Thanks Johan for this I am now getting closer to the solution. Now I the INVITES routed through the port I am after, but for some reason ACK and BYE are routed through 5060 still. Not sure if I have to manually edit port for these commands as well. I have attached here some of the consol logs: PJSIP initialization.. .. 11:29:55.533 pjsua_core.c SIP UDP socket reachable at 10.15.20.147:12345 ... ... INVITE sip:9198542 at domainname.com SIP/2.0 Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj Max-Forwards: 70 From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com Contact: "FadiTest CheTest" <sip:8932153 at 10.15.20.147:12345;ob> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23363 INVITE Route: <sip:gateway.domainname.com:12345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 300 User-Agent: domainname Content-Type: application/sdp Content-Length: 426 ... ... --end msg-- 11:30:48.831 os_core_unix.c Info: possibly re-registering existing thread 11:30:48.834 callhandling.m Call 0 state changed to CALLING 2012-01-19 11:30:48.839 domainname2[4887:6f03] run loop 11:30:49.330 pjsua_core.c TX 1215 bytes Request msg INVITE/cseq=23363 (tdta0xbb9200) to UDP 108.59.2.137:12345: INVITE sip:9198542 at domainname.com SIP/2.0 Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj Max-Forwards: 70 From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com Contact: "FadiTest CheTest" <sip:8932153 at 10.15.20.147:12345;ob> Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23363 INVITE Route: <sip:gateway.domainname.com:12345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 300 User-Agent: domainname Sip, iPhone App v1.1.1 X-LP-GeoLocation: LONDON, gb X-LP-Connection-Type: Wi-Fi Content-Type: application/sdp Content-Length: 426 ... ... --end msg-- 11:30:52.464 pjsua_core.c TX 444 bytes Request msg ACK/cseq=23363 (tdta0xbb5a00) to UDP 108.59.2.137:12345: ACK sip:9198542 at domainname.com SIP/2.0 Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPjm6ezkf5cgXbSZQA.3oc4JTABzqlfaTCj Max-Forwards: 70 From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com;tag=4fdf1f17ba0fc0dcebac7ca3eba0e50d.dc1d Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23363 ACK Route: <sip:gateway.domainname.com:12345;lr> Content-Length: 0 .. .. --end msg-- 11:30:52.482 callhandling.m Call 0 state changed to CALLING 11:30:52.666 pjsua_core.c RX 400 bytes Response msg 100/INVITE/cseq=23364 (rdata0xb7f014) from UDP 108.59.2.137:12345: SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 10.15.20.147:12345;received=188.39.51.2;rport=12345;branch=z9hG4bKPjAPy4b7DDpAGVeJ-Yg51x9xEXyeVttlXo From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23364 INVITE Server: OpenSIPS (1.7.1-notls (x86_64/linux)) Content-Length: 0 ... ... --end msg-- 11:30:52.974 conference.c Port 3 (ringback) transmitting to port 0 (iPhone IO device) 11:30:52.974 callhandling.m Call 0 state changed to EARLY (180 Ringing) 11:30:57.869 udp0xb7de00 Remote RTP address switched to 95.211.119.213:15278 11:30:57.869 udp0xb7de00 Remote RTCP address switched to 95.211.119.213:15279 11:30:59.040 pjsua_core.c RX 1137 bytes Response msg 200/INVITE/cseq=23364 (rdata0xb7f014) from UDP 108.59.2.137:12345: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPjAPy4b7DDpAGVeJ-Yg51x9xEXyeVttlXo From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23364 INVITE Contact: <sip:108.59.2.137:5060;did=de1.b746f8e6> <======== THIS HERE SHOULD BE 12345 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c64f475 2011-01-18 14-36-30 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Session-Expires: 1800;refresher=uac Min-SE: 300 Content-Type: application/sdp Content-Disposition: session Content-Length: 272 Remote-Party-ID: "Outbound Call" <sip:9198542 at 95.211.119.213>;party=calling;privacy=off;screen=no ... ... --end msg-- 11:30:59.055 callhandling.m Call 0 state changed to CONNECTING [Switching to thread 13315] 11:30:59.598 strm0xbc31b4 VAD temporarily disabled 11:30:59.601 strm0xbc31b4 Encoder stream started 11:30:59.601 strm0xbc31b4 Decoder stream started 11:30:59.609 pjsua_media.c Media updates, stream #0: iLBC (sendrecv) 11:30:59.611 conference.c Port 3 (ringback) stop transmitting to port 0 (iPhone IO device) 11:30:59.613 conference.c Port 4 (sip:9198542 at domainname.com) transmitting to port 0 (iPhone IO device) 11:30:59.613 conference.c Port 0 (iPhone IO device) transmitting to port 4 (sip:9198542 at domainname.com) 11:30:59.615 pjsua_core.c TX 386 bytes Request msg ACK/cseq=23364 (tdta0xede000) to UDP 108.59.2.137:5060: <======== THIS HERE SHOULD BE 12345 ACK sip:108.59.2.137:5060;did=de1.b746f8e6 SIP/2.0 Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPjZNZiwUuuXAkSHBRRtjWNjIz2kRQ2tWH4 Max-Forwards: 70 From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23364 ACK Content-Length: 0 --end msg-- 11:30:59.618 callhandling.m Call 0 state changed to CONFIRMED 11:30:59.676 Master/sound Underflow, buf_cnt=0, will generate 1 frame 11:31:00.260 strm0xbc31b4 VAD re-enabled 11:31:05.303 conference.c Port 0 (iPhone IO device) stop transmitting to port 4 (sip:9198542 at domainname.com) 11:31:05.303 conference.c Port 2 (/var/mobile/Applications/4E314999-EBC0-41E9-9587-136EE83AF2BB/domainname2.app/hang-up.wav) transmitting to port 4 (sip:9198542 at domainname.com) 11:31:06.304 conference.c Port 2 (/var/mobile/Applications/4E314999-EBC0-41E9-9587-136EE83AF2BB/domainname2.app/hang-up.wav) stop transmitting to port 4 (sip:9198542 at domainname.com) 11:31:06.304 conference.c Port 0 (iPhone IO device) transmitting to port 4 (sip:9198542 at domainname.com) 11:31:06.305 pjsua_core.c TX 433 bytes Request msg BYE/cseq=23365 (tdta0xee6000) to UDP 108.59.2.137:5060: <======== THIS HERE SHOULD BE 12345 BYE sip:108.59.2.137:5060;did=de1.b746f8e6 SIP/2.0 Via: SIP/2.0/UDP 10.15.20.147:12345;rport;branch=z9hG4bKPj.oGON31LnBSH9GNcVLHPJe4so5fG3LzD Max-Forwards: 70 From: "FadiTest CheTest" <sip:8932153 at domainname.com>;tag=0S4PJkPDy8oOArXzTNchLLQ8X19bCPaJ To: sip:9198542 at domainname.com;tag=9e4a4ZyK11XUa Call-ID: bLCNgGoNpAq89tUEs7IMoS7LzmwMOGml CSeq: 23365 BYE User-Agent: domainname Sip, iPhone App v1.1.1 Content-Length: 0 ... ... > sipProxy = sip:ip:port in the pjsua config? > > BR > > Johan > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > 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