Hvorfor virker ikke h263??? -- Executing Macro("SIP/trond-a669", "dial-direct-sip|20172@tandberg.net|20172") in new stack -- Executing Dial("SIP/trond-a669", "SIP/20172@10.47.20.172|30|Cf") in new stack -- Called 20172@10.47.20.172 -- SIP/10.47.20.172-d71a is ringing -- SIP/10.47.20.172-d71a answered SIP/trond-a669 -- Attempting native bridge of SIP/trond-a669 and SIP/10.47.20.172-d71a -- Executing Macro("SIP/trond-a669", "hangupcall") in new stack -- Executing ResetCDR("SIP/trond-a669", "w") in new stack <-- SIP read from 10.47.5.174:5060: INVITE sip:20172@10.47.8.89 SIP/2.0 Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bK15c9d030edb9b1afaf7e2ffd7a11346d.1;rport Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 100 INVITE Contact: From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: Max-Forwards: 70 Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Content-Type:application/sdp Content-Length:508 v=0 o=tandberg 0 1 IN IP4 10.47.5.174 s=- c=IN IP4 10.47.5.174 b=CT:768 t=0 0 m=audio 5600 RTP/AVP 9 8 0 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 m=video 5602 RTP/AVP 96 34 31 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:96 H263-1998/90000 a=fmtp:96 custom=1024,768,4;custom=800,600,3;custom=640,480,2;cif=1;4cif=2;qcif=1;sqcif=1 a=rtpmap:34 H263/90000 a=fmtp:34 cif=1;4cif=2;qcif=1;sqcif=1 a=rtpmap:31 H261/90000 a=fmtp:31 cif=1;qcif=1 --- (12 headers 21 lines)--- Using INVITE request as basis request - 5200c300d800b600@10.47.5.174 Sending to 10.47.5.174 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 10.47.5.174:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bK15c9d030edb9b1afaf7e2ffd7a11346d.1;rport;received=10.47.5.174 From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: ;tag=as66710ad7 Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 100 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Proxy-Authenticate: Digest realm="asterisk.tsip.lab", nonce="7daac182" Content-Length: 0 --- Scheduling destruction of call '5200c300d800b600@10.47.5.174' in 15000 ms Found user 'trond' asterisk1*CLI> <-- SIP read from 10.47.5.174:5060: ACK sip:20172@10.47.8.89 SIP/2.0 Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bK15c9d030edb9b1afaf7e2ffd7a11346d.1;rport Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 100 ACK From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: ;tag=as66710ad7 User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Content-Length:0 --- (8 headers 0 lines)--- <-- SIP read from 10.47.5.174:5060: INVITE sip:20172@10.47.8.89 SIP/2.0 Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bKeca35c5ee90313af19010bb5cdec4860.1;rport Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 101 INVITE Contact: From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: Max-Forwards: 70 Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Proxy-Authorization: Digest nonce="7daac182", realm="asterisk.tsip.lab", qop="", username="trond", uri="sip:asterisk.tsip.lab", response="d920cd2be5513a7f8dda521812b7c96f", algorithm=MD5 Content-Type:application/sdp Content-Length:508 v=0 o=tandberg 0 1 IN IP4 10.47.5.174 s=- c=IN IP4 10.47.5.174 b=CT:768 t=0 0 m=audio 5600 RTP/AVP 9 8 0 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 m=video 5602 RTP/AVP 96 34 31 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:96 H263-1998/90000 a=fmtp:96 custom=1024,768,4;custom=800,600,3;custom=640,480,2;cif=1;4cif=2;qcif=1;sqcif=1 a=rtpmap:34 H263/90000 a=fmtp:34 cif=1;4cif=2;qcif=1;sqcif=1 a=rtpmap:31 H261/90000 a=fmtp:31 cif=1;qcif=1 --- (13 headers 21 lines)--- Using INVITE request as basis request - 5200c300d800b600@10.47.5.174 Sending to 10.47.5.174 : 5060 (non-NAT) Found user 'trond' Found RTP audio format 9 Found RTP audio format 8 Found RTP audio format 0 Found video format unknown Found video format unknown Found video format unknown Peer audio RTP is at port 10.47.5.174:5600 Peer video RTP is at port 10.47.5.174:5602 Found description format G722 Found description format PCMA Found description format PCMU Found description format H263-1998 Found description format H263 Found description format H261 Capabilities: us - 0x4000c (ulaw|alaw|h261), peer - audio=0xc (ulaw|alaw)/video=0x1c0000 (h261|h263|h263p), combined - 0x4000c (ulaw|alaw|h261) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 20172 in from-internal (domain 10.47.8.89) list_route: hop: Transmitting (no NAT) to 10.47.5.174:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bKeca35c5ee90313af19010bb5cdec4860.1;rport;received=10.47.5.174 From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- Executing Macro("SIP/trond-24b7", "dial-direct-sip|20172@tandberg.net|20172") in new stack -- Executing Dial("SIP/trond-24b7", "SIP/20172@10.47.20.172|30|Cf") in new stack -- Called 20172@10.47.20.172 -- SIP/10.47.20.172-206b is ringing Transmitting (no NAT) to 10.47.5.174:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bKeca35c5ee90313af19010bb5cdec4860.1;rport;received=10.47.5.174 From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: ;tag=as671b5efa Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 --- -- SIP/10.47.20.172-206b answered SIP/trond-24b7 We're at 10.47.8.89 port 8308 Video is at 10.47.8.89 port 6736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40000 (h261) to SDP Reliably Transmitting (no NAT) to 10.47.5.174:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bKeca35c5ee90313af19010bb5cdec4860.1;rport;received=10.47.5.174 From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: ;tag=as671b5efa Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Type: application/sdp Content-Length: 