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Can't Dial Out with A400P Card

Discussion in 'Trunks' started by JRayfield, Oct 13, 2008.

  1. JRayfield New Member

    And it does:

    pbx*CLI> zap show channels
    Chan Extension Context Language MOH Interpret
    pseudo from-internal en default
    1 from-zaptel en default
    2 from-internal en default
    The 'zap show channels' command is deprecated and will be removed in a future release. Please use 'dahdi show channels' instead.
  2. JRayfield New Member

    I have 1 Trunk set up, Zap Identifier g0, with no Dial rules, Maximum Channels set to 1.

    I have 1 Outbound Route set up, with Dial Pattern set to . and Trunk Sequence set to Zap/g0.
  3. jroper Guru

    That's good news.

    In FreePBX, create a zap trunk, put 1 in the trunk identifier.

    Now gow into outbound routes, there will alread be a default route in there with a 9|. make sure the trunk selected is Zap/1

    Then try an outbound call, prefixing it with 9

    Joe
  4. JRayfield New Member

    I still get the "All circuits are busy now" message.

    John
  5. JRayfield New Member

    As I mentioned, the same Trunks and Outbound Routes setup works fine in Trixbox but doesn't work with piaf. I appears that the FXO card isn't be 'accessed' in piaf.

    John
  6. JRayfield New Member

    I went to the Asterisk cli prompt and tried making the outgoing call (actually, I'm connected to the extension port of a legacy PBX, so I dialed '13', another extension on the legacy PBX).

    For some reason, Asterisk 'thinks' that the FXO port/line is 'busy/congested', even though it's not. Here's what I saw:

    ============================
    root@pbx:~ $ rasterisk
    Asterisk 1.4.22, 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 details.
    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.
    =========================================================================
    Connected to Asterisk 1.4.22 currently running on pbx (pid = 2666)
    Verbosity is at least 3
    -- Executing [913@from-internal:1] Macro("SIP/658072-087edf30", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/658072-087edf30", "AMPUSER=658072") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/658072-087edf30", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/658072-087edf30", "1|Set|REALCALLERIDNUM=658072") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/658072-087edf30", "AMPUSER=658072") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/658072-087edf30", "AMPUSERCIDNAME=658072") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/658072-087edf30", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/658072-087edf30", "AMPUSERCID=658072") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/658072-087edf30", "CALLERID(all)="658072" <658072>") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/658072-087edf30", "REALCALLERIDNUM=658072") in new stack
    -- Executing [s@macro-user-callerid:10] ExecIf("SIP/658072-087edf30", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/658072-087edf30", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,20)
    -- Executing [s@macro-user-callerid:20] NoOp("SIP/658072-087edf30", "Using CallerID "658072" <658072>") in new stack
    -- Executing [913@from-internal:2] Set("SIP/658072-087edf30", "_NODEST=") in new stack
    -- Executing [913@from-internal:3] Macro("SIP/658072-087edf30", "record-enable|658072|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/658072-087edf30", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/658072-087edf30", "recordingcheck|20081014-184612|1224027972.26") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20081014-184612|1224027972.26: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/658072-087edf30", "") in new stack
    -- Executing [913@from-internal:4] Macro("SIP/658072-087edf30", "dialout-trunk|2|13||") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/658072-087edf30", "DIAL_TRUNK=2") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/658072-087edf30", "0?sub-pincheck|s|1") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/658072-087edf30", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/658072-087edf30", "DIAL_NUMBER=13") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/658072-087edf30", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/658072-087edf30", "OUTBOUND_GROUP=OUT_2") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/658072-087edf30", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/658072-087edf30", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/658072-087edf30", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/658072-087edf30", "outbound-callerid|2") in new stack
    -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/658072-087edf30", "0|SetCallerPres|") in new stack
    -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/658072-087edf30", "0|Set|REALCALLERIDNUM=658072") in new stack
    -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/658072-087edf30", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,6)
    -- Executing [s@macro-outbound-callerid:6] Set("SIP/658072-087edf30", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:7] Set("SIP/658072-087edf30", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:8] Set("SIP/658072-087edf30", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/658072-087edf30", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,12)
    -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/658072-087edf30", "0|Set|CALLERID(all)=") in new stack
    -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/658072-087edf30", "1?exit") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/658072-087edf30", "") in new stack
    -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/658072-087edf30", "0|AGI|fixlocalprefix") in new stack
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/658072-087edf30", "OUTNUM=13") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/658072-087edf30", "custom=ZAP/1") in new stack
    -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/658072-087edf30", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
    -- Executing [s@macro-dialout-trunk:16] Macro("SIP/658072-087edf30", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/658072-087edf30", "") in new stack
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/658072-087edf30", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/658072-087edf30", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/658072-087edf30", "ZAP/1/13|300|") in new stack
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/658072-087edf30", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/658072-087edf30", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/658072-087edf30", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks") in new stack
    -- Executing [913@from-internal:5] Macro("SIP/658072-087edf30", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/658072-087edf30", "all-circuits-busy-now|noanswer") in new stack
    -- <SIP/658072-087edf30> Playing 'all-circuits-busy-now' (language 'en')
    -- Executing [s@macro-outisbusy:2] Playback("SIP/658072-087edf30", "pls-try-call-later|noanswer") in new stack
    -- <SIP/658072-087edf30> Playing 'pls-try-call-later' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/658072-087edf30", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/658072-087edf30", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/658072-087edf30", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/658072-087edf30", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/658072-087edf30", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/658072-087edf30", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/658072-087edf30", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/658072-087edf30' in macro 'hangupcall'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/658072-087edf30' in macro 'outisbusy'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/658072-087edf30'
    ============================
  7. jroper Guru

