Integration Examples | Developer Documentation

[c2b-sub-dial]
exten => s,1,NoOp()
  same => n,Read(Digits,incoming_welcome,0,,5,500)
  same => n,Dial(PJSIP/${Digits})
  same => n,Hangup()[whatsapp]
exten => _10XX,1,NoOp()
  same => n,Dial(PJSIP/${EXTEN})
  same => n,Hangup();Extensionfor B2C business call through Meta SIP gateway
exten => b2c-sip,1,NoOp()
  same => n,Read(Digits,outgoing_welcome,0,,5,500)
  same => n,Dial(PJSIP/whatsapp/sip:${Digits}@wa.meta.vc);Extension to handle incoming invite requests fromMeta SIP gateway to <wa-business-phone-number>@<asterisk-sip-server-dns>
exten => _+<wa-business-phone-number>,1,Goto(c2b-sub-dial,s,1)

[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
cert_file=/var/lib/asterisk/certs/fullchain.cer
priv_key_file=/var/lib/asterisk/certs/cer.key
method=sslv23
allow_wildcard_certs=yes
external_media_address={external-media-address};External address for SIP signalling
external_signaling_address={external-signaling-address};Network to consider local used for NAT purposes
local_net={local-net}[sdes_endpointtemplate](!)
type=endpoint
context=whatsapp
disallow=all
allow=OPUS
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=no
media_use_received_transport=yes
media_encryption=sdes

[authtemplate](!)
type=auth
auth_type=userpass
password={sip-ua-password}[aortemplate](!)
type=aor
max_contacts=1
remove_existing=yes

[aoridentitytemplate](!)
type=identify
match_header=X-FB-External-Domain: wa.meta.vc

;SDES users
[1000](sdes_endpointtemplate)
auth=1000_auth
aors=1000[1000_auth](authtemplate)
username=1000[1000](aortemplate)[1000](aoridentitytemplate)
endpoint=1000[1001](sdes_endpointtemplate)
auth=1001_auth
aors=1001[1001_auth](authtemplate)
username=1001[1001](aortemplate)[1001](aoridentitytemplate)
endpoint=1001[1002](sdes_endpointtemplate)
auth=1002_auth
aors=1002[1002_auth](authtemplate)
username=1002[1002](aortemplate)[1002](aoridentitytemplate)
endpoint=1002[1003](sdes_endpointtemplate)
auth=1003_auth
aors=1003[1003_auth](authtemplate)
username=1003[1003](aortemplate)[1003](aoridentitytemplate)
endpoint=1003[1004](sdes_endpointtemplate)
auth=1004_auth
aors=1004[1004_auth](authtemplate)
username=1004[1004](aortemplate)[1004](aoridentitytemplate)
endpoint=1004[1005](sdes_endpointtemplate)
auth=1005_auth
aors=1005[1005_auth](authtemplate)
username=1005[1005](aortemplate)[1005](aoridentitytemplate)
endpoint=1005;This endpoint maps to an IVR for C2B calls
[c2b-sip](sdes_endpointtemplate)[c2b-sip](aortemplate)[c2b-sip]
type=identify
endpoint=c2b-sip
match_header=X-FB-External-Domain: wa.meta.vc

;special endpoint forMeta SIP Gateway integration
;This endpoint maps to an IVR for B2C calls
[b2c-sip](sdes_endpointtemplate)[b2c-sip](aortemplate)[whatsapp](sdes_endpointtemplate)
type=endpoint
transport=transport-tls
disallow=all
allow=opus,ulaw,alaw
aors=whatsapp
from_user={wa-business-phone-number}
from_domain={domain-name}
outbound_auth=whatsapp

[whatsapp]
type=aor
contact=sip:wa.meta.vc

[whatsapp]
type=identify
endpoint=whatsapp

[whatsapp]
type=auth
auth_type=digest
password={meta-sip-user-password}
username={wa-business-phone-number}
realm=*

[general];Hostnameor address for the STUN server used for determining the external
; IP address and port an RTP session can be reached at.The port number is; optional.If omitted default value of 3478 will be used.This option is; disabled bydefault.Name resolution occurs at load time,andif DNS is; used, name resolution will occur repeatedly after the TTL expires.;;for example stundaddr=mystun.server.com:3478;
stunaddr=stun.l.google.com:19302

rtpstart=10000
rtpend=60000

Record-Route:<sip:wa.meta.vc;transport=tls;lr>Record-Route:<sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>

