Oct 18 11:17:57 VERBOSE[32180] logger.c: <-- SIP read from 10.1.16.13:5060: INVITE sip:7135795711@10.1.16.11;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac20675368 Max-Forwards: 70 From: "Stoner Sean " ;tag=1c20667331 To: Call-ID: 206666981512000233230@10.1.16.13 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE, UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.00A.024 Content-Type: application/sdp Content-Disposition: session Content-Length: 280 v=0 o=AudiocodesGW 20637400 20637133 IN IP4 10.1.16.13 s=Phone-Call c=IN IP4 10.1.16.13 t=0 0 m=audio 6300 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=rtcp:6301 IN IP4 10.1.16.13 Oct 18 11:17:57 VERBOSE[32180] logger.c: --- (14 headers 13 lines) --- Oct 18 11:17:57 VERBOSE[32180] logger.c: Using INVITE request as basis request - 206666981512000233230@10.1.16.13 Oct 18 11:17:57 VERBOSE[32180] logger.c: Sending to 10.1.16.13 : 5060 (non-NAT) Oct 18 11:17:57 VERBOSE[32180] logger.c: Found no matching peer or user for '10. 1.16.13:5060' Oct 18 11:17:57 VERBOSE[32180] logger.c: Found RTP audio format 0 Oct 18 11:17:57 VERBOSE[32180] logger.c: Found RTP audio format 8 Oct 18 11:17:57 VERBOSE[32180] logger.c: Found RTP audio format 101 Oct 18 11:17:57 VERBOSE[32180] logger.c: Peer audio RTP is at port 10.1.16.13:63 00 Oct 18 11:17:57 DEBUG[32180] chan_sip.c: Peer audio RTP is at port 10.1.16.13:63 00 Oct 18 11:17:57 VERBOSE[32180] logger.c: Found description format PCMU Oct 18 11:17:57 VERBOSE[32180] logger.c: Found description format PCMA Oct 18 11:17:57 VERBOSE[32180] logger.c: Found description format telephone-even t Oct 18 11:17:57 VERBOSE[32180] logger.c: Capabilities: us - 0xc (ulaw|alaw), pee r - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Oct 18 11:17:57 VERBOSE[32180] logger.c: Non-codec capabilities: us - 0x1 (telep hone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Oct 18 11:17:57 DEBUG[32180] chan_sip.c: Checking SIP call limits for device Oct 18 11:17:57 VERBOSE[32180] logger.c: Looking for 7135795711 in from-sip-exte rnal (domain 10.1.16.11;user=phone) Oct 18 11:17:57 DEBUG[32180] chan_sip.c: build_route: Contact hop: Oct 18 11:17:57 VERBOSE[32180] logger.c: list_route: hop: Oct 18 11:17:57 VERBOSE[32180] logger.c: Transmitting (no NAT) to 10.1.16.13:506 0: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac20675368;received=10.1.16.13 From: "Stoner Sean " ;tag=1c20667331 To: Call-ID: 206666981512000233230@10.1.16.13 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing NoOp("SIP/mediant-b725 b4c0", "Received incoming SIP connection from unknown peer to 7135795711") in ne w stack Oct 18 11:17:57 DEBUG[18326] pbx.c: Expression result is '0' Oct 18 11:17:57 DEBUG[18326] pbx.c: Function result is '7135795711' Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing Set("SIP/mediant-b725b 4c0", "DID=7135795711") in new stack Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing Goto("SIP/mediant-b725 b4c0", "s|1") in new stack Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Goto (from-sip-external,s,1) Oct 18 11:17:57 DEBUG[18326] pbx.c: Expression result is '0' Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing GotoIf("SIP/mediant-b7 25b4c0", "0?from-trunk|7135795711|1") in new stack Oct 18 11:17:57 DEBUG[18326] pbx.c: Not taking any branch Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing Set("SIP/mediant-b725b 4c0", "TIMEOUT(absolute)=15") in new stack Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Channel will hangup at 2007-10-1 8 16:18:12 UTC. Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing Answer("SIP/mediant-b7 25b4c0", "") in new stack Oct 18 11:17:57 VERBOSE[18326] logger.c: We're at 10.1.16.11 port 10210 Oct 18 11:17:57 VERBOSE[18326] logger.c: Adding codec 0x4 (ulaw) to SDP Oct 18 11:17:57 VERBOSE[18326] logger.c: Adding codec 0x8 (alaw) to SDP Oct 18 11:17:57 VERBOSE[18326] logger.c: Adding non-codec 0x1 (telephone-event) to SDP Oct 18 11:17:57 VERBOSE[18326] logger.c: Reliably Transmitting (no NAT) to 10.1. 16.13:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac20675368;received=10.1.16.13 From: "Stoner Sean " ;tag=1c20667331 To: ;tag=as4a20a22e Call-ID: 206666981512000233230@10.1.16.13 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 236 v=0 o=root 32104 32104 IN IP4 10.1.16.11 s=session c=IN IP4 10.1.16.11 t=0 0 m=audio 10210 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Oct 18 11:17:57 VERBOSE[18326] logger.c: -- Executing Wait("SIP/mediant-b725 b4c0", "2") in new stack Oct 18 11:17:57 VERBOSE[32180] logger.c: <-- SIP read from 10.1.16.13:5060: ACK sip:7135795711@10.1.16.11 SIP/2.0 Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac21102191 Max-Forwards: 70 From: "Stoner Sean " ;tag=1c20667331 To: ;tag=as4a20a22e Call-ID: 206666981512000233230@10.1.16.13 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE, UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.00A.024 Content-Length: 0 Oct 18 11:17:59 VERBOSE[18326] logger.c: -- Executing Playback("SIP/mediant- b725b4c0", "ss-noservice") in new stack Oct 18 11:17:59 DEBUG[18326] channel.c: Scheduling timer at 160 sample intervals Oct 18 11:17:59 VERBOSE[18326] logger.c: -- Playing 'ss-noservice' (language 'en') Oct 18 11:18:02 VERBOSE[32180] logger.c: --- (10 headers 0 lines) --- Oct 18 11:18:02 DEBUG[32180] chan_sip.c: Stopping retransmission on '2606d23628d dbba253af2a7473efb8bf@10.1.16.11' of Request 102: Match Found Oct 18 11:18:02 VERBOSE[32180] logger.c: Destroying call '2606d23628ddbba253af2a 7473efb8bf@10.1.16.11' Oct 18 11:18:02 VERBOSE[32180] logger.c: <-- SIP read from 10.1.16.13:5060: BYE sip:7135795711@10.1.16.11 SIP/2.0 Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac94867495 Max-Forwards: 70 From: "Stoner Sean " ;tag=1c20667331 To: ;tag=as4a20a22e Call-ID: 206666981512000233230@10.1.16.13 CSeq: 2 BYE Supported: em,timer,replaces,path,early-session,resource-priority Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE, UPDATE User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.00A.024 Reason: Q.850 ;cause=16 Content-Length: 0 Oct 18 11:18:02 VERBOSE[32180] logger.c: --- (12 headers 0 lines) --- Oct 18 11:18:02 VERBOSE[32180] logger.c: Sending to 10.1.16.13 : 5060 (non-NAT) Oct 18 11:18:02 VERBOSE[32180] logger.c: Transmitting (no NAT) to 10.1.16.13:506 0: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.1.16.13;branch=z9hG4bKac94867495;received=10.1.16.13 From: "Stoner Sean " ;tag=1c20667331 To: ;tag=as4a20a22e Call-ID: 206666981512000233230@10.1.16.13 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Oct 18 11:18:02 DEBUG[18326] channel.c: Scheduling timer at 0 sample intervals Oct 18 11:18:02 VERBOSE[18326] logger.c: == Spawn extension (from-sip-external , s, 5) exited non-zero on 'SIP/mediant-b725b4c0' Oct 18 11:18:02 VERBOSE[18326] logger.c: -- Executing NoOp("SIP/mediant-b725 b4c0", "Hangup") in new stack