I am having one of those days. We just replaced an old Asterisk
1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost
everything is working except for some incoming calls directed to a Cisco
SPA-8000. The PSTN trunk is SIP. Only calls coming from the PSTN to a
direct DID that just rings an extension on the SPA get no incoming
audio. All other calls, including calls from the PSTN that go through
the main IVR or operator have audio.
I made sure that the trunk has direct_media=no. I checked the SPA
configuration to make sure it is not using NAT. Only the SPA suffers
from this as regular SIP phones can receive calls from their DID with no
problems. This is the first time I use an SPA analog adapter with
PJSIP. They work great with chan_sip so I do not know what maybe wrong
here. Anyone using an SPA-8000 with PJSIP? Any settings I should check
on the SPA or in Asterisk?
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)8116-9161
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