Daniel, I'm a little confused as to what I'm seeing here. You're bounding through two RFC1918 address networks -- 10.1.0.X and 192.168.2.X. Is this some sort of dual NAT scenario?
Perhaps if you can explain a little more about your network setup. N. Daniel Bareiro wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > SIP wrote: > > >> Daniel, >> > > Hi SIP. > > >> Check your stunaddr setting. Is it misspelled, or do they really use >> stun.exiga.net instead of stun.ekiga.net ? >> > > Thanks to indicate that error to me. I doing the test again. I don't > believe that this solves what I commented before about 192.168.1.2 > direction, but, just in case, I copy the output of debugging when trying > to communicate to ekiga.net. The problem continues persisting after the > correction. > > alderamin*CLI> > <--- SIP read from 10.1.0.65:5060 ---> > INVITE sip:8...@10.1.0.10 SIP/2.0 > Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke > Max-Forwards: 70 > To: <sip:8...@10.1.0.10> > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 709 INVITE > Contact: <sip:2...@10.1.0.65> > Content-Type: application/sdp > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > Supported: replaces,norefersub,100rel > User-Agent: Twinkle/1.2 > Content-Length: 247 > > v=0 > o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 > s=- > c=IN IP4 10.1.0.65 > t=0 0 > m=audio 8000 RTP/AVP 8 0 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > <-------------> > - --- (13 headers 12 lines) --- > Sending to 10.1.0.65 : 5060 (NAT) > Using INVITE request as basis request - > kafgeaflkmsd...@defiant.freesoftware.org > > <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 ---> > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 10.1.0.65;branch=z9hG4bKrqslryke;received=10.1.0.65;rport=5060 > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > To: <sip:8...@10.1.0.10>;tag=as0a3a462b > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 709 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", > nonce="497d879d" > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog > 'kafgeaflkmsd...@defiant.freesoftware.org' in 32000 ms (Method: INVITE) > Found user '201' > alderamin*CLI> > <--- SIP read from 10.1.0.65:5060 ---> > ACK sip:8...@10.1.0.10 SIP/2.0 > Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke > Max-Forwards: 70 > To: <sip:8...@10.1.0.10>;tag=as0a3a462b > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 709 ACK > User-Agent: Twinkle/1.2 > Content-Length: 0 > > > <-------------> > - --- (9 headers 0 lines) --- > alderamin*CLI> > <--- SIP read from 10.1.0.65:5060 ---> > INVITE sip:8...@10.1.0.10 SIP/2.0 > Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKxpraybjr > Max-Forwards: 70 > Proxy-Authorization: Digest > username="201",realm="asterisk",nonce="497d879d",uri="sip:8...@10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5 > To: <sip:8...@10.1.0.10> > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 710 INVITE > Contact: <sip:2...@10.1.0.65> > Content-Type: application/sdp > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > Supported: replaces,norefersub,100rel > User-Agent: Twinkle/1.2 > Content-Length: 247 > > v=0 > o=twinkle 933572867 1938524932 IN IP4 10.1.0.65 > s=- > c=IN IP4 10.1.0.65 > t=0 0 > m=audio 8000 RTP/AVP 8 0 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > <-------------> > - --- (14 headers 12 lines) --- > Sending to 10.1.0.65 : 5060 (NAT) > Using INVITE request as basis request - > kafgeaflkmsd...@defiant.freesoftware.org > Found user '201' > Found RTP audio format 8 > Found RTP audio format 0 > Found RTP audio format 3 > Found RTP audio format 101 > Peer audio RTP is at port 10.1.0.65:8000 > Found audio description format PCMA for ID 8 > Found audio description format PCMU for ID 0 > Found audio description format GSM for ID 3 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0xe > (gsm|ulaw| > alaw)/video=0x0 (nothing), combined - 0xa (gsm|alaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 10.1.0.65:8000 > Looking for 8500 in from-internal (domain 10.1.0.10) > list_route: hop: <sip:2...@10.1.0.65> > > <--- Transmitting (no NAT) to 10.1.0.65:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > To: <sip:8...@10.1.0.10> > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 710 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:8...@10.1.0.10> > Content-Length: 0 > > > <------------> > -- Executing [8...