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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.

SIP realtime account settings with fictional chan_sip trunk values.
SIP realtime account settings with fictional chan_sip trunk values.

Common Realtime Fields

FieldApplies toPurposeFictional example
Use Realtime AccountSIP, PJSIP, IAX2Creates and maintains the Asterisk realtime peer, endpoint, AOR, auth, or IAX row for the provider.checked
TransportSIP, PJSIPSIP 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, PJSIPEnables 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
HostSIP, PJSIP, IAX2Remote provider address. Use an IP address when inbound realtime matching must identify the carrier by source IP.203.0.113.20
PortSIP, PJSIP, IAX2Remote signaling port. Defaults are commonly 5060 for SIP/PJSIP and 4569 for IAX2.5060
Outbound ProxySIP, PJSIPProxy URI or host used for outbound signaling when required by the carrier.sip-proxy.example.invalid
UsernameSIP, PJSIP, IAX2Authentication username sent to the provider. PJSIP stores this in ps_auths.username, chan_sip as defaultuser.docs_sip_user
PasswordSIP, PJSIP, IAX2Authentication secret for the provider. Use a unique carrier-supplied or generated secret.FictionalSecret-2026
QualifySIP, PJSIPEnables SIP OPTIONS reachability checks and controls timeout behavior.Yes
Qualify FrequencySIP, PJSIPInterval in seconds between provider reachability checks.60 secs
CodecsSIP, PJSIP, IAX2Allowed audio codecs. MiRTA PBX stores disallow=all and the selected allow list.alaw, ulaw, g722
Session ExpiresSIP, PJSIPMaximum SIP session refresh interval in seconds.1800
DTMF ModeSIP, PJSIPDTMF signaling method. PJSIP maps RFC 2833 to rfc4733.RFC 2833/RFC 4733
RTP KeepaliveSIP, PJSIPSends RTP keepalive packets to help keep NAT bindings and media paths active.5 seconds

SIP-Only Fields

FieldApplies toPurposeFictional example
From UserSIP onlychan_sip fromuser value used in the SIP From header.+15551230000
From DomainSIP onlychan_sip fromdomain value used when the provider requires a specific domain in the From header.sip.example.invalid
Send RPIDSIP onlyControls whether caller identity is sent using Remote-Party-ID, P-Asserted-Identity, or both.PAI
Trust RPIDSIP onlyControls whether inbound Remote-Party-ID or P-Asserted-Identity data is trusted for caller identity.PAI
Trust Outbound for CallerID PresentationSIP onlyControls how prohibited caller presentation is handled in RPID/PAI headers.Legacy
Session TimersSIP onlychan_sip session timer policy: accept remote requests, originate timers, or refuse timers.Accept
Progress inbandSIP onlyControls whether Asterisk generates in-band progress audio before answer.Yes
NATSIP onlychan_sip NAT behavior. force_rport,comedia is common for symmetric RTP and response routing.force_rport,comedia
Call response timer (ms) - T1SIP onlyMilliseconds to wait for response to SIP messages.500
Call setup timer (ms) - BSIP onlyMaximum milliseconds to wait for call setup before autocongestion when no provisional response arrives.5000
Can ReinviteSIP onlyAllows Asterisk to attempt direct media re-invites. Use No when recording, NAT, or media control should keep Asterisk in the RTP path.No
Allow TransferSIP onlychan_sip allowtransfer. When enabled, the peer may request SIP REFER transfers.No
InsecureSIP onlychan_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

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