The number you dialed is not in service

joyful2 · January 14, 2010, 8:38pm

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

Please help!

joyful2 · January 14, 2010, 8:55pm

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

joyful2 · January 14, 2010, 9:24pm

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.

blanchae · January 31, 2010, 7:48am

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 [s@from-pstn:1] NoOp(“DAHDI/7-1”, “No DID or CID Match”