<include><extensionname="c2b_calls_sip_ivr"><!--Dial plan is selected if the SIP request is coming from Meta--><conditionfield="${sip_from_host}"expression="^wa.meta.vc$"><!--Verify the IP from where the request is coming, compare the IP with the Meta allowlisted IPs--><actionapplication="check_acl"data="${network_addr} whatsapp_allow normal_clearing"/><!--Enable encrypted media using SDES--><actionapplication="set"data="rtp_secure_media=true"/><actionapplication="answer"/><!--Add silence stream for  1 sec so that the media path is established between whatsapp and freeswitch to avoid audio clipping--><actionapplication="playback"data="silence_stream://1000"/><actionapplication="play_and_get_digits"data="2 5 3 7000 # $${base_dir}/sounds/incoming_welcome.wav  $${base_dir}/sounds/incoming_invalid.wav extension \d+"/><!--While the call is being bridged, play a ringtone for the caller--><actionapplication="set"data="ringback=%(2000, 4000, 440.0, 480.0)"/><!--Offer G711 and Opus for FreeSWITCH-SIP UA leg --><actionapplication="export"data="nolocal:absolute_codec_string=PCMA,PCMU,OPUS@48000h@20i"/><actionapplication="bridge"data="user/${extension}"/><actionapplication="hangup"/></condition></extension><extensionname="b2c_calls_ivr"><conditionfield="destination_number"expression="^b2c-sip$"><!--Enable encrypted media using SDES--><actionapplication="set"data="rtp_secure_media=true"/><actionapplication="answer"/><actionapplication="playback"data="silence_stream://1000"/><actionapplication="set"data="caller_id_check=${caller_id_number}"/><actionapplication="play_and_get_digits"data="2 12 3 20000 # $${base_dir}/sounds/outgoing_welcome.wav $${base_dir}/sounds/outgoing_invalid.wav whatsapp_number \d+"/><actionapplication="log"data="INFO [whatsapp_number] is ${whatsapp_number}"/><!--While the call is being bridged, play a ringtone for the caller--><actionapplication="set"data="ringback=%(2000, 4000, 440.0, 480.0)"/><!--Offer only OPUS--><actionapplication="export"data="nolocal:absolute_codec_string=OPUS@48000h@20i,OPUS@8000h@20i"/><!--Bridge the call by calling META SIP with the WA Number--><actionapplication="bridge"data="sofia/gateway/whatsapp/+${whatsapp_number}"/><actionapplication="hangup"/></condition></extension></include>

<!--Gateway configuration for Meta SIP--><!--replace {phone-number},{meta-sip-password} and {domain-name} before starting FreeSWITCH--><include><gatewayname="whatsapp"><paramname="username"value="{phone-number}"/><paramname="password"value="{meta-sip-password}"/><paramname="register"value="false"/><paramname="realm"value="wa.meta.vc"/><paramname="from-user"value="{phone-number}"/><paramname="from-domain"value="{domain-name}"/></gateway></include>

<!--IP addresses from Meta that are allowed to send SIP requests via the gateway. Keep this up to date--><listname="whatsapp_allow"default="deny"><nodetype="allow"cidr="31.13.24.0/21"/><nodetype="allow"cidr="31.13.64.0/18"/><nodetype="allow"cidr="45.64.40.0/22"/><nodetype="allow"cidr="57.141.0.0/21"/><nodetype="allow"cidr="57.141.8.0/22"/><nodetype="allow"cidr="57.141.12.0/23"/><nodetype="allow"cidr="57.144.0.0/14"/><nodetype="allow"cidr="66.220.144.0/20"/><nodetype="allow"cidr="69.63.176.0/20"/><nodetype="allow"cidr="69.171.224.0/19"/><nodetype="allow"cidr="74.119.76.0/22"/><nodetype="allow"cidr="102.132.96.0/20"/><nodetype="allow"cidr="103.4.96.0/22"/><nodetype="allow"cidr="129.134.0.0/16"/><nodetype="allow"cidr="147.75.208.0/20"/><nodetype="allow"cidr="157.240.0.0/16"/><nodetype="allow"cidr="163.70.128.0/17"/><nodetype="allow"cidr="163.77.128.0/17"/><nodetype="allow"cidr="173.252.64.0/18"/><nodetype="allow"cidr="179.60.192.0/22"/><nodetype="allow"cidr="185.60.216.0/22"/><nodetype="allow"cidr="185.89.216.0/22"/><nodetype="allow"cidr="204.15.20.0/22"/></list>