228 v=0 o=root 10114 10114 IN IP4 10.47.8.89 s=session c=IN IP4 10.47.8.89 t=0 0 m=audio 8308 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - m=video 6736 RTP/AVP 31 a=rtpmap:31 H261/90000 --- -- Attempting native bridge of SIP/trond-24b7 and SIP/10.47.20.172-206b asterisk1*CLI> <-- SIP read from 10.47.5.174:5060: ACK sip:20172@10.47.8.89 SIP/2.0 Via: SIP/2.0/UDP 10.47.5.174:5060;branch=z9hG4bKeca35c5ee90313af19010bb5cdec4860.1;rport Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 101 ACK From: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A To: ;tag=as671b5efa Max-Forwards: 70 User-Agent: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Content-Length:0 --- (9 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 10.47.5.174, port 5060 We're at 10.47.8.89 port 8308 Video is at 10.47.8.89 port 6736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40000 (h261) to SDP 13 headers, 11 lines Reliably Transmitting (no NAT) to 10.47.5.174:5060: INVITE sip:trond@10.47.5.174 SIP/2.0 Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK03b08c95 From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Contact: Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 232 v=0 o=root 10114 10115 IN IP4 10.47.20.172 s=session c=IN IP4 10.47.20.172 t=0 0 m=audio 5600 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - m=video 5602 RTP/AVP 31 a=rtpmap:31 H261/90000 --- asterisk1*CLI> <-- SIP read from 10.47.5.174:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK03b08c95 Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 102 INVITE From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE Server: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Content-Type:application/sdp Content-Length:271 v=0 o=tandberg 0 2 IN IP4 10.47.5.174 s=- c=IN IP4 10.47.5.174 b=CT:768 t=0 0 m=audio 5600 RTP/AVP 0 8 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 5602 RTP/AVP 31 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:31 H261/90000 --- (10 headers 15 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found video format unknown Peer audio RTP is at port 10.47.5.174:5600 Peer video RTP is at port 10.47.5.174:5602 Found description format PCMU Found description format PCMA Found description format H261 Capabilities: us - 0x4000c (ulaw|alaw|h261), peer - audio=0xc (ulaw|alaw)/video=0x40000 (h261), combined - 0x4000c (ulaw|alaw|h261) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Nov 16 04:34:46 NOTICE[10120]: chan_sip.c:5723 parse_ok_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway Nov 16 04:34:46 WARNING[10120]: chan_sip.c:5756 parse_ok_contact: Invalid host '' list_route: no route Transmitting (no NAT) to 10.47.5.174:5060: ACK sip:trond@10.47.5.174 SIP/2.0 Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK2482962e From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Contact: Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Destroying call 'b700c3001000e100@10.47.5.174' Nov 16 04:34:47 NOTICE[10120]: chan_sip.c:5723 parse_ok_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway Nov 16 04:34:47 WARNING[10120]: chan_sip.c:5756 parse_ok_contact: Invalid host '' We're at 10.47.8.89 port 8308 Video is at 10.47.8.89 port 6736 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP 13 headers, 13 lines Reliably Transmitting (no NAT) to 10.47.5.174:5060: INVITE sip:trond@10.47.5.174 SIP/2.0 Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK50099626 From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Contact: Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 293 v=0 o=root 10114 10116 IN IP4 10.47.20.172 s=session c=IN IP4 10.47.20.172 t=0 0 m=audio 5600 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - m=video 5602 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 --- (9 headers 0 lines)--- asterisk1*CLI> <-- SIP read from 10.47.5.174:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK50099626 Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 103 INVITE From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Allow: UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE Server: TANDBERG/46 (F3.2Beta10 PAL (TEST SW)) Content-Type:application/sdp Content-Length:330 v=0 o=tandberg 0 3 IN IP4 10.47.5.174 s=- c=IN IP4 10.47.5.174 b=CT:768 t=0 0 m=audio 5600 RTP/AVP 0 8 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 m=video 5602 RTP/AVP 31 34 96 c=IN IP4 10.47.5.174 a=sendrecv a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:96 H263-1998/90000 --- (10 headers 17 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found video format unknown Found video format unknown Found video format unknown Peer audio RTP is at port 10.47.5.174:5600 Peer video RTP is at port 10.47.5.174:5602 Found description format PCMU Found description format PCMA Found description format H261 Found description format H263 Found description format H263-1998 Capabilities: us - 0x4000c (ulaw|alaw|h261), peer - audio=0xc (ulaw|alaw)/video=0x1c0000 (h261|h263|h263p), combined - 0x4000c (ulaw|alaw|h261) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Nov 16 04:34:47 NOTICE[10120]: chan_sip.c:5723 parse_ok_contact: '' is not a valid SIP contact (missing sip:) trying to use anyway Nov 16 04:34:47 WARNING[10120]: chan_sip.c:5756 parse_ok_contact: Invalid host '' list_route: no route Transmitting (no NAT) to 10.47.5.174:5060: ACK sip:trond@10.47.5.174 SIP/2.0 Via: SIP/2.0/UDP 10.47.8.89:5060;branch=z9hG4bK29b0823d From: ;tag=as671b5efa To: "tga1000" ;tag=2300880072008900;epid=TAA00506001E02A Contact: Call-ID: 5200c300d800b600@10.47.5.174 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0