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<DIV><FONT face=Arial size=2>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.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>See you on irc soon.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks,</FONT></DIV>
<DIV><FONT face=Arial size=2>Col</FONT></DIV>
<BLOCKQUOTE
style="PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="BACKGROUND: #e4e4e4; FONT: 10pt arial; font-color: black"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com">Anthony Minessale</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, July 22, 2008 12:10
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] OpenZap
and tones.conf - FXO hangup problem</DIV>
<DIV><BR></DIV>
<DIV dir=ltr>did you play around with the fxo types like kewlstart
etc.<BR><BR>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.<BR><BR><BR>
<DIV class=gmail_quote>On Mon, Jul 21, 2008 at 9:10 AM, Col Ferguson <<A
href="mailto:asterisk@coltect.no-ip.com">asterisk@coltect.no-ip.com</A>>
wrote:<BR>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid">I'm
using zaptel-1.4.9.2.xpp.r5566 (I had this downloaded already for
an<BR>asterisk install). I tried 1.4.11 svn trunk initially, but changed
to<BR><A href="http://1.4.9.2" target=_blank>1.4.9.2</A> in case it was
something to do with the svn trunk code. I have<BR>another box with asterisk
<A href="http://1.4.18.1" target=_blank>1.4.18.1</A> and zaptel
1.4.9.2.xpp.r5566 working<BR>fine.<BR><BR>I am using freeswitch svn trunk,
as I read somewhere in the user list<BR>archives that there were openzap
fixes done recently.<BR><BR>Any other info that may help
?<BR><BR>Thanks,<BR><FONT color=#888888>Col<BR></FONT>
<DIV>
<DIV></DIV>
<DIV class=Wj3C7c><BR><BR>----- Original Message -----<BR>From: Anthony
Minessale<BR>To: <A
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A><BR>Sent:
Monday, July 21, 2008 11:31 PM<BR>Subject: Re: [Freeswitch-users] OpenZap
and tones.conf - FXO hangup problem<BR><BR><BR>What driver are you using
underneath for the FXO.<BR><BR><BR><BR><BR>On Mon, Jul 21, 2008 at 7:44 AM,
Col Ferguson <<A
href="mailto:asterisk@coltect.no-ip.com">asterisk@coltect.no-ip.com</A>><BR>wrote:<BR><BR>Hello
all,<BR>I have installed freeswitch and had a bit of a play over the last
few days<BR>and have a question about the format of the tones.conf file for
OpenZap.<BR><BR>I have a Xorcom Astribank to play with at the moment and
have it working<BR>mostly. I haven't done anything with the SIP part at all
yet.<BR><BR>My basic hurdle at the moment is detecting a hangup on an FXO
port before<BR>any bridged FXS extensions answer.<BR>If I hangup the line I
am ringing into, the FXS extension keeps on ringing,<BR>then when the FXS
extension is answered it is bridged to the FXO port and I<BR>get a dialtone
and access to the FXO PSTN line directly.<BR><BR>I read a post that alluded
to the possibility that having the wrong info in<BR>tones.conf may result in
strange behaviour as openzap doesn't recognise<BR>tones correctly. I could
be completely wrong, and often am, but thats always<BR>been a good way to
learn stuff.<BR><BR><BR>I haven't been able to find any good info on the
format of tones.conf, but<BR>have managed to work out a few things so
far.<BR><BR>I have this is tones.conf for au, and its loading properly, but
I don't know<BR>what its doing exactly.<BR><BR>[au]<BR>generate-dial =>
v=-7;%(1000,0,413,438)<BR>detect-dial => 413,438<BR>generate-ring =>
v=-7;%(400,200,413,438);%(400,2000,413,438)<BR>detect-ring =>
413,438<BR>generate-busy => v=-7;%(375,375,425)<BR>detect-busy =>
425<BR>generate-attn => v=0;%(100,100,1400,2060,2450,2600)<BR>detect-attn
=> 1400,2060,2450,2600<BR>generate-callwaiting-sas =>
v=0;%(300,0,440)<BR>detect-callwaiting-sas =>
440<BR>generate-callwaiting-cas =>
v=0;%(80,0,2750,2130)<BR>detect-callwaiting-cas =>
2750,2130<BR>detect-fail1 => 913.8<BR>detect-fail2 =>
1370.6<BR>detect-fail3 => 776.7<BR><BR>Using the generate-ring line as an
example<BR><BR>generate-ring =>
v=-7;%(400,200,413,438);%(400,2000,413,438)<BR><BR>I think
that<BR>generate-ring is for generating the ring tone used internally by
ring_ready<BR>(and probably other areas I haven't found/noticed yet)<BR>; is
a separator<BR>v=-7 probably sets a volume level ?<BR>%(400,200,413,438) 400
is the time for the tone to be on, 200 is the time<BR>for the tone to be
off, 413 is the first tone played, 438 is the second tone<BR>played
