OpenVox D110P

Discussion in 'General' started by vloose, Mar 25, 2009.

  1. vloose

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    I can't seem to get my OpenVox D110P to work with my T1 circuit. Everywhere I read makes it sound like it should pretty much just work after a hardware detect, but I must be missing something.

    I have tried elastix 1.3.2, 1.4.4 and 1.5rc4 and get the same results from all. I get a busy signal when in and out bound tells me "all circuits are busy."

    Here is a bit from my log if it helps:


    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: Set
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: ExecIf
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Last app: Set|DIAL_TRUNK_OPTIONS=M(setmusic^acc_1)M(setmusic^acc_1)wW
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: Macro
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: GotoIf
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: GotoIf
    [Mar 25 02:01:36] NOTICE[3048] app_dial.c: Hey! chan SIP/537-08b94700's context='macro-dialout-trunk', and exten='s'
    [Mar 25 02:01:36] WARNING[3048] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
    [Mar 25 02:01:36] DEBUG[3048] app_macro.c: Executed application: Dial

    And this is what I see when trying to make an outbound call while having asterisk -r running:

    -- Executing [5555555555@from-internal:1] Set("SIP/537-08b94700", "MOHCLASS=acc_1") in new stack
    -- Executing [5555555555@from-internal:2] Macro("SIP/537-08b94700", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/537-08b94700", "AMPUSER=537") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/537-08b94700", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/537-08b94700", "1|Set|REALCALLERIDNUM=537") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/537-08b94700", "AMPUSER=537") in new stack
    -- Executing [s@macro-user-callerid:5] [1 ;36;40mSet("SIP/537-08b94700", "AMPUSERCIDNAME=Vaughn Loose") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/537-08b94700", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/537-08b94700", "AMPUSERCID=537") in new stack
    -- Executing [s@macro-user-callerid:8] Set("[ 1;35;40mSIP/537-08b94700", "CALLERID(all)="Vaughn Loose" <537>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/537-08b94700", "REALCALLERIDNUM=537") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/537-08b94700", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf[0;3 7;40m("SIP/537-08b94700", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/537-08b94700", "Using CallerID "Vaughn Loose" <537>") in new stack
    -- Executing [5555555555@from-internal:3] Set("SIP/537-08b94700", "_NODEST=") in new stack
    -- Executing [5555555555@from-internal:4] Macro("SIP/537-08b94700", "record-enable|537|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/537-08b94700", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/537-08b94700", "recordingcheck|20090325-020136|1237960896.2") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20090325-020136|1237960896.2: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/537-08b94700", "") in new stack
    -- Executing [5555555555@from-internal:5] Macro("SIP/537-08b94700", "dialout-trunk|2|5555555555||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/537-08b94700", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/537-08b94700", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/537-08b94700", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/537-08b94700", "DIAL_NUMBER=5555555555") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/537-08b94700", "DIAL_TRUNK_OPTIONS=trwW") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/537-08b94700", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/537-08b94700", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/537-08b94700", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/537-08b94700", "DIAL_TRUNK_OPTIONS=wW") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/537-08b94700", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/537-08b94700", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/537-08b94700", "0|Set|REALCALLERIDNUM=537") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/537-08b94700", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/537-08b94700", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/537-08b94700", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/537-08b94700", "TRUNKOUTCID=8149774108") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/537-08b94700", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/537-08b94700", "1|Set|CALLERID(all)=8149774108") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/537-08b94700", "1?exit") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/537-08b94700", "") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/537-08b94700", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/537-08b94700[0 ;37;40m", "OUTNUM=5555555555") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/537-08b94700", "custom=DAHDI/g0") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/537-08b94700", "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^acc_1)wW") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/537-08b94700", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/537-08b94700", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/537-08b94700", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/537-08b94700[0;37; 40m", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/537-08b94700", "DAHDI/g0/5555555555|300|M(setmusic^acc_1)wW") in new stack
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/537-08b94700", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/537-08b94700", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/537-08b94700", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
    -- Executing [5555555555@from-internal:6] Macro("SIP/537-08b94700", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/537-08b94700", "all-circuits-busy-now|noanswer") in new stack
    -- <SIP/537-08b94700> Playing 'all-circuits-busy-now' (language 'en')
    == Spawn extension (macro-outisbusy, s, 1) exited non-zero on 'SIP/537-08b94700' in macro 'outisbusy'
    == Spawn extension (from-internal, 5555555555, 6) exited non-zero on 'SIP/537-08b94700'
    -- Executing [h@from-internal:1] Macro("SIP/537-08b94700", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/537-08b94700", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/537-08b94700", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/537-08b94700", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/537-08b94700", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/537-08b94700", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/537-08b94700", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/537-08b94700' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/537-08b94700'
    elastix*CLI>

