slightly improve N5 Route, adding IBCF Prototype

.env

@@ -130,6 +130,9 @@ SRS_GNB_IP=172.22.0.37
 # GRAFANA
 GRAFANA_IP=172.22.0.39
 
+# IBCF 
+IBCF_IP=172.22.0.140
+
 # UE IPv4 Subnet Range for APN=internet
 UE_IPV4_INTERNET=192.168.100.0/24
 

dns/e164.arpa

@@ -10,5 +10,6 @@ $TTL 1h
 @ IN A DNS_IP
 
 ; Wildcard to match any tel:+xxxx and change to sip:xxxx@IMS_DOMAIN
-* IN NAPTR 10 100 "u" "E2U+sip" "!(^.*$)!sip:\\1@IMS_DOMAIN!" .
-* IN NAPTR 20 100 "u" "E2U+sip" "!(^.*$)!sip:+\\1@IMS_DOMAIN!" .
+*.9.4 IN NAPTR 20 100 "u" "E2U+sip" "!(^.*$)!sip:\\[email protected]_DOMAIN!" .
+; add more country codes to route over IBCF using the same format as above, you may need also to adjust teh dialplan in Asterisk:
+; * IN NAPTR 20 100 "u" "E2U+sip" "!(^.*$)!sip:\\[email protected]_DOMAIN!" .

dns/ims_zone

@@ -27,3 +27,11 @@ hss                     1D IN A         PYHSS_IP
 smsc                    1D IN A         SMSC_IP
 _sip._udp.smsc          1D SRV 0 0 7090 smsc
 _sip._tcp.smsc          1D SRV 0 0 7090 smsc
+
+ibcf                    1D IN A         IBCF_IP
+_sip._udp.ibcf          1D SRV 0 0 5090 ibcf
+_sip._tcp.ibcf          1D SRV 0 0 5090 ibcf
+
+voicemail               1D IN A         IBCF_IP
+_sip._udp.voicemail     1D SRV 0 0 5090 voicemail
+_sip._tcp.voicemail     1D SRV 0 0 5090 voicemail

ibcf/Dockerfile

@@ -0,0 +1,68 @@
+# syntax=docker/dockerfile:1
+
+FROM debian:bullseye
+# Install required packages
+RUN apt-get update \
+    && apt-get upgrade -y \
+    && apt-get install -y \
+        lsb-release \
+        ca-certificates \
+        git \
+        vim \
+        wget \
+        curl \
+        openssh-server \
+        net-tools \
+    && apt-get clean
+RUN apt --no-install-recommends --assume-yes install unzip autoconf automake build-essential pkg-config libedit-dev libjansson-dev libsqlite3-dev uuid-dev libxslt1-dev xmlstarlet \
+    && apt clean
+
+# Download Asterisk source :
+WORKDIR /usr/src
+RUN git clone -b releases/22 https://github.com/asterisk/asterisk.git
+# Adding EVS and AMR support
+# First EVS 
+WORKDIR /usr/src/asterisk
+# install additional DEBs required by Asterisk
+RUN sh contrib/scripts/install_prereq install
+RUN sh contrib/scripts/get_mp3_source.sh
+RUN git clone https://github.com/NUCLEAR-WAR/asterisk-evs.git
+RUN cp --verbose --recursive ./asterisk-evs*/* ./
+RUN patch -p0 <./codec_evs.patch
+RUN wget www.etsi.org/deliver/etsi_ts/126400_126499/126443/16.01.00_60/ts_126443v160100p0.zip
+RUN unzip -qq ts_126443v*.zip
+RUN unzip -qq 26443-*-ANSI-C_source_code.zip
+WORKDIR /usr/src/asterisk/c-code
+RUN chmod +r ./lib_*/*.h
+RUN mkdir /usr/include/3gpp-evs
+RUN cp --verbose --target-directory=/usr/include/3gpp-evs ./lib_*/*.h
+RUN DEBUG=0 RELEASE=1 CFLAGS='-DNDEBUG -fPIC' make
+WORKDIR /usr/src/asterisk/c-code/build
+RUN rm ./decoder.o
+RUN cc -shared -o lib3gpp-evs.so *.o
+RUN cp ./lib3gpp-evs.so /usr/lib/
+WORKDIR /usr/src/asterisk
+RUN patch -p0 <./build_evs.patch
+RUN patch -p0 <./force_limitations.patch
+
+# Now AMR
+RUN apt --assume-yes install libopencore-amrnb-dev libopencore-amrwb-dev libvo-amrwbenc-dev
+RUN wget github.com/traud/asterisk-amr/archive/master.zip
+RUN unzip -qq master.zip
+RUN rm master.zip
+RUN cp --verbose --recursive ./asterisk-amr*/* ./
+RUN patch -p0 <./codec_amr.patch
+RUN patch -p0 <./build_tools.patch
+
+# Run the bootstrap script to re-generate configure files and configure patched Asterisk:
+RUN ./bootstrap.sh
+RUN ./configure
+COPY menuselect.makedeps .
+COPY menuselect.makeopts .
+RUN make && make install
+
+COPY config/* /etc/asterisk/
+
+CMD ["/mnt/ibcf/start-asterisk.sh"]
+
+

