Hi, My setup looks like this: User A (1001), User B (1002) and User C (1003) are registered to 13.13.13.66 via OpenSIPs Proxy 222.222.222.222. User B activated unconditional call forwarding on his phone to User C. Now User A (or any external caller) calls User B. Until last week User A would then be able to talk to User C - the expected behaviour. On Saturday, I updated the freeswitch from version 9fe08675a1 to d1c3f910a6 to get and test some of Steve's fax changes. Since this upgrade, Call-Forwarding / Call Diversion is broken on a level, that I don't understand. User A: INVITE 1002 to B's phone => User B: 302 Moved Temporarily (Contact: <sip:1003@sip.isp.net;user=phone>) => FS: ACK => FS: INVITE 1003 to *B*'s phone! => User B: 404 Not Found => FS: ACK => User A gets signalled the 404 Not Found. packet trace below. I'm aware of http://jira.freeswitch.org/browse/FS-724 http://jira.freeswitch.org/browse/FS-821 but those didn't help me solve my issue: internal.xml: <param name="apply-nat-acl" value="nat.auto"/> internal.xml: <param name="aggressive-nat-detection" value="false"/> I added aggressive-nat-detection=false yesterday, before it wasn't in the config at all. I also tried removing the nat.auto acl temporarily, but that didn't help either. Yesterday afternoon I updated to git c3de9637af, did several bootstrappings, cleans and makes... no change. So today I tried to downgrade to 9fe08675a1, the version I had running until Friday and that was working fine... well, it doesn't now. So I can't even _prove_ it was working before :( Can somebody please point me in the right direction? Any hint in the right direction is much appreciated! Thanks! ====================================================================================== SIP Trace (tcpdump): 16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....[ .>..;^... .......D.fINVITE sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 442 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 teleph 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp E....[..>.4.^... ..one-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623 E.....@.@. . ..^........w8$SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:1003@sip.isp.net;user=phone> Diversion: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;reason="unconditional" Content-Length: 0 16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358 .^....@.>. .......n.fACK sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....\ .>..:^... .......ZgIINVITE sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 466 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-e 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp E....\..>.4.^... ..vent/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551 E..C..@.@..7 ..^......../..SIP/2.0 404 Not Found Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0 16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341 E..q..@.>...^... .......].mACK sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 ======================================================================================