Busy Signal When Dialing Out, Redial Completes Call

Troubleshooting a busy signal when dialing out, if redial completes the call

1. First, check Outgoing Call Rules: go to Setup > Outgoing Calls and check for an Outgoing Call Rule that matches the number dialed, for each call placed (for example, a 10 digit number, or a 9 and 1, plus 10 digits, etc.). If an Outgoing Call Rule is set up, does the extension have permission to use that call rule? Go to Setup > Manage and click pencil icon to modify the extension; look at Outgoing Call Rules tab and verify the call rules are allowed.

2. Are there any Outgoing Call Rules that are missing the provider name, or that have any errors in the Outgoing Call Rule? Is the Outgoing Call Rule Final (YES)?

3. Check Server > Error Logs: do you see an error: "doesn't match any outgoing call rules"?  Is there an error showing the provider may not be working, and was the phone number misdialed?

4. Check Reporting > Call Logs and see if the call appears in the call logs, then click the notepad icon to view the call details. Did the call go out over the provider?

5. Check the digit map setting on the extension. You can try changing to the Universal Digit Map: x.T|*xx.T (go to Setup > Manage > click on pencil icon > Phone Settings tab > Advanced Phone Options > Override Dial Plan: set to YES, replace the entire existing digit map with x.T|*xx.T and Save SIP Extension).

For a Remote Phone Over VPN:

Also, this issue may be seen when you have remote phone connected over a VPN. In some cases, a VPN connection limits the MTU to 1500 and the packet size of the SIP packets sent (such as the INVITE) may exceed the MTU limit, resulting in the firewall rejecting the SIP message with ICMP Fragmentation needed.

Solution

In order to address this issue, you need to increase the MTU size of 1700; this would allow the SIP messages to pass without an issue. See your firewall settings, or refer to the manufacturers documentation for your firewall.

Workaround

If your firewall does not allow the MTU changes, or limits the number to 1500, you can disable codecs on your extension's Phone Settings tab to make the SIP messages smaller (fewer codecs result in a shorter list of supported codecs in the message packets).

How to disable codecs for a single extension:

1. Log into your administrative portal

2. Got to Setup > Manage > click the pencil icon to modify settings for the extension.

3. For Digium phones, click the Phone Settings tab > Advanced Phone Options > scroll down to Codec Settings  > click the red circle under Actions to disable all codecs, except ULAW (set to ON).

4. For a Polycom or other supported phone (Snom), click the Other Manufacturers tab > click the Advanced Phone Options > click the slider to disable all codecs (set to OFF), except ULAW (set to ON).

5. Save SIP Extension (the digium phone will show updating configuration, you may need to manually reboot a Polycom phone). 

To change codec settings for multiple extensions, useBulk-Modify Extensions, with the options listed below:

1. Setup > Manage > click the Bulk-Modify Extensions button.

2. Add the extensions you want to change, in the Phone Extensions search bar, or click the magnifying glass to see a complete extension list.

3. Under the Phone-Extension Fields, select the Phone Settings tab.

4. For Digium phones, select the Digium Phones tab and click Field to modify, then select Audio Codecs from the drop-down, and click the Next button. Under Actions, click the red circle to disable all codecs, except ULAW. The Digium phone should display updating configuration.

5. For Polycom phones, click the Other Manufacturers tab and click Field to modify, then select Audio & Video Codecs from the drop-down, and click Next. Set all options to OFF, except ULAW (set to ON).

6. Click Save Modifications.

7. You may need to reboot the Polycom phone to update the configuration.