Upgrade from FreePBX 13 to Incredible PBX 13 with gvsip

General Help
1

Since GV removed support for Motif/GV, I am trying to switch to Incredible PBX with GVSIP on vmware. I have a SPA232D and SPA122 for 3 extensions using analog phones. Everything was configured and working on FreePBX calling outbound via pstn line. I have set up Incredible PBX with the same configuration as FreePBX with trunks, incoming / outgoing routes, and extensions. SPA232D PSTN and Line 1, SPA122 Line 1 and LIne 2 all register fine. I can receive incoming PSTN and GV calls through the SPA232D and make GV calls out. I am having problems with calling out through SPA232D to standard POTS line. Here is the end of the log file that shows outgoing sip connecting to trunk and bridging to line 1, but exiting due to dialout-trunk macro and hanging up.

[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:21]
ExecIf("SIP/100-00000040", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)2188942441)") in new stack
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf("SIP/100-00000040", "0?customtrunk") in new stack
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-dialout-trunk:23] Dial("SIP/100-00000040", "SIP/1-pstn/2182960712,300,T") in new stack
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] netsock2.c: Using SIP RTP TOS bits 184
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] netsock2.c: Using SIP RTP CoS mark 5
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] app_dial.c: Called SIP/1-pstn/2182960712
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] app_dial.c: SIP/1-pstn-00000041 answered SIP/100-00000040
[2018-10-30 12:20:19] VERBOSE[23082][C-00000021] bridge_channel.c: Channel SIP/1-pstn-00000041 joined 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc>
[2018-10-30 12:20:19] VERBOSE[23081][C-00000021] bridge_channel.c: Channel SIP/100-00000040 joined 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc>
[2018-10-30 12:20:20] VERBOSE[23082][C-00000021] bridge_channel.c: Channel SIP/1-pstn-00000041 left 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc>
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] bridge_channel.c: Channel SIP/100-00000040 left 'simple_bridge' basic-bridge <f8543099-ac21-4989-a665-3a795fd6c9fc>
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on 'SIP/100-00000040' in macro 'dialout-trunk'
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Spawn extension (from-internal, 2182960712, 6) exited non-zero on 'SIP/100-00000040'
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [h@from-internal:1] Macro("SIP/100-00000040", "hangupcall") in new stack
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/100-00000040", "1?theend") in new stack
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/100-00000040", "0?Set(CDR(recordingfile)=)") in new stack
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/100-00000040", "") in new stack
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/100-00000040' in macro 'hangupcall'
[2018-10-30 12:20:20] VERBOSE[23081][C-00000021] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000040'

Any assistance would be greatly appreciated.

2

Did some further tracing with sip log on the SPA232 and found some setting was causing the problem. I did a factory default for voice info and started over with the config and got it working.

3

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