[Freeswitch-users] Hang up not received

Wed Jan 21 01:15:56 PST 2009

Scott, I imagined that it could be an OpenZap problem, but I didn't find an
OpenZap mailing list, so I sent the email to FS list. Do you know where can
I find more information about OpenZap hardware support and developement
status (I have special interest in Loop Start)??

Anthony and Ognjen, I've tried tone detection and thanks to that FS is
detecting hung up, but I faced the problem that tone detector answer the
call...

That's my dialplan:

 <extension name="extension_name">
      <condition field="destination_number" expression="^919999999$">
        <action application="tone_detect" data="busy 425,0 r +100 hangup 16
4"/>
        <action application="bridge"
data="sofia/internal/1003%${server-domain-name},
sofia/internal/1004%${server-domain-name}"/>
      </condition>
    </extension>

When I receive a call from PSTN, tone detection answer the call (the caller
hears only one first tone and then hears "nothing" until I pick up the call
on softphone).

So, I think that tone detection solution does not resolve my problem... Is
there any other possibility to detect hang up without answering the call
(using Loop Start signaling) or have we to wait until OpenZap is completely
developed?

Thanks in advance.

On Tue, Jan 20, 2009 at 10:43 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> 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/
>>> ClueCon http://www.cluecon.com/
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> 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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