SPA-3102 Outbound Calls circuit-busy

I’m helping a friend set up a 3102 in Costa Rica and am running into some troubles getting outbound calls to work through the gateway.

The PBX is running a year or two old version of Asterisk but can successfully make calls through two external SIP trunks as well as an internal ZAP FXO card.

Incoming calls through the SPA-3102 work fine but I’m getting an “all circuits are busy” message when I make outgoing calls.

The SPA-3102 is showing as registered and I can see the device if I do a sip show peers. Has anyone had any experience with this before?

Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/5060-00000b13", "OUTNUM=22****26") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/5060-00000b13", "custom=SIP/1pstn") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/5060-00000b13", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/5060-00000b13", "dialout-trunk-predial-hook,") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/5060-00000b13", "") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/5060-00000b13", "0?bypass,1") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/5060-00000b13", "0?customtrunk") in new stack Nov 4 19:10:05 VERBOSE [15948] pbx.c: -- Executing [s@macro-dialout-trunk:19] Dial("SIP/5060-00000b13", "SIP/1pstn/22****26,300,") in new stack Nov 4 19:10:05 VERBOSE [15948] netsock.c: == Using SIP RTP TOS bits 184 Nov 4 19:10:05 VERBOSE [15948] netsock.c: == Using SIP RTP CoS mark 5 Nov 4 19:10:05 VERBOSE [15948] app_dial.c: -- Called 1pstn/22****26 Nov 4 19:10:05 VERBOSE [15948] app_dial.c: -- SIP/1pstn-00000b14 is circuit-busy Nov 4 19:10:05 VERBOSE [15948] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)

Please provide the SIP dialogue from the INVITE through to the final response, e.g. by using sip set debug on.