    If anything jumps out at anyone, I would really appreciate the help.

    Thanks,

    Vaughn
     
  2. vloose

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    BTW,

    This sample log was while using Elastix 1.5rc4. The logs looked pretty much the same using the earlier versions of Elastix with Zaptel rather than DAHDI.
     
  3. dicko

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    I suggest you make sure the T1 is up and running first with
    at a low level

    zttool (dahdi_tool) at the bash prompt, it should show the link is up with no errors.
    when that is working:

    from asterisk CLI
    dahdi show status
    dahdi show channels
     
  4. vloose

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    Thanks for the response. I'll take a look at that tonight. These are very useful tools I wasn't finding other places other than the dahdi show channels.

    I did see a yellow alarm clear in /var/log/messages when I plugged the T1 in so I assumed it was working but this looks more along the lines of what I need to look at.

    If there is anything else I should look at when I try this again tonight I'm all ears.
     
  5. dicko

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    For a start:
    That your T1 signaling in /etc/zaptel.conf matches the telco's signaling exactly (PRI/em_w or whatever), that g0 is populated with channels in /etc/asterisk/zapata.conf. When it is all working /syncing/active then "tune" the T1 for gain. (worry about that later) and
    dahdi show channel 1 (for example)
    to make sure that channel 1 is responsive, then
    originate zap/1/<anumber> extension <an extension>
    from asterisk CLI (with fingers crossed)

    (if a PRI then pri debug and pri intense from asterisk CLI to confuse the f*&k out of yourself :p )
     
  6. vloose

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    You don't understand how much I appreciate your help.

    I am obviously missing a piece of the puzzle. Something you said in the last post jumped out at me "that g0 is populated with channels in /etc/asterisk/zapata.conf"

    I don't understand that statement, so that might be the piece I am missing.

    These are the only lines in my zapta-channels.conf file:

    group=0,11
    context=from-pstn
    switchtype = national
    signalling = pri_cpe
    channel => 1-23
    group=
    context=default

    And here is my zapta.conf

    [trunkgroups]

    [channels]
    context=from-pstn
    signalling=fxs_ks
    rxwink=300 ; Atlas seems to use long (250ms) winks
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echocancelwhenbridged=no
    faxdetect=incoming
    echotraining=800
    rxgain=0.0
    txgain=0.0
    callgroup=1
    pickupgroup=1

    ;Uncomment these lines if you have problems with the disconection of your analog lines
    ;busydetect=yes
    ;busycount=3


    immediate=no

    #include zapata_additional.conf
    #include zapata-channels.conf
    ~
     
  7. dicko

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    First you need to get dahdi/zaptel config right






    If indeed you have a PRI T1, then /etc/zaptel.conf should include something like

    span=1,1,0,esf,b8zs
    bchan=1-23
    dchan=24


    but the dchan (signalling channel must reflect exactly what you get from the telco) (are you in North America?, if so it is almost certainly 24)

    start with posting /etc/zaptel.conf
     
  8. vloose

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    Yes, North America.