Add line <X-PRE-PROCESS cmd="set" data="rtp_secure_media=mandatory"/> under <include>Replace<X-PRE-PROCESS cmd="set" data="default_password=1234"/>with(substitute {sip_ua_password}with your password)<X-PRE-PROCESS cmd="set" data="default_password={sip-ua-password}"/>Replace<X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>with(substitute {domain-name}with your FreeSWITCH SIP server DNS)<X-PRE-PROCESS cmd="set" data="domain={domain-name}”/>

Replace
  <X-PRE-PROCESS cmd="stun-set" data="external_sip_ip=stun:stun.freeswitch.org"/>
with (substitute {external-ip} with your FreeSWITCH public ip)
  <X-PRE-PROCESS cmd="set" data="external_sip_ip={external-ip}"/>

Replace
  <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
with (substitute {external-ip} with your FreeSWITCH public ip)
  <X-PRE-PROCESS cmd="stun-set" data="external_rtp_ip={external-ip}"/>

<paramname="sip-trace"value="no"/>

<paramname="sip-trace"value="yes"/>

Replace<param name="sip-trace" value="no"/>with<param name="sip-trace" value="yes"/>Replace<param name="tls" value="$${external_ssl_enable}"/>with<param name="tls" value="true"/>Replace<!--<param name="tls-cert-dir" value=""/>-->with<param name="tls-cert-dir" value="/usr/share/freeswitch/etc/freeswitch/certs"/>

/usr/share/janus/bin/janus  --debug-level=6--libnice-debug=on -S stun.l.google.com:19302--log-file=/var/log/janus.log --config=/usr/share/janus/etc/janus/janus.jcfg

<include><extensionname="c2b_calls_ivr"><conditionfield="destination_number"expression="^c2b-sip$"><actionapplication="set"data="rtp_secure_media=true"/><actionapplication="answer"/><!--Add silence stream for  1 sec so that the media path is established between whatsapp and freeswitch to avoid audio clipping. TODO: Investigate if silence can be removed--><actionapplication="playback"data="silence_stream://1000"/><actionapplication="play_and_get_digits"data="2 5 3 7000 # $${base_dir}/sounds/incoming_welcome.wav  $${base_dir}/sounds/incoming_invalid.wav extension \d+"/><!--While the call is being bridged, play a ringtone for the caller--><actionapplication="set"data="ringback=%(2000, 4000, 440.0, 480.0)"/><!--WA calls bridged via Janus through extension 1000 only support OPUS. However, the callee might be restricted to other codecs for example G722--><!--Therefore , don't restrict to OPUS for C2B calls and offer more codecs to the caller. Transcoding between OPUS and the negotiated codec by the caller--><!--will happen in freeswitch--><actionapplication="export"data="nolocal:absolute_codec_string=PCMA,PCMU,OPUS@48000h@20i,G722"/><actionapplication="bridge"data="user/${extension}"/><actionapplication="hangup"/></condition></extension><extensionname="b2c_calls_ivr"><conditionfield="destination_number"expression="^b2c-sip$"><actionapplication="set"data="rtp_secure_media=true"/><actionapplication="answer"/><actionapplication="playback"data="silence_stream://1000"/><actionapplication="set"data="caller_id_check=${caller_id_number}"/><actionapplication="log"data="INFO [caller id ] is ${caller_id_check}"/><actionapplication="play_and_get_digits"data="2 12 3 20000 # $${base_dir}/sounds/outgoing_welcome.wav $${base_dir}/sounds/outgoing_invalid.wav whatsapp_number \d+"/><actionapplication="log"data="INFO [whatsapp_number] is ${whatsapp_number}"/><!--Add the whatsapp number entered by the user as a custom SIP header, Janus will use this WA user number in API request to Meta--><actionapplication="export"data="sip_h_X-WhatsApp-Number=${whatsapp_number"/><!--While the call is being bridged, play a ringtone for the caller--><actionapplication="set"data="ringback=%(2000, 4000, 440.0, 480.0)"/><!--WA calls bridged via Janus through extension 1000 only support OPUS. However, the caller might be restricted to other codecs for example G722--><!--Therefore , don't restrict to OPUS for B2C calls and let caller select other codecs--><!--However, force transcoding to OPUS by only offering OPUS to Janus--><actionapplication="export"data="nolocal:absolute_codec_string=OPUS@48000h@20i,PCMU,PCMA"/><!--Bridge the call to extension 1000 to which capi-calling is registered via Janus to route calls to WhatsApp--><actionapplication="bridge"data="user/1000"/><actionapplication="hangup"/></condition></extension></include>