ibcf/README.md

@@ -0,0 +1,5 @@
+This is a basic kind of IBCF based on Asterisk to handle calls from and to Docker_open5gs, I tried my best to do all the configs, but it still requires your adjustments.
+
+Please read the Comments carefully before starting the container, specially in : pjsip.conf, acl.conf, extenteions.conf and voiemail.conf, those files must be configured otherwise the IBCF will not work properly!!
+
+have fun Testing!

ibcf/config/acl.conf

@@ -0,0 +1 @@
+[ibcf_acl]

ibcf/config/asterisk.conf

@@ -0,0 +1,13 @@
+[options]
+; If we want to start Asterisk with a default verbosity for the verbose
+; or debug logger channel types, then we use these settings (by default
+; they are disabled).
+;verbose = 5
+;debug = 2
+
+; User and group to run asterisk as. NOTE: This will require changes to
+; directory and device permissions.
+;runuser = asterisk		; The user to run as. The default is root.
+;rungroup = asterisk		; The group to run as. The default is root
+
+;defaultlanguage = es

ibcf/config/cdr.conf

@@ -0,0 +1,7 @@
+[general]
+enable=yes
+
+[custom]
+; We log the unique ID as it can be useful for troubleshooting any issues
+; that arise.
+loguniqueid=yes

ibcf/config/cdr_custom.conf

@@ -0,0 +1,4 @@
+[mappings]
+; Our CDR log will be written to /var/log/asterisk/cdr-custom/Master.csv
+; with the following schema.
+Master.csv => ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})},${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})},${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})},${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})},${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})},${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})},${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})},${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})},${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})},${CDR(sequence)}

ibcf/config/confbridge.conf

@@ -0,0 +1 @@
+; All conferences use default settings. This config must be present to load the confbridge application

