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.net channel #freeswitch
We're here every day so whenever you feel like dropping by.... On Mon, Jul 21, 2008 at 9:26 AM, Col Ferguson <[EMAIL PROTECTED]> 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 <[EMAIL PROTECTED]> > *To:* [email protected] > *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 <[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 > > ------------------------------ > > _______________________________________________ > 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
_______________________________________________ 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
