Analog-PSTN-FXO-Grandstream-UCM.

Common Analog Trunk issues in Grandstream UCM63XX/62XX/6510

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Analog-PSTN-FXO-Grandstream-UCM.

Description

This article describes common Analog or PSTN trunk issues that happen in Grandstream PBX UCM63XX/62XX/6510. A list of suggestions is stated to deal with different scenarios in the customer’s setup.

Issue 1: Noise or Echo

  • Try with an analog phone connected to the PSTN line that calls do not produce any static noise
  • Some ADSL setups involve using a splitter or device that modulates the signal and we have seen problems with the signal coming from those setups.
  • Ground the UCM using the grounding screw behind the device to a metal object.
  • Tried changing the codec on the IP phones themselves from the default (PCMU) to G.729A/B and turned on silence suppression
  • Perform PSTN and ACIM detection
  • Reduce RX / TX Gain
  • Select different modes in Echo Cancellation Mode 
  • Please make sure to use other working RJ11 cable
  • Please try to power up the UCM using another working PSU and check if the noise persists.
  • Test with Analog/ SIP/ Softphones to determine the behaviors.

Issue 2: Caller ID not showing or Undetected 0000

  • Confirm with a test that the Caller ID is activated by the Provider; Connect directly an analog phone to the PSTN line and see if the Caller ID is displayed correctly on all incoming calls
  • Ground the UCM properly
  • Increase RX Gain to 6 or 12dB
  • Perform PSTN and ACIM detection
  • No splitter or DSL filter is used before the PSTN line
  • Disable Jitter Butter under PBX Settings
  • Increase the ‘FXO Dial Delay’ to 250ms or 1000ms (max. is 3000ms) 
  • Go to PBX Setting > Analog Hardware > Boost Ringer > Set to Peak
  • Set DTMF Threshold to 1 (revert this change)

Issue 3: FXO port status is not reflected correctly in the Web Interface

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

If the reported FXO port is the only one that doesn’t have an LED or indication lit on the web UI dashboard, it may have a hardware issue and require replacement.

Issue 4: Active Call page is showing disconnected calls

Reason: This could be the low voltage of the inbound analog signal (for eg. disconnect tone) to be detected by UCM. FXO port is a passive port as it requires a busy tone from Telco to disconnect and release the line.

  • Enable Polarity Reversal
  • Disable Caller ID
  • PSTN / ACIM Detection
  • UCM Grounding
  • Increase Current Disconnect Threshold (500 ms)
  • Configured “Tone Country” to custom and set busy tone per your country
  • Increase RX gain (start from ‘6’ or ’12’)
  • Adjust “FXO Frequency Tolerance” to 250Hz.

Issue 5: Delays 

Normally, inbound calls have some delays over the PSTN line if you:

  • configure the Ring Group as Default Destination
  • are dialing from/to a cell phone, the delays are caused by the signaling between SIP > PSTN > GSM

Try these:

a. Select Boost Ring (Peak) under PBX Setting > Interface Setting > Analog Hardware

b. Adjust the Analog Buffer value under DAHDI Settings.

Decrease the buffer size and write frequency accordingly until audio latency and quality are acceptable.

The numbers 4, 8, and 32 indicate the buffer size. The lower it is, the lower the latency.

– Half, Full, and Immediate indicate when data will be written based on how much of the buffer is filled.

– Half – Writes data once the buffer is half-full

– Full – Writes data once the buffer is full

– Immediate – Writes data immediately, regardless of how much buffer is left.

c. Loop backtest to verify the delay

– Connect an FXS1 port to an FXO1 port on the UCM using RJ 11.

– Create an analog trunk for FXO1, and an inbound and outbound route for it.

– The outbound route should be patterned to _X.

– The Inbound Route is set with Default Destination – specific SIP extension.

– Dial out to the FXS1 Extension number from any UCM extension, this allows UCM to route calls out of FXO1 to FXS1 and its destination.

– Verify the delays

Required Debug Logs

Our developers required the following logs to be enabled or turned on under the Maintenance tab:

  • Syslog modules: chan_dahhdi, RTP, AVS (ALL level)
  • Analog Record Trace

Analysis

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

Table 1: DTMF Keypad frequencies
Figure 1: Each number has 2 different frequencies, as indicated by the yellow lines. It is abnormal if there are more than 3 frequencies seen at a number, this could be due to the device faults. You may need to playback the RTP stream or wave file.
Figure 2 shows the received DTMF signal of an inbound call. The signal is weak at the low-frequency numbers causing the UCM to detect the Caller ID even after setting RX Gain to -12. The Service Provider is required to increase the DTMF signal strength to around –10dBm.
Figure 3 shows the good DTMF signal during an inbound call
Figure 4 shows the Analog Signal Raw Data analysis for Caller ID using Audacity
Figure 5 shows the analog signal DTMF before the ring.

The more you know

  1. Analog trunk does not support simultaneous calls over the same copper pair.
  2. Lifeline feature – When UCM is powered off, FXS1 will be connected to FXO1 and FXS2 will be connected to FXO2.  So you can access FXO dialtone via FXS ports. 
  3. Set the CRC to None on the UCM6510’s Digital port to avoid Noise in the PRI trunk. 

References

Analog Signals Analysis