Step-by-Step Guide to Troubleshooting Analog Trunk Issues in Grandstream IP PBX UCM6XXX

Description

This article outlines common Analog and PSTN trunk issues encountered with Grandstream IP PBX UCM63XX/62XX/6510. It provides a list of suggestions to address various scenarios in customer setups.

Issue 1: Noise or Echo

1. Test with an Analog Phone: Connect an analog phone to the PSTN line to check if calls produce static noise.
2. Check ADSL Setups: Some ADSL configurations use splitters or modulation devices that may cause signal issues.
3. Ground the UCM: Use the grounding screw on the back of the device to connect to a metal object.
4. Change Codec Settings: Change the codec on the IP phones from the default (PCMU) to G.729A/B and enable silence suppression.
5. Perform PSTN and ACIM Detection: Ensure both detections are completed to identify potential issues.
6. Adjust RX/TX Gain: Reduce the RX and TX gain settings.
7. Experiment with Echo Cancellation Modes: Select different modes to see if the echo is reduced.
8. Use a Different RJ11 Cable: Ensure the RJ11 cable is functioning properly.
9. Test with a Different Power Supply: Power the UCM with another known working PSU to check if the noise persists.
10. Conduct Tests with Various Phones: Test with Analog, SIP, and Softphones to observe behavior differences.

Issue 2: Caller ID Not Showing or Displayed as 0000

1. Verify Caller ID Activation: Confirm with your provider that Caller ID is activated. Connect an analog phone directly to the PSTN line to check if the Caller ID displays correctly on incoming calls.
2. Ground the UCM: Ensure the UCM is properly grounded.
3. Adjust RX Gain: Increase the RX gain to 6 or 12 dB.
4. Perform PSTN and ACIM Detection: Carry out both detection processes to identify any issues.
5. Check for Splitters or DSL Filters: Ensure no splitters or DSL filters are connected before the PSTN line.
6. Disable Jitter Buffer in the PBX Settings.
7. Increase FXO Dial Delay: Set the FXO Dial Delay to 250ms or 1000ms (maximum is 3000ms).
8. Boost Ringer Setting: Navigate to PBX Settings > Analog Hardware > Boost Ringer and set it to Peak.
9. Set DTMF Threshold: Adjust the DTMF Threshold to 1 and revert this change if necessary.

Issue 3: FXO Port Status Not Displayed Correctly in Web Interface

If the FXO port status in the web UI is grayed out and not lit, perform a loopback test by connecting one end of an RJ11 cable to the UCM’s FXO port and the other end to the corresponding FXS port.

If this FXO port is the only one showing no LED indication on the web UI dashboard, it may indicate a hardware issue that requires replacement.

Issue 4: Active Call Page Displays Disconnected Calls

This issue may arise from a low voltage of the inbound analog signal (e.g., disconnect tone) not being detected by the UCM. The FXO port is passive and requires a busy tone from the telecom provider to disconnect and release the line.

• Enable Polarity Reversal
• Disable Caller ID
• Perform PSTN / ACIM Detection
• Ensure Proper UCM Grounding
• Increase the Current Disconnect Threshold to 500 ms
• Set "Tone Country" to custom and configure the busy tone according to your country
• Increase RX Gain, starting from '6' or '12'
• Adjust "FXO Frequency Tolerance" to 250 Hz

Issue 5: Delays?

Inbound calls may experience delays over the PSTN line in the following scenarios:

• When the Ring Group is configured as the Default Destination.
• When dialing from or to a cell phone, where delays can occur due to signaling between SIP, PSTN, and GSM networks.

Try These Steps:

a. Enable Boost Ring: Go to PBX Settings > Interface Settings > Analog Hardware and select "Boost Ring (Peak)."

b. Adjust Analog Buffer: Under DAHDI Settings, modify the Analog Buffer value. Decrease the buffer size and adjust the write frequency until audio latency and quality are acceptable.

• The numbers 4, 8, and 32 represent the buffer size; lower values result in reduced latency.
• Modes: Half: Writes data once the buffer is half-full. Full: Writes data once the buffer is full. Immediate: Writes data immediately, regardless of the buffer level.

c. Conduct a Loopback Test to Verify Delay:

• Connect the FXS1 port to the FXO1 port on the UCM using an RJ11 cable.
• Create an analog trunk for FXO1, along with inbound and outbound routes.
• Set the outbound route pattern to _X.
• Configure the inbound route with the Default Destination set to a specific SIP extension.
• Dial the FXS1 extension number from any UCM extension. This will allow the UCM to route calls from FXO1 to FXS1 and its destination.
• Verify any delays during the call.

Required Debug Logs for Issue Diagnosis

The following logs are needed to be enabled under the Maintenance tab:

• Maintenance > Syslog: PBX Modules > chan_dahdi (All levels) PBX Modules > RTP (All levels) Process Log Level > AVS (All levels)
• Maintenance > Signaling Troubleshooting > Analog Record Trace > Select the FXO ports and click Start

Analysis

You can download the Adobe Audition application to analyze the Analog trace file. The DTMF signaling frequencies are shown in the table below:

The more you know

1. Analog trunks do not support simultaneous calls over the same copper pair.
2. Lifeline Feature: When the UCM is powered off, FXS1 connects to FXO1 and FXS2 connects to FXO2, allowing access to the FXO dial tone via the FXS ports.
3. To avoid noise in the PRI trunk, set the CRC to "None" on the UCM6510's digital port.