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installation_guide:configuration:start [2016/10/24 21:07] – [3) Voximal accounts] borja | installation_guide:configuration:start [2016/10/25 21:27] – [Call Detail Record] borja | ||
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+ | ===== System Status ===== | ||
+ | The home page after login show you the system status. | ||
+ | |||
+ | * Host name of the server | ||
+ | * Summary : Main modules status | ||
+ | * Interpreter statistics | ||
+ | * Telephony statistics | ||
+ | * Uptime / Load average | ||
+ | |||
+ | {{: | ||
---- | ---- | ||
- | ===== Configuration | + | ===== Main configuration |
To configure the Voximal IVR you have 4 steps to do : | To configure the Voximal IVR you have 4 steps to do : | ||
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---- | ---- | ||
- | ==== 1) Telephony | + | ==== 1) Configure the telephony |
+ | |||
+ | === a) Use the test number and the PIN === | ||
If your server is connected to the internet you can use the free test access to place calls to your server. | If your server is connected to the internet you can use the free test access to place calls to your server. | ||
There is nothing to do. You should only check that the port 4569 is open from/to internet in UDP. | There is nothing to do. You should only check that the port 4569 is open from/to internet in UDP. | ||
+ | === b) Connect a trunk SIP === | ||
- | + | Before you can make external calls or accept incoming calls from outside, you need to setup SIP Trunks. You can choose any VoIP Service providers. | |
- | You can create a SIP trunk with an operator | + | |
You have to enter informations : | You have to enter informations : | ||
* A trunk name : a string to identify you accounts | * A trunk name : a string to identify you accounts | ||
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{{: | {{: | ||
+ | |||
+ | === c) Add an extension === | ||
+ | |||
+ | After you setup your Voxibot, the first thing you do is to add extensions (connect a Phone). The integrated FreePbx allows you to add a couple of different Device types | ||
+ | |||
+ | * Generic SIP Device | ||
+ | * Generic IAX2 Device | ||
+ | * Generic DAHDi Device | ||
+ | * Other Custom Device | ||
+ | |||
+ | Among these types, SIP device is the most common and popular one. | ||
+ | |||
+ | You can give your extension any unique number, Display Name, password, whether allow this extension to accept inbound external calls or can make outbound external calls. can have voicemail or not etc. | ||
---- | ---- | ||
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- **Dial format** : you can define a specific dialout format for outgoing calls. | - **Dial format** : you can define a specific dialout format for outgoing calls. | ||
- **Mark** : you can define a specific mark, that will appear in traces. | - **Mark** : you can define a specific mark, that will appear in traces. | ||
- | - **Speech** : you can specify the use of the ASR. In case of using ASR server, the better way is to set '' | + | - **Speech** : you can specify the use of the ASR. In case of using ASR server, the better way is to set **Automatic** choice. |
- **Max time** : you can set a maximum duration of call. If not setted or equals to 0, the duration is unlimited. | - **Max time** : you can set a maximum duration of call. If not setted or equals to 0, the duration is unlimited. | ||
- **Vxml parameter** : you can set a string to pass it to vxml script. | - **Vxml parameter** : you can set a string to pass it to vxml script. | ||
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---- | ---- | ||
- | ==== 4) Routes configuration ==== | + | ==== 4) Number/Routes configuration ==== |
- | Firstly, you have to define which application | + | You have your DID number and SIP Trunk set up (with the test number, the called number will be 4568 : " |
- | It's done by defining | + | |
- | | + | You can to define which application you want to use by default for all incoming calls. |
- | - __Leave empty Field__ **'' | + | It's done by defining the **any DID/any CID** or **All DIDs** settings in **Connectivity/ |
- | - Select the application to use : **'' | + | |
- | * You can select an existing application | + | |
- | * Or create a new one by clicking "Add new **'' | + | - __Leave empty Field__ **DID Number** |
- | - **Click on submit** button | + | - Select the application to use : **Voximal Application** |
+ | * You can select an existing application | ||
+ | * Or create a new one by clicking "Add new **Voximal application**" | ||
+ | - Click on **[submit]** button. | ||
{{: | {{: | ||
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---- | ---- | ||
+ | ===== Call Detail Record ===== | ||
+ | |||
+ | The CDR Reports allows you to view a report showing the telephone calls made from and received to your system. | ||
+ | You can choose to view a complete history or calls, or to search by date, date range, number called, caller ID, etc. | ||
+ | |||
+ | {{: | ||
+ | |||
+ | ===== Logs ===== | ||
+ | |||
+ | The Asterisk Logfiles Module is an easy way to view portions of the Asterisk Log. However, this Module is only useful when you want to view a very recent event in the Asterisk Log. | ||
+ | |||
+ | You have similar Module for the Voximal log : | ||
+ | |||
+ | {{: | ||
+ | |||
+ | |||
+ | |||
+ | ===== Settings ===== | ||
+ | |||
+ | === a) General === | ||
+ | |||
+ | {{: | ||
+ | |||
+ | === b) TextToSpeech === | ||
+ | |||
+ | {{: | ||
+ | |||
+ | The home page after login show you the system status. | ||
+ | |||
+ | * Host name of the server | ||
+ | * Summary : Main modules status | ||
+ | * Interpreter statistics | ||
+ | * Telephony statistics | ||
+ | * Uptime / Load average | ||
+ | |||
+ | === d) License === | ||
+ | |||
+ | === c) Test number === | ||
+ | |||
+ | {{: | ||
+ | |||
===== Configuration files ===== | ===== Configuration files ===== | ||
* / | * / | ||
* / | * / | ||
+ | |||
+ |