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 <[EMAIL PROTECTED]> 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: [email protected] > 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 <[EMAIL PROTECTED]> > 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 > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> > iax:[EMAIL PROTECTED]/888 > googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> > pstn:213-799-1400 > > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] <[EMAIL PROTECTED]> GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]<[EMAIL PROTECTED]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] <[EMAIL PROTECTED]> iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED]<[EMAIL PROTECTED]> pstn:213-799-1400
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