Author Topic: No incoming phone calls?  (Read 5737 times)

brononius

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No incoming phone calls?
« on: January 08, 2015, 02:19:41 pm »
Hey,

After a reinstallation of 1004, my incoming calls aren't working anymore.
I can call outside, I can call internally, but when you try to dial in, i'm directly forwarded to the voicemail.

When I check my phone devices, all phones are in there, but they're not registred? Not sure if that's normal?

Any clue what I can check?

I did an import of my old databases (pluto & asterisk).
I needed also to rename the files /usr/pluto/bin/configure_cisco79xx.pl since they crash my installation completely. My sql-database went crazy, and in my processes, a lot of these scripts were running.
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thor

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Re: No incoming phone calls?
« Reply #1 on: January 21, 2015, 02:03:49 am »
One thing I've checked on my own system is whether the voicemail offered by my provider is enabled/disabled (I made that mistake once or twice).

In a terminal, after
Code: [Select]
sudo asterisk -rvvvvv, doing a 'sip show peers' or 'sip show registry'. All your phones should be registered. If not, track down why.

Not sure if you could do this since you reinstalled 1004, but compare 'dialplan show' of the old with the new. I've found asterisk to be a bear when tracking down errors.

Also, check the at home, or away status (go into lmce-admin website ->  Security -> Security status) and how it affects the way your telephones answer/ring (Telecom -> Call routing); on that same page, check: "# of seconds to ring before ivr (0=none)".

brononius

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Re: No incoming phone calls?
« Reply #2 on: January 22, 2015, 08:33:45 am »
Hey,

For the moment, i've got 7 internal SIP phones online, with status unmonitored.
And 1 SIP account (SIP-provider), but status unknown. Don't know if this is correct?

Strange thing is that sometimes, I can phone in, sometimes, the SIP provider answer that it's not registred. And sometimes I'm getting the answering of linuxmce (asterisk?).
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brononius

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Re: No incoming phone calls?
« Reply #3 on: January 22, 2015, 10:55:00 am »
My house mode is unarmed.


