Transfering calls with a phone behind a CUBE.

Hello,
My setup is as follows.
I have two softswitches, one behind my CUBE (from now on I call it "M") and another one outside the CUBE (from now on I call it "A").
I can call from A to M and viceversa.
If a phone from A orginates a call to M, that phone from M can transfer that call to another phone from M..but I am not able to transfer that call to a phone from A (that is, a connection between two phones from A)
Softswitch M is a softswitch we are currently testing, and using a CUBE is the only way we can hook it to the network without making lots of changes in the routing tables.
I was wondering if the transfer I am not being able to complete, can be accomplished.
I will be happy to provide you with any info you require to help me with my problem.
Thank you in advance for your help.

Softswitch A is an Asterisk 1.8, so it does.
This is the initial invite, the one that phone registered in A sends to CUBE.
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.6.4:5060;branch=z9hG4bK265d780a
Max-Forwards: 70
From: "Gabriel Lema" <sip:[email protected]>;tag=as54f712b3
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Date: Thu, 12 Jun 2014 12:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 884768917 884768917 IN IP4 10.0.6.4
s=Asterisk PBX 1.8.7.0
c=IN IP4 10.0.6.4
t=0 0
m=audio 10598 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
And this is the one that CUBE sends to M.
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.10.254:5060;branch=z9hG4bK6DBF1889
Remote-Party-ID: "Gabriel Lema" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Gabriel Lema" <sip:[email protected]>;tag=E193EFFC-60F
To: <sip:[email protected]>
Date: Thu, 12 Jun 2014 12:15:09 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0138615386-4049736163-2934674880-0163092859
User-Agent: Cisco-SIPGateway/IOS-15.2.4.M4
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1402575309
Contact: <sip:[email protected]:5060>
Call-Info: <sip:10.0.10.254:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 280
v=0
o=CiscoSystemsSIP-GW-UserAgent 2759 7351 IN IP4 10.0.10.254
s=SIP Call
c=IN IP4 10.0.10.254
t=0 0
m=audio 19398 RTP/AVP 8 0 101 19
c=IN IP4 10.0.10.254
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
Thanks,
Gabriel

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