[2009-09-03 06:20:43.9191] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452614352 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 1 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="480e35a0",uri="sip:22600@192.168.1.93",algorithm=MD5,response="dca7ca176bed2b811e77fc652bbbbfec" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:43.9199] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:43.9207] VERBOSE[22938] netsock.c: == Using SIP RTP TOS bits 24 [2009-09-03 06:20:43.9208] VERBOSE[22938] netsock.c: == Using SIP RTP CoS mark 5 [2009-09-03 06:20:43.9208] VERBOSE[22938] netsock.c: == Using UDPTL TOS bits 24 [2009-09-03 06:20:43.9209] VERBOSE[22938] netsock.c: == Using UDPTL CoS mark 5 [2009-09-03 06:20:43.9209] DEBUG[22938] chan_sip.c: Setting NAT on RTP to Off [2009-09-03 06:20:43.9210] DEBUG[22938] chan_sip.c: Setting NAT on UDPTL to Off [2009-09-03 06:20:43.9210] DEBUG[22938] chan_sip.c: Allocating new SIP dialog for 145261139211200002124@192.168.0.37 - INVITE (With RTP) [2009-09-03 06:20:43.9211] VERBOSE[22938] chan_sip.c: Sending to 192.168.0.37 : 5060 (no NAT) [2009-09-03 06:20:43.9212] DEBUG[22938] chan_sip.c: Initializing initreq for method INVITE - callid 145261139211200002124@192.168.0.37 [2009-09-03 06:20:43.9212] VERBOSE[22938] chan_sip.c: Using INVITE request as basis request - 145261139211200002124@192.168.0.37 [2009-09-03 06:20:43.9213] VERBOSE[22938] chan_sip.c: Found peer '90027' for '90027' from 192.168.0.37:5060 [2009-09-03 06:20:43.9213] DEBUG[22938] chan_sip.c: Setting NAT on RTP to On [2009-09-03 06:20:43.9214] DEBUG[22938] chan_sip.c: Setting NAT on UDPTL to On [2009-09-03 06:20:43.9214] VERBOSE[22938] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452614352;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as50a2f89b Call-ID: 145261139211200002124@192.168.0.37 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20e38f9e" Content-Length: 0 <------------> [2009-09-03 06:20:43.9215] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 40' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:43.9216] VERBOSE[22938] chan_sip.c: Scheduling destruction of SIP dialog '145261139211200002124@192.168.0.37' in 32000 ms (Method: INVITE) [2009-09-03 06:20:43.9378] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> ACK sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452614352 Max-Forwards: 70 From: ;tag=1c1452611704 To: ;tag=as50a2f89b Call-ID: 145261139211200002124@192.168.0.37 CSeq: 1 ACK Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Length: 0 <-------------> [2009-09-03 06:20:43.9383] VERBOSE[22938] chan_sip.c: --- (12 headers 0 lines) --- [2009-09-03 06:20:43.9387] DEBUG[22938] chan_sip.c: Stopping retransmission on '145261139211200002124@192.168.0.37' of Response 1: Match Found [2009-09-03 06:20:43.9426] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:43.9432] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:43.9436] VERBOSE[22938] chan_sip.c: Sending to 192.168.0.37 : 5060 (NAT) [2009-09-03 06:20:43.9439] DEBUG[22938] chan_sip.c: Initializing initreq for method INVITE - callid 145261139211200002124@192.168.0.37 [2009-09-03 06:20:43.9440] VERBOSE[22938] chan_sip.c: Using INVITE request as basis request - 145261139211200002124@192.168.0.37 [2009-09-03 06:20:43.9442] VERBOSE[22938] chan_sip.c: Found peer '90027' for '90027' from 192.168.0.37:5060 [2009-09-03 06:20:43.9445] DEBUG[22938] chan_sip.c: Setting NAT on RTP to On [2009-09-03 06:20:43.9446] DEBUG[22938] chan_sip.c: Setting NAT on UDPTL to On [2009-09-03 06:20:43.9448] VERBOSE[22938] chan_sip.c: Found RTP audio format 0 [2009-09-03 06:20:43.9451] VERBOSE[22938] chan_sip.c: Found RTP audio format 8 [2009-09-03 06:20:43.9453] VERBOSE[22938] chan_sip.c: Found RTP audio format 18 [2009-09-03 06:20:43.9456] VERBOSE[22938] chan_sip.c: Found RTP audio format 96 [2009-09-03 06:20:43.9459] DEBUG[22938] chan_sip.c: Peer doesn't provide T.38 UDPTL [2009-09-03 06:20:43.9460] VERBOSE[22938] chan_sip.c: Peer audio RTP is at port 192.168.0.37:6000 [2009-09-03 06:20:43.9462] VERBOSE[22938] chan_sip.c: Found audio description format PCMU for ID 0 [2009-09-03 06:20:43.9465] VERBOSE[22938] chan_sip.c: Found audio description format PCMA for ID 8 [2009-09-03 