Hello again I am resending this email hoping that someone will be able to help. Cheers Fadi On Thu, Jan 19, 2012 at 11:43 AM, Fadi Chehimi <fchehimi at localphone.com>wrote: > 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 > > > > pjsip mailing list > > pjsip at lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > -------------- next part -------------- An HTML attachment was scrubbed... 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