ibcf/config/extensions.conf

@@ -0,0 +1,176 @@
+
+; extensions.conf - the Asterisk dial plan
+;
+; Static extension configuration file, used by
+; the pbx_config module. This is where you configure all your
+; inbound and outbound calls in Asterisk.
+;
+; This configuration file is reloaded
+; - With the "dialplan reload" command in the CLI
+; - With the "reload" command (that reloads everything) in the CLI
+
+;
+; The "General" category is for certain variables.
+;
+[general]
+;
+; If static is set to no, or omitted, then the pbx_config will rewrite
+; this file when extensions are modified.  Remember that all comments
+; made in the file will be lost when that happens.
+;
+; XXX Not yet implemented XXX
+;
+static=yes
+;
+; if static=yes and writeprotect=no, you can save dialplan by
+; CLI command "dialplan save" too
+;
+writeprotect=yes
+;
+; If autofallthrough is set, then if an extension runs out of
+; things to do, it will terminate the call with BUSY, CONGESTION
+; or HANGUP depending on Asterisk's best guess. This is the default.
+;
+; If autofallthrough is not set, then if an extension runs out of
+; things to do, Asterisk will wait for a new extension to be dialed
+; (this is the original behavior of Asterisk 1.0 and earlier).
+;
+;autofallthrough=no
+;
+;
+;
+; If extenpatternmatchnew is set (true, yes, etc), then a new algorithm that uses
+; Trie
+extenpatternmatchnew=no
+;
+; If clearglobalvars is set, global variables will be cleared
+; and reparsed on a dialplan reload, or Asterisk reload.
+;
+; If clearglobalvars is not set, then global variables will persist
+; through reloads, and even if deleted from the extensions.conf or
+; one of its included files, will remain set to the previous value.
+;
+; NOTE: A complication sets in, if you put your global variables into
+; the AEL file, instead of the extensions.conf file. With clearglobalvars
+; set, a "reload" will often leave the globals vars cleared, because it
+; is not unusual to have extensions.conf (which will have no globals)
+; load after the extensions.ael file (where the global vars are stored).
+; So, with "reload" in this particular situation, first the AEL file will
+; clear and then set all the global vars, then, later, when the extensions.conf
+; file is loaded, the global vars are all cleared, and then not set, because
+; they are not stored in the extensions.conf file.
+;
+clearglobalvars=yes
+;
+; User context is where entries from users.conf are registered.  The
+; default value is 'default'
+;
+
+
+; Welcome to Asterisk Dialplan!! its not complicated, you will get use to it in just a "few" tries
+userscontext=unspecified
+
+
+; Here are some deffined variables that will be used in the dialplan, some of them must be adjusted to your need!
+
+; I use a german Trunk so I have the +49, adjust depand on your country, this is used on incoming, as my provider sends only E164 numbers.
+[globals]
+INTERNAL_DIAL_OPT=,30
+ENUM_PREFIX_FOR_DE=+49
+
+; this this the head number, you need to change it to teh value provided by your providers, in consist of 10 digit, this is important as we will cut the number and normalize it before it enters the IMS Core, there is various names for Head number like : Direct dial in .. prefix numbers..etc so its basiclly the number from your SIPTrunk Provider that can be extended and add digits to it
+
+PHONE_NR_HEAD=1234567890
+
+; Dont change
+ICP_OUT=ICP_LINK_out
+ICP_IN=ims_core
+; change to your provider's provided domain
+SIPTRUNK_DOMAIN=PROVIDER_DOMAIN
+; Dont change
+IMS_CN_DOMAIN=IMS_DOMAIN
+PAI_HEADER=P-Asserted-Identity
+PPI_HEADER=P-Preferred-Identity
+NO_FORK=Request-Disposition
+P_HEADER=P-Early-Media
+X_HEADER=P-MeinSipTrunk
+
+; now the dial plan, here is an explnination so you can adjust what you need to make it able to proccess calls and send it to and from teh IMS Core
+; in "incoming" in my case i used an internal MSISDN for my open5GS Netwrok 0912584710 as you see this is a private self generated number.
+; in normal world there is a shared Database for all operator so they know that a numer belong to a specific operator, in our case the netwrok is private and our
+; numbers are unknown to teh wolrd, so I had to do a trick, this trick consist of the head numer provided from my SIPTrunk provider adding the UE MSISDN to it liek that:
+; CC+HEADER_NUMER+MSSIDN = +49 228 1234567 0912584710 and then cut or add the CC+HEADER_NUMER depanding on the direction :
+; to IMS Core cute CC+HEADER_NUMER, if to outside world then add CC+HEADER_NUMER
+; this is using the ${EXTEN:n} "n" is where the digit that need to stay along with the digits the come after it. in my case its the 0, so the results will be 0912584710 that will
+; be recognized insdie my private network, this need to be adjusted depand on the your needs like the lenth of your Head number and your MSISDN.
