Sending to 63.123.133.45 : 5060 (no NAT) Using INVITE request as basis request - BW190417222170307-1435228656@63.123.133.45 Found peer '7001' Found RTP audio format 0 Found RTP audio format 18 Peer audio RTP is at port 209.244.187.88:60802 Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x104 (ulaw|g729)/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 209.244.187.88:60802 Looking for 7001 in incomingcalls (domain 64.81.140.205) list_route: hop: <--- Transmitting (no NAT) to 63.123.133.45:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 63.123.133.45;branch=z9hG4bK-BroadWorks.63.123.133.45-64.81.140.205V5060-0-826892708-278713361-1174179857222-;received=63.123.133.45 From: "Cell Phone IL";tag=278713361-1174179857222- To: "Larry Butterfield Elvin Yonan Tech Contact" Call-ID: BW190417222170307-1435228656@63.123.133.45 CSeq: 826892708 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [7001@incomingcalls:1] Answer("SIP/7737285106-40b01880", "") in new stack Audio is at 64.81.140.205 port 13822 Adding codec 0x4 (ulaw) to SDP <--- Reliably Transmitting (no NAT) to 63.123.133.45:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 63.123.133.45;branch=z9hG4bK-BroadWorks.63.123.133.45-64.81.140.205V5060-0-826892708-278713361-1174179857222-;received=63.123.133.45 From: "Cell Phone IL";tag=278713361-1174179857222- To: "Larry Butterfield Elvin Yonan Tech Contact";tag=as6fd63c67 Call-ID: BW190417222170307-1435228656@63.123.133.45 CSeq: 826892708 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 186 v=0 o=root 20944 20944 IN IP4 64.81.140.205 s=session c=IN IP4 64.81.140.205 t=0 0 m=audio 13822 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [7001@incomingcalls:2] Wait("SIP/7737285106-40b01880", "2") in new stack <--- SIP read from 63.123.133.45:33844 ---> ACK sip:7001@64.81.140.205 SIP/2.0 Via:SIP/2.0/UDP 63.123.133.45;branch=z9hG4bK-BroadWorks.63.123.133.45-64.81.140.205V5060-0-826892708A278713361-1174179857222- From:"Cell Phone IL";tag=278713361-1174179857222- To:"Larry Butterfield Elvin Yonan Tech Contact";tag=as6fd63c67 Call-ID:BW190417222170307-1435228656@63.123.133.45 CSeq:826892708 ACK Contact: Max-Forwards:10 Content-Length:0 <-------------> --- (9 headers 0 lines) --- [Mar 17 21:28:24] WARNING[20944]: pbx.c:1755 pbx_extension_helper: No application 'DigitTimeout' for extension (incomingcalls, 7001, 3) == Spawn extension (incomingcalls, 7001, 3) exited non-zero on 'SIP/7737285106-40b01880' Scheduling destruction of SIP dialog 'BW190417222170307-1435228656@63.123.133.45' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 63.123.133.45, port 5060 Reliably Transmitting (no NAT) to 63.123.133.45:5060: BYE sip:63.123.133.45:5060 SIP/2.0 Via: SIP/2.0/UDP 64.81.140.205:5060;branch=z9hG4bK245e84f0;rport From: "Larry Butterfield Elvin Yonan Tech Contact";tag=as6fd63c67 To: "Cell Phone IL";tag=278713361-1174179857222- Call-ID: BW190417222170307-1435228656@63.123.133.45 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- <--- SIP read from 63.123.133.45:33844 ---> SIP/2.0 200 OK Via:SIP/2.0/UDP 64.81.140.205:5060;branch=z9hG4bK245e84f0;rport From:"Larry Butterfield Elvin Yonan Tech Contact";tag=as6fd63c67 To:"Cell Phone IL";tag=278713361-1174179857222- Call-ID:BW190417222170307-1435228656@63.123.133.45 CSeq:102 BYE Content-Length:0