Ok. Found the problem. I had started using "sofia/outbound/ xxx...@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/ xxx...@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.gauf...@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: xmbw9pyq5q6l2...@192.168.1.121 > User: u1000...@default > Contact: "u1000009" <sip:u1000...@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: u1000...@default > > Outbound INVITE: > send 1122 bytes to udp/[62.80.XX.XX]:5060 at 17:05:01.740000: > ------------------------------------------------------------------------ > INVITE > sip:0706930...@sipgw2.xxxxx.se<sip%3a0706930...@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:0500650...@85.89.xx.xx>;tag=B7pve7F6eeH7c > To: <sip:0706930...@sipgw2.xxxxx.se <sip%3a0706930...@sipgw2.xxxxx.se>> > Call-ID: 2dcead20-52f5-122d-d3a1-77ca4f97ec23 > CSeq: 123379614 INVITE > Contact: <sip:mod_so...@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:0500650...@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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