I have Asterisk running on a pi with an FXS port to which is connected an analogue phone. Asterisk is registered with a sip service, so the unit is acting as an ATA
Incoming calls ,from the SIP provider to the analogue phone, work fine. Likewise, DTMF tones for outgoing calls are detected…but as soon as the first tone is detected, asterisk hangs up - see below (the result of picking up the analogue handset and pressing “0”):
-- Starting simple switch on 'DAHDI/1-1' [Apr 8 23:00:03] DTMF[C-00000000]: channel.c:4126 __ast_read: DTMF begin '0' received on DAHDI/1-1 [Apr 8 23:00:03] DTMF[C-00000000]: channel.c:4130 __ast_read: DTMF begin ignored '0' on DAHDI/1-1 [Apr 8 23:00:03] DTMF[C-00000000]: channel.c:4040 __ast_read: DTMF end '0' received on DAHDI/1-1, duration 89 ms [Apr 8 23:00:03] DTMF[C-00000000]: channel.c:4110 __ast_read: DTMF end passthrough '0' on DAHDI/1-1 -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1'
I am still getting to grips with the various config files…but any suggestions as to what I might be missing here?