[Freeswitch-users] Hang up not received

Michael Collins msc at freeswitch.org
Wed Jan 21 06:54:20 PST 2009 

On Wed, Jan 21, 2009 at 1:49 AM, Ognjen Seslija <oseslija at gmail.com> wrote:
> When call comes in from Openzap, tone_detect app does pre_answer of a call
> cause it's need media to start detecting tones in the first place. This
> behaviour is something that I see on calls inside my telco when calling from
> analogue lines. I don't think this is a big of deal because ringback
> provided by FS will make caller understand that the call is still in
> progress. One can make its own ringback to sound exactly the same as
> telco's.
>
> I don't think that we'll ever make POTS behave like digital protocols do.

So true!
-MC
>
> Regards,
> Ognjen
>
> On Wed, Jan 21, 2009 at 10:36 AM, Scott Ellis <scott.ellis at novatex.com.au>
> wrote:
>>
>> I had a similar problem, you can use
>> <action application="set" data="ringback=${au-ring}"/>  (I added an "au"
>> ring definition to my vars.xml file)
>>
>> To get what you want.
>>
>> I also had a problem that you get two rings, then an answer then to the
>> system generated ring tone, which was confusing some of our (not to bright)
>> callers.
>>
>> As we don't use callerID I turned that flag off in the openzap.conf.xml
>> file - I thought that this would do what I wanted (answer the instant the
>> call is detected), but the change in the config file does not make it all
>> the way down to the point where it takes action. At this point I hacked the
>> code to get what I wanted. I have to create a JIRA entry with the details
>> yet.
>>
>> As far as I understand, this is the right place for OpenZap, as it is a
>> product of the FS project.
>>
>> Scott
>>
>> Tomás wrote:
>>
>> Scott, I imagined that it could be an OpenZap problem, but I didn't find
>> an OpenZap mailing list, so I sent the email to FS list. Do you know where
>> can I find more information about OpenZap hardware support and developement
>> status (I have special interest in Loop Start)??
>>
>> Anthony and Ognjen, I've tried tone detection and thanks to that FS is
>> detecting hung up, but I faced the problem that tone detector answer the
>> call...
>>
>> That's my dialplan:
>>
>>  <extension name="extension_name">
>>       <condition field="destination_number" expression="^919999999$">
>>         <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}"/>
>>> 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.
>>>>>>
>>>>>> _______________________________________________
>>>>>> Freeswitch-users mailing list
>>>>>> Freeswitch-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
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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
>>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>> IRC: irc.freenode.net #freeswitch
>>>>>
>>>>> FreeSWITCH Developer Conference
>>>>> sip:888 at conference.freeswitch.org
>>>>> iax:guest at conference.freeswitch.org/888
>>>>> googletalk:conf+888 at conference.freeswitch.org
>>>>> pstn:213-799-1400
>>>>>
>>>>> _______________________________________________
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>>>>>
>>>>
>>>
>>>
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>>
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