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Re-fix Bug #705: Asterisk gives tone even if number is not configured
Remove voice_client.sip0.call_lines in the default config. So no voice lines are bound to "Incoming Phone Lines" of any SIP accounts by default. Reload FXS channel before chan_sip when voice configuration changes.
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parent
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commit
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2 changed files with 6 additions and 7 deletions
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@ -174,5 +174,4 @@ config sip_service_provider 'sip0'
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option redial '*66'
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option redial '*66'
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option is_fax '0'
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option is_fax '0'
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option transport 'udp'
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option transport 'udp'
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option call_lines '0'
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@ -259,7 +259,7 @@ read_lines()
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local loffset=$(ubus -t 1 call voice.asterisk platform | jsonfilter -e @.lineoffset)
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local loffset=$(ubus -t 1 call voice.asterisk platform | jsonfilter -e @.lineoffset)
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loffset=${loffset:-0}
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loffset=${loffset:-0}
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config_get call_lines $1 call_lines "0"
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config_get call_lines $1 call_lines
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for line in $call_lines ; do
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for line in $call_lines ; do
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@ -1844,7 +1844,6 @@ configure_tel()
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#
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#
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configure_tel_line()
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configure_tel_line()
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{
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{
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echo "Configuring $CHANNELNAME line $1"
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local extension
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local extension
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local sip_provider
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local sip_provider
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local codecs
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local codecs
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@ -2240,16 +2239,17 @@ stop_service() {
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reload_service() {
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reload_service() {
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start
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start
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#stop
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# turn off voice led; asterisk will turn it on if there is a registered account
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# turn off voice led; asterisk will turn it on
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# if there is a registered account
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ubus call led.voice1 set '{"state":"off"}'
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ubus call led.voice1 set '{"state":"off"}'
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# FXS channel module must be reloaded before sip module. Otherwise some attributes like
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# line's registration state which is updated by SIP module through callback might be
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# reset.
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asterisk -rx "$(getChipVendor) reload"
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asterisk -rx "config reload $ASTERISKDIR/sip.conf"
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asterisk -rx "config reload $ASTERISKDIR/sip.conf"
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sleep 1
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sleep 1
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asterisk -rx "core reload"
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asterisk -rx "core reload"
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asterisk -rx "dialplan reload"
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asterisk -rx "dialplan reload"
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asterisk -rx "$(getChipVendor) reload"
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}
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}
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service_triggers() {
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service_triggers() {
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