asterisk1*CLI> set debug 4 No such command 'set debug' (type 'help' for help) asterisk1*CLI> set verbose 4 No such command 'set verbose' (type 'help' for help) asterisk1*CLI> sip debug SIP Debugging enabled asterisk1*CLI> <-- SIP read from 10.69.255.246:5060: INVITE sip:404@10.69.255.251;transport=UDP SIP/2.0 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404" Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70c0-68c33729 Max-Forwards: 70 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Type: application/sdp Content-Length: 293 v=0 o=rtp 1159872671864569000 1159872671864569000 IN IP4 10.69.255.246 s=- c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 18 4 98 2 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 g726-16/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 12 lines) --- Sending to 10.69.255.246 : 5060 (NAT) Using INVITE request as basis request - 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 Reliably Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70c0-68c33729;received=10.69.255.246 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404";tag=as018643f9 Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="inspiredbroadcast.net", nonce="5bf52f8c" Content-Length: 0 --- Scheduling destruction of SIP dialog '100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246' in 32000 ms (Method: INVITE) Found user 'tulip2' asterisk1*CLI> <-- SIP read from 10.69.255.246:5060: ACK sip:404@10.69.255.251;transport=UDP SIP/2.0 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404";tag=as018643f9 Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 1 ACK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70c0-68c33729 Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 10.69.255.246:5060: INVITE sip:404@10.69.255.251;transport=UDP SIP/2.0 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404" Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 2 INVITE Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70ea-714e0e46 Max-Forwards: 70 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Proxy-Authorization: Digest username="tulip2",realm="inspiredbroadcast.net",nonce="5bf52f8c",uri="sip:404@10.69.255.251;transport=UDP",response="27473db4436adbae2dcc62977a2589fa",algorithm=MD5 Content-Type: application/sdp Content-Length: 293 v=0 o=rtp 1159872671864569000 1159872671864569000 IN IP4 10.69.255.246 s=- c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 18 4 98 2 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:98 g726-16/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (13 headers 12 lines) --- Sending to 10.69.255.246 : 5060 (NAT) Using INVITE request as basis request - 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 Found user 'tulip2' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 98 Found RTP audio format 2 Found RTP audio format 101 Peer audio RTP is at port 10.69.255.246:5004 Found description format G729 for ID 18 Found description format G723 for ID 4 Found description format g726-16 for ID 98 Found description format g726-32 for ID 2 Found description format telephone-event for ID 101 Capabilities: us - 0xfa3 (g723|gsm|g726|adpcm|lpc10|g729|speex|ilbc), peer - audio=0x901 (g723|g726|g729)/video=0x0 (nothing), combined - 0x901 (g723|g726|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.69.255.246:5004 Looking for 404 in from-sip (domain 10.69.255.251) list_route: hop: Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70ea-714e0e46;received=10.69.255.246 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404" Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- -- Executing [404@from-sip:1] Macro("SIP/tulip2-09bbaf68", "ext|SIP/tulip4") in new stack -- Executing [s@macro-ext:1] Dial("SIP/tulip2-09bbaf68", "SIP/tulip4|15") in new stack Audio is at 10.69.255.251 port 22150 Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x1 (g723) to SDP Adding codec 0x400 (ilbc) to SDP Adding codec 0x80 (lpc10) to SDP Adding codec 0x200 (speex) to SDP Adding codec 0x20 (adpcm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.69.255.242:5060: INVITE sip:tulip4@10.69.255.242:5060 SIP/2.0 Via: SIP/2.0/UDP 10.69.255.251:5060;branch=z9hG4bK7a883cd2;rport From: "Tulip ATA port 2" ;tag=as2a8c71eb To: Contact: Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 03 Oct 2006 11:51:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 531 v=0 o=root 15893 15893 IN IP4 10.69.255.246 s=session c=IN IP4 10.69.255.246 t=0 0 m=audio 5004 RTP/AVP 18 3 2 4 97 7 110 5 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=ptime:20 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=ptime:30 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:7 LPC/8000 a=ptime:20 a=rtpmap:110 speex/8000 a=ptime:20 a=rtpmap:5 DVI4/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Called tulip4 [Oct 3 12:51:37] WARNING[15971]: channel.c:2983 ast_channel_make_compatible: No path to translate from SIP/tulip4-09bc00b8(2) to SIP/tulip2-09bbaf68(256) asterisk1*CLI> <-- SIP read from 10.69.255.242:5060: SIP/2.0 180 Ringing From: "Tulip ATA port 2";tag=as2a8c71eb To: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK7a883cd2 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Length: 