----- Original Message -----
From: Anthony
Minessale
To:
freeswitch-users@lists.freeswitch.orgSent:
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@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
_______________________________________________
Freeswitch-users
mailing list
Freeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
--
Anthony
Minessale II
FreeSWITCH
http://www.freeswitch.org/ClueCon
http://www.cluecon.com/AIM: anthm
MSN:anthony_minessale@hotmail.comGTALK/JABBER/
PAYPAL:anthony.minessale@gmail.com
IRC:
irc.freenode.net
#freeswitch
FreeSWITCH Developer Conference
sip:888@conference.freeswitch.orgiax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.orgpstn:213-799-1400
_______________________________________________
Freeswitch-users
mailing list
Freeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
_______________________________________________
Freeswitch-users
mailing list
Freeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org