INCOMING ROUTES !!!

Discussion in 'General' started by emenny81, Nov 11, 2009.

  1. emenny81

    Joined:
    Nov 5, 2009
    Messages:
    24
    Likes Received:
    0
    I have two inbound routes from different did numbers, lets say did 1 and did 2.

    did 1 is routed to an IVR, this route works perfect.

    did 2 is routed to a direct extension, this doesn't work.

    When I call did 1 and dial the extension it works, but when I try to point any did to a direct extension I get the following error msg, I never had this issue but I recently upgrade to 1.6, clean install.

    Please Help this is driving me crazy.

    Thanks



    Verbosity is at least 4
    -- Executing [did2@from-sip-external:1] NoOp("SIP/MeOut-08444b50", "Received incoming SIP connection from unknown peer to did2") in new stack
    -- Executing [did2@from-sip-external:2] Set("SIP/MeOut-08444b50", "DID=did2") in new stack
    -- Executing [did2@from-sip-external:3] Goto("SIP/MeOut-08444b50", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/MeOut-08444b50", "1?from-trunk|did2|1") in new stack
    -- Goto (from-trunk,did2,1)
    -- Executing [did2@from-trunk:1] Set("SIP/MeOut-08444b50", "__FROM_DID=did2") in new stack
    -- Executing [did2@from-trunk:2] Gosub("SIP/MeOut-08444b50", "app-blacklist-check|s|1") in new stack
    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/MeOut-08444b50", "") in new stack
    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/MeOut-08444b50", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:3] Return("SIP/MeOut-08444b50", "") in new stack
    -- Executing [did2@from-trunk:3] ExecIf("SIP/MeOut-08444b50", "0 |Set|CALLERID(name)=19493268122") in new stack
    -- Executing [did2@from-trunk:4] Set("SIP/MeOut-08444b50", "FAX_RX=disabled") in new stack
    -- Executing [did2@from-trunk:5] Set("SIP/MeOut-08444b50", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [did2@from-trunk:6] SetCallerPres("SIP/MeOut-08444b50", "allowed_not_screened") in new stack
    -- Executing [did2@from-trunk:7] Goto("SIP/MeOut-08444b50", "from-did-direct|446|1") in new stack
    -- Goto (from-did-direct,446,1)
    -- Executing [446@from-did-direct:1] Macro("SIP/MeOut-08444b50", "exten-vm|novm|446") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/MeOut-08444b50", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/MeOut-08444b50", "AMPUSER=19493268122") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/MeOut-08444b50", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/MeOut-08444b50", "1|Set|REALCALLERIDNUM=19493268122") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/MeOut-08444b50", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/MeOut-08444b50", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/MeOut-08444b50", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/MeOut-08444b50", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/MeOut-08444b50", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/MeOut-08444b50", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/MeOut-08444b50", "Using CallerID "HI" <1949xxxxxxxx>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/MeOut-08444b50", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/MeOut-08444b50", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/MeOut-08444b50", "EXTTOCALL=446") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/MeOut-08444b50", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/MeOut-08444b50", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/MeOut-08444b50", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/MeOut-08444b50", "record-enable|446|IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/MeOut-08444b50", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/MeOut-08444b50", "recordingcheck|20091110-174958|1257904198.25") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20091110-174958|1257904198.25: Inbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/MeOut-08444b50", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/MeOut-08444b50", "dial||tr|446") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/MeOut-08444b50", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/MeOut-08444b50", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
    dialparties.agi: Starting New Dialparties.agi
    == Parsing '/etc/asterisk/manager.conf': Found
    == Parsing '/etc/asterisk/manager_additional.conf': Found
    == Parsing '/etc/asterisk/manager_custom.conf': Found
    == Manager 'admin' logged on from 127.0.0.1
    dialparties.agi: Caller ID name is 'HI' number is '1949xxxxxxxx'
    dialparties.agi: Methodology of ring is 'none'
    -- dialparties.agi: Added extension 446 to extension map
    > dialparties.agi: Extension 446 has call screening off
    -- dialparties.agi: Extension 446 cf is disabled
    -- dialparties.agi: Extension 446 do not disturb is disabled
    > dialparties.agi: extnum 446 has: cw: 1; hascfb: 0 [] hascfu: 0 []
    dialparties.agi: ExtensionState: 0
    -- dialparties.agi: dbset CALLTRACE/446 to 1949xxxxxxx
    -- dialparties.agi: Filtered ARG3: 446
    == Manager 'admin' logged off from 127.0.0.1
    -- AGI Script dialparties.agi completed, returning 0
    -- Executing [s@macro-dial:7] Dial("SIP/MeOut-08444b50", "SIP/446||tr") in new stack
    -- Couldn't call 446
    == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [s@macro-dial:8] Set("SIP/MeOut-08444b50", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-dial:9] GosubIf("SIP/MeOut-08444b50", "0?CHANUNAVAIL|1") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/MeOut-08444b50", "0?exit|return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/MeOut-08444b50", "SV_DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/MeOut-08444b50", "0?docfu|1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/MeOut-08444b50", "0?docfb|1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/MeOut-08444b50", "DIALSTATUS=CHANUNAVAIL") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/MeOut-08444b50", "Voicemail is novm") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/MeOut-08444b50", "1?s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] NoOp("SIP/MeOut-08444b50", "IVR_RETVM: IVR_CONTEXT: ") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] GotoIf("SIP/MeOut-08444b50", "0?exit|1") in new stack
    -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] PlayTones("SIP/MeOut-08444b50", "congestion") in new stack
    == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited non-zero on 'SIP/MeOut-08444b50' in macro 'exten-vm'
    == Spawn extension (from-did-direct, 446, 1) exited non-zero on 'SIP/MeOut-08444b50'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    elastix*CLI>
     
