[Asterisk-Users] Odd internal vs. External dialplan issue

[Steven]

I have the following in my extensions.conf

[ext-local]
exten => _53XX,1,Wait(2)
exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom
exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,)

This is used to match inbound caller-id for my legacy PBX.
It works fine for inbound calls, but not for internal SIP calls.

If I call from a SIP phone that is also in [ext-local], it looks like it is calling, but never connects.

excerpt from log when called from pstn zap PRI:
Apr 28 14:18:16 VERBOSE[28452] logger.c:     -- Called g2/5386
Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/27-1 to read format slin
Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to write format slin
Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/2-1 to read format slin
Apr 28 14:18:16 DEBUG[28452] channel.c: Set channel Zap/27-1 to write format slin
Apr 28 14:18:16 DEBUG[11073] devicestate.c: Changing state for Zap/27 - state 2 (In use)
Apr 28 14:18:16 DEBUG[28457] app_queue.c: Device 'Zap/27' changed to state '2' (In use)
Apr 28 14:18:17 DEBUG[11111] chan_zap.c: Enabled echo cancellation on channel 27
Apr 28 14:18:17 DEBUG[11073] channel.c: Avoiding initial deadlock for 'Zap/27-1'
Apr 28 14:18:17 VERBOSE[28452] logger.c:     -- Zap/27-1 is ringing

excerpt from log when called from internal SIP extension:
Apr 28 14:18:25 VERBOSE[28477] logger.c:     -- Called g2/5386
Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel Zap/27-1 to read format ulaw
Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to write format ulaw
Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel SIP/5665-e60f to read format ulaw
Apr 28 14:18:25 DEBUG[28477] channel.c: Set channel Zap/27-1 to write format ulaw
Apr 28 14:18:25 DEBUG[28482] app_queue.c: Device 'Zap/27' changed to state '2' (In use)
Apr 28 14:18:25 DEBUG[28477] rtp.c: Ooh, format changed from unknown to ulaw

I never get a ringing log entry if dialed from SIP.
This SIP phone can call other extensions in asterisk as well as native (voicemail) and PSTN calls out ZAP/g0.

I have tried various dial strings ( like the Dial command instead of the macro) and they all work for incoming PSTN calls and not
for SIP.

I am at a loss where to find the problem.

Please advise.