-- Accepting call from '7327041000' to '7327049020' on channel 0/1, span 1 -- Executing [7327049020@default:1] Macro("Zap/1-1", "writeincomingsipcallid") in new stack -- Executing [s@macro-writeincomingsipcallid:1] Set("Zap/1-1", "CDR(SIPCALLID1)=") in new stack [Nov 14 09:03:37] DEBUG[18998]: app_macro.c:337 _macro_exec: Executed application: Set -- Executing [7327049020@default:2] Goto("Zap/1-1", "incoming|7327049020|1") in new stack -- Goto (incoming,7327049020,1) -- Executing [7327049020@incoming:1] Dial("Zap/1-1", "SIP/7327049020@fsa-fsdev") in new stackAudio is at 64.19.145.13 port 11446Adding codec 0x4 (ulaw) to SDPReliably Transmitting (NAT) to 64.19.145.4:5060: INVITE sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK741278f1;rport From: "7327041000" ;tag=as7180474c To: Contact: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 14 Nov 2008 14:03:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 184 v=0 o=root 17371 17371 IN IP4 64.19.145.13 s=session c=IN IP4 64.19.145.13 t=0 0 m=audio 11446 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 7327049020@fsa-fsdev <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK741278f1;received=64.19.145.13;rport=5060 From: "7327041000" ;tag=as7180474c To: ;tag=as2bacdfea Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2312001b" Content-Length: 0 <------------->--- (11 headers 0 lines) ---Transmitting (NAT) to 64.19.145.4:5060: ACK sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK741278f1;rport From: "7327041000" ;tag=as7180474c To: ;tag=as2bacdfea Contact: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ---Audio is at 64.19.145.13 port 11446Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 64.19.145.4:5060: INVITE sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7b0d8297;rport From: "7327041000" ;tag=as7180474c To: Contact: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="fsdev", realm="asterisk", algorithm=MD5, uri="sip:7327049020@64.19.145.4", nonce="2312001b", response="07cb9db7895a40e19a0df37b63d14ee6" Date: Fri, 14 Nov 2008 14:03:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 184 v=0 o=root 17371 17372 IN IP4 64.19.145.13 s=session c=IN IP4 64.19.145.13 t=0 0 m=audio 11446 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ---sdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7b0d8297;received=64.19.145.13;rport=5060 From: "7327041000" ;tag=as7180474c To: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------->--- (11 headers 0 lines) ---sdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7b0d8297;received=64.19.145.13;rport=5060 From: "7327041000" ;tag=as7180474c To: ;tag=as3e45f899 Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------->--- (11 headers 0 lines) --- -- SIP/fsa-fsdev-0a1d7da8 is ringingsdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7b0d8297;received=64.19.145.13;rport=5060 From: "7327041000" ;tag=as7180474c To: ;tag=as3e45f899 Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 178 v=0 o=root 832 832 IN IP4 64.19.145.4 s=session c=IN IP4 64.19.145.4 t=0 0 m=audio 17798 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------->--- (12 headers 10 lines) ---Found RTP audio format 0Peer audio RTP is at port 64.19.145.4:17798Found audio description format PCMU for ID 0Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)Peer audio RTP is at port 64.19.145.4:17798list_route: hop: [Nov 14 09:03:39] DEBUG[17443]: chan_sip.c:6044 reqprep: Strict routing enforced for session 226ef2ce735857d5636fd6560bae4abf@64.19.145.13set_destination: Parsing for address/port to send toset_destination: set destination to 64.19.145.4, port 5060Transmitting (NAT) to 64.19.145.4:5060: ACK sip:7327049020@64.19.145.4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7b665e53;rport From: "7327041000" ;tag=as7180474c To: ;tag=as3e45f899 Contact: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 ---sdev*CLI> <--- SIP read from 64.19.145.4:5060 ---> INVITE sip:7327041000@64.19.145.13 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK1c6f90e0;rport From: ;tag=as3e45f899 To: "7327041000" ;tag=as7180474c Contact: Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 182 v=0 o=root 832 833 IN IP4 64.19.186.173 s=session c=IN IP4 64.19.186.173 t=0 0 m=audio 10000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------->--- (13 headers 10 lines) ---Sending to 64.19.145.4 : 5060 (NAT)Found RTP audio format 0Peer audio RTP is at port 64.19.186.173:10000Found audio description format PCMU for ID 0Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)Peer audio RTP is at port 64.19.186.173:10000 <--- Transmitting (NAT) to 64.19.145.4:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 64.19.145.4:5060;branch=z9hG4bK1c6f90e0;received=64.19.145.4;rport=5060 From: ;tag=as3e45f899 To: "7327041000" ;tag=as7180474c Call-ID: 226ef2ce735857d5636fd6560bae4abf@64.19.145.13 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- SIP/fsa-fsdev-0a1d7da8 answered Zap/1-1