<html><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Also, make sure your using the trunk of freeswitch.<div><br></div><div>Mike</div><div><br><div><div>On Jul 21, 2008, at 10:10 AM, Anthony Minessale wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite"><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="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"> 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 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