I am trying to make install the latest freeswitch SVN trunk on my OSX system, but having difficulty with the mod-pocketsphinx. I got the following error: *making install mod_pocketsphinx usage: mv [-f | -i | -n] [-v] source target mv [-f | -i | -n] [-v] source ... directory make[6]: *** [/opt/local/freeswitch/grammar/model/wsj1] Error 64 make[5]: *** [install] Error 1 make[4]: *** [mod_pocketsphinx-install] Error 1 make[3]: *** [install-recursive] Error 1 * Could anybody shed some light on this what's wrong with this? I checked my target directory *opt/local/freeswitch/grammar/model/wsj1, nothing is there. Thanks, Chris *
Let me test on OS X and i'll fix that. I added the 16k support to it lastnight i must have flubbed something up. /b On Jul 18, 2008, at 7:53 AM, unknown wrote:
Could anybody shed some light on this what's wrong with this? I checked my target directory opt/local/freeswitch/grammar/model/wsj1, nothing is there.
Brian West sip:brian@freeswitch.org
I have corrected the typo. Sorry about that ;) /b On Jul 18, 2008, at 7:53 AM, unknown wrote:
I am trying to make install the latest freeswitch SVN trunk on my OSX system, but having difficulty with the mod-pocketsphinx. I got the following error:
making install mod_pocketsphinx usage: mv [-f | -i | -n] [-v] source target mv [-f | -i | -n] [-v] source ... directory make[6]: *** [/opt/local/freeswitch/grammar/model/wsj1] Error 64 make[5]: *** [install] Error 1 make[4]: *** [mod_pocketsphinx-install] Error 1 make[3]: *** [install-recursive] Error 1
Could anybody shed some light on this what's wrong with this? I checked my target directory opt/local/freeswitch/grammar/model/wsj1, nothing is there.
Thanks,
Chris _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
Thanks Brian, the install is OK now. However when I tried the pizza demo I stopped at very beginning. see the snapshot from console log: 2008-07-18 16:40:33 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/remote/2002@x.y.z.g [3ff6ecf6-425d-4af4-aecf-32a38ee18a86] 2008-07-18 16:40:33 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing unknown->1888@public 2008-07-18 16:40:33 [NOTICE] mod_spidermonkey.c:2050 session_answer() Channel [sofia/remote/2002@x.y.z.g] has been answered 2008-07-18 16:40:35 [WARNING] mod_pocketsphinx.c:225 pocketsphinx_asr_close() Port Closed. 2008-07-18 16:40:35 [NOTICE] switch_ivr_async.c:1786 switch_ivr_detect_speech() Hangup sofia/remote/2002@x.y.z.g [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-18 16:40:35 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/remote/2002@x.y.z.g) Ended 2008-07-18 16:40:35 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/remote/2002@x.y.z.g[CS_HANGUP] Could you please guide through what could be the problem? Thanks, Chris
console loglevel debug Bet the grammar files aren't in the right place. /b On Jul 18, 2008, at 3:46 PM, unknown wrote:
Thanks Brian, the install is OK now. However when I tried the pizza demo I stopped at very beginning. see the snapshot from console log:
2008-07-18 16:40:33 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/remote/2002@x.y.z.g [3ff6ecf6-425d-4af4-aecf-32a38ee18a86] 2008-07-18 16:40:33 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing unknown->1888@public 2008-07-18 16:40:33 [NOTICE] mod_spidermonkey.c:2050 session_answer() Channel [sofia/remote/2002@x.y.z.g] has been answered 2008-07-18 16:40:35 [WARNING] mod_pocketsphinx.c:225 pocketsphinx_asr_close() Port Closed. 2008-07-18 16:40:35 [NOTICE] switch_ivr_async.c:1786 switch_ivr_detect_speech() Hangup sofia/remote/2002@x.y.z.g [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-18 16:40:35 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 5 (sofia/remote/2002@x.y.z.g) Ended 2008-07-18 16:40:35 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/remote/2002@x.y.z.g [CS_HANGUP]
Could you please guide through what could be the problem? Thanks,
Chris _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Brian West sip:brian@freeswitch.org
Also I had another bug in the makefile I fixed that... copy and paste error. :( Would have only affected the installation of the 16khz acoustical model. /b On Jul 18, 2008, at 3:46 PM, unknown wrote:
Thanks Brian, the install is OK now. However when I tried the pizza demo I stopped at very beginning. see the snapshot from console log:
Hi Brian, thanks for your tips. I have the pizza demo working now, but looks like you have one typo in the ps_pizza.js when doing the speciality confirmation, the wav file name is different from the script. See my error below: 2008-07-20 21:43:19 [INFO] SpeechTools.jm:150 console_log() ----TAKEOUT 2008-07-20 21:43:25 [INFO] SpeechTools.jm:150 console_log() ----YES 2008-07-20 21:43:36 [INFO] SpeechTools.jm:150 console_log() ----LARGE HUGE 2008-07-20 21:43:42 [INFO] SpeechTools.jm:150 console_log() ----LARGE 2008-07-20 21:43:54 [INFO] SpeechTools.jm:150 console_log() ----TOSSED 2008-07-20 21:44:00 [INFO] SpeechTools.jm:150 console_log() ----YES 2008-07-20 21:44:08 [INFO] SpeechTools.jm:150 console_log() ----SPECIALTY 2008-07-20 21:44:10 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/opt/local/freeswitch/sounds/en/us/pizza/ConfirmSpecialty.wav] [System error : No such file or directory.] 2008-07-20 21:44:20 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/opt/local/freeswitch/sounds/en/us/pizza/ConfirmSpecialty.wav] [System error : No such file or directory.] the actual wav file name is ConfirmSpeciality.wav So one patch is required. Thanks, Chris On Fri, Jul 18, 2008 at 10:21 PM, Brian West <brian@freeswitch.org> wrote:
Also I had another bug in the makefile I fixed that... copy and paste error. :( Would have only affected the installation of the 16khz acoustical model.
