Take a look at this draft, there are several possibilities described.
The call transfer is a feature of the end devices (UAs, GWs ...), not
the SIP proxy
regards,
klaus
Gregory D. Burns wrote:
Hello Group,
I have a problem I hope I can solve by using SER. Here it is.
1. Person A calls person B
Person A ---[PBX]à[SER]à VoIP à [SIP Gateway] à Telco à Person B
2. Person A calls person C and second line. ( Person B is on Hold)
Person A ---[PBX]à[SER]à VoIP à [SIP Gateway] à Telco à Person C
3. Now we want to be about to Connect person B and person C, with out
Person A needing to be on the line.
Person B à Telco à [SIP Gateway] à VoIP à [SER] àVoIP à [SIP Gateway] à
Telco à Person C
It does not sound that hard, but I could not think of a way to make it
work. Does any one have any ideas?
-Greg
VoIpin
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