I would like get your feedback on the proposed change to SIP Call Flow module.
We had a number of internal debates on how to improve SIP Call Flow Module and make it even more powerful and still easy to use module. However, we can not agree on the best approach to achieve that so I am turning to you for help.
One of the most common questions we get is "how can I configure a failover if my SIP/ATA device is offline" ?
Right now SIP Call Flow module has only one [N/A] (no answer) outgoing connector and sets Call Flow variable with SIP transfer status (offline, not answered, busy etc).
While it is possible to use NESTED IF call flow item to evaluate the result and route handle the call based on the status it is not intuitive for most and makes call flow look more complex than it is.
We consider a number of changes
1. Split SIP Call Flow module into 3 separate Call Flow modules
* SIP DEVICE - to transfer to SIP device registered with Anveo
* SIP URI - to transfer to some SIP URI
* SIP RING GROUP - to work with ring group of SIP devices connected to Anveo the same way the current SIP Call Flow deals with Ring Groups
2. SIP DEVICE Call Flow module will have 4 outgoing connectors
* [N/A] - a call was not answered
* [BUSY] - SIP/ATA device is busy
* [ON CALL] - SIP/ATA device has another call
* [OFFLINE] - SIP/ATA is offline
3. SIP URI module will have 1 outgoing connectors
4. SIP RING GROUP module will have 1 outgoing connector
What do you think? Please share your feedback as we always welcome new ideas.