    Running out of ideas here. Anyone else with any ideas?

    You've got the channel displayed in asterisk, everything would appear to be fine, I'm assuming lspci and cat /proc/interrupts/ and cat /proc/zaptel/* shows the card in place.

    The A400 is a clone of the Digium TDM400, and I've used a few of them, and they have been as good as gold, with no issues. Like you, I wonder whether an update to zaptel has "broken" the clone card.

    I suspect we are back to the beginning again, trying an earlier version of Zaptel and Asterisk.

    Joe
  8. JRayfield New Member

    Everything looks good. I tend to agree - I think I need to try going back to Zaptel 1.4.11. That version did work in Trixbox. Maybe that will fix this.

    ============================
    lspci shows it:
    ============================
    00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
    ============================

    ============================
    cat /proc/interrupts/ shows this:
    ============================
    root@pbx:~ $ cat /proc/interrupts
    CPU0
    0: 7212858 IO-APIC-edge timer
    1: 9 IO-APIC-edge i8042
    6: 3 IO-APIC-edge floppy
    7: 2 IO-APIC-edge parport0
    8: 1 IO-APIC-edge rtc
    9: 0 IO-APIC-level acpi
    12: 104 IO-APIC-edge i8042
    15: 64423 IO-APIC-edge ide1
    169: 17723 IO-APIC-level sata_via
    177: 0 IO-APIC-level ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb4, uhci_hcd:usb5
    185: 7194231 IO-APIC-level wctdm
    193: 48790 IO-APIC-level eth0
    201: 0 IO-APIC-level VIA8237
    NMI: 0
    LOC: 7213080
    ERR: 0
    MIS: 0
    ============================

    ============================
    cat /proc/zaptel/* shows this:
    ============================
    root@pbx:~ $ cat /proc/zaptel/*
    Span 1: WCTDM/0 "Wildcard TDM400P REV E/F Board 1" (MASTER)
    1 WCTDM/0/0 FXSKS (In use)
    2 WCTDM/0/1 FXOKS (In use)
    3 WCTDM/0/2
    4 WCTDM/0/3
    ============================
  9. JRayfield New Member

    Here's some of the output when I run dmesg. The Echo Canceller shown here is MG2. In Trixbox, it showed OSLEC. That wouldn't have a bearing on this problem, would it?

    John

    ============================
    Zapata Telephony Interface Registered on major 196
    Zaptel Version: 1.4.12.1
    Zaptel Echo Canceller: MG2
    ACPI: PCI Interrupt 0000:00:0a.0[A] -> GSI 19 (level, low) -> IRQ 185
    Freshmaker version: 71
    Freshmaker passed register test
    Module 0: Installed -- AUTO FXO (FCC mode)
    Module 1: Installed -- AUTO FXS/DPO
    Module 2: Not installed
    Module 3: Not installed
    Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules)
    ============================
  10. JRayfield New Member

    Is this normal?

