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

Anthony Minessale anthony.minessale at gmail.com
Mon Jul 21 08:20:03 PDT 2008


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.netchannel #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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>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> 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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>> http://www.freeswitch.org
>>
>>
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>> http://www.freeswitch.org
>>
>>
>
>
> --
> 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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