@from-internal:1] Dial("SIP/201-090ffff0", > "SIP/ekiga/500|20|r)") in new stack > Video is at 192.168.1.2 port 10112 > Audio is at 192.168.1.2 port 12592 > Adding codec 0x8 (alaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding codec 0x40000 (h261) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 86.64.162.35:5060: > INVITE sip:5...@ekiga.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport > From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8 > To: <sip:5...@ekiga.net> > Contact: <sip:2...@192.168.1.2> > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 17 Aug 2009 21:30:25 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 331 > > v=0 > o=root 4959 4959 IN IP4 192.168.1.2 > s=session > c=IN IP4 192.168.1.2 > b=CT:384 > t=0 0 > m=audio 12592 RTP/AVP 8 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > m=video 10112 RTP/AVP 31 > a=rtpmap:31 H261/90000 > a=sendrecv > > - --- > -- Called ekiga/500 > > <--- Transmitting (no NAT) to 10.1.0.65:5060 ---> > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP > 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > To: <sip:8...@10.1.0.10>;tag=as37d19c71 > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 710 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:8...@10.1.0.10> > Content-Length: 0 > > > <------------> > alderamin*CLI> > <--- SIP read from 86.64.162.35:5060 ---> > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport=10003;received=190.51.105.123 > From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8 > To: <sip:5...@ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918 > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 102 INVITE > Proxy-Authenticate: Digest realm="ekiga.net", > nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f" > Server: Kamailio (1.4.0-notls (i386/linux)) > Content-Length: 0 > > > <-------------> > - --- (9 headers 0 lines) --- > Transmitting (NAT) to 86.64.162.35:5060: > ACK sip:5...@ekiga.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport > From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8 > To: <sip:5...@ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918 > Contact: <sip:2...@192.168.1.2> > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Content-Length: 0 > > > - --- > Video is at 192.168.1.2 port 10112 > Audio is at 192.168.1.2 port 12592 > Adding codec 0x8 (alaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding codec 0x40000 (h261) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (NAT) to 86.64.162.35:5060: > INVITE sip:5...@ekiga.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport > From: "Hector Bareiro" <sip:2...@192.168.1.2>;tag=as7bab61b8 > To: <sip:5...@ekiga.net> > Contact: <sip:2...@192.168.1.2> > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Proxy-Authorization: Digest username="danib", realm="ekiga.net", > algorithm=MD5, uri="sip:5...@ekiga.net", > nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f", > response="152416b836f298095455859a7c3f1696" > Date: Mon, 17 Aug 2009ideo 10112 RTP/AVP 31 > a=rtpmap:31 H261/90000 > a=sendrecv > > - --- > alderamin*CLI> > <--- SIP read from 86.64.162.35:5060 ---> > SIP/2.0 100 Giving a try > Via: SIP/2.0/UDP > 192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport=10003;received=190.51.105.123 > From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8 > To: <sip:5...@ekiga.net> > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 103 INVITE > Server: Kamailio (1.4.0-notls (i386/linux)) > Content-Length: 0 > > > <-------------> > - --- (8 headers 0 lines) --- > alderamin*CLI> > <--- SIP read from 86.64.162.35:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.1.2:5060;received=190.51.105.123;branch=z9hG4bK6e8ff8ba;rport=10003 > Record-Route: <sip:86.64.162.35;lr=on> > From: "Hector Bareiro" <sip:2...@192.168.1.2:5060>;tag=as7bab61b8 > To: <sip:5...@ekiga.net>;tag=as38bf28ad > Call-ID: 6149754405a8a52d5cae9ad92c813...@192.168.1.2 > CSeq: 103 INVITE > User-Agent: Ekiga.NET > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:5...