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god.nirvana at gmail.com
Guest

Posted: Thu Jun 04, 2009 3:58 am Post subject: [Freeswitch-users] busy tone detect issue

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

mike at jerris.com
Guest

Posted: Thu Jun 04, 2009 4:06 am Post subject: [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:

Quote:

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"/>



</settings>
<pri_spans>


<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"/>


<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"/>

</pri_spans>

<analog_spans>




<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"/>









<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"/>









<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"/>









<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"/>





</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 ([email]sofia/maqian/6789@njpbx.vicp.net[/email])"/>
</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

god.nirvana at gmail.com
Guest

Posted: Thu Jun 04, 2009 9:28 am Post subject: [Freeswitch-users] busy tone detect issue

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: [span zt FXO1]name => OpenZAP-FXO1number => 1fxo-channel => 1[span zt FXO2]name => OpenZAP-FXO2number => 2fxo-channel => 2 [span zt FXO3]name => OpenZAP-FXO3number => 3fxo-channel => 3 [span zt FXO4]name => OpenZAP-FXO4number => 4fxo-channel => 4 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

dujinfang at gmail.com
Guest

Posted: Thu Jun 04, 2009 9:00 pm Post subject: [Freeswitch-users] busy tone detect issue

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:

Quote:

msc at freeswitch.org
Guest

Posted: Fri Jun 05, 2009 12:16 am Post subject: [Freeswitch-users] busy tone detect issue

On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

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

dujinfang at gmail.com
Guest

Posted: Fri Jun 05, 2009 4:12 am Post subject: [Freeswitch-users] busy tone detect issue

On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:

Quote:

On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

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.

Quote:

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dujinfang at gmail.com
Guest

Posted: Fri Jun 05, 2009 9:13 pm Post subject: [Freeswitch-users] busy tone detect issue

On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:

Quote:

On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

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

Quote:

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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gcd at i.ph
Guest

Posted: Fri Jun 05, 2009 10:04 pm Post subject: [Freeswitch-users] busy tone detect issue

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 (dujinfang@gmail.com)> wrote:

Quote:

On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:

Quote:

On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

I'm using openzap analog with tone_detect, it works(conference not tested). however,Â 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.

1)Â Don't know why but the similar zaptel.conf works on asterisk. I guess tone_detect in FS isÂ equivalent to busydetect=yes in Asterisk(zapata.conf)Â .Â

zaptel.conf
# Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER)Â
fxsks=1
fxsks=2
fxsks=3
fxsks=4

# Span 2: WCTDM/1 "Wildcard TDM400P REV I Board 2"Â
fxoks=5
fxoks=6
fxoks=7
fxoks=8

# Global data

loadzone Â Â Â Â = us
defaultzone Â Â = 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 isÂ weird, why not they just match the care name and reverse that internally?Â

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.Â Is there any equivalent configuration in FS like echocanccelwhenbridged=no in asterisk? can I set busydetect and echocancelwhenbridged and other options like thisÂ <param name="enable-analog-option" value="call-swap"/> ?

[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64

Quote:

dujinfang at gmail.com
Guest

Posted: Mon Jun 08, 2009 1:32 am Post subject: [Freeswitch-users] busy tone detect issue

On Jun 6, 2009, at 11:03 AM, Nandy Dagondon wrote:

Quote:

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. Smile

Quote:

-nandy

On Sat, Jun 6, 2009 at 10:11 AM, dujinfang <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

On Jun 5, 2009, at 1:14 PM, Michael Collins wrote:

Quote:

On Thu, Jun 4, 2009 at 6:59 PM, seven <dujinfang@gmail.com (dujinfang@gmail.com)> wrote:

Quote:

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