    ==============================
    pbx*CLI> zap show status
    Description Alarms IRQ bpviol CRC4
    Wildcard TDM400P REV E/F Board 1 OK 0 0 0
    The 'zap show status' command is deprecated and will be removed in a future release. Please use 'dahdi show status' instead.
    ==============================
  11. JRayfield New Member

    Joe, I found someone else (over on the OpenVox forum) having a similar problem - no dial out on an A400P card. They're running CentOS 5.2, Kernal 2.6.18-92.1.10.el5 #1, and zaptel driver version 1.4.12.1 on Asterisk 1.4.18 - freePBX 2.4.0. This makes me suspect that Zaptel 1.4.12.1 may have a problem with the A400P. I think I need to go back to Zaptel 1.4.11. Can you help me do this (hopefull this will work this time <G>)?

    John
  12. jroper Guru

    As offered before, I'd be happy to SSH in and install an earlier zaptel version, although the instructions posted earlier should have worked just fine.

    As a reward a small donation to PiaF would be nice on sucessful completion.

    I have to say that all the vital signs look good, and this should just work. So we may suspect that that Zaptel has changed in some way.

    At this point, it would be interesting to know 2 things.
    1. Will the A400 work on an earlier version of Zaptel, (compiled from source), thus proving nothing amiss with the card.
    2. Do Digium TDM400P's still work properly with this version of Asterisk.


    Joe
  13. JRayfield New Member

    Let me try a couple of other things and I'll get back.

    A small donation is definitely in order. I also want to visit with you (and probably your partners) later about a commercial project that I would like to look into, using PBX in a Flash.

    John
  14. JRayfield New Member

    Joe, I removed zaptel again and tried re-installing version 1.4.11. I get these errors when I try to 'make':

    ============================
    In file included from /usr/src/zaptel/kernel/xpp/xpd.h:26,
    from /usr/src/zaptel/kernel/xpp/card_fxo.c:27:
    /usr/src/zaptel/kernel/xpp/xdefs.h:117: error: conflicting types for âboolâ
    include/linux/types.h:36: error: previous declaration of âboolâ was here
    make[4]: *** [/usr/src/zaptel/kernel/xpp/card_fxo.o] Error 1
    make[3]: *** [/usr/src/zaptel/kernel/xpp] Error 2
    make[2]: *** [_module_/usr/src/zaptel/kernel] Error 2
    make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.6.el5-i686'
    make[1]: *** [modules] Error 2
    make[1]: Leaving directory `/usr/src/zaptel'
    make: *** [all] Error 2
    ============================
  15. JRayfield New Member

    I figured out what was causing the compile error - it was the xpp interface. I chose not to compile that interface in 'menuselect' and zaptel 1.4.11 compiled just fine.

    By the time that I get done with this project, I'll know a lot more about Linux and Asterisk then I'd ever planned on learning (and I may have a whole lot less hair, too <G>).

    John
  16. JRayfield New Member

    So, I got Zaptel version 1.4.11 to compile properly by leaving out xpp. I recompiled Asterisk. Everything looks good. My A400P card is recognized by Asterisk (ztcfg -vvv shows the card correctly). But I still have the problem of not being able to dial out through the card.

    John
  17. jroper Guru

    Aah yes, I'd forgotten about the xdefs bug - here is the problem and the fix, commenting out the errant line, http://bugs.digium.com/view.php?id=12889

    Try recompiling with the fix in place described above, and that would explain why my earlier instructions would not have worked, and it did not downgrade as expected, because Zaptel never compiled properly.

    Joe
  18. JRayfield New Member

    Joe, I'm not familiar (yet) with some of the syntax used in the xdefs.h file. What do the '+' and '-' signs mean? Are these showing the lines that I should 'add' and 'take out' from the original file? Or something else?

    John
  19. JRayfield New Member

    Never mind, I figured it out. :)

    John
  20. JRayfield New Member

    Well, I got Zaptel 1.4.11 compiled and installed and re-compiled Asterisk. Everything comes back up fine, except that I still can't make outbound calls through the A400P card (FXO port). So the problem apparently isn't the Zaptel version.

    Based on what I'm seeing here at the CLI prompt, it looks like Asterisk 'thinks' that the line is 'busy'. Am I understanding this information correctly?

    Any more ideas?

    ============================
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/658072-09e76468", "ZAP/1/13|300|") in new stack
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/658072-09e76468", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    ============================

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