From 291266f50044381bbb40b9f37e00de3c4080ac6e Mon Sep 17 00:00:00 2001
From: Yalu Zhang <yalu.zhang@iopsys.eu>
Date: Fri, 7 Jun 2019 16:31:51 +0200
Subject: [PATCH] Re-fix Bug #705: Asterisk gives tone even if number is not
configured
Set the parameter "context" in asterisk channel configuration file according to
voice_client.sipX.call_lines, i.e. "Incoming Phone Lines".
Also 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.
---
voice-client/files/etc/config/voice_client | 1 -
voice-client/files/etc/init.d/voice_client | 58 +++++++++++++++++-----
2 files changed, 46 insertions(+), 13 deletions(-)
diff --git a/voice-client/files/etc/config/voice_client b/voice-client/files/etc/config/voice_client
index dc5816eee..d60de1e08 100644
--- a/voice-client/files/etc/config/voice_client
+++ b/voice-client/files/etc/config/voice_client
@@ -174,5 +174,4 @@ config sip_service_provider 'sip0'
option redial '*66'
option is_fax '0'
option transport 'udp'
- option call_lines '0'
diff --git a/voice-client/files/etc/init.d/voice_client b/voice-client/files/etc/init.d/voice_client
index 6b3238a65..4540e67e6 100755
--- a/voice-client/files/etc/init.d/voice_client
+++ b/voice-client/files/etc/init.d/voice_client
@@ -259,7 +259,7 @@ read_lines()
local loffset=$(ubus -t 1 call voice.asterisk platform | jsonfilter -e @.lineoffset)
loffset=${loffset:-0}
- config_get call_lines $1 call_lines "0"
+ config_get call_lines $1 call_lines
for line in $call_lines ; do
@@ -1839,12 +1839,49 @@ configure_tel()
sed -i "s/|DTMFCOMPATIBILITY|/$dtmfcompatibility/" $WORKDIR/$LINENAME.tmp
}
+# Return all SIP accounts to which a voice line is bound as per "voice_client.sipX.call_lines",
+# i.e. Incoming Phone Lines
+get_sip_accounts_for_incoming_call()
+{
+ local tel_line="$1"
+ local sip_accounts=""
+ local line_name_len=${#LINENAME}
+ local lnum=${tel_line:${line_name_len}}
+ local section cfgtype call_lines
+
+ for section in ${CONFIG_SECTIONS}; do
+ config_get cfgtype "$section" TYPE
+ [ "$cfgtype" != "sip_service_provider" ] && continue
+
+ config_get call_lines "$section" call_lines
+ # Note: voice_client.sipX.call_lines contains voice line index which starts with 0
+ # for all platforms, e.g. "0", "1", or "0 1"
+ [ -n "$call_lines" ] && echo "$call_lines" | grep -wq "$lnum" && {
+ local user enabled
+
+ config_get user $section user
+ config_get enabled $section enabled
+
+ [ "$section" != "-" -a "$enabled" == "1" -a -n "$user" ] && {
+ # Append this sip_acount to the tel line
+ if [ -z "$sip_accounts" ]; then
+ sip_accounts="$section"
+ else
+ sip_accounts="$sip_accounts $section"
+ fi
+ }
+ }
+ done
+
+ echo "$sip_accounts"
+}
+
+
#
# Configure settings for individual line in $LINENAME.conf
#
configure_tel_line()
{
- echo "Configuring $CHANNELNAME line $1"
local extension
local sip_provider
local codecs
@@ -1852,13 +1889,10 @@ configure_tel_line()
local autodial
local autodial_timeout
local dialtone_timeout
- local sip_provider_user
- local sip_provider_enabled
local callwaiting
local clir
config_get extension $1 extension
- config_get sip_provider $1 sip_account
config_get autodial $1 autodial
config_get autodial_timeout $1 autodial_timeout
config_get dialtone_timeout $1 dialtone_timeout
@@ -1866,9 +1900,8 @@ 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 "$sip_provider_user" ] ; then
+ sip_provider=$(get_sip_accounts_for_incoming_call $1)
+ if [ -z "$sip_provider" ] ; then
sip_provider="local_extensions"
fi
@@ -2240,16 +2273,17 @@ 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() {
--
GitLab