Call dropped after 1 ring A400P FXO-100 rev1.2

Trixbox 2.8.0.4. OpenVox A400P with two FXO-100 and two FXS-100. They are in the order for ports 1 - 4 FXO FXO FXS FXS. Port 1 is FXO-100 rev1.2, and port 2 FXO-100 rev1.3. I purchased the rev1.3 about a month ago. I have two analog lines from the provider. I can call out from both, even tested simultaneously. I can call in on the FXO rev1.3 but the port 1 with the FXO rev1.2 rings once and hangs up. The caller does not get disconnected. I compared the sections of the log from /var/log/asterisk/full for the calls in on both lines.

I just have excerpts from when the incoming call works and when it does not. Most of the similar lines were omitted.

Works FXO-100 rev1.3 in port 2
[Oct 22 20:10:24] VERBOSE[12989] logger.c: -- Starting simple switch on 'DAHDI/2-1'
[Oct 22 20:10:27] NOTICE[12989] chan_dahdi.c: Got event 18 (Ring Begin)...
[Oct 22 20:10:30] NOTICE[12989] chan_dahdi.c: Got event 2 (Ring/Answered)...
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- Executing [s@from-pstn:1] Set("DAHDI/2-1", "__FROM_DID=s") in new stack
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- Executing [s@from-pstn:2] Gosub("DAHDI/2-1", "app-blacklist-check,s,1") in new stack
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- Executing [s@app-blacklist-check:1] GotoIf("DAHDI/2-1", "0?blacklisted") in new stack
...
... All lines are similar
...
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- Executing [s@macro-dial:38] Dial("DAHDI/2-1", "SIP/200,10,trM(auto-blkvm)") in new stack
[Oct 22 20:10:30] VERBOSE[12989] logger.c: == Using SIP RTP TOS bits 184
[Oct 22 20:10:30] VERBOSE[12989] logger.c: == Using SIP RTP CoS mark 5
[Oct 22 20:10:30] VERBOSE[12989] logger.c: == Using SIP VRTP TOS bits 136
[Oct 22 20:10:30] VERBOSE[12989] logger.c: == Using SIP VRTP CoS mark 6
[Oct 22 20:10:30] DEBUG[2746] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- Called 200
[Oct 22 20:10:30] VERBOSE[12989] logger.c: -- SIP/200-00000020 is ringing
[Oct 22 20:10:40] VERBOSE[12989] logger.c: -- Nobody picked up in 10000 ms
[Oct 22 20:10:40] VERBOSE[12989] logger.c: -- Executing [s@macro-dial:39] Set("DAHDI/2-1", "HuntLoop=1") in new stack
[Oct 22 20:10:40] VERBOSE[12989] logger.c: -- Executing [s@macro-dial:40] GotoIf("DAHDI/2-1", "1?a46") in new stack
[Oct 22 20:10:40] VERBOSE[12989] logger.c: -- Goto (macro-dial,s,42)
...
Continues on to ring groups etc...

Does not work FXO-100 rev1.2 in port 1
[Oct 22 20:22:36] VERBOSE[13022] logger.c: -- Starting simple switch on 'DAHDI/1-1'
[Oct 22 20:22:44] WARNING[13022] chan_dahdi.c: CallerID returned with error on channel 'DAHDI/1-1'
[Oct 22 20:22:44] VERBOSE[13022] logger.c: -- Executing [s@from-pstn:1] Set("DAHDI/1-1", "__FROM_DID=s") in new stack
[Oct 22 20:22:44] VERBOSE[13022] logger.c: -- Executing [s@from-pstn:2] Gosub("DAHDI/1-1", "app-blacklist-check,s,1") in new stack
[Oct 22 20:22:44] VERBOSE[13022] logger.c: -- Executing [s@app-blacklist-check:1] GotoIf("DAHDI/1-1", "0?blacklisted") in new stack
...
... All lines are similar
...
Oct 22 20:22:44] VERBOSE[13022] logger.c: -- Executing [s@macro-dial:38] Dial("DAHDI/1-1", "SIP/200,10,trM(auto-blkvm)") in new stack
[Oct 22 20:22:44] VERBOSE[13022] logger.c: == Using SIP RTP TOS bits 184
[Oct 22 20:22:44] VERBOSE[13022] logger.c: == Using SIP RTP CoS mark 5
[Oct 22 20:22:44] VERBOSE[13022] logger.c: == Using SIP VRTP TOS bits 136
[Oct 22 20:22:44] VERBOSE[13022] logger.c: == Using SIP VRTP CoS mark 6
[Oct 22 20:22:44] VERBOSE[13022] logger.c: -- Called 200
[Oct 22 20:22:44] DEBUG[2746] pbx.c: FONALITY: This thread has already held the conlock, skip locking
[Oct 22 20:22:44] VERBOSE[13022] logger.c: -- SIP/200-0000002f is ringing
[Oct 22 20:22:52] VERBOSE[13022] logger.c: == Spawn extension (macro-dial, s, 38) exited non-zero on 'DAHDI/1-1' in macro 'dial'
[Oct 22 20:22:52] VERBOSE[13022] logger.c: == Spawn extension (ext-group, 600, 14) exited non-zero on 'DAHDI/1-1'
[Oct 22 20:22:52] VERBOSE[13022] logger.c: -- Hungup 'DAHDI/1-1'

Any suggestion will be appreciated

Can you write a dialplan for port 1 rev 1.2? Just to find out if this module is all right in hardware. like:

exten => s, 1, answer()
exten => s, n, Playback(demo-instruct)
exten => s, n, Hangup

When you dial in from port 1, asterisk play a voice file "demo-instruct" for you, please try and send me
the log file.