I'm seeing an issue during a bridge where the '200' from freeswitch is not being ACK'd and Freeswitch is correctly dropping the connection. However I belive that the Contact header in the 200 packet is causing the problem. Am I missing a config parameter? If I back out the revision 5208 change to mod_sofia.c from the trunk source then the problem goes away. (SIPTAG_CONTACT_STR(tech_pvt->to_uri) back to SIPTAG_CONTACT_STR(tech_pvt->profile->url)). My test setup is (Freeswitch is behind a NAT): External SIP -> Freeswitch -> Gizmo The freeswitch config for the extension is: <extension name="test"> <condition field="destination_number" expression="^test$"> <action application="bridge" data="sofia/gizmo/17470355893@proxy01.sipphone.com"/> </condition> </extension> I've put a full copy of the log here: http://pastebin.freeswitch.org/3008 My internet facing profile is: <profile name="internet"> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${global_codec_prefs}"/> <param name="codec-ms" value="20"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="sip-ip" value="$${bind_server_ip}"/> <param name="accept-blind-reg" value="true"/> <param name="nonce-ttl" value="60"/> <param name="ext-rtp-ip" value="80.189.98.58"/> <param name="ext-sip-ip" value="80.189.98.58"/> <param name="sip-domain" value="sip.dalethatcher.com"/> </settings> </profile> With the current trunk the SIP packet that does not get a response is: send 1182 bytes to udp/[193.111.201.32]:5060 at 13:05:15.136189: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 193.111.201.32;branch=z9hG4bK614f.db2cc426.0 Via: SIP/2.0/UDP 193.111.200.182;branch=z9hG4bK614f.9dbef354.0 Via: SIP/2.0/UDP 194.145.189.10:3670;branch=z9hG4bK00E0F556495005A857E00000230F Record-Route: <sip:193.111.201.32;lr=on;ftag=00E0F556495005A857E00000222A> Record-Route: <sip:193.111.200.182;lr=on;ftag=00E0F556495005A857E00000222A> From: <sip:02085555129@194.145.189.10>;tag=00E0F556495005A857E00000222A To: <sip:442070968842@193.111.200.182>;tag=r5ae93vm8aK1p Call-ID: 00E0F556495005A857E000001549@194.145.189.10 CSeq: 35816 INVITE Contact: <sip:442070968842@193.111.200.182:5060> User-Agent: FreeSWITCH(mod_sofia) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Supported: 100rel, precondition Content-Type: application/sdp Content-Disposition: session Content-Length: 290 v=0 o=FreeSWITCH 6489556414011969222 9057524047013517960 IN IP4 192.168.93.5 s=FreeSWITCH c=IN IP4 80.189.98.58 t=0 0 a=sendrecv m=audio 10002 RTP/AVP 0 96 13 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=rtpmap:13 CN/8000 a=ptime:20 m=image 0 UDPTL 9 ------------------------------------------------------------------------ Backing out the change modifies the Contact header to: Contact: <sip:mod_sofia@80.189.98.58:5060> Which gets an ACK packet and doesn't drop the call. thanks, - Dale