The number you dialed is not in service

I installed FreePBX with a 12 port analog card. The first 5 ports are FXS and rest 7 are FXO ports. dahdi_genconf correctly generated the dahdi-channel.conf. I can dial out from extensions to outside through SIP route. I can also call each other on those 5 analog FXS ports. But when incoming call ringing, Asterisk announces “The number you dialed is not in service, please check your number and try again”. The phone line service does not have caller ID service, but I defined the CID in the truck section, also defined the Zap Channle DIDs and Inbound Route.

Following is the /var/log/asterisk/full log about that call progress:
[Jan 14 12:30:52] VERBOSE[6115] chan_dahdi.c: – Starting simple switch on ‘DAHDI/7-1’
[Jan 14 12:30:55] NOTICE[6115] chan_dahdi.c: Got event 18 (Ring Begin)…
[Jan 14 12:30:58] NOTICE[6115] chan_dahdi.c: Got event 2 (Ring/Answered)…
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:1] NoOp(“DAHDI/7-1”, "Entering from-zaptel with DID == ") in new stac
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[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:2] Ringing(“DAHDI/7-1”, “”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:3] Set(“DAHDI/7-1”, “DID=s”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:4] NoOp(“DAHDI/7-1”, “DID is now s”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:5] GotoIf(“DAHDI/7-1”, “0?zapok:notzap”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Goto (from-zaptel,s,6)
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:6] Goto(“DAHDI/7-1”, “from-pstn,s,1”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Goto (from-pstn,s,1)
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:1] NoOp(“DAHDI/7-1”, “No DID or CID Match”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:2] Answer(“DAHDI/7-1”, “”) in new stack
[Jan 14 12:30:58] VERBOSE[6115] pbx.c: – Executing [[email protected]:3] Wait(“DAHDI/7-1”, “2”) in new stack
[Jan 14 12:31:00] VERBOSE[6115] pbx.c: – Executing [[email protected]:4] Playback(“DAHDI/7-1”, “ss-noservice”) in new stack
[Jan 14 12:31:00] VERBOSE[6115] file.c: – <DAHDI/7-1> Playing ‘ss-noservice.ulaw’ (language ‘en’)
[Jan 14 12:31:05] VERBOSE[6115] pbx.c: – Executing [[email protected]:5] SayAlpha(“DAHDI/7-1”, “”) in new stack
[Jan 14 12:31:05] VERBOSE[6115] pbx.c: – Executing [[email protected]:6] Hangup(“DAHDI/7-1”, “”) in new stack
[Jan 14 12:31:05] VERBOSE[6115] pbx.c: == Spawn extension (from-pstn, s, 6) exited non-zero on ‘DAHDI/7-1’
[Jan 14 12:31:05] VERBOSE[6115] pbx.c: – Executing [[email protected]:1] Hangup(“DAHDI/7-1”, “”) in new stack
[Jan 14 12:31:05] VERBOSE[6115] pbx.c: == Spawn extension (from-pstn, h, 1) exited non-zero on ‘DAHDI/7-1’
[Jan 14 12:31:05] VERBOSE[6115] chan_dahdi.c: – Hungup ‘DAHDI/7-1’

Please help!

I just updated the freepbx core and all the latest packages through module admin, it still doing the same :frowning:

In FreePBX Asterisk Info page, it shows under Channels:

Channel Location State Application(Data)
0 active channels
0 active calls

Should I see all the analog FXO ports as active channels here? dahdi-scan showing them all good. dahdi-channels.conf showing them all in use. I am confused where to look for the problem? If anyone need me to run any debug code, I will be more than happy to do that --if that could help.

It is the Asterisk log file and is very large. Serach for the term dahdi and you will see the dahdi hardware drivers registering with the Asterisk channels. Any problems will show up as errors such as mismatched signaling fxo_ls vs. fxo_ks. Lots of good clues in there.

From the Asterisk CLI> dahdi show channels

It should show which channels have registered. What is your inbound route? According to the log file, it says Executing [[email protected]:1] NoOp(“DAHDI/7-1”, “No DID or CID Match”