Problem after install Elastix 1.1

Discussion in 'General' started by danardf, Jun 23, 2008.

  1. danardf

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    I went to TB2.6 to Elastix 1.1 and ho surprise.!

    I have informed all extensions SIP (as before), and I can not call that the extensions Xlite ....
    All other extensions as linksys (SPA3102 PAP2), do not work. (occupied)

    I have not changed my setup

    Besides, I have the option FR in sip.conf, but the sounds are still EN!

    Result ... not Elastix not working properly.

    I'm disappointed, because before it was operating under TB26.<br><br>Post edited by: danardf, at: 2008/06/22 21:54
     
  2. danardf

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    hmmm.
    For information, this problem is for exemple:
    phone 102 (Xlite) calling 103 (Linksys)
    I have this message:
    The..nanana..is not aviable, please try again.

    On 103 extension there is not VM.
    If VM is enabled, my call is hangup.

    problem into dialplan? :unsure:

    Help me please. :(
     
  3. danardf

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    I reload the french voices into sounds root diretory (var/lib/asterisk/sounds) is ok.
     
  4. danardf

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    With sip debug peeer 103:

     
  5. Bob

    Bob

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    Danardf,

    How did you migrate??

    Was it a completely fresh install of Elastix 1.1 with everything put back by hand in the GUI??
    OR
    Did you restore the Freepbx backup from TB2.6 to the new Elastix 1.1 install?

    Regards

    Bob<br><br>Post edited by: Bob, at: 2008/06/23 02:00
     
  6. danardf

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    yes, it's a fresh install with everything put back by hand in the GUI.

    Can i update asterisk version 1.4.21.
    Recompiling option "make sample"?

    If you have some solution....

    Thanks
     
  7. lek

    lek Guest

    danardf,

    Are you sure you typed "sip debug..."?. I can't see the sip packages in your text.

    Additionally, please post the output of the command "show hints" on the CLI.
     
  8. danardf

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    i'm very sure!
    sip debug peer 103

    show hints :
     
  9. danardf

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    ha yes.... excuse me.
    it's a part of sip debug.

    the complete trace is:

    oupss. :unsure:
     
  10. danardf

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    i have to make an asterisk update to 1.4.21 + addons.
    Still the same problem.
    I had to make a backup Elastix and after update, i make a restore the last configuration.

    wath's append?

    I do not want to return to TB2.6
     
  11. CleveJ

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    Hi Danardf,

    I do understand how you feel, but I can assure you when you have it all up and running you will be so pleased and happy you will think of nothing else but Elastix, so please don't be upset we here will assist you one way or another.

    I am doing a fresh install of Elastix 1.1 Stable right now to test it as I go. I do not have any of the problems you have at all it works without any problems.

    So lets start!!

    Can you please answer the following.

    1. With your fresh install did you update FreePBX? if so what did you update it to.
    2. From a extension can you dial *60 and see if it will repeat the time.

    Cheers<br><br>Post edited by: CleveJ, at: 2008/06/23 18:10
     
  12. danardf

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    ok thanks for your help ;)
    The problem is that, i don't have the phone now because the Elastix is down.
    But..... i think this gonna be ok, with your help. :)

    i try to looking for by google and i think that the problem become chan_sip.so like this link:

    http://bugs.digium.com/view.php?id=9329

    For answer at your question, yes after installed Elastix, i had to make an >yum update. and make also freepbx.

    after execute the debug methode by digium, i have this part:

    I must to reinstall quickly elastix 1.1 and not updating system.
    And, the same time, i try this also.
     
  13. danardf

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    hmmmm whooooo:

    I have a way....

    I try to change some codecs by extension and now is ok.
    But...but....

    In my exemple (befor)
    (102) - codec = ilbc, gsm only
    (103) - codec = alaw only.

    the calls is ok. with old configuration.

    I must to put alaw into (102) for the call is up.

    funy no?

    Normaly, call 102 to 103.
    102 (ilbc)---> asterisk translate ilbc to alaw and ----> 103 (Alaw)
    So 2 codecs alive. NO?
     
  14. CleveJ

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    Good to see you have it fixed, you must remember that the SPA device does not support the ilbc or gsm codec, xlite has the ilbc codec, so the best way to do this is to have all you LAN extensions using g711 u or a as they are all local you should be ok.
     
  15. danardf

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    i'm ok with you, but befor (with the last configuration on TB26), i had not this problem.

    Asterisk make a twin compression. (ILBC<--->ALAW).

    I want to find this

    For exemple, if i have some interanet extension on the LAN ans one extension on the WAN with a little band switch. I must to use only one codec gsm on ilbc.

    I must to do make a call from (102 / gsm) to (SPA3102 / Alaw), .... no? otherise it's unthinkable.
     
  16. CleveJ

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    Hi Danardf,

    I have found your problem, Elastix 1.1 Stable does not have the ilbc codec so this is your problem. What you have to do is install the codec and reboot your Elastix server and it should work. To make sure at Cli do "core show translation" you will see tha ilbc has all dashes (-).

    Please post back soon as I have to go to bed soon.

    Cheers
     
  17. danardf

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    i replace all ilbc codec by gsm and speex.
    All is ok now.... :)

    pfffiuuuuu.

    One more try, outing by SPA. (Trunk SPA3102)

    Ok thanks very very much.:woohoo:

    I didn't think that ilbc was not installed. :(

    info. the result core:

     
  18. danardf

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    and now.....
    I can make outgoing call by my SPA3102, but i can't take incoming call. :angry:

    with trace debug, like this:

    in my trunk freebox, i had insecure=very, i change by invite, but no result. :unsure:

    i don't understand that's append :unsure:
    there is a "SIP/2.0 404 Not Found" into the trace SIP.!!!
     
  19. Bob

    Bob

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    Just in case anyone else is reading this thread.

    It is Asterisk that has removed the inclusion of the ILBC Codec as of a particular version 1.4.19. More info at:

    http://www.asterisk.org/node/48469

    Naturally if you were running it with Trixbox, then they may have an earlier version of Asterisk e.g. pre 1.4.19, or you may have updated that kept it included from 1.4.18 to 1.4.19...who knows...

    But unless you have a pressing need for ILBC, then best to ignore it entirely....

    Regards
    Bob
     
  20. danardf

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    yes... now it's ok.

    The problem was to SPA configuration.
    into PSTN line, dialplan:
    befor - (S0<:pstn@193.107.20.38>)
    after - (S0<:193.107.20.38>)

    Allow Anonymous Inbound SIP Calls = yes

    There are a little difference between TB26 and Elastix.!!!

    Thanks for your help
     

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