@mlaihk wrote:
I am using and OBI202 acting as an FXO for my Freepbx 14 (asterisk 14). Everything is connected via PJSIP, together with 3 SIP trunks to an ITSP.
Everything works except for the OBI202 FXO with incoming calls. It will ring and if not picked up by an extension within the first ring, incoming call will get disconnected.In the asterisk Log, I am seeing errors like this:
app_dial.c: Connected line update to PJSIP/obi28907870-000000d3 prevented.before it immediately goes to the hangup script with code 127.
Out going calls using the OBI202 FXO is fine, so is incoming and outgoing on the other SIP trunks.
The OBI202 FXO and the Freepbx14 box is on the same network and has the same RTP ports configured so not likely a NAT problem.
Any help or insight will be much appreciated!
Marshall
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