[Freeswitch-users] Hang up not received

Ognjen Seslija oseslija at gmail.com
Tue Jan 20 13:43:06 PST 2009


Ok, as discussed with Tony on IRC channel I followed his directions which
lead to a successfull outcome (like it always does I might add :).

One has to use tone_detect app in FreeSWITCH dialplan in order to check for
busy tones coming from the PSTN side and if matched fire a hangup
application. This is the snippet of my test dp that does the trick (from
extension Local_extensions in default.xml):

<anti-action application="tone_detect" data="busy 425,0 r +100 hangup 16
4"/>
<anti-action application="bridge" data="
user/${dialed_extension}@${domain_name}"/<user/$%7Bdialed_extension%7D@$%7Bdomain_name%7D%22/>
>
This means that FS will listen to freq of 425 Hz and wait for 4 positive
detection to fire up hangup app with code 16 which is NORMAL_CLEARING (425
Hz is the freq telco here uses; for other countries I suggest getting the
ITU world tones pdf file and check there):

2009-01-20 22:32:46 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
TONE busy HIT 1/4
2009-01-20 22:32:47 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
TONE busy HIT 2/4
2009-01-20 22:32:48 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
TONE busy HIT 3/4
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1262 tone_detect_callback()
TONE busy HIT 4/4
2009-01-20 22:32:50 [DEBUG] switch_ivr_async.c:1268 tone_detect_callback()
TONE busy DETECTED
2009-01-20 22:32:50 [NOTICE] mod_dptools.c:584 hangup_function() Hangup
OpenZAP/1:1/1 [CS_EXECUTE] [NORMAL_CLEARING]

Regards,
Ognjen

On Tue, Jan 20, 2009 at 8:45 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> I tried similar setup with my analog card (X100P) and I'm having same
> issue. Call is not hungup on the oz side once the caller ends. My telco
> doesn't use polarity reversal for singalling hang up so I'm guess I'm stuck
> to detecting busy tone from the telco side. I'll try to modify tones.conf
> accordingly.
>
> Regards,
> Ognjen
> (sekil)
>   On Tue, Jan 20, 2009 at 6:05 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> This is a common issue with analog phones even traditional answering
>> machines suffer from it.
>> I'm sure you must have had an answering machine at some point that has
>> dial tone as the message it receives.
>>
>> Unless FreeSWITCH has some hint that the call has hungup it will not stop
>> trying to complete the call.
>>
>> If the other side is sending a busy tone to indicate hangup it's possible
>> to use the tone_detect app to pick
>> up on the tones and abort the call.
>>
>> Another thing you could do if you have unlimited inbound is explicitly
>> answer the call in the dialplan before
>> you call your sip phones this will give you a more profound hangup
>> detection but it will make every call count
>> even when nobody answers.
>>
>>
>>
>>   On Tue, Jan 20, 2009 at 10:46 AM, Tomás <tomasborrella at gmail.com>wrote:
>>
>>>  Hi all,
>>>
>>> I'm configuring my home PBX using FreeSwitch. I'm using a X101P card
>>> configured as FXO (conected to analog PSTN line) and I have several IP
>>> phones and softphones conected to FreeSwitch.
>>>
>>> I can call from an IP phone to other IP phone (the same with the
>>> softphones) and also from an IP phone (or softphone) to an external number
>>> thought PSTN.
>>>
>>> When I call from an external analog phone to FreeSwitch, I bridge the
>>> call to all internal IP phones and softphones and they ring, but the problem
>>> is that when I hang up the call in the external phone, all internal phones
>>> (IP phones and softphones) keeps ringing...
>>>
>>> I'm pretty sure the problem is that FreeSwitch don't receive the hang up,
>>> because I cann't see anything on the log.
>>>
>>> I've also created my own tones.conf for my country (Spain) but I'm not
>>> sure if it's ok (but I have the same problem with hang up)
>>>
>>> I've googled the list, and I've found several people with a similar
>>> problem but no solution...
>>>
>>> That's my pastebin with the most importants printouts and config files:
>>> http://pastebin.freeswitch.org/6822
>>>
>>> Thank you very much in advance.
>>>
>>> _______________________________________________
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>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
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>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
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>
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