CONFIGURATION GUIDE FOR ASTERISK PBX PDF

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Once you have set up and configured Asterisk, you can use the following details to start making calls. These details are visible on your customer control panel if you have been allocated a SIP trunk. Asterisk sip. You should have something similar to the following in your extensions.

Please check your configuration thoroughly as VoiceHost does not accept any liability for fraudulent use of your Asterisk system. Inbound telephone numbers need to be mapped in the Asterisk system in E. Troubleshooting Asterisk configurations Calls to VoiceHost error with "all circuits busy" or "congestion" - this is the default configuration of Asterisk regardless of the actual error generated unless Asterisk is updated to send back the real error rather than the changed error.

Just because a trunk is showing as registered does not mean it will authenticate correctly. For inbound calls, ensure the mapping is correct and a viable destination set in the PBX, e.

The codecs in the SIP trunk configuration within Asterisk need to be aligned to use one of the above codecs. As a minimum, offer both alaw and ulaw codecs when sending and receiving calls for widest compatibility with fixed line and mobile operators. Inbound calls to Asterisk fail with SIP error Request Timeout - check the inbound number is mapped in the system correctly, if necessary the SIP trunk on the portal can be configured to strip the plus when sending calls to Asterisk, e.

Check the trunk is registered. Ascertain how long the error took to come back if it was immediate the trunk is usually unregistered if it took a few seconds the number is usually not mapped correctly.

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