/b
On Jul 18, 2008, at 3:46 PM, unknown wrote:
Thanks Brian, the install is OK now. However when I tried the pizza demo I stopped at very beginning. see the snapshot from console log:
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Its already spelled correctly in the wav file from the website last I checked. I'm having them all re-recorded anyway. /b On Jul 20, 2008, at 9:01 PM, unknown wrote:
Hi Brian, thanks for your tips. I have the pizza demo working now, but looks like you have one typo in the ps_pizza.js when doing the speciality confirmation, the wav file name is different from the script. See my error below: 2008-07-20 21:43:19 [INFO] SpeechTools.jm:150 console_log() ---- TAKEOUT 2008-07-20 21:43:25 [INFO] SpeechTools.jm:150 console_log() ----YES 2008-07-20 21:43:36 [INFO] SpeechTools.jm:150 console_log() ---- LARGE HUGE 2008-07-20 21:43:42 [INFO] SpeechTools.jm:150 console_log() ----LARGE 2008-07-20 21:43:54 [INFO] SpeechTools.jm:150 console_log() ----TOSSED 2008-07-20 21:44:00 [INFO] SpeechTools.jm:150 console_log() ----YES 2008-07-20 21:44:08 [INFO] SpeechTools.jm:150 console_log() ---- SPECIALTY 2008-07-20 21:44:10 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/opt/local/freeswitch/sounds/en/us/pizza/ ConfirmSpecialty.wav] [System error : No such file or directory.] 2008-07-20 21:44:20 [ERR] mod_sndfile.c:175 sndfile_file_open() Error Opening File [/opt/local/freeswitch/sounds/en/us/pizza/ ConfirmSpecialty.wav] [System error : No such file or directory.]
the actual wav file name is ConfirmSpeciality.wav
So one patch is required.
Thanks,
Chris
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: freeswitch-users@lists.freeswitch.org 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 <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.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 <asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 <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.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Also, make sure your using the trunk of freeswitch. Mike On Jul 21, 2008, at 10:10 AM, Anthony Minessale wrote:
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 <asterisk@coltect.no-ip.com
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: freeswitch-users@lists.freeswitch.org 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 <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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
OK, thanks. I will get organised and try tomorrow. Thanks, Col ----- Original Message ----- From: Anthony Minessale To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, July 22, 2008 12:27 AM Subject: Re: [Freeswitch-users] OpenZap and tones.conf - FXO hangup problem 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 <asterisk@coltect.no-ip.com> 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 To: freeswitch-users@lists.freeswitch.org 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 <asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 <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.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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 To: freeswitch-users@lists.freeswitch.org 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 <asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 <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.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ------------------------------------------------------------------------------ _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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 <asterisk@coltect.no-ip.com> 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 <anthony.minessale@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *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 <asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 <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.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
we lost you on irc, can you tell us the ip and port On Mon, Jul 21, 2008 at 9:27 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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.netchannel #freeswitch
We're here every day so whenever you feel like dropping by....
On Mon, Jul 21, 2008 at 9:26 AM, Col Ferguson <asterisk@coltect.no-ip.com> 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 <anthony.minessale@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *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 <asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 < 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.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
but be sure to send it to just me or the whole world will know it ;) On Mon, Jul 21, 2008 at 10:20 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
we lost you on irc, can you tell us the ip and port
On Mon, Jul 21, 2008 at 9:27 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
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 <asterisk@coltect.no-ip.com> 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 <anthony.minessale@gmail.com> *To:* freeswitch-users@lists.freeswitch.org *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 < asterisk@coltect.no-ip.com> 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: freeswitch-users@lists.freeswitch.org 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 < 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.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org 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 Freeswitch-users@lists.freeswitch.org 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:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (5)
-
Anthony Minessale
-
Brian West
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Col Ferguson
-
Michael Jerris
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unknown