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%3A0706930XXX@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 To: <sip:0706930821@sipgw2.XXXXX.se <sip%3A0706930821@sipgw2.XXXXX.se>> Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 CSeq: 123379614 INVITE Contact: <sip:mod_sofia@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@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