You should not need to create an additional trunk in the 3CX system for this. Based on the shared information I think that something similar to this should do the trick for you.
controller E1 0/1/?
pri-group timeslots 1-31
!
interface Serial0/1/?:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
isdn calling-number 3660010
!
voice-port 0/1/?:15
connection plar <number you want to use for PLAR>
!
voice class uri 3CX sip
host ipv4:10.10.10.1
host ipv4:<any other IP that the 3CX system might have>
!
voice class e164-pattern-map 1
description E164 Pattern Map for called number to 3CX
e164 0901
e164 <any other directory number that you have on 3CX that you want to send calls to from the gateway>
!
voice class server-group 1
ipv4 10.10.10.1 preference 1
ipv4 <any other IP that the 3CX system might have> preference 2
description Inbound calls to 3CX
!
voice class sip-options-keepalive 1
description Used for Server Group SIP OPTIONS PING
!
dial-peer voice 30 pots
description Outgoing calls to PSTN
destination-pattern <any pattern to use for your new E1 for outbound calls to PSTN>
port 0/1/?:15
forward-digits all
!As an option to add dial-peer 30 you could reconfigure dial-peer 3 to use a trunk group instead of individual a port
!
dial-peer voice 4 voip
description Outgoing calls to 3CX
no destination-pattern 0901
destination e164-pattern-map 1
signaling forward none
session protocol sipv2
no session target ipv4:10.10.10.1
session server-group 1
voice-class sip options-keepalive profile 1
voice-class sip rel1xx supported "100rel"
no voice-class sip pass-thru content unsupp
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2 voip
description Incoming calls from 3CX
translation-profile incoming AutoAttendant
session protocol sipv2
no session target ipv4:10.10.10.1 !This is an outbound dial-peer setting, no need to have on an inbound dial-peer
no incoming called-number .%
incoming uri via 3CX !Better match than what you had before with incoming called-number
voice-class sip rel1xx supported "100rel"
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
All this depends on the version of IOS you have, for your gateway model my recommendation would be to use version 15.7(3)M8.
For additional in-dept information on IOS call routing have a look at this marvelous document.