06:20:43.9468] VERBOSE[22938] chan_sip.c: Found audio description format G729 for ID 18 [2009-09-03 06:20:43.9470] VERBOSE[22938] chan_sip.c: Found audio description format telephone-event for ID 96 [2009-09-03 06:20:43.9473] VERBOSE[22938] chan_sip.c: Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-09-03 06:20:43.9476] VERBOSE[22938] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-03 06:20:43.9479] VERBOSE[22938] chan_sip.c: Peer audio RTP is at port 192.168.0.37:6000 [2009-09-03 06:20:43.9481] DEBUG[22938] chan_sip.c: Checking SIP call limits for device 90027 [2009-09-03 06:20:43.9486] VERBOSE[22938] chan_sip.c: Looking for 22600 in default-sip (domain 192.168.1.93) [2009-09-03 06:20:43.9490] VERBOSE[22938] chan_sip.c: list_route: hop: [2009-09-03 06:20:43.9494] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:43.9499] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:43.9509] DEBUG[23000] pbx.c: Launching 'NoOp' [2009-09-03 06:20:43.9510] VERBOSE[23000] pbx.c: -- Executing [22600@default-sip:1] NoOp("SIP/90027-0a502a50", "-(B)------------------------------------- context default_from_local ---------------") in new stack ... [2009-09-03 06:20:44.2840] DEBUG[23000] pbx.c: Launching 'Dial' [2009-09-03 06:20:44.2841] VERBOSE[23000] pbx.c: -- Executing [22600@local_dial:87] Dial("SIP/90027-0a502a50", "SIP/90028,30,gtiU(agent_call_answer^22600)") in new stack [2009-09-03 06:20:44.2845] DEBUG[23000] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw) [2009-09-03 06:20:44.2850] VERBOSE[23000] netsock.c: == Using SIP RTP TOS bits 24 [2009-09-03 06:20:44.2854] VERBOSE[23000] netsock.c: == Using SIP RTP CoS mark 5 [2009-09-03 06:20:44.2856] VERBOSE[23000] netsock.c: == Using UDPTL TOS bits 24 [2009-09-03 06:20:44.2859] VERBOSE[23000] netsock.c: == Using UDPTL CoS mark 5 [2009-09-03 06:20:44.2863] DEBUG[23000] chan_sip.c: Allocating new SIP dialog for 25879a5d598557802d70330b30313796@127.0.0.1 - INVITE (With RTP) [2009-09-03 06:20:44.2864] DEBUG[23000] chan_sip.c: Setting NAT on RTP to On [2009-09-03 06:20:44.2865] DEBUG[23000] chan_sip.c: Setting NAT on UDPTL to On [2009-09-03 06:20:44.2866] DEBUG[23000] chan_sip.c: OBPROXY: Not applying OBproxy to this call ... variables ... [2009-09-03 06:20:44.3005] DEBUG[23000] chan_sip.c: Outgoing Call for 90028 [2009-09-03 06:20:44.3011] DEBUG[23000] chan_sip.c: ** Our capability: 0xc7f9afe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140) Video flag: False Text flag: False [2009-09-03 06:20:44.3012] DEBUG[23000] chan_sip.c: ** Our prefcodec: 0x4 (ulaw) [2009-09-03 06:20:44.3013] VERBOSE[23000] chan_sip.c: Audio is at 192.168.1.93 port 48238 [2009-09-03 06:20:44.3018] VERBOSE[23000] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:44.3021] VERBOSE[23000] chan_sip.c: Adding codec 0x2 (gsm) to SDP [2009-09-03 06:20:44.3024] VERBOSE[23000] chan_sip.c: Adding codec 0x8 (alaw) to SDP [2009-09-03 06:20:44.3027] VERBOSE[23000] chan_sip.c: Adding codec 0x10 (g726aal2) to SDP [2009-09-03 06:20:44.3030] VERBOSE[23000] chan_sip.c: Adding codec 0x20 (adpcm) to SDP [2009-09-03 06:20:44.3033] VERBOSE[23000] chan_sip.c: Adding codec 0x40 (slin) to SDP [2009-09-03 06:20:44.3036] VERBOSE[23000] chan_sip.c: Adding codec 0x80 (lpc10) to SDP [2009-09-03 06:20:44.3069] VERBOSE[23000] chan_sip.c: Adding codec 0x200 (speex) to SDP [2009-09-03 06:20:44.3072] VERBOSE[23000] chan_sip.c: Adding codec 0x800 (g726) to SDP [2009-09-03 06:20:44.3075] VERBOSE[23000] chan_sip.c: Adding codec 0x1000 (g722) to SDP [2009-09-03 06:20:44.3078] VERBOSE[23000] chan_sip.c: Adding codec 0x8000 (slin16) to SDP [2009-09-03 06:20:44.3081] VERBOSE[23000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:44.3084] DEBUG[23000] chan_sip.c: Initializing initreq for method INVITE - callid 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 [2009-09-03 06:20:44.3085] VERBOSE[23000] chan_sip.c: Reliably Transmitting (NAT) to 192.168.0.37:5060: INVITE sip:90028@192.168.0.