callback problem (not working)

magicyes

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#1
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
 

rejil.rajan

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#2
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
 

magicyes

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#3
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!!
 

rejil.rajan

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#4
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.
 

magicyes

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#5
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
 

rejil.rajan

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#6
Can you try putting the callback number as an extension and see if it is working
 

magicyes

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#7
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
 

rejil.rajan

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#8
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
 

magicyes

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#9
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!
 

magicyes

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rejil.rajan

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magicyes

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#12
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!
 

rejil.rajan

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#13
Can you try adding the lines in the equivalent of zapata-auto.conf

dtmfmode=inband
relaxdtmf=yes

Should improve the dtmf response
 

magicyes

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#14
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
 

rejil.rajan

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#15
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
 

magicyes

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#16
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
 

dicko

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#17
It's an ongoing asterisk problem,

try

dtmfmode=auto

or upgrade to asterisk >= 1.4.25
 

magicyes

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#18
hi thanks for the input

I tired with dtmfmode=auto but nothing...

will try the upgrade
thank you
 

magicyes

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#19
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!
 

magicyes

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#20
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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