Примеры интеграции | Документация для разработчиков

[ 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 () ; Расширение для  бизнес  - звонков B2C через шлюз Meta  SIP  exten = >  b2c - sip , 1 , NoOp ( ) 
  same  = >  n , Read ( Digits , outgoing_welcome , 0 , 5 , 500 ) 
  same  = >  n , Dial ( PJSIP / whatsapp / sip : $ { Digits}@wa.meta.vc ) ; Расширение для обработки входящих запросов  приглашения  от шлюза Meta SIP  на  <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 } ;​​​​ ​ ​​​​​​​ Сеть,  которую следует считать  локальной,  используется  для  целей NAT 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 ; Эта  конечная точка сопоставляется с IVR  для  вызовов C2B
 [ c2b - sip ]( sdes_endpointtemplate ) [ c2b - sip ]( aortemplate ) [ c2b - sip ] 
type = identify endpoint = c2b - sip match_header = X - FB - External - Domain :  wa . meta . vc

 ; специальная конечная точка  для  интеграции
 с Meta  SIP  Gateway ; Эта  конечная точка соответствует IVR  для  звонков B2C
 [ 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 =*

[ общие ] ; 

Имя хоста или  адрес   STUN-сервера, используемого  для определения внешнего IP - 
адреса  и
 порта ,  по  которому можно получить  доступ к RTP-сессии . Номер порта  является  необязательным  . Если он  опущен  ,  будет  использоваться  значение  по умолчанию 3478. Эта  опция  отключена по умолчанию .  Разрешение  имен происходит во  время загрузки , и если используется  DNS  , разрешение имен будет происходить многократно после истечения  TTL . ; ; например  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--><conditionполе="${sip_from_host}"выражение="^wa.meta.vc$"><!--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>

Добавьте  строку  < X - PRE - PROCESS cmd = "set"  data = "rtp_secure_media=mandatory" />  под  Замените < X - PRE - PROCESS cmd = "set"  data = "default_password=1234" /> на ( замените  { sip_ua_password } на  ваш пароль ) < X - PRE - PROCESS cmd = "set"  data = "default_password={sip-ua-password}" /> Замените < X - PRE - PROCESS cmd = "set"  data = "domain=$${local_ip_v4}" /> на ( замените  { domain - name } на  DNS  вашего  SIP-сервера FreeSWITCH ) < X - PRE - PROCESS cmd = "set"  data = "domain={domain-name}"/> Замените  stun - set " data=" external_sip_ip = stun : stun . freeswitch . org "/> с (замените {external-ip} на ваш публичный IP-адрес FreeSWITCH)  set " data=" external_sip_ip ={ external - ip } "/> Замените  stun - set " data=" external_rtp_ip = stun : stun . freeswitch . org "/> на (замените {external-ip} на ваш публичный IP-адрес FreeSWITCH)  stun - set " data=" external_rtp_ip ={ external - ip } "/><include>

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

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

Замените<param name="sip-trace" value="no"/>на<param name="sip-trace" value="yes"/>Замените<param name="tls" value="$${external_ssl_enable}"/>на<param name="tls" value="true"/>Замените<!--<<param name="tls-cert-dir" value=""/>-->с<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 ] ; Установить  заголовки для канала вызываемого абонента 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 }) ; Перед началом бизнес -  звонка  добавьте заголовок WA клиента для хранения номера пользователя WA, полученного  из  цифр, введенных агентом  ( 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  ; Включить / отключить  отладку SIP . Допустимые  параметры: yes | no [transport-tcp] type = transport protocol = tcp bind = 0.0.0.0; Внешний IP - адрес для обработки RTP 
external_media_address = { external - media - address }  ;  Внешний адрес  для сигнализации SIP external_signaling_address = { external - signaling -  address  }  ; Сеть , которую следует считать локальной для целей NAT local_net  =   {  local -  net } [ endpointtemplate ] ( ! ) type 
= endpoint context = default 
disallow = all allow = OPUS , g722 , g729 , ulaw
 ; Нет  звука  , если  direct_media  установлено  на 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
 ; расширение  1000 используется RtpEngine  для  организации  звонков WhatsApp
 ; WhatsApp  поддерживает только 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 )

... ; Остальное  содержимое опущено  для  краткости

 [ 1000 ]( endpointtemplate ) 
disallow = all
 ; расширение  1000 используется SIP UA модуля  интеграции   для организации   звонков
 WhatsApp ; WhatsApp  поддерживает   только OPUS allow = OPUS auth = 1000 _auth aors = 1000 
dtls_auto_generate_cert = yes webrtc = yes
 ; Установка  webrtc = yes  — это  сокращение  для  установки следующих параметров : ;  use_avpf = yes
 ;  media_encryption = dtls
 ;  dtls_verify = fingerprint
 ;  dtls_setup = actpass
 ;  ice_support = yes
 ;  media_use_received_transport = yes
 ;  rtcp_mux = yes