I've found that when you make one sip call throught spa3102 when it redirects to fxo port (pstn) it tells inmediatly that the call has been answered always. It does not wait to know the remote party status to send it to the sip caller so it is impossible to get billing info from pstn calls. Do you know if there is any way to change this way of working?
Please verify the firmware version in this unit, the latest firmware in this device is 5.1.10, most of the time the firmware correct this issues.
Make sure that the prefered codec is 711u if this does not work try 729a. This option is under PSTN LINE then under AUDIO CONFIGURATION, if this does not take care your issues please contact us at 866-606-1866 US/CANADA.
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