[Freeswitch-users] OpenZap and tones.conf - FXO hangup problem

Anthony Minessale anthony.minessale at gmail.com
Mon Jul 21 11:21:13 EDT 2008 

but be sure to send it to just me or the whole world will know it ;)


On Mon, Jul 21, 2008 at 10:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> we lost you on irc, can you tell us the ip and port
>
>
> On Mon, Jul 21, 2008 at 9:27 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> labbed up just means let us ssh to the box while you create the problem =D
>>
>> to get on irc go to our homepage and look for "join us on irc" on the
>> right hand side of the page, make up a nick name and press the "join irc"
>> button.  Otherwise just get your own client and point it at
>> irc.freedode.net channel #freeswitch
>>
>> We're here every day so whenever you feel like dropping by....
>>
>>
>>
>> On Mon, Jul 21, 2008 at 9:26 AM, Col Ferguson <asterisk at coltect.no-ip.com>
>> wrote:
>>
>>>  I'd love to, but I was up till 3 last night playing with it and wife is
>>> getting mad. I haven't tried anything apart from kewlstart as that works in
>>> asterisk. Also I have never used irc, and have no idea how to "lab it up".
>>> I'll have a go for a little while then will have to go to bed.
>>>
>>> See you on irc soon.
>>>
>>> Thanks,
>>> Col
>>>
>>> ----- Original Message -----
>>>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>  *Sent:* Tuesday, July 22, 2008 12:10 AM
>>> *Subject:* Re: [Freeswitch-users] OpenZap and tones.conf - FXO hangup
>>> problem
>>>
>>> did you play around with the fxo types like kewlstart etc.
>>>
>>> maybe if you would like to join irc and lab it up for us, we can have a
>>> look at it on your box in real time.
>>>
>>>
>>> On Mon, Jul 21, 2008 at 9:10 AM, Col Ferguson <
>>> asterisk at coltect.no-ip.com> wrote:
>>>
>>>> I'm using zaptel-1.4.9.2.xpp.r5566 (I had this downloaded already for an
>>>> asterisk install). I tried 1.4.11 svn trunk initially, but changed to
>>>> 1.4.9.2 in case it was something to do with the svn trunk code. I have
>>>> another box with asterisk 1.4.18.1 and zaptel 1.4.9.2.xpp.r5566 working
>>>> fine.
>>>>
>>>> I am using freeswitch svn trunk, as I read somewhere in the user list
>>>> archives that there were openzap fixes done recently.
>>>>
>>>> Any other info that may help ?
>>>>
>>>> Thanks,
>>>> Col
>>>>
>>>>
>>>> ----- Original Message -----
>>>> From: Anthony Minessale
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Sent: Monday, July 21, 2008 11:31 PM
>>>> Subject: Re: [Freeswitch-users] OpenZap and tones.conf - FXO hangup
>>>> problem
>>>>
>>>>
>>>> What driver are you using underneath for the FXO.
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Jul 21, 2008 at 7:44 AM, Col Ferguson <
>>>> asterisk at coltect.no-ip.com>
>>>> wrote:
>>>>
>>>> Hello all,
>>>> I have installed freeswitch and had a bit of a play over the last few
>>>> days
>>>> and have a question about the format of the tones.conf file for OpenZap.
>>>>
>>>> I have a Xorcom Astribank to play with at the moment and have it working
>>>> mostly. I haven't done anything with the SIP part at all yet.
>>>>
>>>> My basic hurdle at the moment is detecting a hangup on an FXO port
>>>> before
>>>> any bridged FXS extensions answer.
>>>> If I hangup the line I am ringing into, the FXS extension keeps on
>>>> ringing,
>>>> then when the FXS extension is answered it is bridged to the FXO port
>>>> and I
>>>> get a dialtone and access to the FXO PSTN line directly.
>>>>
>>>> I read a post that alluded to the possibility that having the wrong info
>>>> in
>>>> tones.conf may result in strange behaviour as openzap doesn't recognise
>>>> tones correctly. I could be completely wrong, and often am, but thats
>>>> always
>>>> been a good way to learn stuff.
>>>>
>>>>
>>>> I haven't been able to find any good info on the format of tones.conf,
>>>> but
>>>> have managed to work out a few things so far.
>>>>
>>>> I have this is tones.conf for au, and its loading properly, but I don't
>>>> know
>>>> what its doing exactly.
>>>>
>>>> [au]
>>>> generate-dial => v=-7;%(1000,0,413,438)
>>>> detect-dial => 413,438
>>>> generate-ring => v=-7;%(400,200,413,438);%(400,2000,413,438)
>>>> detect-ring => 413,438
>>>> generate-busy => v=-7;%(375,375,425)
>>>> detect-busy => 425
