Sip read: INVITE sip:0321;npdi@10.0.0.174;user=phone SIP/2.0 Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_28hx From: ;tag=1_1146_f49869_1547 To: Call-ID: 1512502930@callagent.adomain.local CSeq: 1 INVITE Max-Forwards: 70 Supported: 100rel,precondition,timer Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,REFER,UPDATE User-Agent: BTS10200/900-04.02.00.V04(SIA) Content-Length: 397 Content-Type: application/sdp v=0 o=- 46974 0 IN IP4 66.220.74.1 s=Cisco SDP 0 c=IN IP4 66.220.74.1 t=0 0 m=audio 17528 RTP/AVP 0 8 99 18 2 102 103 104 105 106 107 4 101 a=rtpmap:99 G.729b/8000 a=rtpmap:102 G.726-24/8000 a=rtpmap:103 G.726-16/8000 a=rtpmap:104 G.723.1-H/8000 a=rtpmap:105 G.723.1a-H/8000 a=rtpmap:106 G.723.1-L/8000 a=rtpmap:107 G.723.1a-L/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 15 lines Using latest request as basis request Sending to 192.168.20.6 : 5060 (non-NAT) Found no matching peer or user for '192.168.10.6:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 99 Found RTP audio format 18 Found RTP audio format 2 Found RTP audio format 102 Found RTP audio format 103 Found RTP audio format 104 Found RTP audio format 105 Found RTP audio format 106 Found RTP audio format 107 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 66.220.74.1:17528 Peer audio RTP is at port 66.220.74.1:17528 Found description format G.729b Found description format G.726-24 Found description format G.726-16 Found description format G.723.1-H Found description format G.723.1a-H Found description format G.723.1-L Found description format G.723.1a-L Found description format telephone-event Capabilities: us - 0x4 (ulaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Check for res for is not a local user Looking for 0321 in from-sip build_route: Contact hop: list_route: hop: Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_28hx From: ;tag=1_1146_f49869_1547 To: Call-ID: 1512502930@callagent.adomain.local CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.20.6:5060 -- Executing Dial("SIP/callagent.adomain.local-08125180", "SIP/103@10.0.0.170") in new stack Outgoing Call for 103 103 is not a local user We're at 10.0.0.174 port 16738 Answering/Requesting with root capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:103@10.0.0.170 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.174:5060;branch=z9hG4bK1b20aad2 From: "8649804718" ;tag=as2fb0d460 To: Contact: Call-ID: 2627e65927387d6c152bb3f929ece2ea@10.0.0.174 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 13 Apr 2005 22:06:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 216 v=0 o=root 2968 2968 IN IP4 10.0.0.174 s=session c=IN IP4 10.0.0.174 t=0 0 m=audio 16738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 10.0.0.170:5060 -- Called 103@10.0.0.170 Sip read: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.174:5060;branch=z9hG4bK1b20aad2 From: "8649804718" ;tag=as2fb0d460 To: ;tag=395B3A2C-1432 Date: Wed, 13 Apr 2005 22:06:31 GMT Call-ID: 2627e65927387d6c152bb3f929ece2ea@10.0.0.174 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 10 headers, 0 lines (Provisional) Stopping retransmission (but retaining packet) on '2627e65927387d6c152bb3f929ece2ea@10.0.0.174' Request 102: Found Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.0.0.174:5060;branch=z9hG4bK1b20aad2 From: "8649804718" ;tag=as2fb0d460 To: ;tag=395B3A2C-1432 Date: Wed, 13 Apr 2005 22:06:31 GMT Call-ID: 2627e65927387d6c152bb3f929ece2ea@10.0.0.174 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER Allow-Events: telephone-event Contact: Content-Length: 0 12 headers, 0 lines (Provisional) Stopping retransmission (but retaining packet) on '2627e65927387d6c152bb3f929ece2ea@10.0.0.174' Request 102: Found -- SIP/10.0.0.170-17b4 is ringing Transmitting (no NAT): SIP/2.0 180 Ringing Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_28hx From: ;tag=1_1146_f49869_1547 To: ;tag=as7ad7479a Call-ID: 1512502930@callagent.adomain.local CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.20.6:5060 Sip read: SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 10.0.0.174:5060;branch=z9hG4bK1b20aad2 From: "8649804718" ;tag=as2fb0d460 To: ;tag=395B3A2C-1432 Date: Wed, 13 Apr 2005 22:06:31 GMT Call-ID: 2627e65927387d6c152bb3f929ece2ea@10.0.0.174 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 INVITE Allow-Events: telephone-event Contact: , Diversion: ;reason=no-answer Content-Length: 0 12 headers, 0 lines Acked pending invite 102 Stopping retransmission on '2627e65927387d6c152bb3f929ece2ea@10.0.0.174' of Request 102: Found -- Got SIP response 300 "Multiple Choices" back from 10.0.0.170 Found 302 Redirect to extension '1002' Transmitting: ACK sip:103@10.0.0.170 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.174:5060;branch=z9hG4bK1b20aad2 From: "8649804718" ;tag=as2fb0d460 To: ;tag=395B3A2C-1432 Contact: Call-ID: 2627e65927387d6c152bb3f929ece2ea@10.0.0.174 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 10.0.0.170:5060 -- Now forwarding SIP/callagent.adomain.local-08125180 to 'Local/1002@from-sip' (thanks to SIP/10.0.0.170-17b4) update_user_counter(103) - decrement outUse counter 103 is not a local user -- Executing Answer("Local/1002@from-sip-0667,2", "") in new stack -- Local/1002@from-sip-0667,1 answered SIP/callagent.adomain.local-08125180 We're at 10.0.0.174 port 13578 Answering with preferred capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_28hx From: ;tag=1_1146_f49869_1547 To: ;tag=as7ad7479a Call-ID: 1512502930@callagent.adomain.local CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 2968 2968 IN IP4 10.0.0.174 s=session c=IN IP4 10.0.0.174 t=0 0 m=audio 13578 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - to 192.168.20.6:5060 -- Executing Wait("Local/1002@from-sip-0667,2", "2") in new stack Sip read: ACK sip:0321@10.0.0.174 SIP/2.0 Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_5nj1 From: ;tag=1_1146_f49869_1547 To: ;tag=as7ad7479a Call-ID: 1512502930@callagent.adomain.local CSeq: 1 ACK Max-Forwards: 70 Content-Length: 0 8 headers, 0 lines Stopping retransmission on '1512502930@callagent.adomain.local' of Response 1: Found Destroying call '2627e65927387d6c152bb3f929ece2ea@10.0.0.174' Planning to masquerade SIP/callagent.adomain.local-08125180 into the structure of Local/1002@from-sip-0667,2 Done planning to masquerade Local/1002@from-sip-0667,2 into the structure of SIP/callagent.adomain.local-08125180 Actually Masquerading SIP/callagent.adomain.local-08125180(6) into the structure of Local/1002@from-sip-0667,2(6) Got clone lock on 'SIP/callagent.adomain.local-08125180' at 0x812cd68 Putting channel SIP/callagent.adomain.local-08125180 in 4/4 formats Released clone lock on 'Local/1002@from-sip-0667,2' Didn't get a frame from channel: Local/1002@from-sip-0667,2 Bridge stops bridging channels Local/1002@from-sip-0667,2 and Local/1002@from-sip-0667,1 Exiting with DIALSTATUS=ANSWER. == Spawn extension (from-sip, 0321, 1) exited non-zero on 'Local/1002@from-sip-0667,2' Done Masquerading SIP/callagent.adomain.local-08125180 (6) -- Executing VoiceMail("SIP/callagent.adomain.local-08125180", "u0321") in new stack No entry in voicemail config file for '0321' Sip read: BYE sip:0321@10.0.0.174 SIP/2.0 Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_086a From: ;tag=1_1146_f49869_1547 To: ;tag=as7ad7479a Call-ID: 1512502930@callagent.adomain.local CSeq: 2 BYE Max-Forwards: 70 Content-Length: 0 8 headers, 0 lines Sending to 192.168.20.6 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP callagent.adomain.local:5060;branch=z9hG4bK_1146_086a From: ;tag=1_1146_f49869_1547 To: ;tag=as7ad7479a Call-ID: 1512502930@callagent.adomain.local CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0 to 192.168.20.6:5060 update_user_counter() - decrement inUse counter is not a local user Destroying call '1512502930@callagent.adomain.local'