Ok. Found the problem. I had started using "sofia/outbound/XXXXXX@sipgw2.XXXX.se"  as bridge destination to try to get outbound_caller_id_name/outbound_caller_id_number working.
It works if I use the correct profile name, "sofia/internal/XXXXXX@sipgw2.XXXX.se"

When do FS use outbound_caller_id instead of effective_caller_id?

On Mon, Nov 23, 2009 at 6:08 PM, Jonas Gauffin <jonas.gauffin@gmail.com> wrote:
Hello

I got the following setup: Phones -> FreeSwitch -> NAT -> Internet -> Gateway

And I'm struggling to get NAT working properly. I'm running freeswitch with the "-nonat" option and have tried different ext-rtp-ip/ext-sip-ip combinations in external/internal profiles.
The From header seems to be correct while contact header and SDP uses local ip? Please help me configure everything correctly.

Currently I have this setup:

API CALL [sofia(status profile external)] output:
========================================================
Name                    external
Domain Name             N/A
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.1.110
Ext-RTP-IP              85.89.XX.XX
SIP-IP                  192.168.1.110
Ext-SIP-IP              85.89.XX.XX
OUTBOUND-PROXY          N/A
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false

API CALL [sofia(status profile default)] output:
========================================================
Name                    default
Domain Name             N/A
Alias Of                internal
Context                 public
Challenge Realm         auto_from
RTP-IP                  192.168.1.110
Ext-RTP-IP              85.89.XX.XX
SIP-IP                  192.168.1.110
OUTBOUND-PROXY          N/A
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            false
STUN-AUTO-DISABLE       false

Sample phone registration:
Call-ID:        Xmbw9PyQ5Q6L2MnQ@192.168.1.121
User:           u1000009@default
Contact:        "u1000009" <sip:u1000009@192.168.1.121:6094>
Agent:          IP PHONE 3 V1.58.004 CFG0
Status:         Registered(UDP)(unknown) EXP(2009-11-23 19:26:40)
Host:           jonas-PC
IP:             192.168.1.121
Port:           6094
Auth-User:      u1000009
Auth-Realm:     default
MWI-Account:    u1000009@default

Outbound INVITE:
send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000:
   ------------------------------------------------------------------------
   INVITE sip:0706930XXX@sipgw2.XXXXX.se SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.110;rport;branch=z9hG4bKB72B75aKmSyBp
   Max-Forwards: 69
   From: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;tag=B7pve7F6eeH7c
   Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23
   CSeq: 123379614 INVITE
   Call-Info: <answer-after=400>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-UNKNOWN
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 293
   X-FS-Support: update_display
   Remote-Party-ID: "Kundtjänst Arne" <sip:0500650XXX@85.89.XX.XX>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1258970915 1258970916 IN IP4 192.168.1.110
   s=FreeSWITCH
   c=IN IP4 192.168.1.110
   t=0 0
   m=audio 24986 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20

Many thanks,
  Jonas