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bK6e2651d3;rport Max-Forwards: 70 From: "Victor Negru" ;tag=as0adae4d8 To: Contact: Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.5 Date: Thu, 03 Sep 2009 13:20:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 500 v=0 o=root 563256592 563256592 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 48238 RTP/AVP 0 3 8 112 5 10 7 110 111 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:112 AAL2-G726-32/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:10 L16/8000 a=rtpmap:7 LPC/8000 a=rtpmap:110 speex/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:44.3093] DEBUG[23000] chan_sip.c: Trying to put 'INVITE sip' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:44.3095] VERBOSE[23000] app_dial.c: -- Called 90028 [2009-09-03 06:20:44.3534] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bK6e2651d3;rport From: "Victor Negru" ;tag=as0adae4d8 To: ;tag=1c1453140395 Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 102 INVITE Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Length: 0 <-------------> [2009-09-03 06:20:44.3535] VERBOSE[22938] chan_sip.c: --- (10 headers 0 lines) --- [2009-09-03 06:20:44.3537] DEBUG[22938] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '43cb19045a7789b61c5f9e010cabf05e@192.168.1.93' Request 102: Found [2009-09-03 06:20:44.3587] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bK6e2651d3;rport From: "Victor Negru" ;tag=as0adae4d8 To: ;tag=1c1453140395 Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Length: 0 <-------------> [2009-09-03 06:20:44.3588] VERBOSE[22938] chan_sip.c: --- (11 headers 0 lines) --- [2009-09-03 06:20:44.3589] DEBUG[22938] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '43cb19045a7789b61c5f9e010cabf05e@192.168.1.93' Request 102: Found [2009-09-03 06:20:44.3598] VERBOSE[23000] app_dial.c: -- SIP/90028-0a4d3480 is ringing [2009-09-03 06:20:44.3603] VERBOSE[23000] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:44.3608] DEBUG[23000] chan_sip.c: Trying to put 'SIP/2.0 18' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:44.4552] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:44.4553] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:44.4554] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:44.4555] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:44.4556] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:45.4552] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:45.4553] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:45.4554] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:45.4555] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:45.4556] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:45.6617] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bK6e2651d3;rport From: "Victor Negru" ;tag=as0adae4d8 To: ;tag=1c1453140395 Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 102 INVITE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 233 v=0 o=AudiocodesGW 1453154429 1453154345 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6010 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:45.6618] VERBOSE[22938] chan_sip.c: --- (12 headers 11 lines) --- [2009-09-03 06:20:45.6619] DEBUG[22938] chan_sip.c: Acked pending invite 102 [2009-09-03 06:20:45.6620] DEBUG[22938] chan_sip.c: Stopping retransmission on '43cb19045a7789b61c5f9e010cabf05e@192.168.1.93' of Request 102: Match Found [2009-09-03 06:20:45.6621] VERBOSE[22938] chan_sip.c: Found RTP audio format 0 [2009-09-03 06:20:45.6622] VERBOSE[22938] chan_sip.c: Found RTP audio format 101 [2009-09-03 06:20:45.6633] DEBUG[22938] chan_sip.c: Peer doesn't provide T.38 UDPTL [2009-09-03 06:20:45.6634] VERBOSE[22938] chan_sip.c: Peer audio RTP is at port 192.168.0.37:6010 [2009-09-03 06:20:45.6635] VERBOSE[22938] chan_sip.c: Found audio description format PCMU for ID 0 [2009-09-03 06:20:45.6635] VERBOSE[22938] chan_sip.c: Found audio description format telephone-event for ID 101 [2009-09-03 06:20:45.6636] VERBOSE[22938] chan_sip.c: Capabilities: us - 0xc7f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [2009-09-03 06:20:45.6636] VERBOSE[22938] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [2009-09-03 06:20:45.6637] VERBOSE[22938] chan_sip.c: Peer audio RTP is at port 192.168.0.37:6010 [2009-09-03 06:20:45.6638] VERBOSE[22938] chan_sip.c: list_route: hop: [2009-09-03 06:20:45.6639] DEBUG[22938] chan_sip.c: Strict routing enforced for session 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 [2009-09-03 06:20:45.6640] VERBOSE[22938] chan_sip.c: set_destination: Parsing for address/port to send to [2009-09-03 06:20:45.6640] VERBOSE[22938] chan_sip.c: set_destination: set destination to 192.168.0.37, port 5060 [2009-09-03 06:20:45.6641] VERBOSE[22938] chan_sip.c: Transmitting (NAT) to 192.168.0.37:5060: ACK sip:90028@192.168.0.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.93:5060;branch=z9hG4bK42f95319;rport Max-Forwards: 70 From: "Victor Negru" ;tag=as0adae4d8 To: ;tag=1c1453140395 Contact: Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.5 Content-Length: 0 --- [2009-09-03 06:20:45.6641] DEBUG[22938] chan_sip.c: Trying to put 'ACK sip:90' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:45.6663] VERBOSE[23000] app_dial.c: -- SIP/90028-0a4d3480 answered SIP/90027-0a502a50 [2009-09-03 06:20:45.6664] DEBUG[23000] app_stack.c: Channel SIP/90028-0a4d3480 has no datastore, so we're allocating one. [2009-09-03 06:20:45.6666] DEBUG[23000] app_stack.c: Setting 'ARG1' to '22600' [2009-09-03 06:20:45.6667] DEBUG[23000] pbx.c: Launching 'Set' [2009-09-03 06:20:45.6667] VERBOSE[23000] pbx.c: -- Executing [s@agent_call_answer:1] Set("SIP/90028-0a4d3480", "LOCAL(agent_answered)=22600") in new stack [2009-09-03 06:20:45.6668] DEBUG[23000] pbx.c: Launching 'NoOp' [2009-09-03 06:20:45.6669] VERBOSE[23000] pbx.c: -- Executing [s@agent_call_answer:2] NoOp("SIP/90028-0a4d3480", "--(B)--------------------------- agent_call_answer(22600)------------------------") in new stack ... dialplan ... [2009-09-03 06:20:45.7419] DEBUG[23000] pbx.c: Launching 'Return' [2009-09-03 06:20:45.7420] VERBOSE[23000] pbx.c: -- Executing [s@agent_call_answer:14] Return("SIP/90028-0a4d3480", "") in new stack [2009-09-03 06:20:45.7424] DEBUG[23000] pbx.c: Launching 'NoOp' [2009-09-03 06:20:45.7425] VERBOSE[23000] pbx.c: -- Executing [s@app_dial_gosub_virtual_context:1] NoOp("SIP/90028-0a4d3480", "") in new stack [2009-09-03 06:20:45.7429] VERBOSE[23000] pbx.c: -- Auto fallthrough, channel 'SIP/90028-0a4d3480' status is 'UNKNOWN' [2009-09-03 06:20:45.7432] DEBUG[23000] app_dial.c: Gosub exited with status 0 [2009-09-03 06:20:45.7439] DEBUG[23000] chan_sip.c: SIP answering channel: SIP/90027-0a502a50 [2009-09-03 06:20:45.7439] DEBUG[23000] chan_sip.c: Setting framing from config on incoming call [2009-09-03 06:20:45.7440] DEBUG[23000] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2009-09-03 06:20:45.7441] DEBUG[23000] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-09-03 06:20:45.7441] VERBOSE[23000] chan_sip.c: Audio is at 192.168.1.93 port 43820 [2009-09-03 06:20:45.7445] VERBOSE[23000] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:45.7448] VERBOSE[23000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:45.7451] VERBOSE[23000] chan_sip.c: <--- Reliably Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-03 06:20:45.7476] DEBUG[23000] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:45.7481] DEBUG[23000] rtp.c: Got RTCP report of 44 bytes [2009-09-03 06:20:45.7483] DEBUG[23000] rtp.c: Ooh, format changed from unknown to ulaw [2009-09-03 06:20:45.7485] DEBUG[23000] rtp.c: Created smoother: format: 4 ms: 20 len: 160 [2009-09-03 06:20:46.7469] VERBOSE[22938] chan_sip.c: Retransmitting #1 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:46.7477] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:47.4550] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:47.4558] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:47.4562] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:47.4566] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:47.4571] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:47.4573] DEBUG[22938] chan_sip.c: Setting framing from config on incoming call [2009-09-03 06:20:47.4574] DEBUG[22938] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2009-09-03 06:20:47.4575] DEBUG[22938] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-09-03 06:20:47.4576] VERBOSE[22938] chan_sip.c: Audio is at 192.168.1.93 port 43820 [2009-09-03 06:20:47.4580] VERBOSE[22938] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:47.4583] VERBOSE[22938] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:47.4586] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482281 