SIP Provider
Use SIP technology for chan_sip provider trunks. SIP providers can create and maintain a global sipfriends realtime row when Use Realtime Account is enabled.
Common Realtime Fields
| Field | Applies to | Purpose | Fictional example |
|---|
| Use Realtime Account | SIP, PJSIP, IAX2 | Creates and maintains the Asterisk realtime peer, endpoint, AOR, auth, or IAX row for the provider. | checked |
| Transport | SIP, PJSIP | SIP signaling transport. For PJSIP this maps to transport objects such as transport-udp, transport-tcp, or transport-tls. | UDP or TLS |
| RTP Encryption (SRTP) | SIP, PJSIP | Enables SRTP media encryption when supported by the provider. PJSIP stores this as media_encryption=sdes. | No for standard UDP SIP, Yes for TLS/SRTP trunks |
| Host | SIP, PJSIP, IAX2 | Remote provider address. Use an IP address when inbound realtime matching must identify the carrier by source IP. | 203.0.113.20 |
| Port | SIP, PJSIP, IAX2 | Remote signaling port. Defaults are commonly 5060 for SIP/PJSIP and 4569 for IAX2. | 5060 |
| Outbound Proxy | SIP, PJSIP | Proxy URI or host used for outbound signaling when required by the carrier. | sip-proxy.example.invalid |
| Username | SIP, PJSIP, IAX2 | Authentication username sent to the provider. PJSIP stores this in ps_auths.username, chan_sip as defaultuser. | docs_sip_user |
| Password | SIP, PJSIP, IAX2 | Authentication secret for the provider. Use a unique carrier-supplied or generated secret. | FictionalSecret-2026 |
| Qualify | SIP, PJSIP | Enables SIP OPTIONS reachability checks and controls timeout behavior. | Yes |
| Qualify Frequency | SIP, PJSIP | Interval in seconds between provider reachability checks. | 60 secs |
| Codecs | SIP, PJSIP, IAX2 | Allowed audio codecs. MiRTA PBX stores disallow=all and the selected allow list. | alaw, ulaw, g722 |
| Session Expires | SIP, PJSIP | Maximum SIP session refresh interval in seconds. | 1800 |
| DTMF Mode | SIP, PJSIP | DTMF signaling method. PJSIP maps RFC 2833 to rfc4733. | RFC 2833/RFC 4733 |
| RTP Keepalive | SIP, PJSIP | Sends RTP keepalive packets to help keep NAT bindings and media paths active. | 5 seconds |
SIP-Only Fields
| Field | Applies to | Purpose | Fictional example |
|---|
| From User | SIP only | chan_sip fromuser value used in the SIP From header. | +15551230000 |
| From Domain | SIP only | chan_sip fromdomain value used when the provider requires a specific domain in the From header. | sip.example.invalid |
| Send RPID | SIP only | Controls whether caller identity is sent using Remote-Party-ID, P-Asserted-Identity, or both. | PAI |
| Trust RPID | SIP only | Controls whether inbound Remote-Party-ID or P-Asserted-Identity data is trusted for caller identity. | PAI |
| Trust Outbound for CallerID Presentation | SIP only | Controls how prohibited caller presentation is handled in RPID/PAI headers. | Legacy |
| Session Timers | SIP only | chan_sip session timer policy: accept remote requests, originate timers, or refuse timers. | Accept |
| Progress inband | SIP only | Controls whether Asterisk generates in-band progress audio before answer. | Yes |
| NAT | SIP only | chan_sip NAT behavior. force_rport,comedia is common for symmetric RTP and response routing. | force_rport,comedia |
| Call response timer (ms) - T1 | SIP only | Milliseconds to wait for response to SIP messages. | 500 |
| Call setup timer (ms) - B | SIP only | Maximum milliseconds to wait for call setup before autocongestion when no provisional response arrives. | 5000 |
| Can Reinvite | SIP only | Allows Asterisk to attempt direct media re-invites. Use No when recording, NAT, or media control should keep Asterisk in the RTP path. | No |
| Allow Transfer | SIP only | chan_sip allowtransfer. When enabled, the peer may request SIP REFER transfers. | No |
| Insecure | SIP only | chan_sip authentication matching option. Port, Invite is common for IP-authenticated trunks. | Port, Invite |
Fictional SIP Example
Name: Docs Telecom SIP Trunk
Peer Name: docs_telecom_sip
Technology: SIP
Host: 203.0.113.20
Port: 5060
Username: docs_sip_user
From User: +15551230000
From Domain: sip.example.invalid
NAT: force_rport,comedia
Insecure: Port, Invite
