callback problem (not working)

Discussion in 'General' started by magicyes, Jul 7, 2009.

  1. magicyes

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    I've been using elastix for nearly two months
    and I was able to successfully setup everything (IVR, inbound and outbound routes, several trunks etc..)

    but I am stuck with callback
    I set up everything as it should be done as far as I can see, and I call and get the busy tone but I am not able to get the callback, I tried several times but nothing ever happens

    I attach the screenshot of the callback I setup

    and follow is the asterisk log I get and I am not really able what's going wrong
    any help would be super appreciated! :)

    Code:
    Jul 7 18:57:56  	VERBOSE  	[28192] logger.c:  	
    
    -- edited
    
    
    [​IMG]
     
  2. rejil.rajan

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    Hi

    Does 0 take you to your PSTN Trunk. Please ensure you have to prefix the proper Dial prefix before the number if the right trunk has to be chosen
     
  3. magicyes

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    Hello Rejil
    Thank you for the answer

    I have no PSTN trunk... I am running elastix on a hosted VPS with voip only trunks

    is that the problem? a routing problem?

    how can I get the system to call me back according to the outbound routes setup? is this possible?

    Thank you very much!!
     
  4. rejil.rajan

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    Just put the right prefix before the number. How do you make calls to this number if it is from a phone connected to Elastix. Place the same numbers in the callback number field. It should solve your problem.
     
  5. magicyes

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    But this is exactly how the callback is setup now.
    then this is not the problem

    I put the number to call back in the exact same format that I use to dial it from ring groups, phones, follow me, and it works everywhere, except when using callback

    any other ideas / help would be super appreciated
    Thank you again
     
  6. rejil.rajan

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    Can you try putting the callback number as an extension and see if it is working
     
  7. magicyes

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    same behavior
    I get the proper busy tone, it seems that the system is executing the callback, but no call is ever received

    here is the log

    edited
     
  8. rejil.rajan

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    Login into the backend /var/www/html/admin/modules/callback. Take a copy of functions.inc.php, then edit functions.inc.php

    Go to line 46 and modify it to

    $ext->add('callback', $item['callback_id'], '', new ext_system((empty($asterisk_conf['astvarlib']) ? '/var/lib/asterisk' : $asterisk_conf['astvarlib']).'/bin/callback ${CALL} ${DESTINATION} ${SLEEP} >> /tmp/callbacklog &'));

    Recreate the callback number deleting the existing one
     
  9. magicyes

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    hi

    line 46 starts as follows:

    Code:
     if (is_array($dest) && empty($dest)) {
                    return $destlist;
            }
    
    
    shall I just remove that?
    this is quite unclear to me, forgive me but I am fairly new to command prompts and code

    I attached the file you suggested to edit
    if you could edit or show me in a clearer way, it would very much appreciated
    otherwise thank you anyway for your efforts in helping me!
     
  10. magicyes

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  11. rejil.rajan

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  12. magicyes

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    hi Rejil

    thank you sooooo much for your help on this issue so far
    I now got the callback to successfully call me back, works great

    unofrtunately I got a problem with the DISA, it's weird, every number I dial is repeated twice
    so if I dial 12345
    I see from the log 1122334455
    anyway I am gonna get some paid support now to get this fixed! :)
    Thank you very much for your help again!
     
  13. rejil.rajan

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    Can you try adding the lines in the equivalent of zapata-auto.conf

    dtmfmode=inband
    relaxdtmf=yes

    Should improve the dtmf response
     
  14. magicyes

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    hey

    I had support to look into it, they also came to the conclusion that the dtmf was the problem and they suggested I use dtmfmode=rfc2833 on the trunk

    I did that and it didn't work
    then I searched some forums and found that Betamax providers (I use actionvoip and smslisto) need dtmfmode=inband
    I tiried setting various things in the trunk outbound settings but still no luck

    this is how it is now:
    host=sip.smslisto.com
    dtmf=inband
    dtmfmode=inband
    disallow=all
    allow=ulaw&alaw&g729

    I looked for the file you mentioned "zapata-auto.conf" but I don't have such file in my installation
    the only file I find related to zapata is "zapata.conf.template"

    is that where I am supposed to put the 2 lines you suggested?
    or perhaps in the sip.conf?

    thank you again
     
  15. rejil.rajan

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    Ok, so your using a SIP trunk for termination. zapata (now called dahdi) is for PSTN trunks.

    Try removing the line dtmf=inband. Add also the line relaxdtmf=yes and check it out
     
  16. magicyes

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    just tried as you said, here is how my outbound info on the trunk looks like:

    type=peer
    host=sip.actionvoip.com
    relaxdtmf=yes
    dtmfmode=inband
    disallow=all
    allow=ulaw&alaw&g729
    canreinvite=no
    fromdomain=stun.actionvoip.com

    but I am having exactly the same as before, every DTMF is received twice
     
  17. dicko

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    It's an ongoing asterisk problem,

    try

    dtmfmode=auto

    or upgrade to asterisk >= 1.4.25
     
  18. magicyes

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    hi thanks for the input

    I tired with dtmfmode=auto but nothing...

    will try the upgrade
    thank you
     
  19. magicyes

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    one thing:
    if I update asterisk will it break my elastix / freepbx configuration?
    (i.e. will I loose my routes, trunks, extensions etc.. and more importantly the codecs I installed?)

    Thank you!
     
  20. magicyes

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    hey just wanted to update the thread as I seem to have fixed the issue!!

    after several tests this seems to work like a charm

    relaxdtmf=yes
    dtmfmode=rfc2833

    thanks to the info I found here: http://www.voip-info.org/wiki/view/Asterisk+DTMF

    "Duplicate DTMF

    * Change the setting relaxdtmf= and see if that helps
    * If you appear to be receiving doubled DTMF signals then you are likely to get both inband and RFC2833 or SIP INFO signalling on your Asterisk box; you will want to make the sening party use only one of these two methods.
    * Another possible cause would be that your telephony card has hardware-based DMTF detection turned on, coupled with the fact that Asterisk also detects the DTMF signal(s). "

    hope this helps if others are having similar issues!!
    thank you rejil and dicko for your help! :)
     

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