Asterisk is an open-source project. The company Digium provides the infrastructure to contribute, although it competes with products like Switchvox. In Asterisk, that company considers everything, even non-functional features like software usability, software security, and documentation, as best-effort. A graphical frontend exists: Sangoma FreePBX. Asterisk combines several technologies via a single core engine. To access that core, every technology with its own protocol gets its own channel driver. VoIP/SIP got two channels drivers, the older chan_sip and the newer chan_pjsip. However, if you want the latest, consider SignalWire FreeSWITCH …
Although Asterisk is a Back-to-Back User Agent (B2BUA), the example below shows just one back = client side, the external registration at a provider. The sever side gets activated via a TLS transport with cert_file and priv_key_file.
| pjsip.conf: | [global] type = global keep_alive_interval = 0 ; seconds, TCP-PSH as keep-alive mechanism [my_registration] type = registration outbound_auth = my_auth client_uri = sip:user@host contact_user = user server_uri = sip:host\;transport=tls line = yes ; required for "endpoint" endpoint = my_endpoint ; link required for "qualify_frequency" [my_transport_template](!) type = transport protocol = tls ca_list_path = /etc/ssl/certs/ verify_server = yes method = sslv23 cos = 3 tos = cs5 [my_transport_v4](my_transport_template) bind = 0.0.0.0 [my_transport_v6](my_transport_template) bind = [::] [my_auth] type = auth password = password username = user [my_endpoint] type = endpoint from_domain = host from_user = user media_encryption = sdes media_encryption_optimistic = yes aors = my_aor ; link required for "qualify_frequency" cos_audio = 5 cos_video = 4 tos_audio = ef tos_video = af41 [my_aor] type = aor contact = sip:user@host qualify_frequency = 0 ; seconds, SIP-OPTION as keep-alive mechanism which is RTP/AVP with crypto In the example above, you have to replace user (five times), password, and host (four times). And yes, all those settings are for one client connection. |
| DNS-NAPTR: | missing, see ASTERISK-29111 |
| IP Port Source: | not the actual port but the port of bind (default 5061) in the SIP headers Via and Contact, see ASTERISK-29190 Mitigation: unknown; service has to ignore it and re-use the TCP based connection instead |
| Responsible Disclosure: | via E-mail |
| Firmware Update: | missing Automation Newsletter via mailing list |