<paramname="sip-trace"value="no"/>

<paramname="sip-trace"value="yes"/>

/usr/bin/rtpengine --listen-ng={local-ip}:22222--interface={local-ip}\!{public-ip}-f -E

[handler];Set headers on callee channel
exten => addheader,1,Set(PJSIP_HEADER(add,X-WhatsApp-Number)=${DIGITS})
same => n,Return()[default]
exten => _10XX,1,NoOp()
same => n,Dial(PJSIP/${EXTEN})
same => n,Hangup()

exten => b2c-sip,1,NoOp()
same => n,Read(Digits,outgoing_welcome,0,,5,500)
same => n,Set(GLOBAL(DIGITS)=${Digits});Before starting a business initiated call, add customer WA header to store the WA user number captured from agent entered digits (DTMF)
same => n,Dial(PJSIP/1000,,b(handler^addheader^1))
same => n,Hangup()

exten => c2b-sip,1,NoOp()
same => n,Read(Digits,incoming_welcome,0,,5,500)
same => n,Dial(PJSIP/${Digits})
same => n,Hangup()

[global]
type=global
debug=yes ;Enable/Disable SIP debug logging.Valid options include yes|no[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0;External IP address to usein RTP handling
external_media_address={external-media-address};External address for SIP signalling
external_signaling_address={external-signaling-address};Network to consider local used for NAT purposes
local_net={local-net}[endpointtemplate](!)
type=endpoint
context=default
disallow=all
allow=OPUS,g722,g729,ulaw
;No audio if direct_media isset to yes
direct_media=no
rtp_symmetric=yes
use_avpf=yes
media_encryption=sdes
media_use_received_transport=yes
rtcp_mux=yes

[authtemplate](!)
type=auth
auth_type=userpass
password={sip-ua-password}[aortemplate](!)
type=aor
max_contacts=1
remove_existing=yes

[1000](endpointtemplate)
disallow=all
;extension 1000is used byRtpEngine to bridge whatsapp calls
;WhatsApp only support OPUS
allow=OPUS
auth=1000_auth
aors=1000[1000_auth](authtemplate)
username=1000[1000](aortemplate)[1001](endpointtemplate)
auth=1001_auth
aors=1001[1001_auth](authtemplate)
username=1001[1001](aortemplate)[1002](endpointtemplate)
auth=1002_auth
aors=1002[1002_auth](authtemplate)
username=1002[1002](aortemplate)[1003](endpointtemplate)
auth=1003_auth
aors=1003[1003_auth](authtemplate)
username=1003[1003](aortemplate)[1004](endpointtemplate)
auth=1004_auth
aors=1004[1004_auth](authtemplate)
username=1004[1004](aortemplate)[1005](endpointtemplate)
auth=1005_auth
aors=1005[1005_auth](authtemplate)
username=1005[1005](aortemplate)

...;Rest of content omitted for brevity

[1000](endpointtemplate)
disallow=all
;extension 1000is used by SIP UA of the integration module to bridge WhatsApp calls
;WhatsApp only support OPUS
allow=OPUS
auth=1000_auth
aors=1000
dtls_auto_generate_cert=yes
webrtc=yes
;Setting webrtc=yes is a shortcut for setting the following options:; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes