-----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. alderamin*CLI> <--- SIP read from 10.1.0.65:5060 ---> INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: <sip:[email protected]> From: "Hector" <sip:[email protected]>;tag=typwm Call-ID: [email protected] CSeq: 709 INVITE Contact: <sip:[email protected]> 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 - [email protected] <--- 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:[email protected]>;tag=typwm To: <sip:[email protected]>;tag=as0a3a462b Call-ID: [email protected] 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 '[email protected]' in 32000 ms (Method: INVITE) Found user '201' alderamin*CLI> <--- SIP read from 10.1.0.65:5060 ---> ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.65;rport;branch=z9hG4bKrqslryke Max-Forwards: 70 To: <sip:[email protected]>;tag=as0a3a462b From: "Hector" <sip:[email protected]>;tag=typwm Call-ID: [email protected] 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:[email protected] 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:[email protected]",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5 To: <sip:[email protected]> From: "Hector" <sip:[email protected]>;tag=typwm Call-ID: [email protected] CSeq: 710 INVITE Contact: <sip:[email protected]> 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 - [email protected] 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:[email protected]> <--- 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:[email protected]>;tag=typwm To: <sip:[email protected]> Call-ID: [email protected] CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport From: "Hector Bareiro" <sip:[email protected]>;tag=as7bab61b8 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]>;tag=typwm To: <sip:[email protected]>;tag=as37d19c71 Call-ID: [email protected] CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected]:5060>;tag=as7bab61b8 To: <sip:[email protected]>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918 Call-ID: [email protected] 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK651d88ba;rport From: "Hector Bareiro" <sip:[email protected]>;tag=as7bab61b8 To: <sip:[email protected]>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.1918 Contact: <sip:[email protected]> Call-ID: [email protected] 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK6e8ff8ba;rport From: "Hector Bareiro" <sip:[email protected]>;tag=as7bab61b8 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="danib", realm="ekiga.net", algorithm=MD5, uri="sip:[email protected]", nonce="4a89cd1d00000360086ff884818a5d318b81c0d065d2743f", response="152416b836f298095455859a7c3f1696" 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 4960 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 - --- 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:[email protected]:5060>;tag=as7bab61b8 To: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]:5060>;tag=as7bab61b8 To: <sip:[email protected]>;tag=as38bf28ad Call-ID: [email protected] CSeq: 103 INVITE User-Agent: Ekiga.NET Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]: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;lr=on> set_destination: Parsing <sip:86.64.162.35;lr=on> for address/port to send to set_destination: set destination to 86.64.162.35, port 5060 Transmitting (NAT) to 86.64.162.35:5060: ACK sip:[email protected]:5081 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK776caeb0;rport Route: <sip:86.64.162.35;lr=on> From: "Hector Bareiro" <sip:[email protected]>;tag=as7bab61b8 To: <sip:[email protected]>;tag=as38bf28ad Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 - --- -- SIP/ekiga-090cb900 answered 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:[email protected]>;tag=typwm To: <sip:[email protected]>;tag=as37d19c71 Call-ID: [email protected] CSeq: 710 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:[email protected]> 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:[email protected] 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:[email protected]",response="9cb53107d4d15b7a2e7df8599e851b80",algorithm=MD5 To: <sip:[email protected]>;tag=as37d19c71 From: "Hector" <sip:[email protected]>;tag=typwm Call-ID: [email protected] 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:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.1.0.10:5060;branch=z9hG4bK3c1a71ce;rport From: "asterisk" <sip:[email protected]>;tag=as51f657b5 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] 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/UDP 10.1.0.10:5060;rport=5060;branch=z9hG4bK3c1a71ce To: <sip:[email protected]>;tag=pecxh From: "asterisk" <sip:[email protected]>;tag=as51f657b5 Call-ID: [email protected] 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 '[email protected]' 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:[email protected]>;tag=as2a2e4c13 To: <sip:ekiga.net> Contact: <sip:[email protected]> Call-ID: [email protected] 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:[email protected]:5060>;tag=as2a2e4c13 To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.c092 Call-ID: [email protected] CSeq: 102 OPTIONS Server: Kamailio (1.4.0-notls (i386/linux)) Content-Length: 0 <-------------> - --- (8 headers 0 lines) --- Really destroying SIP dialog '[email protected]' Method: OPTIONS Thanks for your reply. Regards, Daniel -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkqJzZ4ACgkQZpa/GxTmHTdjngCbBalIiS+iSsTXDLfS49xJbMUT T64AoIPzTPiCB3sfTmbOGXJ8pomaGgDp =heDa -----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
