX Status or Extension/Trunk > VoIP Trunks and confirm the trunk shows as Registered or Available.
  2. Place a test outbound call — Pick up a phone registered to your UCM and dial an external number. Confirm you hear ringing and can establish two-way audio.
  3. Test an inbound call — Call one of your AIVO Connect DID numbers from a mobile or external phone. Verify the call reaches the correct extension or ring group.
  4. Review CDR records — Go to CDR > CDR in the UCM panel to verify both test calls appear with correct duration and status.
  5. Verify DTMF — During a call, press keypad digits and confirm they are transmitted correctly (test with voicemail access or an IVR menu).

Troubleshooting

Trunk shows “Unreachable”: Verify the hostname, port, and credentials are correct. Check that your UCM can reach the internet and that outbound UDP/TCP on port 5060 (or 5061) is not blocked by your firewall.

One-way audio: This is typically caused by NAT. In the UCM admin, go to PBX Settings > SIP Settings > NAT and configure the External Host to your public IP. Ensure RTP ports (10000–20000 UDP) are forwarded through your firewall/router.

Calls drop after 30 seconds: Disable the SIP ALG on your router, as it can interfere with SIP signaling. Also verify that the UCM's NAT settings are correct and that SIP re-INVITE packets can pass through your firewall.

Codec mismatch errors: If you see “488 Not Acceptable Here” errors in the SIP trace, revisit the codec settings and ensure at least G.711u or G.722 is enabled on both sides.

Need help? Contact our team at contact-sales or check the SIP network information page for full technical details.