Some extra info when a calls enters, and when it's forwarded to voicemail

Code: [Select]
== Using SIP RTP CoS mark 5
    -- Executing [121231212@from-trunk:1] Set("SIP/121231212-00000011", "__FROM_DID=121231212")
    -- Executing [121231212@from-trunk:2] Set("SIP/121231212-00000011", "PAI="1234121212" <sip:1234121212@91.208.12.133>")
    -- Executing [121231212@from-trunk:3] Set("SIP/121231212-00000011", "PAF="1234121212" <sip:1234121212@91.208.12.133>;tag=as1b68bc71")
    -- Executing [121231212@from-trunk:4] Set("SIP/121231212-00000011", "CURRENT_PAI_LENGTH=43")
    -- Executing [121231212@from-trunk:5] Set("SIP/121231212-00000011", "CURRENT_PAF_LENGTH=58")
    -- Executing [121231212@from-trunk:6] gotoif("SIP/121231212-00000011", "1 ? 11")
    -- Goto (from-trunk,121231212,11)
    -- Executing [121231212@from-trunk:11] noop("SIP/121231212-00000011", "config p asserted id ")
    -- Executing [121231212@from-trunk:12] set("SIP/121231212-00000011", "tmpcid=1234121212@91.208.12.133>")
    -- Executing [121231212@from-trunk:13] Set("SIP/121231212-00000011", "tmpcid=1234121212")
    -- Executing [121231212@from-trunk:14] Set("SIP/121231212-00000011", "CALLERID(num)=1234121212")
    -- Executing [121231212@from-trunk:15] Noop("SIP/121231212-00000011", "Incoming call from 1234121212")
    -- Executing [121231212@from-trunk:16] Set("SIP/121231212-00000011", "FAX_RX=")
    -- Executing [121231212@from-trunk:17] Set("SIP/121231212-00000011", "CALLERID(ani)=")
    -- Executing [121231212@from-trunk:18] Goto("SIP/121231212-00000011", "custom-linuxmce,102,1")
    -- Goto (custom-linuxmce,102,1)
    -- Executing [102@custom-linuxmce:1] AGI("SIP/121231212-00000011", "lmce-phonebook-lookup.agi")
    -- Launched AGI Script /usr/share/asterisk/agi-bin/lmce-phonebook-lookup.agi
    -- AGI Script Executing Application: (NoOp) Options: (Checking if name exists for 1234121212 in phone book and attempting to add caller name)
    -- AGI Script Executing Application: (NoOp) Options: (Number not found in phone book)
    -- <SIP/121231212-00000011>AGI Script lmce-phonebook-lookup.agi completed, returning 0
    -- Executing [102@custom-linuxmce:2] AGI("SIP/121231212-00000011", "lmce-gethousemode.agi")
    -- Launched AGI Script /usr/share/asterisk/agi-bin/lmce-gethousemode.agi
    -- AGI Script Executing Application: (Set) Options: (HOUSEMODE=1)
    -- <SIP/121231212-00000011>AGI Script lmce-gethousemode.agi completed, returning 0
    -- Executing [102@custom-linuxmce:3] Set("SIP/121231212-00000011", "CHANNEL(language)='en'")
    -- Executing [102@custom-linuxmce:4] Goto("SIP/121231212-00000011", "from-lmce-custom,102-hm1,1")
    -- Goto (from-lmce-custom,102-hm1,1)
    -- Executing [102-hm1@from-lmce-custom:1] Dial("SIP/121231212-00000011", "Local/200@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trusted&Local/@trust")
    -- Called Local/200@trusted
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [200@trusted:1] Macro("Local/200@trusted-686f;2", "exten-vm,novm,200")
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [s@macro-exten-vm:1] Macro("Local/200@trusted-686f;2", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("Local/200@trusted-686f;2", "user-callerid: 1234121212 1234121212") in new stack
    -- Executing [s@macro-user-callerid:2] Set("Local/200@trusted-686f;2", "AMPUSER=1234121212") in new stack
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [s@macro-user-callerid:3] GotoIf("Local/200@trusted-686f;2", "1?report") in new stack
    -- Goto (macro-user-callerid,s,13)
    -- Executing [s@macro-user-callerid:13] NoOp("Local/200@trusted-686f;2", "TTL:  ARG1: novm") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("Local/200@trusted-686f;2", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:15] Set("Local/200@trusted-686f;2", "__TTL=64") in new stack
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [s@macro-user-callerid:16] GotoIf("Local/200@trusted-686f;2", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("Local/200@trusted-686f;2", "Using CallerID "1234121212" <1234121212>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("Local/200@trusted-686f;2", "FROMCONTEXT=exten-vm") in new stack
    -- Executing [s@macro-exten-vm:3] Set("Local/200@trusted-686f;2", "VMBOX=novm") in new stack
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [s@macro-exten-vm:4] Set("Local/200@trusted-686f;2", "EXTTOCALL=200") in new stack
    -- Executing [s@macro-exten-vm:5] Set("Local/200@trusted-686f;2", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("Local/200@trusted-686f;2", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("Local/200@trusted-686f;2", "RT=""") in new stack
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
    -- Executing [s@macro-exten-vm:8] Macro("Local/200@trusted-686f;2", "record-enable,200,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("Local/200@trusted-686f;2", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("Local/200@trusted-686f;2", "recordingcheck,20150122-104211,1421919731.105") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trusted while calling Local channel
    -- Couldn't call Local/@trusted
[Jan 22 10:42:11] NOTICE[11641]: chan_local.c:899 local_call: No such extension/context @trust while calling Local channel
    -- Couldn't call Local/@trust
 recordingcheck,20150122-104211,1421919731.105: Inbound recording not enabled
    -- <Local/200@trusted-686f;2>AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("Local/200@trusted-686f;2", "No recording needed") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("Local/200@trusted-686f;2", "dial,"",tr,200") in new stack
    -- Executing [s@macro-dial:1] GotoIf("Local/200@trusted-686f;2", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("Local/200@trusted-686f;2", "dialparties.agi") in new stack
    -- Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Starting New Dialparties.agi
 dialparties.agi: Caller ID name is '1234121212' number is '1234121212'
       > dialparties.agi: USE_CONFIRMATION:  'FALSE'
       > dialparties.agi: RINGGROUP_INDEX:   ''
 dialparties.agi: Methodology of ring is  'none'
    -- dialparties.agi: Added extension 200 to extension map
       > dialparties.agi: Extension 200 has call screening off
    -- dialparties.agi: Extension 200 cf is disabled
    -- dialparties.agi: Extension 200 do not disturb is disabled
       > dialparties.agi: extnum 200 has:  cw: 0; hascfb: 0 [] hascfu: 0 []
 dialparties.agi: EXTENSION_STATE: 4 (UNAVAILABLE)
 dialparties.agi: Extension 200 has ExtensionState: 4
    -- dialparties.agi: Checking CW and CFB status for extension 200
    -- dialparties.agi: dbset CALLTRACE/200 to 1234121212
    -- dialparties.agi: Filtered ARG3: 200
    -- <Local/200@trusted-686f;2>AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:10] Dial("Local/200@trusted-686f;2", "SIP/200,"",tr") in new stack
[Jan 22 10:42:11] WARNING[11656]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dial:11] Set("Local/200@trusted-686f;2", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:10] Set("Local/200@trusted-686f;2", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:11] GosubIf("Local/200@trusted-686f;2", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("Local/200@trusted-686f;2", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:13] Set("Local/200@trusted-686f;2", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:14] NoOp("Local/200@trusted-686f;2", "Voicemail is 'novm'") in new stack
    -- Executing [s@macro-exten-vm:15] GotoIf("Local/200@trusted-686f;2", "1?s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] PlayTones("Local/200@trusted-686f;2", "congestion") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] Congestion("Local/200@trusted-686f;2", "10") in new stack
    -- Local/200@trusted-686f;1 is circuit-busy
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 2) exited non-zero on 'Local/200@trusted-686f;2' in macro 'exten-vm'
  == Everyone is busy/congested at this time (1:0/1/0)
  == Spawn extension (trusted, 200, 1) exited non-zero on 'Local/200@trusted-686f;2'
    -- Executing [h@trusted:1] Macro("Local/200@trusted-686f;2", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("Local/200@trusted-686f;2", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("Local/200@trusted-686f;2", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("Local/200@trusted-686f;2", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("Local/200@trusted-686f;2", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("Local/200@trusted-686f;2", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("Local/200@trusted-686f;2", "") in new stack
  == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/200@trusted-686f;2' in macro 'hangupcall'
  == Spawn extension (trusted, h, 1) exited non-zero on 'Local/200@trusted-686f;2'
    -- Executing [102-hm1@from-lmce-custom:2] Goto("SIP/121231212-00000011", "from-lmce-custom,102-hm1-CONGESTION,1")
    -- Goto (from-lmce-custom,102-hm1-CONGESTION,1)
    -- Executing [102-hm1-CONGESTION@from-lmce-custom:1] Goto("SIP/121231212-00000011", "from-lmce-custom,100,1")
    -- Goto (from-lmce-custom,100,1)
    -- Executing [100@from-lmce-custom:1] Voicemail("SIP/121231212-00000011", "100")
    -- <SIP/121231212-00000011> Playing 'vm-intro.gsm' (language ''en'')
  == Spawn extension (from-lmce-custom, 100, 1) exited non-zero on 'SIP/121231212-00000011'


