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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.

Reviewed 2026-10-13Current 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 testedAdvancedDocumentation based · not lab tested

Freeze the disputed answer and expected field

Record timestamp and timezone, caller and callee channel names, unique and linked identifiers, endpoint or trunk, Call-ID, INVITE CSeq, answering destination, final status, header name, requested instance, returned value and the dialplan location that consumed it. Preserve a sanitized working exchange when one exists.

Classify the first symptom: unknown function, handler never ran, no 200 answer, empty value, wrong field instance, wrong outbound leg, provisional or error response expected, value lost after return, duplicate name collapsed by enumeration or consumer rejected a correct value.

Confirm version, module and message scope

Verify the running Asterisk branch and maintenance release meet the documented 16.28.0, 18.14.0 or 19.6.0 floors and that res_pjsip_header_funcs is loaded. Then prove the expected field belongs to the 200 response to an outbound INVITE.

A header seen in an inbound request, outbound INVITE, provisional response, final failure, REGISTER or later in-dialog request belongs to another interface or observation path. Do not diagnose channel inheritance when the expected message is outside the documented response scope.

Prove the handler ran on the answered callee

Inspect the active dialplan and a bounded call trace. Confirm Dial() used uppercase U, the routine started at s,1, executed after this callee answered and returned before bridging. Parallel dialing, queues, Local channels, forwarding and transfers can create several outbound channels; the linked identifier does not merge their response headers.

Lowercase b runs before dialing and is appropriate for request preparation, not for reading a future 200 response. Uppercase B runs on the caller. Moving the same expression among these handlers without mapping channel lifetime can produce another empty result.

Enumerate names and test repeated instances

Use PJSIP_RESPONSE_HEADERS() with the smallest useful prefix to establish which names the function exposes on the selected answered channel. Then read the exact name and explicit instance number with PJSIP_RESPONSE_HEADER(). Compare both to the sanitized 200 response for that Call-ID and CSeq.

Enumeration returns unique names, not a value list or instance count. If multiple Via, Record-Route or custom fields exist, preserve their wire order and test the intended number rather than treating the first value as the complete result.

Trace the value across the channel boundary

If the callee handler reads the correct value but later caller dialplan sees nothing, inspect how the variable is named and propagated. A callee-local variable does not become caller state merely because the channels share one call. Copy only the required value through the deployment-approved inheritance or shared-state mechanism and define when it is cleared.

Keep this handoff separate from SIP proof. A variable that survives does not establish that the peer authenticated, honored a route, changed connected identity or maintained the same value in later messages.

Correct one owner and repeat the same call path

Correct only the demonstrated boundary: supported release, loaded module, 200-response expectation, selected callee, uppercase U placement, Gosub entry or return, exact field name, instance number or post-handler variable transfer. Repeat the same authorized route and retain the response and consumer outcome independently.

Check a no-answer or rejected call and an unrelated route so missing values remain distinguishable from wrong values. Remove temporary protocol logging and protect identities and route data. No PBX, module, dialplan, channel, request, response, header, endpoint, bridge or call was inspected or tested for this diagnostic.

Sources & applicability

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

Examples require adaptation to your topology. No live PBX or hardware testing is claimed.

Continue your research

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.

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Guide · Current Asterisk PJSIP header, inheritable-header, header-list, header-parameter, response-header and pre-dial-handler documentation reviewed October 8, 2026; PJSIP_INHERITABLE_HEADER is documented from 20.19.0, 22.9.0 and 23.3.0, and no PBX, dialplan, channel, header, provider, endpoint, request, response, registration, authentication exchange, availability, compatibility or call was inspected or tested

Asterisk PJSIP outbound headers: channel ownership, inheritance and proof

Put a required SIP header on the channel that will send it, distinguish direct from inherited headers and verify the emitted request.

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Guide · Asterisk subroutine documentation and Asterisk 22 application references reviewed October 12, 2026; exact application syntax, channel behavior and available functions remain dependent on the installed Asterisk branch and modules, and no PBX, dialplan, channel, stack frame, handler, endpoint, bridge, header, call or media path was inspected or tested

Asterisk Gosub subroutines: stack, scope and channel ownership

Use Gosub and Return while keeping stack frames, arguments, return values and caller- or callee-channel execution independently observable.

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Diagnostic · Current Asterisk PJSIP header, inheritable-header, header-list, header-parameter, response-header and pre-dial-handler documentation reviewed October 8, 2026; PJSIP_INHERITABLE_HEADER is documented from 20.19.0, 22.9.0 and 23.3.0, and no PBX, dialplan, channel, header, provider, endpoint, request, response, registration, authentication exchange, availability, compatibility or call was inspected or tested

Asterisk PJSIP required outbound header is missing or duplicated

Trace one outbound request through the caller, callee, handler and header store before changing a trunk or copying inbound data.

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Troubleshooting · Asterisk subroutine documentation and Asterisk 22 application references reviewed October 12, 2026; exact application syntax, channel behavior and available functions remain dependent on the installed Asterisk branch and modules, and no PBX, dialplan, channel, stack frame, handler, endpoint, bridge, header, call or media path was inspected or tested

Asterisk Gosub does not return or runs on the wrong channel

Trace one Gosub from invocation and stack frame through arguments, branch, Return, resumed priority and the caller or callee channel that owns its effects.

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SIP response reference

SIP 200 OK

The request succeeded under the responding system’s processing. Its practical meaning depends on the request method identified by CSeq.

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Guide · Wireshark 4.x / current upstream tool documentation; inspect installed versions

VoIP packet-capture methodology

Select interfaces, time windows and paired capture points that distinguish network loss from missing evidence.

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Guide · Current Asterisk logging, runtime command, logger.conf, debug-collection, call-identifier and console-verbosity documentation reviewed September 30, 2026; no process, logger, trace, storage path or call tested

Asterisk logging without permanent debug overload

Map configured destinations, console levels, call identifiers and a bounded protocol trace without leaving sensitive debug collection active.

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