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add listen property to config (#65)

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* add listen property to config

* fix links

* more links fix
This commit is contained in:
Dave Horton
2023-05-10 13:14:24 -04:00
committed by GitHub
parent 780d08d255
commit 1f7d2f9c61
3 changed files with 4 additions and 3 deletions
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1
markdown/docs/webhooks/config.md
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@@ -40,6 +40,7 @@ You can use the following attributes in the `config` command:
| bargeIn.minDigits | Minimum number of dtmf digits expected to gather. Defaults to 1. | no |
| bargeIn.maxDigits | Maximum number of dtmf digits expected to gather | no |
| bargeIn.interDigitTimeout | Amount of time to wait between digits after minDigits have been entered.| no |
| listen | a nested [listen](/docs/webhooks/listen) action, which allows recording of the call from this point forward by streaming the audio to a remote server over a websocket connection | no |
|amd|enable answering machine detection; see [answering machine detection](/docs/supporting-articles/answering-machine-detection) for details|no|
|record|options to manage [call recording using SIPREC](/docs/supporting-articles/siprec-client)|no|
|record.action|"startCallRecording", "stopCallRecording", "pauseCallRecording", or "resumeCallRecording"|yes|

2
markdown/docs/webhooks/dial.md
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@@ -62,7 +62,7 @@ You can use the following attributes in the `dial` command:
| dtmfCapture | an array of strings that represent dtmf sequence which, when detected, will trigger a mid-call notification to the application via the configured `dtmfHook` | no |
| dtmfHook | a webhook to call when a dtmfCapture entry is matched. This is a notification only -- no response is expected, and any desired actions must be carried out via the REST updateCall API. | no|
| headers | an object containing arbitrary sip headers to apply to the outbound call attempt(s) | no |
| listen | a nested [listen](#listen) action, which will cause audio from the call to be streamed to a remote server over a websocket connection | no |
| listen | a nested [listen](/docs/webhooks/listen) action, which will cause audio from the call to be streamed to a remote server over a websocket connection | no |
| referHook | webhook to invoke when an incoming SIP REFER is received on a dialed call. If the application wishes to accept and process the REFER, the webhook application should simply return an HTTP status code 200 with no body, and jambonz will send a SIP 202 Accepted. Otherwise, any HTTP non-success status will cause jambonz to send a SIP response to the REFER with the same status code. <br/><br/>Note that jambonz will send the 202 Accepted and do nothing further. It is the responsibility of the third-party application to then outdial a new call and bridge the other leg, presumably by using the REST API. See [this example app](https://github.com/jambonz/sip-blind-transfer) for more details.| no|
| target | array of to 10 [destinations](#target-types) to simultaneously dial. The first person (or entity) to answer the call will be connected to the caller and the rest of the called numbers will be hung up.| yes |
| timeLimit | max length of call in seconds | no |

4
markdown/docs/webhooks/listen.md
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@@ -6,7 +6,7 @@ jambonz does not have a 'record' verb. This is by design, for data privacy reaso
Instead, jambonz provides the **listen** verb, where an audio stream(s) can be forked and sent in real-time to your application for processing.
The listen verb can also be nested in a [dial](#dial) verb, which allows the audio for a call between two parties to be sent to a remote websocket server.
The listen verb can also be nested in a [dial](/docs/webhooks/dial) or [config](/docs/webhooks/config) verb, which allows the audio for a call between two parties to be sent to a remote websocket server.
To utilize the listen verb, the customer must implement a websocket server to receive and process the audio. The endpoint should be prepared to accept websocket connections with a subprotocol name of 'audio.jambonz.org'.
@@ -37,7 +37,7 @@ You can use the following options in the `listen` action:
| playBeep | true, false whether to play a beep at the start of the listen operation. Default: false | no |
| sampleRate | sample rate of audio to send (allowable values: 8000, 16000, 24000, 48000, or 64000). Default: 8000 | no |
| timeout | the number of seconds of silence that terminates the listen operation.| no |
| transcribe | a nested [transcribe](#transcribe) verb | no |
| transcribe | a nested [transcribe](/docs/webhooks/transcribe) verb | no |
| url | url of remote server to connect to | yes |
| wsAuth.username | HTTP basic auth username to use on websocket connection | no |
| wsAuth.password | HTTP basic auth password to use on websocket connection | no |