Contents / Telephony

Troubleshooting: Telephony

No sound in browser telephony

  • Check whether your browser recognizes the audio device:

Select the device, restart your browser, and clear cache. If there is more than one device on the list, try selecting them one by one and open your browser for a call test each time.

  • Use Google Chrome to avoid errors.
  • Try to refresh the page in the browser and clear cache using Ctrl+F5 or an alternative method. Clear cookies and get reauthorized in the system.
  • Reboot your router.
  • Some software (antivirus, firewall) can block the telephony ports. Try taking the audio test with this software disabled.
  • Check whether the ports are open (see below).


No incoming/outbound call connection

  • Check whether the SIP telephony ports are open

How to open/forward ports for SIP devices:

1. Open outgoing SIP traffic from IP phone:
outgoing TCP from IP phone address to all addresses, from port 5060 to ports 1024-65535.

2. Open incoming SIP traffic to IP phone:
incoming TCP to IP phone address from all addresses, to port 5060 from ports 1024-65535.

3. Open outgoing RTP traffic from IP phone:
outgoing UDP from IP phone address to all addresses, from ports: {default port from device settings} — {default port from device settings + 10} to ports 1024-65535.

4. Open incoming RTP traffic to IP phone:
incoming UDP to IP phone address from all addresses, to ports {default port from device settings} — {default port from device settings + 10} from ports 1024-65535.

5. Open outgoing traffic from IP phone:
outgoing UDP from IP phone address to all addresses, from port 69 to ports 1024-65535.

6. Open incoming traffic to IP phone:
incoming UDP to IP phone address from all addresses, to port 69 from ports 1024-65535.

The default port from device settings for CISCO ATA-186 is 16380-16390; 5000-5010 — for Grandstream, and 16500-16510 — for the examples above. You can also open (or forward in case you are behind NAT) all the UDP ports from 5000 to 65535.

 

Browser call connection is taking too long

Connection can be slowed down by several factors:

1) Internet connection speed (including ping);

2) routing features (depending on your network hardware and that of your Internet provider);

3) the connection speed of your IP telephony provider.

You can improve the routing quality by making changes to connection settings.

In case call completion is taking too long (due to network/router configuration), go to Main menu — Settings — Integrations — Telephony — Advanced (upper tab) and disable "Use ICE and STUN servers to establish connection". This can speed up virtual PABX server connection, but can also result in total loss of connection, if you are behind NAT:


You may also be interested in: Reminders

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