Route call to external voice rec system

[Scottyjohn]

Hi all,
Looking to route an inbound call out via a SIP trunk to an external system for some voice rec, where the external system will return results in the SIP Headers.  My question is, how do I ensure that even though routing transfers the call, once complete on the external system, the routing maintains control of the call to carry on further routing and treatments etc?

[cavagnaro]

You don't transfer a call for Voice Rec, you create a conf via MCP.
Check MCP recording integration with 3rd party solutions.

[Scottyjohn]

Thanks but we dont want to use GVP as such, so would that be possible using base SIP Server functionality?

[cavagnaro]

No need for GVP, just the MCP.
SIP Server is not a Media Server, you need something, MCP, to do such functions for Agent conversation recording.

[Scottyjohn]

Sorry Cav,
I meant, voice recognition system not recording! Apologis!

[cavagnaro]

ASR? Like an IVR?
Or Voice Transcription?


Can you describe your scenario to understand a little bit better?

[Scottyjohn]

Hi,
So SIP Server transfer to another SIPServer (non-genesys) which is the entry point into the stack of the speech recognition / IVR application.  The SR/IVR will return metadata within the SIP Headers back to Genesys SIP Server, where they will be used to steer further routing.  My questions are, is it better to use "refer-enabled=true" and oosp true?  And is there a way to send the call to the SR/IVR using basic SIP Server functionality to wait for a response to come back?  Or once its transfered to the system, its gone, and has to come back in as a fresh call to SIPServer, albeit with KV pairs gathered from the SIP Headers?

[cavagnaro]

Ok, so you will send to another SIP Server, that SIP Server, say Asterisk for example, has no clue on what Attach Data is, what it knows and may handle (focus on the MAY) is X- headers used for integration. So Genesys SIP Server will send attach data in this way and it is responsibility of Asterisk (example) to treat that information if possible.
Then you need to decide, do a transfer or a Conference...I guess you are building your flow with ORS or IRD? I don't remember any way to do a conference here or via SCXML


What I would do is to transfer the call to Asterisk
Asterisk will return the call with X-Genesys- headers too so Genesys SIP Server can identify them as attach data


but here of course the return would be to a 2nd RP to make things much easier




Not sure if helps, but maybe this gives further ideas
https://docs.genesys.com/Documentation/SIPS/8.1.1/IntegrationReferenceManual/AsteriskMediaServ