    Just switched the t1 over and results are:

    elastix*CLI> dahdi show channels
    Chan Extension Context Language MOH Interpret
    pseudo default default

    elastix*CLI> dahdi show status
    Description Alarms IRQ bpviol CRC4
    Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0

    This is what /etc/dahdi/system.conf looks like:

    # Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF RED
    span=1,1,0,esf,b8zs
    # termtype: te
    bchan=1-23
    dchan=24
    echocanceller=oslec,1-23

    # Global data

    loadzone = us
    defaultzone = us
     
  9. dicko

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    What version of Elastix are you currently running?

    you might need to investigate all the zaptel files (as they used to be called) replacing dahdi instead of zaptel

    I am as yet waiting for 1.5 to download , and all bets are off from me unless we are back on 1.3 (never did a real PRI on 1.5 yet).
    !!

    Please hold until I have a platform to investigate on . . .

    (from all your postings there should have been 23 dahdi channels)
     
  10. vloose

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    I was working with ver. 1.5rc4. That's why I was subbing the dahdi config files for zaptel, but I'm sure I am adding more confusion for both of us by doing that. I'm going to drop back to 1.3.2 this morning so I am on common ground.

    Thanks,

    Vaughn
     
  11. rafael

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  12. vloose

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    Wait a minute!!!

    I just read a post about DID's and saw under FreePBX a button for "Zap channel DID's." Do I need to set this up? Could it be this simple????

    I reinstalled 1.3 and now show dahdi channels looks a little more normal. (T1 not currently plugged in)

    phones*CLI> dahdi show channels
    Chan Extension Context Language MOH Interpret
    pseudo default default
    1 from-pstn default
    2 from-pstn default

    ..etc down to channel 23
     
  13. vloose

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    Well, still no luck. In 1.3.2

    phones*CLI> dahdi show channel 1
    Channel: 1>
    File Descriptor: 10
    Span: 1
    Extension:
    Dialing: no
    Context: from-pstn
    Caller ID:
    Calling TON: 0
    Caller ID name:
    Destroy: 0
    InAlarm: 1
    Signalling Type: ISDN PRI
    Radio: 0
    Owner: <None>
    Real: <None>
    Callwait: <None>
    Threeway: <None>
    Confno: -1
    Propagated Conference: -1
    Real in conference: 0
    DSP: no
    Relax DTMF: no
    Dialing/CallwaitCAS: 0/0
    Default law: ulaw
    Fax Handled: no
    Pulse phone: no
    Echo Cancellation: 128 taps unless TDM bridged, currently OFF
    PRI Flags:
    PRI Logical Span: Implicit
    Hookstate (FXS only): Onhook

    dahdi show channels:
    Chan Extension Context Language MOH Interpret
    pseudo default default
    1 from-pstn default
    2 from-pstn default
    etc...


    zttools:

    Alarms Span
     
  14. dicko

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    But much better (23?) channels and showing reasonable status

    now dahdi_tool from bash

    and watch the behavior as you plug the T1 in and out


    (should change from red alarm to OK. after sometimes quite a few seconds as the framing and low level protocols sync up)
     
  15. vloose

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    Yep, zttool I see the alarm go yellow then green.

    Still no calls in or out.

    I noticed "InAlarm: 1" when I do a "dahdi show channel X"

    I'm trying to research what are common causes of that right now.
     
  16. dicko

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    then from asterisk CLI:
    originate zap/1/13235551212 extension (the one on your desk)
    and watch both the CLI to wwatch for errors, and also zttool (select the span)

    this will call 13235551212 on zap channel 1 (the rx/tx flags on the channel will be diagnostic) after ringing your phone
     
  17. vloose

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    That returns:

    No such command 'zap/1/13235551212 extension 537' (type 'help zap/1/13235551212 extension' for other possible commands)

    for me.
     
  18. dicko

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    I don't believe I have confirmation from you that you do indeed have an isdn PRI rate T1, yet, ( and not a good old fashioned 24 channel AT&T T1) can you so confirm?
     
  19. vloose

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    Something else I find odd.

    When I change signalling=pri_cpe to signalling=em_w, in /etc/asterisk/zapata-channels.conf I lose my zap and dahdi tools in asterisk CLI.

    Does this mean anything?
     
  20. dicko

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    you missed the "originate", type that at the asterisk CLI for syntax help
    from prompt:

    pbx*>originate (enter)
     

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