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Voice/Video/Message calling via SIP URI over Internet

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Set up your own SIP address, place Audio/Video/SMS calls as sending mail via email address. e.g. sip:[email protected]

#Install and clean up default config files.

  1. apt install asterisk asterisk-opus (asterisk-opus is optional to enable opus codes)
  2. mv /etc/asterisk /etc/asterisk.sample #Do NOT use the default configs.
  3. cd /etc
  4. git clone https://github.com/basncy/my-sip-address.git asterisk
  5. chown -R asterisk:asterisk asterik #keep the owner same as default.
  6. systemctl restart asterisk

#Minimal settings

  1. on router, do port mapping. udp 6060 defined in pjsip.conf tcp 6060 defined in pjsip.conf udp 8000-9000 defined in rtp.conf
  2. edit pjsip.conf put your LAN into localnet put your public ip address or domain name into external_signaling_address

#(Option) Advanced settings.

  1. Keep your DDNS A recored updated.
  2. Add SRV DNS record if listen on none standard port: _sip._udp.sip.ddns.org 0 0 6060 sip.ddns.org
  3. Add SRV DNS record if listen on none standard port: _sip._tcp.sip.ddns.org 0 0 6060 sip.ddns.org

#Client.

  1. linphone is open source with full features, download from linphone.org

#DEBUG asterisk -rvvvv pjsip set logger host 192.168.0.1 rtp set debug on

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Voice/Video/Message calling via SIP URI over Internet

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