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-03 06:20:47.4592] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:47.7476] VERBOSE[22938] chan_sip.c: Retransmitting #2 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:47.7483] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:49.7469] VERBOSE[22938] chan_sip.c: Retransmitting #3 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:49.7476] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:51.4547] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:51.4548] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:51.4549] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:51.4550] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:51.4551] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:51.4552] DEBUG[22938] chan_sip.c: Setting framing from config on incoming call [2009-09-03 06:20:51.4553] DEBUG[22938] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2009-09-03 06:20:51.4553] DEBUG[22938] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-09-03 06:20:51.4554] VERBOSE[22938] chan_sip.c: Audio is at 192.168.1.93 port 43820 [2009-09-03 06:20:51.4554] VERBOSE[22938] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:51.4555] VERBOSE[22938] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:51.4555] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482282 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-03 06:20:51.4556] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:51.5467] DEBUG[23000] rtp.c: Got RTCP report of 44 bytes [2009-09-03 06:20:53.7466] VERBOSE[22938] chan_sip.c: Retransmitting #4 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:53.7467] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:55.4544] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:55.4545] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:55.4546] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:55.4547] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:55.4548] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:55.4549] DEBUG[22938] chan_sip.c: Setting framing from config on incoming call [2009-09-03 06:20:55.4550] DEBUG[22938] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2009-09-03 06:20:55.4550] DEBUG[22938] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-09-03 06:20:55.4551] VERBOSE[22938] chan_sip.c: Audio is at 192.168.1.93 port 43820 [2009-09-03 06:20:55.4551] VERBOSE[22938] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:55.4552] VERBOSE[22938] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:55.4552] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482283 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-03 06:20:55.4553] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:56.5261] DEBUG[23000] rtp.c: Got RTCP report of 44 bytes [2009-09-03 06:20:57.7474] VERBOSE[22938] chan_sip.c: Retransmitting #5 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:20:57.7475] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:59.4544] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> INVITE sip:22600@192.168.1.93;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973 Max-Forwards: 70 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Contact: Supported: em,100rel,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90027",realm="asterisk",nonce="20e38f9e",uri="sip:22600@192.168.1.93",algorithm=MD5,response="567f5224397d621284046542967bfa88" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Content-Type: application/sdp Content-Length: 301 v=0 o=AudiocodesGW 1452607236 1452607151 IN IP4 192.168.0.37 s=Phone-Call c=IN IP4 192.168.0.37 t=0 0 m=audio 6000 RTP/AVP 0 8 18 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=sendrecv <-------------> [2009-09-03 06:20:59.4545] VERBOSE[22938] chan_sip.c: --- (14 headers 14 lines) --- [2009-09-03 06:20:59.4546] VERBOSE[22938] chan_sip.c: Ignoring this INVITE request [2009-09-03 06:20:59.4547] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Length: 0 <------------> [2009-09-03 06:20:59.4548] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 10' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:59.4558] DEBUG[22938] chan_sip.c: Setting framing from config on incoming