+; after doing that we need to send the call to the ICSCF as we dont know where the user is registred, the rest will be done insdie the IMS Core+Open5GS
+;
+; its preconfigured here on digit 13 ( UK users please change it! not to trigger any bad luck!) 
+
+
+[incoming]
+
+;              +49               228 1234567   + any
+exten => _${ENUM_PREFIX_FOR_DE}${PHONE_NR_HEAD}X.,1,Verbose(incoming call from ICP from ${CALLERID(num)} calling to ${EXTEN:13})
+same => n,Verbose(check for exten ${EXTEN} 8 ${EXTEN:8} 9 ${EXTEN:9} 10 ${EXTEN:10} 11 ${EXTEN:11} 13 ${EXTEN:13} )
+; cutting until digi 13               CUT0912584710 set some headers and send to icscf to reach the users
+same => n,Dial(PJSIP/${ICP_IN}/sip:${EXTEN:13}@${IMS_CN_DOMAIN},120,b(outgoing^ims_inc^1))
+same => n,Hangup()
+
+[outgoing]
+
+; Due to diffrent way that the devices do calls there is two way defined here eather the call will start with 0 ( in teh IMS is not possbile) or with +
+
+; here is with 0, its not applicale, our IMS is 3gpp compitable so it has always E164
+exten => _0X.,1,Verbose(outgoing conext leading null ${EXTEN} calling from IMS User ${CALLERID(num)} )
+same => n,Dial(PJSIP/${ICP_OUT}/sip:${ENUM_PREFIX_FOR_DE}${EXTEN:1}@${SIPTRUNK_DOMAIN},120,b(outgoing^headers^1))
+same => n,Hangup()
+
+
+; here is the used one as we use E164 in IMS
+exten => _${ENUM_PREFIX_FOR_DE}X.,1,Verbose(outgoing Call leading +49 calling from IMS User ${CALLERID(num)} to ${EXTEN} )
+same => n,Verbose(Sending INVITE to ICP)
+; we set the head number and add the MSISDN to it then set it in the "From" header
+same => n,Set(CALLERID(num)=${ENUM_PREFIX_FOR_DE}${PHONE_NR_HEAD}${CALLERID(num)})
+; setting some headers and sends the INVITE to the Provdier/ICP
+same => n,Dial(PJSIP/${ICP_OUT}/sip:${EXTEN}@${SIPTRUNK_DOMAIN},120,b(outgoing^ims_out^1))
+same => n,Hangup()
+
+; this headers used in outgoing 
+exten => ims_out,1,Verbose(Setting required headers in outgoing INVITE)
+; setting the P-Preferred-Identity and optionally P-Preferred-Identity ( PAI is only a Networ thing) here we do the the as we done with From:
+; adding the MSISDN ot the head number before sending it out.
+same => n,Set(PJSIP_HEADER(add,${PPI_HEADER})=<sip:${ENUM_PREFIX_FOR_DE}${PHONE_NR_HEAD}${CONNECTEDLINE(num)}@${SIPTRUNK_DOMAIN}>)
+same => n,Set(PJSIP_HEADER(add,${PAI_HEADER})=<sip:${ENUM_PREFIX_FOR_DE}${PHONE_NR_HEAD}${CONNECTEDLINE(num)}@${SIPTRUNK_DOMAIN}>)
+; Setting P-Early-Media to Supported, Asterisk know nothing about RFC5009, but its required by 3GPP, so we can only cheat here sorry!
+same => n,Set(PJSIP_HEADER(add,${P_HEADER})=Supported)
+same => n,Set(PJSIP_HEADER(add,Supported)=199)
+same =>	n,Return()
+
+; for incoming we need to set the PAI, some devices need this header
+exten => ims_inc,1,Verbose(Setting required headers in incoming INVITE)
+same => n,Set(PJSIP_HEADER(add,${PAI_HEADER})=<tel:${CONNECTEDLINE(num)}>)
+same => n,Set(PJSIP_HEADER(add,${PAI_HEADER})=<sip:${CONNECTEDLINE(num)}@${IMS_CN_DOMAIN}>)
+same => n,Set(PJSIP_HEADER(add,${P_HEADER})=Supported)
+same => n,Set(PJSIP_HEADER(add,Supported)=199)
+same =>	n,Return()
+
+[IMS_INC]
+
+include => outgoing
+
+;;;;;;; Voicemail IMS, this is a WIP, need to add appreciate IFC for it, dont forget to confgure your MSISDN in the voicemail.conf!!
+
+; configure teh start of your number instaed of "09125" eg. if your MSISDN start with 09125123456 then exten => _09125X.
+exten => _09125X.,1,Verbose(1, Caller ${CALLERID(all)} has entered Voicemail for ${EXTEN} from IMS )
+same => n,VoiceMail(${EXTEN})
+
+exten => 500,1,Ringing()
+exten => 500,n,Wait(2)
+exten => 500,n,Answer
+exten => 500,n,Wait(2)
+exten => 500,n,VoiceMailMain(${CALLERID(num)})
+exten => 500,n,HangUp()
+exten => 500,n,PlayBack(vm-goodbye)
+exten => 500,n,HangUp() 
+

ibcf/config/indications.conf

@@ -0,0 +1,21 @@
+[general]
+country = de
+
+[de]
+description = Germany
+; Reference: http://www.itu.int/ITU-T/inr/forms/files/tones-0203.pdf
+ringcadence = 1000,4000
+dial = 425
+busy = 425/480,0/480
+ring = 425/1000,0/4000
+congestion = 425/240,0/240
+callwaiting = !425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,!0/5000,!425/200,!0/200,!425/200,0
+; DIALRECALL - not specified
+dialrecall = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
+; RECORDTONE - not specified
+record = 1400/80,0/15000
+info = 950/330,1400/330,1800/330,0/1000
+stutter = 425+400
+
+; Additional country configurations can be found in the Asterisk source
+; at /configs/samples/indications.conf.sample

ibcf/config/logger.conf

@@ -0,0 +1,9 @@
+[general]
+
+[logfiles]
+
+console = verbose,notice,warning,error
+
+;messages = notice,warning,error
+;full = verbose,notice,warning,error,debug
+;security = security