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Fix Bug #705: Asterisk gives tone even if number is not configured

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Set voice_client.lineX.sip_account correctly according to "voice_client.sipX.call_lines"
This commit is contained in:
Yalu Zhang 2019-05-17 09:19:09 +02:00
parent 5a4c07278c
commit 44164d950b
1 changed files with 61 additions and 7 deletions
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68
voice-client/files/etc/init.d/voice_client
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@ -1917,9 +1917,11 @@ configure_tel_line()
config_get clir $1 clir
#If line is configured with no sip_provider, or sip_provider is disabled, use local_extensions context
config_get sip_provider_user $sip_provider user
config_get sip_provider_enabled $sip_provider enabled
if [ "$sip_provider" == "-" -o -z "$sip_provider_enabled" -o "$sip_provider_enabled" == "0" -o -z "$sip_provider_user" ] ; then
[ -n "$sip_provider" ] && {
config_get sip_provider_user $sip_provider user
config_get sip_provider_enabled $sip_provider enabled
}
if [ -z "$sip_provider" -o "$sip_provider" == "-" -o -z "$sip_provider_enabled" -o "$sip_provider_enabled" == "0" -o -z "$sip_provider_user" ] ; then
sip_provider="local_extensions"
fi
@ -2062,6 +2064,48 @@ pbx_fix_ownership()
chown $ASTUSER:$ASTGROUP -R $ASTDIRSRECURSIVE
}
# Set voice_client.LINE_x.sip_account correctly as per "voice_client.sipX.call_lines"
set_line_sip_account()
{
local tel_line=$1
local old_sip_account=
local new_sip_account=
local line_name=$(getLineName)
local line_name_len=${#line_name}
local lnum=${tel_line:${line_name_len}}
local chan_name=$(getChannelName)
local ___type="sip_service_provider"
local section cfgtype call_lines chan_num
config_get old_sip_account $tel_line sip_account
# Note: On Intel platform, tel_line's section name starts with 0, e.g. tapi0.
# But channel index in voice_client.sipX.call_lines starts with 1, e.g. TAPI/1
if [ "$line_name" == "tapi" ]; then
chan_num=$((lnum+1))
else
chan_num=$lnum
fi
for section in ${CONFIG_SECTIONS}; do
config_get cfgtype "$section" TYPE
[ -n "$___type" -a "x$cfgtype" != "x$___type" ] && continue
config_get call_lines "$section" call_lines
# Note: tel_line's section name starts with 0, e.g. tapi0 or brcm0.
# But channel index in voice_client.sipX.call_lines starts with 1, e.g. TAPI/1
[ -n "$call_lines" ] && echo "$call_lines" | grep -q "$chan_name/$chan_num" && {
# Append this sip_acount to the tel line
if [ -z "$new_sip_account" ]; then
new_sip_account="$section"
else
new_sip_account="$new_sip_account $section"
fi
}
done
[ "$old_sip_account" != "$new_sip_account" ] && uci_set voice_client $tel_line sip_account "$new_sip_account"
}
#
# Calculate a name for each $(getLineName) line, depending on port type
# and number. The name is used to make UI look better.
@ -2158,6 +2202,12 @@ start_service() {
# Load config file
config_load voice_client
########################################################
# Set line sip_account according to sip accounts'
# settings
########################################################
config_foreach set_line_sip_account tel_line
########################################################
# Set line names according to board parameters
# and delete non-existing lines from voice_client config
@ -2166,6 +2216,9 @@ start_service() {
config_foreach set_line_name tel_line $maxlinenum
uci_commit voice_client
# Reload the config since it has changed
config_load voice_client
#######################################
# Create temporary files from templates
#######################################
@ -2291,16 +2344,17 @@ stop_service() {
reload_service() {
start
#stop
# turn off voice led; asterisk will turn it on
# if there is a registered account
# turn off voice led; asterisk will turn it on if there is a registered account
ubus call led.voice1 set '{"state":"off"}'
# FXS channel module must be reloaded before sip module. Otherwise some attributes like
# line's registration state which is updated by SIP module through callback might be
# reset.
asterisk -rx "$(getChipVendor) reload"
asterisk -rx "config reload $ASTERISKDIR/sip.conf"
sleep 1
asterisk -rx "core reload"
asterisk -rx "dialplan reload"
asterisk -rx "$(getChipVendor) reload"
}
service_triggers() {