Incoming on PSTN Not Working

[gmcfad]

I know this is an old issue - but I just powered up my PBXIAF a couple of weeks ago and ran all the updates (after the system had been down for several months while I was playing around with Trixbox) and now my inbound PSTN calls drop as soon as they are answered.

Used the tutorial from:
(http://nerdvittles.com/index.php?p=65).

========================================

Below is my extensions_custom.conf entry and data pulled from my "full" log file.
************************************************
;next extension (99) is to handle incoming PSTN calls
exten => 99,1,GotoIf($["${CALLERIDNUM:0:2}" = "00"]?2:3)
exten => 99,2,SetCIDNum(${CALLERIDNUM:2})
exten => 99,3,SetMusicOnHold(default)
exten => 99,4,Answer
exten => 99,5,Wait(1)
exten => 99,6,Background(pls-wait-connect-call)
exten => 99,7,Set(TIMEOUT(digit)=2)
exten => 99,8,Set(TIMEOUT(response)=2)
exten => t,1,Answer
exten => t,2,Wait(1)
exten => t,3,Background(pls-hold-while-try)
exten => t,4,Dial(SIP/2006,20,m)
exten => t,5,VoiceMail(2006@default)
exten => t,6,Hangup
exten => i,1,Answer
exten => i,2,Wait(1)
exten => i,3,Playback(wrong-try-again-smarty)
exten => i,3,Playback(yes-dear)
exten => i,4,Goto(99,5)


************************************************** ******

root@pbx:/var/log/asterisk $ vi full.3
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:1] GotoIf("SIP/asterisk-08bdb500", "0?2:3") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Goto (from-internal,99,3)
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:3] SetMusicOnHold("SIP/asterisk-08bdb500", "default") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:4] Answer("SIP/asterisk-08bdb500", "") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:5] Wait("SIP/asterisk-08bdb500", "1") in new stack
[Feb 14 15:29:25] VERBOSE[19184] logger.c: -- Executing [99@from-internal:6] BackGround("SIP/asterisk-08bdb500", "pls-wait-connect-call") in new stack
[Feb 14 15:29:25] VERBOSE[19184] logger.c: -- <SIP/asterisk-08bdb500> Playing 'pls-wait-connect-call' (language 'en')
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [99@from-internal:7] Set("SIP/asterisk-08bdb500", "TIMEOUT(digit)=2") in new stack
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Digit timeout set to 2
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [99@from-internal:8] Set("SIP/asterisk-08bdb500", "TIMEOUT(response)=2") in new stack
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Response timeout set to 2
[Feb 14 15:29:27] VERBOSE[19184] logger.c: == Auto fallthrough, channel 'SIP/asterisk-08bdb500' status is 'UNKNOWN'
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [h@from-internal:1] Macro("SIP/asterisk-08bdb500", "hangupcall") in new stack


==========================================

I've spent quite a bit of time trying to fix this. I'm kinda at a loss now. Thanks in advance for any suggestions.

 

[jroper]

-- Executing [99@from-internal:6] BackGround("SIP/asterisk-0a1c8e08", "custom/welcome") in new stack
-- <SIP/asterisk-0a1c8e08> Playing 'custom/welcome' (language 'en')
== Spawn extension (from-internal, 99, 7) exited non-zero on 'SIP/asterisk-0a1c8e08'
-- Executing [h@from-internal:1] Macro("SIP/asterisk-0a1c8e08", "hangupcall") in new stack

Hi

From that section above, it would look like its possible that the welcome file is not there, or not playable, as it hangs up right after. The /var/log/asterisk/full log - as viewed via the FreePBX interface is far more descriptive that the CLI output.

Therefore handy hint for the day, do your debugging from the log file, not the asterisk CLI.

Check first that the call comes in, by directing it to an extension, then point the incoming route back to your message, and dial 7777 to simulate an incoming call.

Joe

 

[gmcfad]

My bad - copied someone else's post in an effort to save time.

Below is my extensions_custom.conf entry and data pulled from my "full" log file.
************************************************
;next extension (99) is to handle incoming PSTN calls
exten => 99,1,GotoIf($["${CALLERIDNUM:0:2}" = "00"]?2:3)
exten => 99,2,SetCIDNum(${CALLERIDNUM:2})
exten => 99,3,SetMusicOnHold(default)
exten => 99,4,Answer
exten => 99,5,Wait(1)
exten => 99,6,Background(pls-wait-connect-call)
exten => 99,7,Set(TIMEOUT(digit)=2)
exten => 99,8,Set(TIMEOUT(response)=2)
exten => t,1,Answer
exten => t,2,Wait(1)
exten => t,3,Background(pls-hold-while-try)
exten => t,4,Dial(SIP/2006,20,m)
exten => t,5,VoiceMail(2006@default)
exten => t,6,Hangup
exten => i,1,Answer
exten => i,2,Wait(1)
exten => i,3,Playback(wrong-try-again-smarty)
exten => i,3,Playback(yes-dear)
exten => i,4,Goto(99,5)


********************************************************

root@pbx:/var/log/asterisk $ vi full.3
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:1] GotoIf("SIP/asterisk-08bdb500", "0?2:3") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Goto (from-internal,99,3)
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:3] SetMusicOnHold("SIP/asterisk-08bdb500", "default") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:4] Answer("SIP/asterisk-08bdb500", "") in new stack
[Feb 14 15:29:24] VERBOSE[19184] logger.c: -- Executing [99@from-internal:5] Wait("SIP/asterisk-08bdb500", "1") in new stack
[Feb 14 15:29:25] VERBOSE[19184] logger.c: -- Executing [99@from-internal:6] BackGround("SIP/asterisk-08bdb500", "pls-wait-connect-call") in new stack
[Feb 14 15:29:25] VERBOSE[19184] logger.c: -- <SIP/asterisk-08bdb500> Playing 'pls-wait-connect-call' (language 'en')
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [99@from-internal:7] Set("SIP/asterisk-08bdb500", "TIMEOUT(digit)=2") in new stack
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Digit timeout set to 2
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [99@from-internal:8] Set("SIP/asterisk-08bdb500", "TIMEOUT(response)=2") in new stack
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Response timeout set to 2
[Feb 14 15:29:27] VERBOSE[19184] logger.c: == Auto fallthrough, channel 'SIP/asterisk-08bdb500' status is 'UNKNOWN'
[Feb 14 15:29:27] VERBOSE[19184] logger.c: -- Executing [h@from-internal:1] Macro("SIP/asterisk-08bdb500", "hangupcall") in new stack

 

[jroper]

Hi
[Feb 14 15:29:27] VERBOSE[19184] logger.c: == Auto fallthrough, channel 'SIP/asterisk-08bdb500' status is 'UNKNOWN'

It seems to be neither timing out, and therefore going into the t destinations, nor hearing an invalid digit, therefore hanging up.

I don't really understand what you are trying to achieve here.

What's the aim of the context?

Joe

 

[gmcfad]

I'm trying to get all incoming PSTN calls to ring extension 2006 - which is my home phone. I have an incoming VOIP line that is ringing to an Auto Atendent and it's working fine. I copied all of this from the orginal Nerd Vittles tutorial on how to set up an Asterisk box with a SPA3000