is it the Zaptel 1.4 or the x100p??

Discussion in 'General' started by AU Troll, Jan 22, 2008.

  1. AU Troll

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    I have been running Asterisk for a couple of yesrs and have used the old x100p cards with little problem.

    I have noticed that since the upgrade of asterisk to v 1.4.15+ version and Zaptel ver 1.4.6 I am having trouble getting the cards to behave.

    With Asterisk 1.2 and zap v1.2 the x100p handled the lines exactly as it should. With the new stuff, the x100p will NOT detect distant end hang up (i.e., I call my PSTN line from my mobile, asterisk answers, but when i hang up the mobile in the middle of a call, asterisk continues to hold the line open).

    I have googled my arse off looking for an answer, and can't find one that fixes it.

    my zapata.conf is:

    [trunkgroups]

    [channels]
    language=en
    echocancel=yes
    echocancelwhenbridged=no
    faxdetect=both

    ; Including file containing the suggested configuration
    ; generated by the hardware detector tool
    ; Remove this line if you don't want this feature
    #include zapata-channels.conf

    and my zapata-channels.conf is:

    ; Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit
    ; Zaptel Channels Configurations (zapata.conf)
    ;
    ; This is not intended to be a complete zapata.conf. Rather, it is intended
    ; to be #include-d by /etc/zapata.conf that will include the global settings
    ;

    ; Span 1: WCFXO/0 "Wildcard X100P Board 1"
    ;;; line="1 WCFXO/0/0"
    signalling=fxs_ks


    callerid=asreceived
    group=0
    context=from-pstn
    channel => 1
    busydetect=no
    busycount=6
    callprogress=yes
    rxwink=300
    rxgain=7.0
    txgain=1.0
    immediate=yes
    relaxdtmf=yes
    hanguponpolarityswitch=yes

    You will note that I have tried both busydetect and hanguponpolarityswitch with no change in behavior.

    Can ANYONE shed light on this? I really need the newest asterisk for the fax capability and Elastix works perfect for it except for this one small and annoying snag...thanks..

    Mark
     
  2. Usuarioforum

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    Where are you from? If you are from a country that use polarity reversal for hangup line, you have a problem with this card. It don't detect the polarity.

    With an openvox (a400) or digium card, you didn't have problem.

    Cheers.
     
  3. dalybrian

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    Well fortunately i don't have that issue, the issue i have is with incoming caller id ( all call comes in as private or unknown ) and setting up a incoming call route to ring two extension based on calls coming in on the Zap channel. Does any know if the new Elastix will correct this issue.

    PS. I a big fan of Elastix especially after Trixbox whole dial home thing & hearing them wanting to put AD's in their Trixbox CE and i love the fax feature on Elastix.
     
  4. AU Troll

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    I am from Australia.

    In doing some further work with Elastix 0.9.2, another strange thing is happening. I have two sip accounts with the same provider. I can only register one sip line. If I try to register both of them, the registrations time out and I can not get on line at all. I can register one line at a time, and it does not matter which one. With both, the registrations fail, and asterisk fails to see my sip extensions as well...This is not happening with my old system which is an EasyVoxBox v.11

    Concerning the hangup issue, until i can get the above working, then the hangup is just a nagging thing...this one is serious...

    Any ideas?

    Mark
     
  5. Usuarioforum

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    The line is analog or digital? Do you have callerid=asreceived in zaptel?

    Analog lines don't know about incoming did. You must to route call from zaptel 1,2... no grom did.
     
  6. AU Troll

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    the zap is handling an analog line...(believe that is what telstra has here)....

    Anyays..know about the zaptel trunk settings...and yes I do have the callerid+asreceived in the config file...

    I might be something else, but until i can get my sip lines to register properly, i am not concerned with the disconnection issue...

    Thanks,

    Mark
     
  7. dingoland

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    Hi,

    For your 2 SIP lines not registering at the same time, did you asked your ISP for the SIP rules ?
    Maybe you cannot register 2 lines from the same IP/MAC address.
    I have the same config than you, 2 SIP lines from the same provider and i need to add in the sip.conf in the general section the line:
    defaultexpirey=1800

    else, i cannot register my SIP accounts because of permanent timeout.

    Regards<br><br>Post edited by: dingoland, at: 2008/01/27 06:54
     
  8. AU Troll

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    I am thinking it is related more to either Asterisk 1.4.17 or the FreePBX v2.3.1 than anything else. With the EasyVoxBox using Asterisk 1.4.11 and FreePBX 2.3.0, I have no issues registering both lines.

    I will give the default expiry a try and see what happens.

    Thanks.

    Mark
     
  9. AU Troll

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    Update: Added the defaultexpirey to the sip.conf file. No change. Can log in only one SIP account at a time, can not log in both my accounts. When I try both accounts, get time outs and my sip extensions become "unreachable"....

    Mark
     
  10. dingoland

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    Hi,

    It seems to be linked to the release :(
    Do you tried with another release of the Elastix ? not the last but the 0.9.1 for example to be sure ?

    Regards
     
  11. AU Troll

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    Nope...but don't have time to play with it...so will just hang out till the release 1 gets here and try that...but in the mean time, i can set my customer up with the 0.9.2 as he will only have the one account i have to log into, and everything else (specially the fax), works properly.....

    So play the wait and see game here....

    Mark
     
  12. AU Troll

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    Ok...just did some further researcing and questioning and found out that asterisk v 1.4.17 anad v 1.4.14 have a lot of bugs that are supposed to be fixed with v 1.4.18..hope Elastix includes this in the new release..

    mark
     

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