Classify one recipient before changing the group
Use one authorized test page and record the group extension, calling extension, intended member, exact phone model, account, firmware, provisioning owner, timestamp and visible result. Separate no call at the endpoint, ordinary ringing, one ring then answer, immediate answer with silence, answer with one-way audio and the expected speaker page. Those outcomes fail at different stages.
Compare one working and one failing recipient in the same small test group when possible. Confirm whether both were idle, busy, in DND or already handling another page. Sangoma's version-13-labelled guide makes Skip, Force and Whisper busy handling explicit and notes that phones can react differently when already in use. Do not generalize an idle-phone result to a busy endpoint.
Prove group membership and the generated dial path
In the Paging and Intercom module, preserve the effective page extension, selected devices, announcement, busy-extension choice, duplex setting and whether Page Pro or multicast is involved. Submit and Apply Config through the supported GUI, then inspect the generated dialplan and active channels without editing generated output. The current FreePBX 17 paging source retains its own auto-answer mapping data, while the public administration guide is explicitly labelled for version 13; record the installed module version and effective output instead of assuming the older screen exactly matches.
If the failing extension never receives an outbound INVITE, follow page-group membership, endpoint reachability, busy handling and generated dialplan first. A local phone option cannot repair a call that the PBX did not attempt. Keep ordinary SIP paging, Page Pro RTP multicast and an external paging adapter as separate call paths.
Capture the exact outbound INVITE and auto-answer signal
Use a short, authorized PJSIP protocol capture and correlate the outbound INVITE by Call-ID, destination and timestamp. Preserve every Alert-Info or Call-Info value and compare it with the working endpoint; do not normalize punctuation, brackets or case in the evidence. Asterisk's PJSIP_HEADER function can add an outbound header only on the outbound channel, and its pre-dial handler pattern is different from changing the caller channel.
Sangoma phones document model-specific Alert-Info mappings such as normal, ring-answer, answer and visual, and current firmware also documents separate volume and interrupt headers. Those values are evidence for Sangoma phones, not universal SIP semantics for Yealink, Poly, Grandstream, analog adapters or overhead paging devices. Do not add several vendor values at once: the endpoint must receive the exact signal its own current documentation and provisioning expect.
Follow the endpoint's local mapping and provisioning owner
On the failing device, inspect the active account, auto-answer or intercom permission, accepted header pattern, speaker or headset target, privacy controls, DND and busy-call policy. Then identify whether local UI, End Point Manager, vendor redirect service, configuration file or another platform owns that value. A correct screen value that the next provision overwrites is not a durable repair.
Compare the downloaded active configuration and firmware-specific behavior with the working phone of the same model before copying across models. Rebuild or reprovision only the smallest affected template, preserve the prior value and define a rollback. A PBX reload, factory reset or firmware change without the received-header and endpoint-mapping evidence can erase useful state without addressing the cause.
Separate auto-answer from audio, volume and interruption
If the phone answers but the page is silent, auto-answer succeeded. Continue with negotiated SDP, RTP addresses, codec, media direction, firewall path, speaker selection, mute state and announcement timing. One-way audio and no-audio guides address that later stage; changing the auto-answer header cannot repair missing RTP.
Likewise, a low speaker level is not a failed page. Sangoma documents a separate forced paging-volume control, and its busy-interrupt behavior can hold an existing call on supported phone firmware. Treat volume, barge, privacy, current-call interruption and acoustic feedback as distinct acceptance decisions. Do not enable interruption broadly merely to make an idle test auto-answer.
Retest the permitted states and retain safety boundaries
After correcting the demonstrated owner, retest the same endpoint idle, busy, in DND if policy permits, after reprovision, after phone restart and after an approved firmware or module update. Verify the intended answer mode, announcement, one-way or duplex choice, both media directions where applicable, hangup and restoration of any interrupted call. Expand membership only after the small result is repeatable.
Document who may originate pages, which rooms receive them, the expected busy behavior, volume limits, privacy notice, fallback and change owner. Ordinary FreePBX paging and endpoint auto-answer are not represented as a supervised life-safety or certified emergency-notification system. No PBX, page group, endpoint, header, firmware, audio path or live page was tested for this guide.
core show version
module show like res_pjsip
dialplan show <page-extension>@from-internal
pjsip show endpoint <endpoint>
pjsip set logger on
# Place one authorized test page; retain only the affected Call-ID and outbound INVITE.
pjsip set logger offSources & applicability
Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.
- Sangoma: Paging and Intercom user guide ↗
- FreePBX paging module, release 17.0 ↗
- Sangoma phones: Ringing Types and Intercom ↗
- Sangoma phones: Force Paging Volume ↗
- Asterisk PJSIP_HEADER function ↗
- Asterisk PJSIP configuration logger ↗
Examples require adaptation to your topology. No live PBX or hardware testing is claimed.