Some extra info when a call enters, and i'm hearing that the number is not in service:

Code: [Select]
  == Using SIP RTP CoS mark 5
    -- Executing [121231212@from-sip-external:1] NoOp("SIP/91.208.12.133-00000017", "Received incoming SIP connection from unknown peer to 121231212") in new stack
    -- Executing [121231212@from-sip-external:2] Set("SIP/91.208.12.133-00000017", "DID=121231212") in new stack
    -- Executing [121231212@from-sip-external:3] Goto("SIP/91.208.12.133-00000017", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/91.208.12.133-00000017", "0?from-trunk,121231212,1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/91.208.12.133-00000017", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2015-01-22 10:55:54.605 CET.
    -- Executing [s@from-sip-external:3] Answer("SIP/91.208.12.133-00000017", "") in new stack
    -- Executing [s@from-sip-external:4] Wait("SIP/91.208.12.133-00000017", "2") in new stack
    -- Executing [s@from-sip-external:5] Playback("SIP/91.208.12.133-00000017", "ss-noservice") in new stack
    -- <SIP/91.208.12.133-00000017> Playing 'ss-noservice.gsm' (language 'en')
  == Spawn extension (from-sip-external, s, 5) exited non-zero on 'SIP/91.208.12.133-00000017'
    -- Executing [h@from-sip-external:1] NoOp("SIP/91.208.12.133-00000017", "Hangup") in new stack
« Last Edit: January 22, 2015, 10:59:49 am by brononius »
Version: linuxMCE 1404, running virtual on ESXi

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thor

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Re: No incoming phone calls?
« Reply #4 on: March 02, 2015, 07:46:14 pm »
I've looked at your post twice now and realize that maybe you would benefit from the following (which I used sucessfully on my system):

Solution (http://forum.linuxmce.org/index.php?topic=12861.msg92498#msg92498) and this is quoting from that post:

"Apparently the number detection is not yet working...so asterisk thinks it is an anonymous call. In /etc/extensions.conf the second line under globals is set to no

 ALLOW_SIP_ANON = no

this kicks any phone call into internal voice message that sounds exactly like a phone provider message informing the dialer that the number is not in service. If you change this to

ALLOW_SIP_ANON = yes

and the do a core restart gracefully inside of asterisk, external phone calls will now be allowed to go in to the system

Of course you are still not "home" yet as the system then tells you that pluto default voice mail does not exist and then just hangs not knowing what to do..."

brononius

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Re: No incoming phone calls?
« Reply #5 on: March 09, 2015, 07:31:25 am »
I've changed the ALLOW_SIP_ANON to 'yes'.
And for the moment, i've got the feeling that all my calls are entering. Just hitting each time the voicemail of asterisk.

The strange thing is that in linuxmce, the phones are not registred.
But in asterisk, they are...
Version: linuxMCE 1404, running virtual on ESXi

Orbiters: ASUS eeePAD, Nexus 5, Huwai, web
Automation: EIB technology, KNX IP ROUTER 750
Phones: Cisco 7912-7940-7960
Camera's: Foscam POE