  2. jgutierrez

    Joined:
    Feb 28, 2008
    Messages:
    5,737
    Likes Received:
    0
    Is extension 446 registered when you call into your DID?
    asterisk -rx "sip show peers"
     
  3. emenny81

    Joined:
    Nov 5, 2009
    Messages:
    24
    Likes Received:
    0
    yeah its registered, its pretty weird.

    elastix*CLI> sip show peers
    sip show peers
    Name/username Host Dyn Nat ACL Port Status
    MeOut/515429950 38.99.70.232 5060 Unmonitored
    MDOut/124625523 38.99.70.232 5060 Unmonitored
    515429950 38.99.70.232 5060 Unmonitored
    446/446 10.1.1.227 D N A 23230 OK (102 ms)
    124625523 38.99.70.232 5060 Unmonitored
    5 sip peers [Monitored: 1 online, 0 offline Unmonitored: 4 online, 0 offline]
    elastix*CLI>
     
  4. jgutierrez

    Joined:
    Feb 28, 2008
    Messages:
    5,737
    Likes Received:
    0
    As what I can see from the logs, it does directs the call int othe extension. For some reason, the extension is not receiving the incoming call. Try setting the inbound route into a voicemail, announce, or into another extension (ip phone and then into a softphone).

    It seems to me that it is something related to the ip device that you are using for extension 446.
     
  5. emenny81

    Joined:
    Nov 5, 2009
    Messages:
    24
    Likes Received:
    0
    Im using eyebeam softphone and I've tried pointing it to another account I've setup a virtual fax which is iax and I still get the same error.

    I will try zoiper tomorrow to see what I get.


    thanks for your help.
     
  6. Patrick_elx

    Joined:
    Dec 14, 2008
    Messages:
    1,120
    Likes Received:
    0
    go to unembedded freepbx, check that all upgrades are ok.
    Then in go to each extension, route, trunk, ringroup etc... and just save the setting for each and eventually apply/reload.

    I had this problem before with something that looked like an unsynchronized state between the freepbx database and the .conf files.
     
  7. donhwyo

    Joined:
    Aug 8, 2008
    Messages:
    293
    Likes Received:
    0
    446/446 10.1.1.227 D N A 23230 OK (102 ms)

    Is that ip on your local network? 102ms looks like it is not so maybe it is a routing or forwarding issue.

    Don
     
  8. emenny81

    Joined:
    Nov 5, 2009
    Messages:
    24
    Likes Received:
    0
    that was just a test, I used my softphone to try it, but now Im using a fax extension and I still cant route the calls to the fax extension. Here are the logs.

    [root@elastixfax etc]# asterisk -rvvvv
    Asterisk 1.4.26.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for detail s.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    == Parsing '/etc/asterisk/asterisk.conf': Found
    == Parsing '/etc/asterisk/extconfig.conf': Found
    Connected to Asterisk 1.4.26.1 currently running on elastixfax (pid = 2770)
    Verbosity was 3 and is now 4
    -- Executing [19497433360@from-sip-external:1] NoOp("SIP/MDOut-b6c02358", "Received incoming SIP connection from unknown peer to DID1") in new stack
    -- Executing [19497433360@from-sip-external:2] Set("SIP/MDOut-b6c02358", "DID=DID1") in new stack
    -- Executing [19497433360@from-sip-external:3] Goto("SIP/MDOut-b6c02358", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/MDOut-b6c02358", "0?from-trunk|DID1|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/MDOut-b6c02358", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-12-02 19:27:49 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/MDOut-b6c02358", "") in newstack
    -- Executing [s@from-sip-external:4] Wait("SIP/MDOut-b6c02358", "2") in newstack
    -- Executing [s@from-sip-external:5] Playback("SIP/MDOut-b6c02358", "ss-noservice") in new stack
    -- <SIP/MDOut-b6c02358> Playing 'ss-noservice' (language 'en')
    == Spawn extension (from-sip-external, s, 5) exited non-zero on 'SIP/MDOut-b6c02358'
    -- Executing [h@from-sip-external:1] NoOp("SIP/MDOut-b6c02358", "Hangup") in new stack
    -- Executing [h@from-sip-external:2] Set("SIP/MDOut-b6c02358", "DID=s") in new stack
    -- Executing [h@from-sip-external:3] Goto("SIP/MDOut-b6c02358", "s|1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/MDOut-b6c02358", "0?from-trunk|s|1") in new stack
    -- Executing [s@from-sip-external:2] Set("SIP/MDOut-b6c02358", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2009-12-02 19:27:55 UTC.
    -- Executing [s@from-sip-external:3] Answer("SIP/MDOut-b6c02358", "") in new stack
    == Spawn extension (from-sip-external, s, 3) exited non-zero on 'SIP/MDOut-b6c02358'
    elastixfax*CLI>
     
  9. emenny81

    Joined:
    Nov 5, 2009
    Messages:
    24
    Likes Received:
    0
    I found the problem, it was a codec issue, I removed g729 from the trunk settings.
     

Share This Page