backupPBX*CLI> <-- SIP read from 192.168.0.59:5060: INVITE sip:123@192.168.0.25;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.59:5060 From: ;tag=920185343 To: Call-ID: 2153418752@192.168.0.59 CSeq: 1 INVITE Contact: User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Expires: 300 Content-Length: 245 Content-Type: application/sdp v=0 o=12 7905 7905 IN IP4 192.168.0.59 s=ATA186 Call c=IN IP4 192.168.0.59 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (11 headers 11 lines)--- Using INVITE request as basis request - 2153418752@192.168.0.59 Sending to 192.168.0.59 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.0.59:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59 From: ;tag=920185343 To: ;tag=as259d571b Call-ID: 2153418752@192.168.0.59 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5fa099b4" Content-Length: 0 --- Scheduling destruction of call '2153418752@192.168.0.59' in 15000 ms Found user '12' backupPBX*CLI> <-- SIP read from 192.168.0.59:5060: ACK sip:123@192.168.0.25;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59 From: ;tag=920185343 To: ;tag=as259d571b Call-ID: 2153418752@192.168.0.59 CSeq: 1 ACK User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Content-Length: 0 --- (8 headers 0 lines)--- backupPBX*CLI> <-- SIP read from 192.168.0.59:5060: INVITE sip:123@192.168.0.25;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.59:5060 From: ;tag=920185343 To: Call-ID: 2153418752@192.168.0.59 CSeq: 2 INVITE Contact: User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="ea44407c623ffceac38ac33b63c53f3e" Expires: 300 Content-Length: 245 Content-Type: application/sdp v=0 o=12 7907 7907 IN IP4 192.168.0.59 s=ATA186 Call c=IN IP4 192.168.0.59 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --- (12 headers 11 lines)--- Using INVITE request as basis request - 2153418752@192.168.0.59 Sending to 192.168.0.59 : 5060 (non-NAT) Found user '12' Found RTP audio format 18 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.59:16384 Found description format G729 Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 123 in mycontext (domain 192.168.0.25) list_route: hop: Transmitting (no NAT) to 192.168.0.59:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59 From: ;tag=920185343 To: Call-ID: 2153418752@192.168.0.59 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- We're at 192.168.0.25 port 12296 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59 From: ;tag=920185343 To: ;tag=as4843cf8d Call-ID: 2153418752@192.168.0.59 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 285 v=0 o=root 6796 6796 IN IP4 192.168.0.25 s=session c=IN IP4 192.168.0.25 t=0 0 m=audio 12296 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- backupPBX*CLI> <-- SIP read from 192.168.0.59:5060: ACK sip:123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.59:5060 From: ;tag=920185343 To: ;tag=as4843cf8d Call-ID: 2153418752@192.168.0.59 CSeq: 2 ACK User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="ea44407c623ffceac38ac33b63c53f3e" Content-Length: 0 --- (9 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.59, port 5060 Reliably Transmitting (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of call '2153418752@192.168.0.59' in 32000 ms Retransmitting #1 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #2 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #3 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #4 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Retransmitting #5 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- backupPBX*CLI> <-- SIP read from 192.168.0.59:5060: BYE sip:123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.59:5060 From: ;tag=920185343 To: ;tag=as4843cf8d Call-ID: 2153418752@192.168.0.59 CSeq: 3 BYE User-Agent: Cisco ATA 186 v2.16.2 ata18x (030909a) Proxy-Authorization: Digest username="12",realm="asterisk",nonce="5fa099b4",uri="sip:123@192.168.0.25",response="27e636e8556778130daec79ac26fba75" Content-Length: 0 --- (9 headers 0 lines)--- Sending to 192.168.0.59 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.0.59:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.59:5060;received=192.168.0.59 From: ;tag=920185343 To: ;tag=as4843cf8d Call-ID: 2153418752@192.168.0.59 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Retransmitting #6 (no NAT) to 192.168.0.59:5060: CANCEL :123@192.168.0.25 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.25:5060;branch=z9hG4bK2b86a88a;rport From: To: ;tag=920185343 Contact: Call-ID: 2153418752@192.168.0.59 CSeq: 101 CANCEL User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Jul 4 12:49:25 WARNING[6786]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 2153418752@192.168.0.59 for seqno 101 (Non-critical Request) Destroying call '2153418752@192.168.0.59'