> For clean Markdown of any page, append .md to the page URL.
> For a complete documentation index, see https://docs.jambonz.org/changelog/2026/10/4/llms.txt.
> For AI client integration (Claude Code, Cursor, etc.), connect to the MCP server at https://docs.jambonz.org/_mcp/server.

#### 11.1.6

jambonz Commercial 11.1.6

**11.1.6** adds alert notifications, so alerts reach you as they happen, and lets
customers opt out of support audio capture. It adds Speechify and KugelAudio
text-to-speech, Zoom speech-to-text, ElevenLabs `eleven_v3`/`eleven_v4` streaming
and AssemblyAI Universal-3.6 Pro. The API server now rate-limits per account and
per caller. On AWS, scale-in drains SIP, RTP and feature servers by reading the
instance metadata service instead of subscribing to SNS. The release also fixes
a set of [agent](/verbs/verbs/agent), answering machine detection and
[gather](/verbs/verbs/gather) issues.

#### New Features & Improvements

1. **Alert notifications** — every alert on an account's Alerts page can also be
   pushed, as it happens, to up to five destinations: a JSON webhook, Slack,
   PagerDuty or email. Add them in the account's Settings → Notifications, and
   send a test to any one destination. An account is notified about the same
   type of alert at most once an hour. The same destinations can be managed
   through the API (`/Accounts/{AccountSid}/AlertHooks`).
2. **Support audio capture opt-out** — jambonz support keeps SIP and RTP from
   calls for a few days to troubleshoot call quality. An account can now turn
   this off in Settings → Privacy, and an organization can allow it for all
   accounts, disable it for all accounts, or leave it to each account. Recent
   Calls SIP traces are not affected.
3. **Speechify text-to-speech** — `speechify` is available as a TTS vendor, with
   speech credentials in the portal and API.
4. **KugelAudio text-to-speech** — `kugelaudio` is available as a TTS vendor,
   including streaming TTS.
5. **Zoom Scribe speech-to-text** — Zoom is available as an STT vendor.
6. **ElevenLabs `eleven_v3` and `eleven_v4`** — streaming TTS with these models,
   including `eleven_v3_conversational` and its audio tags (such as `[laughs]`),
   uses ElevenLabs' Text to Dialogue stream. The synthesizer language is now sent
   to ElevenLabs instead of being detected on each turn, and the models are
   listed in the portal.
7. **AssemblyAI Universal-3.6 Pro** — new AssemblyAI credentials default to
   `universal-3-6-pro`, and `assemblyAiOptions` accepts `languageCodes` and
   `voiceFocus`. Newer models are now sent 16 kHz audio instead of 8 kHz.
8. **OpenAI GPT-Live** — the GPT-Live speech-to-speech integration uses OpenAI's
   GA protocol.
9. **Per-account API rate limits** — the API server counts requests per account
   for call creation (4000/min) and call status reads (600/min), per caller
   address for everything else, and per address for sign-in and sign-up. Counts
   are shared by all API server workers, and a client over a limit receives
   `429`. Every limit can be changed with an environment variable.
10. **SRTP for TLS-registered devices** — calls to a SIP client that registered
    over TLS are offered SDES-SRTP, so encrypted signaling gets encrypted media.
    Set `JAMBONES_DISABLE_SRTP_FOR_TLS` to keep plain RTP. (jambonz.cloud keeps
    plain RTP for now.)
11. **AWS scale-in without SNS** — SIP servers, RTP servers and feature servers
    drain on scale-in by polling the instance metadata service for their target
    lifecycle state. They no longer subscribe to an SNS topic or open a port for
    it. When a SIP server holding the carrier registrations scales in, it hands
    them to another SIP server before it stops. Registrations now use a Call-ID
    per SIP server, so a carrier sees a move between servers as a new
    registration.
12. **Enterprise user limits** — an organization's user limit can be raised per
    organization, and the portal's Users view follows the limit the API reports.
13. **SIPREC application in the new portal** — the account setting for the
    application that handles SIPREC calls is back in the new portal.
14. **Self-hosted updater** — the update client upgrades itself when a release
    requires it and can apply several releases in one run.
15. **Security hardening** — service-provider users can no longer write
    operator-only organization or account fields; carrier changes by non-admin
    users stay within their own organization or account; and calls from a
    registration trunk's address now match only carriers that belong to the
    called SIP realm's account.
16. **Fewer caller details in logs** — caller transcripts and DTMF digits are
    logged at `debug`, not `info`.