?<BR>%(400,2000,413,438) 400 is the time for the tone to be on, 2000 is the
time<BR>for the tone to be off, 413 is the first tone played, 438 is the
second tone<BR>played ?<BR><BR>(I found some code in switch.conf.xml that
sets up ring tones that showed<BR>using the two lots of settings, and this
sounds right for me in Australia. I<BR>got the freqs from a Sipura, asterisk
source and a web site<BR><A href="http://www.3amsystems.com"
target=_blank>www.3amsystems.com</A>)<BR><BR>detect-ring is used to look for
a specific tone ?<BR><BR>So what then is the information in generate-attn
=><BR>v=0;%(100,100,1400,2060,2450,2600) doing ?<BR>Also what are
detect-fail1,2,3 for ?<BR><BR>Is there anywhere to set a disconnect tone
?<BR>As far as I can tell, Australia uses the busy tone to indicate a
hangup,<BR>which sometimes comes after a period of silence.<BR><BR><BR>In
case I am completely off track my dialplan is below. All Zap channels
are<BR>corresponding numbers, ie channel 1 is number 1 etc. Channel 1-8 are
FXS,<BR>9-14 ate FXS but input/outputs, 15-22 are FXS and 23-30 are
FXO.<BR><BR>This is from
/usr/local/freeswitch/conf/dialplan/extensions/home.xml<BR><BR>Please point
out anything silly in here.<BR><BR><extension
name="out-zap-channel-7"><BR> <condition
field="destination_number" expression="^(7)$"><BR> <action
application="ring_ready"/><BR> <action application="bridge"
data="openzap/7/1"/><BR> </condition><BR></extension><BR><BR><extension
name="out-zap-channel-8"><BR> <condition
field="destination_number" expression="^(8)$"><BR> <action
application="ring_ready"/><BR> <action application="bridge"
data="openzap/8/1"/><BR> </condition><BR></extension><BR><BR><BR><extension
name="in-zap-channel-27"><BR> <condition
field="destination_number" expression="^(27)$"><BR>
<!--<action application="set"
data="hangup_after_bridge=true"/>--><BR><!--Couldn't see a
difference--><BR> <!--<action application="set"
data="effective_caller_id_name=6055<BR>Line"/>-->
<!--fiddling--><BR> <!--<action
application="set"<BR>data="effective_caller_id_number=6055"/>--><BR><!--fiddling--><BR>
<!--<action application="tone_detect" data="busy 425 r +5
hangup<BR>normal_clearing"/>--> <!--Really thought this
might work--><BR> <!--<action
application="answer"/>--><BR><!--Tried a simple ivr and symptoms
same. ivr bridges call--><BR> <!--<action application="sleep"
data="2000"/>--><BR><!--hangup incoming call before answering with
FXS--><BR> <!--<action application="ivr"
data="coltect_ivr"/>--><BR><!--and FXS still rings--><BR>
<action application="bridge" data="openzap/8/1"/><BR>
<!--<action application="transfer"
data="8"/><BR> </condition><BR></extension><BR><BR>Thanks,<BR>Col<BR><BR><BR>_______________________________________________<BR>Freeswitch-users
mailing list<BR><A
href="mailto:Freeswitch-users@lists.freeswitch.org">Freeswitch-users@lists.freeswitch.org</A><BR><A
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href="http://www.freeswitch.org"
target=_blank>http://www.freeswitch.org</A><BR><BR><BR><BR><BR>--<BR>Anthony
Minessale II<BR><BR>FreeSWITCH <A href="http://www.freeswitch.org/"
target=_blank>http://www.freeswitch.org/</A><BR>ClueCon <A
href="http://www.cluecon.com/"
target=_blank>http://www.cluecon.com/</A><BR><BR>AIM: anthm<BR><A
href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</A><BR>GTALK/JABBER/<A
href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</A><BR>IRC:
<A href="http://irc.freenode.net" target=_blank>irc.freenode.net</A>
#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</A><BR><A
href="http://iax:guest@conference.freeswitch.org/888"
target=_blank>iax:guest@conference.freeswitch.org/888</A><BR><A
href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</A><BR>pstn:213-799-1400<BR><BR><BR><BR>_______________________________________________<BR>Freeswitch-users
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clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/">http://www.freeswitch.org/</A><BR>ClueCon <A
href="http://www.cluecon.com/">http://www.cluecon.com/</A><BR><BR>AIM:
anthm<BR><A
href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</A><BR>GTALK/JABBER/<A
href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</A><BR>IRC:
<A href="http://irc.freenode.net">irc.freenode.net</A>
#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</A><BR><A
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</DIV>
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