[Freeswitch-users] NAT problem

[Jonas Gauffin]

Ok. Found the problem. I had started using "sofia/outbound/
XXXXXX at 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 at 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 at 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 at 192.168.1.121
> User:           u1000009 at default
> Contact:        "u1000009" <sip:u1000009 at 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 at default
>
> Outbound INVITE:
> send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000:
>    ------------------------------------------------------------------------
>    INVITE sip:0706930XXX at sipgw2.XXXXX.se<sip%3A0706930XXX at 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 at 85.89.XX.XX>;tag=B7pve7F6eeH7c
>    To: <sip:0706930821 at sipgw2.XXXXX.se <sip%3A0706930821 at sipgw2.XXXXX.se>>
>    Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23
>    CSeq: 123379614 INVITE
>    Contact: <sip:mod_sofia at 192.168.1.110:5060>
>    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 at 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
>
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