#### Bug Fixes

* **Agent verb** — the caller's turn ends normally when Krisp is used only to
  predict interruptions; speech that arrives while the assistant is talking is
  held for the next turn instead of being dropped; interrupt prediction only
  interrupts the assistant while it is speaking; and VAD is used only for turn
  timing when the STT vendor reports the start of speech itself.
* **Answering machine detection** — empty final transcripts no longer count as a
  human, Deepgram endpointing defaults to 150 ms during detection, and built-in
  voicemail hints are used when none are configured.
* **Gather with Deepgram** — an interim word that Deepgram later drops no longer
  delays every following `UtteranceEnd`, and a deferred `UtteranceEnd` returns
  the transcript as soon as it is satisfied.
* **Soniox** — v5 sub-word tokens are joined correctly ("I have Optum.", not
  "I  have  O pt um.").
* **Google Gemini transcription** — `SMART` mode no longer fails with
  `languageCodes` set.
* **Deepgram warm connections** — options that change between transcriptions
  (for example new Flux keyterms) are applied when a parked connection is
  reused.
* **TTS** — a [say](/verbs/verbs/say) no longer hangs when the TTS vendor rejects
  the request, and streaming TTS starts cleanly after a barge-in.
* **Conference** — `distributeDtmf` and `speakOnlyTo` work without another join
  option set.
* **Media timeout** — `JAMBONES_MEDIA_TIMEOUT_MS` now ends calls on mediajam
  feature servers.
* **Dialogflow** — a [dialogflow](/verbs/verbs/dialogflow) verb with no TTS vendor
  no longer looks up TTS credentials.
* **Debug logging** — every leg of a call follows the account's debug log
  setting.
* **SIP** — a re-INVITE rejected with `422` no longer leaks resources in
  drachtio; licensed SBCs issue a fresh session token with each provisional and
  final response, so calls that ring longer than 40 seconds keep their audio; TLS registrations record
  their transport; and calls forwarded to a `sips:` Request-URI use a `sips:`
  Contact and From.
* **Billing** — after a failed renewal the portal shows the suspended
  subscription and asks for a new card, instead of offering an upgrade that
  created a second subscription.
* **Portal** — the call logs panel tells a failed request apart from a call with
  no logs.
* **License keys** — whitespace pasted with a license key is trimmed.

#### Upgrade Notes (self-hosted)

* **Database** — 11.1.6 adds the `alert_hooks` table and three columns; run the
  database upgrade before starting the new API server.
* **AWS scale-in** — set `AWS_LIFECYCLE_DRAIN=1` on `inbound`, `sbc-sip-sidecar` and
  `feature-server`, and on `sbc-rtpengine-sidecar` on dedicated RTP servers. Allow the instance roles
  `autoscaling:DescribeAutoScalingInstances` and
  `autoscaling:DescribeLifecycleHooks` (in addition to
  `autoscaling:CompleteLifecycleAction`). `AWS_SNS_TOPIC_ARN` no longer turns on
  draining, and the SNS topics and their open ports can be removed.
* **API server behind a reverse proxy** — set `JAMBONES_TRUST_PROXY=1` and have
  the proxy send `X-Forwarded-For`. Without both, all callers share one
  rate-limit counter.
* **Recordings** — the API server's built-in recording websocket is removed;
  recordings are handled by `upload_recordings`.
* **Audio capture opt-out** — the Privacy settings appear in hosted portal
  builds with `JAMBONES_OPT_OUT_VOIPMONITOR=true`. The opt-out only takes effect
  if voipmonitor is configured with `norecord-header = yes`.

#### Component Versions

| Component            | 11.1.5             | 11.1.6             |
| -------------------- | ------------------ | ------------------ |
| jambonz              | 11.1.5             | **11.1.6**         |
| mediajam             | 0.5.8              | **0.5.10**         |
| drachtio             | 10.1.5             | **10.1.6**         |
| rtpengine            | 14.1.1.8-jambonz14 | 14.1.1.8-jambonz14 |
| jambonz OSS (sbc-\*) | 0.9.12             | **0.9.14**         |
| upload-recordings    | 1.9.1              | 1.9.1              |
| monitoring-agent     | 10.2.2             | 10.2.2             |
| pcap-server          | 1.0.4              | 1.0.4              |
| heplify-server       | 1.0.4              | 1.0.4              |

#### Availability

**jambonz.cloud** - available now

**self-hosted** - coming soon