Solved!
Thank you all for your support.
Configuration is as follows:
[color=red][b]On Genesys side[/b][/color];
[u][b]SIP Server Application[/b][/u]
Annex/TServer/enable-unknown-gateway=true
[u][b]Trunk[/b][/u]
1. Created trunk for Asterisk, where Annex/TServer is
[font=courier]"contact" = "<Asterisk IP>:5060"
"prefix" = "<DID at Asterisk>"
"record" = "false"
"refer-enabled" = "false"[/font]
2. Created trunk for Routing Point (RP) with following AnnexTServer:
[font=courier]"contact" = "<Genesys SIP Server IP>:5060"
"prefix" = "<RP DN >"
"record" = "false"
"refer-enabled" = "false"[/font]
[color=red][b]On Asterisk side[/b][/color];
[i]sip.conf[/i]
[font=courier][genesys]
type = friend
callerid = genesys
context = callcenter
nat = yes
qualify = yes
insecure = invite,port
host = <IP of Genesys SIP Server>
dtmfmode = rfc2833
canreinvite = no
qualify = yes
disallow = all
allow = ulaw
allow = alaw
allow = gsm
usereqphone = yes
transport = tcp[/font]
[i]extensions.conf[/i]
[font=courier][incoming]
include = default
[callcenter]
include = incoming
include = default
exten = _00.,1,NoOP("Outbound Call")
exten = _00.,2,Dial(SIP/${EXTEN:2}@genesys)[/font]
Using your soft phone application, which already registered to the Asterisk server, when you dial [b]00<RP DN>[/b], your call goes to RP over Genesys SIP Server.
[b]P.S.[/b] You need to reload SIP and Extension settings in Asterisk after having [i]sip.conf[/i] and [i]extensions.conf[/i] modified.
