localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> <--- SIP read from 192.168.1.8:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> <--- SIP read from 192.168.1.8:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'uccgserv' logged on from 192.168.1.20 Audio is at 192.168.1.55 port 15964 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.1.21:5060 : INVITE sip:1000@192.168.1.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport From: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343 To: Contact: Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 04 Apr 2007 18:00:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 251 v=0 o=root 26722 26722 IN IP4 192.168.1.55 s=session c=IN IP4 192.168.1.55 t=0 0 m=audio 15964 RTP/AVP 3 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=sendrecv --- localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport From: "1000" ;tag=as7cdeb343 To: Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.17 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- SIP read from 192.168.1.21:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport From: "1000" ;tag=as7cdeb343 To: < sip:1000@192.168.1.21>;tag=b8682c6199346e89 Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.17 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- == Parsing '/etc/asterisk/manager.conf': Found == Manager 'uccgserv' logged on from 192.168.1.20 == Manager 'uccgserv' logged off from 192.168.1.20 localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK0cf88385;rport From: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343 To: ;tag=b8682c6199346e89 Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 102 INVITE User-Agent: Grandstream BT120 1.0.8.17 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Supported: replaces Content-Length: 155 v=0 o=1000 8000 8000 IN IP4 192.168.1.21 s=SIP Call c=IN IP4 192.168.1.21 t=0 0 m=audio 5004 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 <-------------> --- (12 headers 9 lines) --- Found RTP audio format 0 Peer audio RTP is at port 192.168.1.21:5004 Found description format PCMU for ID 0 Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.21:5004 list_route: hop: < sip:1000@192.168.1.21> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.21, port 5060 Transmitting (no NAT) to 192.168.1.21:5060: ACK sip:1000@192.168.1.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK4a3f2985;rport From: "1000" ;tag=as7cdeb343 To: ;tag=b8682c6199346e89 Contact: Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- > Channel SIP/1000-087a05b8 was answered. == Manager 'uccgserv' logged off from 192.168.1.20 == Starting SIP/1000-087a05b8 at outbound,2000,1 failed so falling back to exten 's' == Starting SIP/1000-087a05b8 at outbound,s,1 still failed so falling back to context 'default' [Apr 4 14:00:39] WARNING[27497]: pbx.c:2405 __ast_pbx_run: Channel 'SIP/1000-087a05b8' sent into invalid extension 's' in context 'default', but no invalid handler Scheduling destruction of SIP dialog ' 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55' in 32000 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.1.21, port 5060 Reliably Transmitting (no NAT) to 192.168.1.21:5060 : BYE sip:1000@192.168.1.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.55:5060;branch=z9hG4bK237c155b;rport rom: "1000" < sip:1000@192.168.1.55>;tag=as7cdeb343 To: ;tag=b8682c6199346e89 Contact: < sip:1000@192.168.1.55> Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.55:5060 ;branch=z9hG4bK237c155b;rport From: "1000" ;tag=as7cdeb343 To: ;tag=b8682c6199346e89 Call-ID: 3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55 CSeq: 103 BYE User-Agent: Grandstream BT120 1.0.8.17 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3c028fcd79ef17be2710bf9b626bf7a5@192.168.1.55' Method: INVITE localhost*CLI> <--- SIP read from 192.168.1.21:5060 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- localhost*CLI> <--- SIP read from 192.168.1.8:5060 --->