I've loaded FreePBX on a RasPi and configured it as a MeshPhone using the instructions provided by Mark, N2MH.
I can dial local extensions but not outbound extensions.
Dialing '73 or 73#' does not produce a second dial tone. See the results below. I've changed my IP for security purposes.
I'm way in over my level of understanding with this project so any help will be appreciated.
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [73@from-internal:1] DISA("SIP/3100-00000022", "no-password,CallSigns,,") in new stack
WARNING[1125]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission 686953f-c0a80101-1-1e@my.ip.add.res for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
WARNING[1125]: chan_sip.c:4092 retrans_pkt: Hanging up call 686953f-c0a80101-1-1e@my.ip.add.res - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (from-internal, 73, 1) exited non-zero on 'SIP/3100-00000022'
-- Executing [h@from-internal:1] Macro("SIP/3100-00000022", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/3100-00000022", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/3100-00000022", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] NoOp("SIP/3100-00000022", " monior file= ") in new stack
-- Executing [s@macro-hangupcall:5] AGI("SIP/3100-00000022", "attendedtransfer-rec-restart.php,,") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
-- <SIP/3100-00000022>AGI Script attendedtransfer-rec-restart.php completed, returning 0
-- Executing [s@macro-hangupcall:6] Hangup("SIP/3100-00000022", "") in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'SIP/3100-00000022' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3100-00000022'
David - N5MXI
In extensions_custom.conf, make sure that you have uncommented the various # include statements at the top of the page. In particular, for dial-by-callsign (which is accessed by 73)
3. Reload asterisk and make a test call
In addition, make sure callsigns.conf is present.
And, as AK4FA asks, does 78 or 78# get you a second dial tone?
Mark, N2MH