hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 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 openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> <pri_spans> <span name="PRI_1"> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug --> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span name="PRI_2"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> <!-- one entry here per openzap span --> <analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="2"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="3"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="4"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans> </configuration> when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. <extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789@njpbx.vicp.net"/> </condition> </extension> restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana
Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote:
hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues
my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 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
openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> <pri_spans> <span name="PRI_1"> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug --> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span name="PRI_2"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> <!-- one entry here per openzap span --> <analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="2"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="3"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="4"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans> </configuration>
when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased.
but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan.
<extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789@njpbx.vicp.net"/> </condition> </extension>
restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR
hi,thanks for your reply. my openzap.conf like this: [span zt FXO1] name => OpenZAP-FXO1 number => 1 fxo-channel => 1 [span zt FXO2] name => OpenZAP-FXO2 number => 2 fxo-channel => 2 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 3 [span zt FXO2] name => OpenZAP-FXO2 number => 3 fxo-channel => 4 i have a 4 fxo ports TDM400. 2009-06-04 god.nirvana 发件人： Michael Jerris 发送时间： 2009-06-04 17:06:05 收件人： freeswitch-users 抄送： 主题： Re: [Freeswitch-users] busy tone detect issue Are you having this issue on your analog or pri lines? what does your openzap.conf look like? Mike On Jun 4, 2009, at 4:28 AM, god.nirvana wrote: hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 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 openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> <pri_spans> <span name="PRI_1"> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug --> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span name="PRI_2"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> <!-- one entry here per openzap span --> <analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="2"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="3"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="4"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans> </configuration> when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. <extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789@njpbx.vicp.net"/> </condition> </extension> restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR
I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then. On Jun 4, 2009, at 4:28 PM, god.nirvana wrote:
hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues
my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 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
openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> <pri_spans> <span name="PRI_1"> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug --> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> <span name="PRI_2"> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> <!-- one entry here per openzap span --> <analog_spans> <span id="1"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="2"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="3"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> <span id="4"> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/> <!-- regex to stop dialing when it matches --> <!--<param name="dial-regex" value="5555"/>--> <!-- regex to stop dialing when it does not match --> <!--<param name="fail-dial-regex" value="^5"/>--> </span> </analog_spans> </configuration>
when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased.
but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan.
<extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789@njpbx.vicp.net"/> </condition> </extension>
restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M .Q 2009-06-04 god.nirvana _______________________________________________ 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
On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com> wrote:
I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
Dujinfang, Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.) Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks. -MC
On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com> wrote: I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
Dujinfang,
Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.)
Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks.
-MC
Thank you for the detailed explain MC. so the ks means kewlstart, it already set, but no luck. anyway, the tone_detect works for me, less worry about that. Thanks again.
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On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com> wrote: I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
Dujinfang,
Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.)
1) Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS is equivalent to busydetect=yes in Asterisk(zapata.conf) . zaptel.conf # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4 # Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" fxoks=5 fxoks=6 fxoks=7 fxoks=8 # Global data loadzone = us defaultzone = us zapata.conf usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes I agree the FXO and FXS signaling is weird, why not they just match the care name and reverse that internally? 2) Another essue is if I dial out from a FXO port from a local extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo on asterisk. the zt.conf as below and I tried to change the echo_cancel_level to 32 or 128 got no much difference. Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like this <param name="enable-analog-option" value="call-swap"/
?
[defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64
Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks.
-MC
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dujinfang, hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on *. oslec works w/ FS, too. -nandy On Sat, Jun 6, 2009 at 10:11 AM, dujinfang <dujinfang@gmail.com> wrote:
On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com> wrote:
I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
Dujinfang,
Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.)
1) Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS is equivalent to busydetect=yes in Asterisk(zapata.conf) .
zaptel.conf # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4
# Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" fxoks=5 fxoks=6 fxoks=7 fxoks=8
# Global data
loadzone = us defaultzone = us
zapata.conf
usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes
I agree the FXO and FXS signaling is weird, why not they just match the care name and reverse that internally?
2) Another essue is if I dial out from a FXO port from a local extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo on asterisk. the zt.conf as below and I tried to change the echo_cancel_level to 32 or 128 got no much difference. Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like this <param name="enable-analog-option" value="call-swap"/> ?
[defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64
Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks.
-MC
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On Jun 6, 2009, at 11:03 AM, Nandy Dagondon wrote:
dujinfang,
hv u tried OSLEC? it's really reduced echo even on the cheapy X100P card on *. oslec works w/ FS, too.
Thanks, will try. :)
-nandy
On Sat, Jun 6, 2009 at 10:11 AM, dujinfang <dujinfang@gmail.com> wrote:
On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:
On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com> wrote: I'm using openzap analog with tone_detect, it works(conference not tested). however, according to the asterisk book, Kewlstart can detect the busy tone and disconnect the circuit. does anyone knows how to configure kewlstart with freeswitch/openzap? guess we don't need tone_detect then.
Dujinfang,
Your telco must support "kewlstart" signaling for this to be effective. The telco probably calls it something different, like "disconnect supervision" or "drop in loop current" or "battery reversal" or something like that. In any case, if the signaling is supported then you need to set up your zaptel.conf with the appropriate signaling type, which is either fxoks or fxsks. (I can never remember because zaptel does it backwards where if you have an FXO port then it uses FXS signaling but if you have an FXS port it uses FXO signaling. Stupidity, to be sure, so be aware of it.)
1) Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS is equivalent to busydetect=yes in Asterisk(zapata.conf) .
zaptel.conf # Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) fxsks=1 fxsks=2 fxsks=3 fxsks=4
# Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2" fxoks=5 fxoks=6 fxoks=7 fxoks=8
# Global data
loadzone = us defaultzone = us
zapata.conf
usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no busydetect=yes
I agree the FXO and FXS signaling is weird, why not they just match the care name and reverse that internally?
2) Another essue is if I dial out from a FXO port from a local extension(sip and zap), I can hear much echo on FreeSWITCH but not much echo on asterisk. the zt.conf as below and I tried to change the echo_cancel_level to 32 or 128 got no much difference. Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like this <param name="enable-analog-option" value="call- swap"/> ?
[defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64
Find the sample zaptel.conf that comes with the zaptel package and search it for fxsks or fxoks and you'll see some notes on how to set it up for your analog trunks.
-MC
_______________________________________________ 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
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participants (6)
-
dujinfang
-
god.nirvana
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Michael Collins
-
Michael Jerris
-
Nandy Dagondon
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seven