[May 27 12:29:28] --- (0 headers 0 lines) Nat keepalive --- [May 27 12:29:32] <--- SIP read from 10.65.0.203:5060 ---> INVITE sip:569@10.0.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-gfdzfsz7b049;rport From: "4903" ;tag=n63i67lmv4 To: Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 1 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 364 v=0 o=root 144451695 144451695 IN IP4 10.65.0.203 s=call c=IN IP4 10.65.0.203 t=0 0 m=audio 53192 RTP/AVP 18 8 0 3 9 2 4 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [May 27 12:29:32] --- (18 headers 17 lines) --- [May 27 12:29:32] Sending to 10.65.0.203 : 5060 (NAT) [May 27 12:29:32] Using INVITE request as basis request - 3c3111252fab-hxxqgpdtvohc [May 27 12:29:32] <--- Reliably Transmitting (no NAT) to 10.65.0.203:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-gfdzfsz7b049;received=10.65.0.203;rport=5060 From: "4903" ;tag=n63i67lmv4 To: ;tag=as642e6123 Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="521dde3a" Content-Length: 0 <------------> [May 27 12:29:32] Scheduling destruction of SIP dialog '3c3111252fab-hxxqgpdtvohc' in 32000 ms (Method: INVITE) [May 27 12:29:32] Found user '4903' [May 27 12:29:32] <--- SIP read from 10.65.0.203:5060 ---> ACK sip:569@10.0.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-gfdzfsz7b049;rport From: "4903" ;tag=n63i67lmv4 To: ;tag=as642e6123 Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 1 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [May 27 12:29:32] --- (9 headers 0 lines) --- [May 27 12:29:32] <--- SIP read from 10.65.0.203:5060 ---> INVITE sip:569@10.0.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-5yoyewzabedp;rport From: "4903" ;tag=n63i67lmv4 To: Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 2 INVITE Max-Forwards: 70 Contact: ;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.1.30 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="4903",realm="asterisk",nonce="521dde3a",uri="sip:569@10.0.10.1;user=phone",response="9bf614e32733075d907b67453b2652a6",algorithm=MD5 Content-Type: application/sdp Content-Length: 364 v=0 o=root 144451695 144451695 IN IP4 10.65.0.203 s=call c=IN IP4 10.65.0.203 t=0 0 m=audio 53192 RTP/AVP 18 8 0 3 9 2 4 101 a=rtpmap:18 g729/8000 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:3 gsm/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [May 27 12:29:32] --- (19 headers 17 lines) --- [May 27 12:29:32] Sending to 10.65.0.203 : 5060 (NAT) [May 27 12:29:32] Using INVITE request as basis request - 3c3111252fab-hxxqgpdtvohc [May 27 12:29:32] Found user '4903' [May 27 12:29:32] Found RTP audio format 18 [May 27 12:29:32] Found RTP audio format 8 [May 27 12:29:32] Found RTP audio format 0 [May 27 12:29:32] Found RTP audio format 3 [May 27 12:29:32] Found RTP audio format 9 [May 27 12:29:32] Found RTP audio format 2 [May 27 12:29:32] Found RTP audio format 4 [May 27 12:29:32] Found RTP audio format 101 [May 27 12:29:32] Peer audio RTP is at port 10.65.0.203:53192 [May 27 12:29:32] Found audio description format g729 for ID 18 [May 27 12:29:32] Found audio description format pcma for ID 8 [May 27 12:29:32] Found audio description format pcmu for ID 0 [May 27 12:29:32] Found audio description format gsm for ID 3 [May 27 12:29:32] Found audio description format g722 for ID 9 [May 27 12:29:32] Found audio description format g726-32 for ID 2 [May 27 12:29:32] Found audio description format g723 for ID 4 [May 27 12:29:32] Found audio description format telephone-event for ID 101 [May 27 12:29:32] Got unsupported a:fmtp in SDP offer [May 27 12:29:32] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x190f (g723|gsm|ulaw|alaw|g726|g729|g722)/video=0x0 (nothing), combined - 0x10e (gsm|ulaw|alaw|g729) [May 27 12:29:32] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 27 12:29:32] Peer audio RTP is at port 10.65.0.203:53192 [May 27 12:29:32] Looking for 569 in clbr (domain 10.0.10.1) [May 27 12:29:32] Reliably Transmitting (no NAT) to 10.65.0.244:5060: NOTIFY sip:4900@10.65.0.244:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.10.1:5060;branch=z9hG4bK57b611e7;rport From: ;tag=as24b359f9 To: ;tag=nljsnba270 Contact: Call-ID: 3c267018b42c-v2pd049uveva CSeq: 120 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: dialog Content-Type: application/dialog-info+xml Subscription-State: active Content-Length: 206 confirmed --- [May 27 12:29:32] Extension Changed 4903[clbr] new state InUse for Notify User 4900 (queued) [May 27 12:29:32] <--- Reliably Transmitting (no NAT) to 10.65.0.203:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-5yoyewzabedp;received=10.65.0.203;rport=5060 From: "4903" ;tag=n63i67lmv4 To: ;tag=as642e6123 Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [May 27 12:29:32] NOTICE[16784]: chan_sip.c:13968 handle_request_invite: Call from '4903' to extension '569' rejected because extension not found. [May 27 12:29:32] Scheduling destruction of SIP dialog '3c3111252fab-hxxqgpdtvohc' in 32000 ms (Method: INVITE) [May 27 12:29:32] Extension Changed 4903[clbr] new state Idle for Notify User 4900 (queued) [May 27 12:29:32] <--- SIP read from 10.65.0.244:5060 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.0.10.1:5060;branch=z9hG4bK57b611e7;rport=5060 From: ;tag=as24b359f9 To: ;tag=nljsnba270 Call-ID: 3c267018b42c-v2pd049uveva CSeq: 120 NOTIFY Content-Length: 0 <-------------> [May 27 12:29:32] --- (7 headers 0 lines) --- [May 27 12:29:32] SIP Response message for INCOMING dialog NOTIFY arrived [May 27 12:29:32] <--- SIP read from 10.65.0.203:5060 ---> ACK sip:569@10.0.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.65.0.203:5060;branch=z9hG4bK-5yoyewzabedp;rport From: "4903" ;tag=n63i67lmv4 To: ;tag=as642e6123 Call-ID: 3c3111252fab-hxxqgpdtvohc CSeq: 2 ACK Max-Forwards: 70 Contact: ;flow-id=1 Content-Length: 0 <-------------> [May 27 12:29:32] --- (9 headers 0 lines) --- [May 27 12:29:33] <--- SIP read from 10.65.0.203:5060 ---> <-------------> [May 27 12:29:33] --- (0 headers 0 lines) Nat keepalive ---