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SIP INVITE explained

Read the request that starts or changes a session and identify the point where call setup fails.

Reviewed 2026-10-13Foundational

A request for a session

An initial INVITE asks another participant to establish a session. Its request URI, routing information and offered media parameters help determine where it goes and what the receiver can accept. An INVITE within an existing dialog can modify an established session, such as a media change.

Do not treat every INVITE in a trace as a new customer call. Retransmissions and in-dialog requests have different roles. Use transaction and dialog identifiers to understand which exchange you are reading.

Compare the offer with the answer
An offer proposes media addresses, codecs and parameters. The answer selects compatible media details or the request is rejected. RTP must then follow the negotiated path.

A negotiated codec does not prove packet delivery. Check both session description and actual media packets.

Follow responses in order

Provisional responses indicate progress without final success. A final success response to INVITE is followed by ACK. A rejection should be interpreted on the leg that produced it, with any useful response details and associated SDP.

A PBX may originate a separate INVITE toward a carrier, using a different Call-ID. A failure on that leg can be translated into another response toward the phone. Follow both sides before attributing the failure to the handset.

Separate destination and media rejection

A request aimed at an unavailable destination raises routing and reachability questions. A request rejected because offered media cannot be accepted raises codec, SDP or encryption-policy questions. Changing a password will not repair every unsuccessful INVITE.

Capture a bounded example

Record dialed digits before normalization, the resulting destination, the selected route and the response source. Redact private identities before sharing. After a repair, test the original call and a related route, because a broad pattern change can fix one destination while breaking another.

Keep the answered 200 response distinct from earlier messages

Asterisk response-header functions expose fields from the 200 response to an outbound INVITE. The focused guide does not generalize that result to provisional responses, failures or later in-dialog requests.

Treat an Asterisk header list as one INVITE view

Use the PJSIP inbound-header inventory guide to distinguish unique names from repeated rows, values and other SIP messages.

Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

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Guide · Current Asterisk PJSIP_RESPONSE_HEADER, PJSIP_RESPONSE_HEADERS, pre-bridge-handler and Dial documentation plus the project source implementation reviewed October 13, 2026; the functions are documented from Asterisk 16.28.0, 18.14.0 and 19.6.0, and installed branch, maintenance release, channel topology, peer response, header set, dialplan and consumer policy remain deployment specific; no PBX, module, dialplan, channel, request, response, header, endpoint, provider, bridge or call was inspected or tested

Asterisk PJSIP response headers: channel, answer and pre-bridge timing

Read named headers from the answered outbound INVITE response on the callee PJSIP channel without confusing request headers, provisional responses or caller state.

Read guide →
Troubleshooting · Current Asterisk PJSIP_RESPONSE_HEADER, PJSIP_RESPONSE_HEADERS, pre-bridge-handler and Dial documentation plus the project source implementation reviewed October 13, 2026; the functions are documented from Asterisk 16.28.0, 18.14.0 and 19.6.0, and installed branch, maintenance release, channel topology, peer response, header set, dialplan and consumer policy remain deployment specific; no PBX, module, dialplan, channel, request, response, header, endpoint, provider, bridge or call was inspected or tested

Asterisk PJSIP response header is empty or from the wrong response

Trace the exact answered outbound INVITE, PJSIP callee channel, pre-bridge handler, header match and instance before changing request headers or copying caller state.

Read guide →
Guide · Current Asterisk PJSIP_HEADERS and PJSIP_HEADER documentation, Asterisk project source and RFC 3261 reviewed October 13, 2026; PJSIP_HEADERS is documented from Asterisk 16.20.0 and 18.6.0, and installed branch, maintenance release, module, channel topology, request, header set, endpoint policy and dialplan remain deployment specific; no PBX, module, dialplan, channel, request, header, endpoint, provider, packet or call was inspected or tested

Asterisk PJSIP inbound header inventory: names, prefixes and instances

Enumerate unique header names from the current INVITE, then read the intended value and repeated instance without treating a name list as a packet capture.

Read guide →
Troubleshooting · Current Asterisk PJSIP_HEADERS and PJSIP_HEADER documentation, Asterisk project source and RFC 3261 reviewed October 13, 2026; PJSIP_HEADERS is documented from Asterisk 16.20.0 and 18.6.0, and installed branch, maintenance release, module, channel topology, request, header set, endpoint policy and dialplan remain deployment specific; no PBX, module, dialplan, channel, request, header, endpoint, provider, packet or call was inspected or tested

Asterisk PJSIP header list is empty, incomplete or misleading

Trace function availability, INVITE scope, current channel, prefix, unique-name behavior and repeated instances before changing endpoint trust or outbound headers.

Read guide →