@86.64.162.35:5081> > Content-Type: application/sdp > Content-Length: 310 > > v=0 > o=root 9963 9963 IN IP4 86.64.162.35 > s=session > c=IN IP4 86.64.162.35 > b=CT:384 > t=0 0 > m=audio 10400 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > m=video 14488 RTP/AVP 31 > a=rtpmap:31 H261/90000 > a=sendrecv > > <-------------> > - --- (13 headers 16 lines) --- > Found RTP audio format 8 > Found RTP audio format 101 > Found RTP video format 31 > Peer audio RTP is at port 86.64.162.35:10400 > Found audio description format PCMA for ID 8 > Found audio description format telephone-event for ID 101 > Found video description format H261 for ID 31 > Capabilities: us - 0x4040a (gsm|alaw|ilbc|h261), peer - audio=0x40008 > (alaw| > h261)/video=0x40000 (h261), combined - 0x40008 (alaw|h261) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 86.64.162.35:10400 > Peer video RTP is at port 86.64.162.35:14488 > list_route: hop: <sip:86.64.162.35;lwered SIP/201-090ffff0 > Audio is at 10.1.0.10 port 12994 > Adding codec 0x8 (alaw) to SDP > Adding codec 0x2 (gsm) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > > <--- Reliably Transmitting (no NAT) to 10.1.0.65:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 10.1.0.65;branch=z9hG4bKxpraybjr;received=10.1.0.65;rport=5060 > rom: "Hector" <sip:2...@10.1.0.10>;tag=typwm > To: <sip:8...@10.1.0.10>;tag=as37d19c71 > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 710 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:8...@10.1.0.10> > Content-Type: application/sdp > Content-Length: 255 > > v=0 > o=root 4959 4959 IN IP4 10.1.0.10 > s=session > c=IN IP4 10.1.0.10 > t=0 0 > m=audio 12994 RTP/AVP 8 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > <------------> > alderamin*CLI> > <--- SIP read from 10.1.0.65:5060 ---> > ACK sip:8...@10.1.0.10 SIP/2.0 > Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKpnwhfosw > Max-Forwards: 70 > Proxy-Authorization: Digest > username="201",realm="asterisk",nonce="497d879d",uri="sip:8...@10.1.0.10",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5 > To: <sip:8...@10.1.0.10>;tag=as37d19c71 > From: "Hector" <sip:2...@10.1.0.10>;tag=typwm > Call-ID: kafgeaflkmsd...@defiant.freesoftware.org > CSeq: 710 ACK > User-Agent: Twinkle/1.2 > Content-Length: 0 > > > <-------------> > - --- (10 headers 0 lines) --- > Reliably Transmitting (no NAT) to 10.1.0.65:5060: > OPTIONS sip:2...@10.1.0.65 SIP/2.0 > Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK3c1a71ce;rport > From: "asterisk" <sip:aster...@10.1.0.10>;tag=as51f657b5 > To: <sip:2...@10.1.0.65> > Contact: <sip:aster...@10.1.0.10> > Call-ID: 2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 17 Aug 2009 21:30:29 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > - --- > alderamin*CLI> > <--- SIP read from 10.1.0.65:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UD=5060;branch=z9hG4bK3c1a71ce > To: <sip:2...@10.1.0.65>;tag=pecxh > From: "asterisk" <sip:aster...@10.1.0.10>;tag=as51f657b5 > Call-ID: 2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10 > CSeq: 102 OPTIONS > Accept: application/sdp > Accept-Encoding: identity > Accept-Language: en > Allow: > INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE > Server: Twinkle/1.2 > Supported: replaces,norefersub,100rel > Content-Length: 0 > > > <-------------> > - --- (13 headers 0 lines) --- > Really destroying SIP dialog > '2096f1ac21f419aa029d7ccb5d8de...@10.1.0.10' > Method: OPTIONS > Reliably Transmitting (NAT) to 86.64.162.35:5060: > OPTIONS sip:ekiga.net SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2268d402;rport > From: "asterisk" <sip:aster...@192.168.1.2>;tag=as2a2e4c13 > To: <sip:ekiga.net> > Contact: <sip:aster...@192.168.1.2> > Call-ID: 1a9ce3ad2f18ecf129c457d527603...@192.168.1.2 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Mon, 17 Aug 2009 21:30:52 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Length: 0 > > > - --- > alderamin*CLI> > <--- SIP read from 86.64.162.35:5060 ---> > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.1.2:5060;branch=z9hG4bK2268d402;rport=10003;received=190.51.105.123 > From: "asterisk" <sip:aster...@192.168.1.2:5060>;tag=as2a2e4c13 > To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c092 > Call-ID: 1a9ce3ad2f18ecf129c457d527603...@192.168.1.2 > CSeq: 102 OPTIONS > Server: Kamailio (1.4.0-notls (i386/linux)) > Content-Length: 0 > > > <-------------> > - --- (8 headers 0 lines) --- > Really destroying SIP dialog > '1a9ce3ad2f18ecf129c457d527603...@192.168.1.2' > Method: OPTIONS > > > > > Thanks for your reply. > > Regards, > Daniel > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (GNU/Linux) > > iEYEARECAAYFAkqLPfgACgkQZpa/GxTmHTdrGgCfYSVRonoXKAdgYU2bWp4ZibA0 > ic8AmwQUlEPB1VTthUp3WF+6dP5maU7P > =Av+8 > -----END PGP SIGNATURE----- > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users