0 --- (10 headers 0 lines) --- -- SIP/tulip4-09bc00b8 is ringing Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70ea-714e0e46;received=10.69.255.246 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404";tag=as4494aa01 Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 --- asterisk1*CLI> <-- SIP read from 10.69.255.242:5060: SIP/2.0 200 OK From: "Tulip ATA port 2";tag=as2a8c71eb To: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 102 INVITE Via: SIP/2.0/UDP 10.69.255.251:5060;rport=5060;branch=z9hG4bK7a883cd2 Supported: replaces Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=rtp 1159872672466606000 1159872672466606000 IN IP4 10.69.255.242 s=- c=IN IP4 10.69.255.242 t=0 0 m=audio 5004 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 9 lines) --- [Oct 3 12:51:39] DEBUG[15966]: chan_sip.c:2005 __sip_ack: Acked pending invite 102 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 10.69.255.242:5004 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0xfa3 (g723|gsm|g726|adpcm|lpc10|g729|speex|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 10.69.255.242:5004 [Oct 3 12:51:39] DEBUG[15966]: chan_sip.c:4993 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x100 (g729) and not 0x2 (gsm) [Oct 3 12:51:39] WARNING[15966]: channel.c:2641 set_format: Unable to find a codec translation path from g729 to gsm [Oct 3 12:51:39] WARNING[15966]: channel.c:2641 set_format: Unable to find a codec translation path from g729 to gsm list_route: hop: [Oct 3 12:51:39] DEBUG[15966]: chan_sip.c:5412 reqprep: Strict routing enforced for session 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 set_destination: Parsing for address/port to send to set_destination: set destination to 10.69.255.242, port 5060 Transmitting (no NAT) to 10.69.255.242:5060: ACK sip:tulip4@10.69.255.242:5060 SIP/2.0 Via: SIP/2.0/UDP 10.69.255.251:5060;branch=z9hG4bK6dd8bf22;rport From: "Tulip ATA port 2" ;tag=as2a8c71eb To: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 Contact: Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/tulip4-09bc00b8 answered SIP/tulip2-09bbaf68 Audio is at 10.69.255.251 port 17340 Adding codec 0x100 (g729) to SDP Adding codec 0x800 (g726) to SDP Adding codec 0x1 (g723) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 10.69.255.246:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-4522409f-dcc70ea-714e0e46;received=10.69.255.246 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404";tag=as4494aa01 Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 339 v=0 o=root 15893 15893 IN IP4 10.69.255.242 s=session c=IN IP4 10.69.255.242 t=0 0 m=audio 5004 RTP/AVP 18 2 4 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no =ptime:20*CLI> a=rtpmap:2 G726-32/8000 a=ptime:20 a=rtpmap:4 G723/8000 a=ptime:30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- -- Native bridging SIP/tulip2-09bbaf68 and SIP/tulip4-09bc00b8 asterisk1*CLI> <-- SIP read from 10.69.255.246:5060: ACK sip:404@10.69.255.251 SIP/2.0 From: "Line 2";tag=100f43e8-f6ff450a-13c4-4522409f-78e12f20-4522409f To: "404";tag=as4494aa01 Call-ID: 100f7c48-f6ff450a-13c4-4522409f-15f56db-4522409f@10.69.255.246 CSeq: 2 ACK Via: SIP/2.0/UDP 10.69.255.246:5060;rport;branch=z9hG4bK-452240a2-dcc7932-187fce57 Max-Forwards: 70 Contact: Proxy-Authorization: Digest username="tulip2",realm="inspiredbroadcast.net",nonce="5bf52f8c",uri="sip:404@10.69.255.251;transport=UDP",response="27473db4436adbae2dcc62977a2589fa",algorithm=MD5 Content-Length: 0 --- (10 headers 0 lines) --- asterisk1*CLI> <-- SIP read from 10.69.255.242:5060: INVITE sip:402@10.69.255.251 SIP/2.0 From: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 To: "Tulip ATA port 2";tag=as2a8c71eb Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 1 INVITE Via: SIP/2.0/UDP 10.69.255.242:5060;rport;branch=z9hG4bK-452240a5-dcc85cb-148ce816 Max-Forwards: 70 Supported: replaces Contact: Content-Type: application/sdp Content-Length: 294 v=0 o=rtp 1159872672466606000 159872672466606000 IN IP4 10.69.255.242 s=- c=IN IP4 10.69.255.242 t=0 0 m=image 5004 UDPTL t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxMaxBuffer:1024 a=T38FaxMaxDatagram:238 a=T38FaxRateManagement:transferredTCF a=T38FaxUdpEC:t38UDPRedundancy --- (11 headers 12 lines) --- Sending to 10.69.255.242 : 5060 (NAT) [Oct 3 12:51:42] WARNING[15966]: chan_sip.c:4637 process_sdp: Unsupported SDP media type in offer: image 5004 UDPTL t38 Transmitting (NAT) to 10.69.255.242:5060: SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 10.69.255.242:5060;branch=z9hG4bK-452240a5-dcc85cb-148ce816;received=10.69.255.242;rport=5060 From: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 To: "Tulip ATA port 2";tag=as2a8c71eb Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 --- asterisk1*CLI> <-- SIP read from 10.69.255.242:5060: ACK sip:402@10.69.255.251 SIP/2.0 From: ;tag=100f5b18-f2ff450a-13c4-452240a0-1b199bb9-452240a0 To: "Tulip ATA port 2";tag=as2a8c71eb Call-ID: 1815b4a24b8121bb1e6878fe37c2d72f@10.69.255.251 CSeq: 1 ACK Via: SIP/2.0/UDP 10.69.255.242:5060;rport;branch=z9hG4bK-452240a5-dcc85cb-148ce816 Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines) --- asterisk1*CLI>