Alcatel Unleashed

I have a problem using a SIP trunk to a remote node.
Locally the SIP trunk works correctly but when using it on a remote node I have no dial tone.

create:

1.-sip trunk mode ABC-F for sip gateway
2.-sip trunk mode isdn all countries for external sip gateway

node 1 with the sip trunk local
DELIVERY h1.301
Patch identification: 38
Dynamic patch identification: c

i have the domain 0 with call extra domain without compresion
│ Trunk Group Id : 11 │
│ │
│ Trunk Group Type + T2 │
│ Trunk Group Name : SIP │
│ UTF-8 Trunk Group Name : ------------------------------------------- │
│ Number Compatible With : 3 │
│ Remote Network : 15 │
│ Shared Trunk Group + False │
│ Special Services + Nothing │
│ Node number : 1 │
│ Transcom Trunk Group + False │
│ Auto.reserv.by Attendant + False │
│ Overflow trunk group No. : -1 │
│ Tone on seizure + True │
│ Private Trunk Group + False │
│ Q931 signal variant + ISDN all countries │
│ SS7 signal variant + No variant │
│ Channel selection type + Quantum │
│ auto.DTMF dialing on outgoing call + NO │
│ T2 Specificity + SIP │
│ Homogenous network for direct RTP + NO │
│ Public Network Category : 31 │
│ DDI transcoding + True │
│ Can support UUS in SETUP + True │


┌─Consult/Modify: Network Routing Table─────────────────────────┐

│ Network Number : 15 │
│ │
│ Rank of First Digit to be Sent : 1 │
│ Incoming identification prefix : -------- │
│ Protocol Type + ABC_F │
│ Numbering Plan Descriptor Id : 11 │
│ ARS Route list : 0 │
│ Schedule number : -1 │
│ ATM Address Id : -1 │
│ Network call prefix : -------- │
│ Town Name : -------------------- │
│ Send Town Name + False │
│ Associated SIP gateway : 0 │
│ Unable_UTF8 + True │
│ │
└───────────────────────────────────────────────────────────────┘


┌─Consult/Modify: External Gateways────────────────────────────────────────────────────────┐
│ │

│ Instance : 0 │
│ │
│ Gateway Name : Cable │
│ Remote domain : 10.32.2.1 │
│ PCS IP address : ----------------------------------------------- │
│ Port number : 5060 │
│ Transport type + UDP │
│ RFC3262 forced use + False │
│ Belonging domain : -------------------------------------------------- │
│ Registration Id : -------------------------------------------------- │
│ Registration ID in P_Asserted + True │
│ Registration timer : 0 │
│ Outbound Proxy : -------------------------------------------------- │
│ Supervision timer : 0 │
│ Trunk group number : 11 │
│ Pool Number : -1 │
│ Outgoing realm : -------------------------------------------------- │
│ Outgoing username : -------------------------------------------------- │
│ │
│ Outgoing Password : ---------- │
│ Confirm : ---------- │
│ │
│ Incoming username : -------------------------------------------------- │
│ │
│ Incoming Password : ---------- │
│ Confirm : ---------- │
│ │
│ RFC 3325 supported by the distant + True │
│ DNS type + DNS A │
│ First DNS IP Address : ----------------------------------------------- │
│ Second DNS IP Address : ----------------------------------------------- │
│ SDP in 18x + True │
│ Minimal authentication method + None │
│ INFO method for remote extension + False │
│ Send only trunk group algo + True │
│ To EMS + False │
│ Dynamic Payload type for dtmf : 97 │
│ │
└──────────────────────────────────────────────────────────────────────────────────────────┘

node 2

Release:
DELIVERY i1.605
Patch identification: 14
Dynamic patch identification: e
i have the domain 0 with call extra domain without compresion

│ Trunk Group Type + T2 │
│ Trunk Group Name : SIP MID │
│ UTF-8 Trunk Group Name : ------------------------------------------- │
│ Number Compatible With : 0 │
│ Remote Network : 15 │
│ Shared Trunk Group + False │
│ Special Services + Nothing │
│ Node number : 1 │
│ Transcom Trunk Group + False │
│ Auto.reserv.by Attendant + False │
│ Overflow trunk group No. : -1 │
│ Tone on seizure + True │
│ Private Trunk Group + False │
│ Q931 signal variant + ISDN all countries │
│ SS7 signal variant + No variant │
│ Channel selection type + Quantum │
│ auto.DTMF dialing on outgoing call + NO │
│ T2 Specificity + SIP │
│ Homogenous network for direct RTP + NO │
│ Public Network Category : 31 │
│ DDI transcoding + True │
│ Can support UUS in SETUP + True │



┌─Consult/Modify: Network Routing Table─────────────────────────┐

│ Network Number : 15 │
│ │
│ Rank of First Digit to be Sent : 1 │
│ Incoming identification prefix : -------- │
│ Protocol Type + ABC_F │
│ Numbering Plan Descriptor Id : 11 │
│ ARS Route list : 0 │
│ Schedule number : -1 │
│ ATM Address Id : -1 │
│ Network call prefix : -------- │
│ Town Name : -------------------- │
│ Send Town Name + False │
│ Associated SIP gateway : 0 │
│ Unable_UTF8 + True │
│ │
└───────────────────────────────────────────────────────────────┘

use a OXE as node 3 with patch i160533c to connect to node 1 is the one with the local SIP trunk. I made changes only at the node 3, the compression option of cards Intip and gd is in without compression and the overflow vpn change ip type of compression default to g711 and with those options remained functional with ring back tone.
Node 1 to 2 where I have problems ringback apply the same patch and copy the settings but I still have the same problem, someone could tell me what other options check since all sip, ars, cdt, npd, ip parameters and internode links from node 2 and 3 are equal.