>>>> generate-attn => v=0;%(100,100,1400,2060,2450,2600)
>>>> detect-attn => 1400,2060,2450,2600
>>>> generate-callwaiting-sas => v=0;%(300,0,440)
>>>> detect-callwaiting-sas => 440
>>>> generate-callwaiting-cas => v=0;%(80,0,2750,2130)
>>>> detect-callwaiting-cas => 2750,2130
>>>> detect-fail1 => 913.8
>>>> detect-fail2 => 1370.6
>>>> detect-fail3 => 776.7
>>>>
>>>> Using the generate-ring line as an example
>>>>
>>>> generate-ring => v=-7;%(400,200,413,438);%(400,2000,413,438)
>>>>
>>>> I think that
>>>> generate-ring is for generating the ring tone used internally by
>>>> ring_ready
>>>> (and probably other areas I haven't found/noticed yet)
>>>> ; is a separator
>>>> v=-7 probably sets a volume level ?
>>>> %(400,200,413,438) 400 is the time for the tone to be on, 200 is the
>>>> time
>>>> for the tone to be off, 413 is the first tone played, 438 is the second
>>>> tone
>>>> played ?
>>>> %(400,2000,413,438) 400 is the time for the tone to be on, 2000 is the
>>>> time
>>>> for the tone to be off, 413 is the first tone played, 438 is the second
>>>> tone
>>>> played ?
>>>>
>>>> (I found some code in switch.conf.xml that sets up ring tones that
>>>> showed
>>>> using the two lots of settings, and this sounds right for me in
>>>> Australia. I
>>>> got the freqs from a Sipura, asterisk source and a web site
>>>> www.3amsystems.com)
>>>>
>>>> detect-ring is used to look for a specific tone ?
>>>>
>>>> So what then is the information in generate-attn =>
>>>> v=0;%(100,100,1400,2060,2450,2600) doing ?
>>>> Also what are detect-fail1,2,3 for ?
>>>>
>>>> Is there anywhere to set a disconnect tone ?
>>>> As far as I can tell, Australia uses the busy tone to indicate a hangup,
>>>> which sometimes comes after a period of silence.
>>>>
>>>>
>>>> In case I am completely off track my dialplan is below. All Zap channels
>>>> are
>>>> corresponding numbers, ie channel 1 is number 1 etc. Channel 1-8 are
>>>> FXS,
>>>> 9-14 ate FXS but input/outputs, 15-22 are FXS and 23-30 are FXO.
>>>>
>>>> This is from /usr/local/freeswitch/conf/dialplan/extensions/home.xml
>>>>
>>>> Please point out anything silly in here.
>>>>
>>>> <extension name="out-zap-channel-7">
>>>>  <condition field="destination_number" expression="^(7)$">
>>>>   <action application="ring_ready"/>
>>>>   <action application="bridge" data="openzap/7/1"/>
>>>>  </condition>
>>>> </extension>
>>>>
>>>> <extension name="out-zap-channel-8">
>>>>  <condition field="destination_number" expression="^(8)$">
>>>>   <action application="ring_ready"/>
>>>>   <action application="bridge" data="openzap/8/1"/>
>>>>  </condition>
>>>> </extension>
>>>>
>>>>
>>>> <extension name="in-zap-channel-27">
>>>>  <condition field="destination_number" expression="^(27)$">
>>>>   <!--<action application="set" data="hangup_after_bridge=true"/>-->
>>>> <!--Couldn't see a difference-->
>>>>   <!--<action application="set" data="effective_caller_id_name=6055
>>>> Line"/>-->                    <!--fiddling-->
>>>>   <!--<action application="set"
>>>> data="effective_caller_id_number=6055"/>-->
>>>> <!--fiddling-->
>>>>   <!--<action application="tone_detect" data="busy 425 r +5 hangup
>>>> normal_clearing"/>-->   <!--Really thought this might work-->
>>>>   <!--<action application="answer"/>-->
>>>> <!--Tried a simple ivr and symptoms same. ivr bridges call-->
>>>>   <!--<action application="sleep" data="2000"/>-->
>>>> <!--hangup incoming call before answering with FXS-->
>>>>   <!--<action application="ivr" data="coltect_ivr"/>-->
>>>> <!--and FXS still rings-->
>>>>   <action application="bridge" data="openzap/8/1"/>
>>>>   <!--<action application="transfer" data="8"/>
>>>>  </condition>
>>>> </extension>
>>>>
>>>> Thanks,
>>>> Col
>>>>
>>>>
>>>> _______________________________________________
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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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>>>>
>>>
>>>
>>>
>>> --
>>> 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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>>>
>>>
>>
>>
>> --
>> 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
>>
>
>
>
> --
> 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
>



-- 
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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