call [2009-09-03 06:20:59.4559] DEBUG[22938] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [2009-09-03 06:20:59.4560] DEBUG[22938] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [2009-09-03 06:20:59.4560] VERBOSE[22938] chan_sip.c: Audio is at 192.168.1.93 port 43820 [2009-09-03 06:20:59.4561] VERBOSE[22938] chan_sip.c: Adding codec 0x4 (ulaw) to SDP [2009-09-03 06:20:59.4561] VERBOSE[22938] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [2009-09-03 06:20:59.4562] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482284 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [2009-09-03 06:20:59.4562] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:20:59.9660] DEBUG[23000] rtp.c: Got RTCP report of 44 bytes [2009-09-03 06:21:01.7471] VERBOSE[22938] chan_sip.c: Retransmitting #6 (NAT) to 192.168.0.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1452643973;received=192.168.0.37 From: ;tag=1c1452611704 To: ;tag=as41890e37 Call-ID: 145261139211200002124@192.168.0.37 CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Require: timer Session-Expires: -1;refresher=uas Contact: Content-Type: application/sdp Content-Length: 260 v=0 o=root 1440482280 1440482280 IN IP4 192.168.1.93 s=Asterisk PBX 1.6.1.5 c=IN IP4 192.168.1.93 t=0 0 m=audio 43820 RTP/AVP 0 96 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [2009-09-03 06:21:01.7472] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:21:03.1606] VERBOSE[22938] chan_sip.c: <--- SIP read from UDP://192.168.0.37:5060 ---> BYE sip:22609@192.168.1.93 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1476668181 Max-Forwards: 70 From: ;tag=1c1453140395 To: "Victor Negru" ;tag=as0adae4d8 Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 1 BYE Contact: Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Authorization: Digest username="90028",realm="asterisk",nonce="4f79b115",uri="sip:22609@192.168.1.93",algorithm=MD5,response="0f1f0d05d2b3fc24c768eff271a12ed2" User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.4.80A.025.004 Reason: Q.850 ;cause=16 Content-Length: 0 <-------------> [2009-09-03 06:21:03.1607] VERBOSE[22938] chan_sip.c: --- (14 headers 0 lines) --- [2009-09-03 06:21:03.1608] DEBUG[22938] chan_sip.c: Initializing initreq for method BYE - callid 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 [2009-09-03 06:21:03.1608] VERBOSE[22938] chan_sip.c: Sending to 192.168.0.37 : 5060 (NAT) [2009-09-03 06:21:03.1613] VERBOSE[22938] chan_sip.c: <--- Transmitting (NAT) to 192.168.0.37:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.37;branch=z9hG4bKac1476668181;received=192.168.0.37 From: ;tag=1c1453140395 To: "Victor Negru" ;tag=as0adae4d8 Call-ID: 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 CSeq: 1 BYE Server: Asterisk PBX 1.6.1.5 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [2009-09-03 06:21:03.1613] DEBUG[22938] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 192.168.0.37:5060 [2009-09-03 06:21:03.1616] DEBUG[23000] channel.c: Didn't get a frame from channel: SIP/90028-0a4d3480 [2009-09-03 06:21:03.1617] DEBUG[23000] channel.c: Bridge stops bridging channels SIP/90027-0a502a50 and SIP/90028-0a4d3480 [2009-09-03 06:21:03.1621] DEBUG[23000] cdr_addon_mysql.c: Inserting a CDR record. [2009-09-03 06:21:03.1621] DEBUG[23000] cdr_addon_mysql.c: SQL command as follows: INSERT INTO cdr_post (calldate,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) VALUES ('2009-09-03 06:20:43','22609','22600','local_dial','SIP/90027-0a502a50','SIP/90028-0a4d3480','Dial','SIP/90028,30,gtiU(agent_call_answer^22600)','20','18','ANSWERED','3','1251984043.4','1251984043.4','#call_id=1251984043.4-1#agent=22600#t_id=35422#t_ext=90028#t_ext_id=34575#t_wg=1#t_dpt=1#s_type=agent#s_num=22609#s_id=35686#s_ext=90027#s_ext_id=34574#s_wg=1#s_dpt=1') [2009-09-03 06:21:03.1626] DEBUG[23000] channel.c: Hanging up channel 'SIP/90028-0a4d3480' [2009-09-03 06:21:03.1627] DEBUG[23000] chan_sip.c: Hangup call SIP/90028-0a4d3480, SIP callid 43cb19045a7789b61c5f9e010cabf05e@192.168.1.93 [2009-09-03 06:21:03.1627] DEBUG[23000] chan_sip.c: update_call_counter(90028) - decrement call limit counter on hangup [2009-09-03 06:21:03.1895] DEBUG[23000] rtp.c: Channel '' has no RTP, not doing anything