pbx*CLI> pbx*CLI> -- Executing [1100057@default:1] Dial("SIP/1100055-0819be40", "SIP/1100057|3600|t") in new stack Audio is at 122.254.93.32 port 56728 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 122.254.x.y:56197: INVITE sip:1100057@192.168.0.173;user=phone SIP/2.0 Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK7f40d880;rport From: "1100055" ;tag=as68a4a3b5 To: Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 19 Feb 2008 01:04:40 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 266 v=0 o=root 11609 11609 IN IP4 122.254.93.32 s=session c=IN IP4 122.254.93.32 t=0 0 m=audio 56728 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 1100057 pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK7f40d880;rport From: "1100055" ;tag=as68a4a3b5 To: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.8.33 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK7f40d880;rport From: "1100055" ;tag=as68a4a3b5 To: ;tag=e27c880befa39442 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.8.33 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/1100057-0821b008 is ringing [Feb 19 09:04:42] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 0d7c6cb32453b82336f5bb4413713ff1@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK7f40d880;rport From: "1100055" ;tag=as68a4a3b5 To: ;tag=e27c880befa39442 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 102 INVITE User-Agent: Grandstream HT487 1.0.8.33 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Supported: replaces, timer Content-Length: 216 v=0 o=1100057 8000 8000 IN IP4 192.168.0.173 s=SIP Call c=IN IP4 192.168.0.173 t=0 0 m=audio 5004 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (12 headers 11 lines) --- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.173:5004 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.173:5004 list_route: hop: [Feb 19 09:04:47] DEBUG[11619]: chan_sip.c:5871 reqprep: Strict routing enforced for session 28be12706c806a484697cd6a042ac3cc@122.254.93.32 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.173, port 5060 Transmitting (NAT) to 122.254.x.y:56197: ACK sip:1100057@192.168.0.173;user=phone SIP/2.0 Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK42fdff38;rport From: "1100055" ;tag=as68a4a3b5 To: ;tag=e27c880befa39442 Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1100057-0821b008 answered SIP/1100055-0819be40 [Feb 19 09:04:47] NOTICE[18357]: rtp.c:787 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 203.91.112.200 [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 6f9007ed14fb1bc1204fcd7e071c829a@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 4a52d91577b9d97726c392e52cdcb0db@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 5d37c910142d8454279ec4456b552124@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 06d330454ef75e577ad8b34d604e9f7a@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 34bc478d119cda9c402560e8572795ca@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 04e440b3404822345c7701f773003ec4@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 5410e668393250813f9016982f201e86@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 4171467d5977779137efcfa20242a8b7@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 3efa25786a61c9b12345da9e007da3af@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 4f22336b22da2e7b4b29859451fcb5da@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:48] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 5434337c4ce4ab5006fb11453bb758d2@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> INVITE sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK2e63b26f5866a946 From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Supported: replaces, timer Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16034 INVITE User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 353 v=0 o=1100057 8000 8001 IN IP4 192.168.0.173 s=SIP Call c=IN IP4 192.168.0.173 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (13 headers 15 lines) --- Sending to 122.254.x.y : 56197 (NAT) [Feb 19 09:04:52] WARNING[11619]: chan_sip.c:5083 process_sdp: Unsupported SDP media type in offer: image 5004 udptl t38 <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK2e63b26f5866a946;received=122.254.x.y From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16034 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> ACK sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK2e63b26f5866a946 From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16034 ACK User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> INVITE sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKbc1d210928fbd059 From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Supported: replaces, timer Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16035 INVITE User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 216 v=0 o=1100057 8000 8002 IN IP4 192.168.0.173 s=SIP Call c=IN IP4 192.168.0.173 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 11 lines) --- Sending to 122.254.x.y : 56197 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.173:5004 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.173:5004 [Feb 19 09:04:52] DEBUG[11619]: chan_sip.c:5480 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x4 (ulaw) and not 0x8 (alaw) <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKbc1d210928fbd059;received=122.254.x.y From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16035 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 122.254.93.32 port 56728 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKbc1d210928fbd059;received=122.254.x.y From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16035 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 242 v=0 o=root 11609 11610 IN IP4 122.254.93.32 s=session c=IN IP4 122.254.93.32 t=0 0 m=audio 56728 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> ACK sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK712d3e0ed6a2872c From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16035 ACK User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- [Feb 19 09:04:53] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 4885e9bd1cba1d3a699ec88a7099d9cf@122.254.93.32 for seqno 102 (Non-critical Request) [Feb 19 09:04:54] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 3a90e76a7b7969e14f8e1e7e2bb0fc0c@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> INVITE sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK79cd75b57e87ba96 From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Supported: replaces, timer Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16036 INVITE User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 353 v=0 o=1100057 8000 8003 IN IP4 192.168.0.173 s=SIP Call c=IN IP4 192.168.0.173 t=0 0 m=image 5004 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:9600 a=T38FaxFillBitRemoval:0 a=T38FaxTranscodingMMR:0 a=T38FaxTranscodingJBIG:0 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:400 a=T38FaxMaxDatagram:280 a=T38FaxUdpEC:t38UDPRedundancy <-------------> --- (13 headers 15 lines) --- Sending to 122.254.x.y : 56197 (NAT) [Feb 19 09:04:55] WARNING[11619]: chan_sip.c:5083 process_sdp: Unsupported SDP media type in offer: image 5004 udptl t38 <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK79cd75b57e87ba96;received=122.254.x.y From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16036 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> ACK sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bKc97592ccc3a5b043 From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16035 ACK User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> INVITE sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK8adbe011f4bad31b From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Contact: Supported: replaces, timer Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16037 INVITE User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 216 v=0 o=1100057 8000 8004 IN IP4 192.168.0.173 s=SIP Call c=IN IP4 192.168.0.173 t=0 0 m=audio 5004 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (13 headers 11 lines) --- <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK8adbe011f4bad31b;received=122.254.x.y From: ;tag=e27c880befa39442 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16037 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> ACK sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK8adbe011f4bad31b From: "1100055" ;tag=as68a4a3b5 To: "1100055" ;tag=as68a4a3b5 Contact: Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16037 ACK User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> BYE sip:1100055@122.254.93.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK4aa6e952754cbb69 From: "1100055" ;tag=as68a4a3b5 To: "1100055" ;tag=as68a4a3b5 Supported: replaces, timer Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16038 BYE User-Agent: Grandstream HT487 1.0.8.33 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 122.254.x.y : 56197 (NAT) <--- Transmitting (NAT) to 122.254.x.y:56197 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.173;branch=z9hG4bK4aa6e952754cbb69;received=122.254.x.y From: "1100055" ;tag=as68a4a3b5 To: "1100055" ;tag=as68a4a3b5 Call-ID: 28be12706c806a484697cd6a042ac3cc@122.254.93.32 CSeq: 16038 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> == Spawn extension (default, 1100057, 1) exited non-zero on 'SIP/1100055-0819be40' -- Executing [h@default:1] NoOp("SIP/1100055-0819be40", "TX: REMOTESTATIONID is ") in new stack -- Executing [h@default:2] NoOp("SIP/1100055-0819be40", "TX: PHASEESTATUS is ") in new stack [Feb 19 09:04:55] DEBUG[18357]: chan_sip.c:5871 reqprep: Strict routing enforced for session 02631C7A4A4B38158FBC0B01E5775272@boldsoft.mn Really destroying SIP dialog '28be12706c806a484697cd6a042ac3cc@122.254.93.32' Method: BYE [Feb 19 09:05:04] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 5793afa82589340f5d802fcd206cb715@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- Reliably Transmitting (NAT) to 122.254.x.y:56197: OPTIONS sip:1100057@192.168.0.173;user=phone SIP/2.0 Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK3b827bfa;rport From: "asterisk" ;tag=as369a1fdf To: Contact: Call-ID: 3b1834b82609787d1525800e6874fa85@122.254.93.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 19 Feb 2008 01:05:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK3b827bfa;rport From: "asterisk" ;tag=as369a1fdf To: ;tag=441630d9005a03ca Call-ID: 3b1834b82609787d1525800e6874fa85@122.254.93.32 CSeq: 102 OPTIONS User-Agent: Grandstream HT487 1.0.8.33 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '3b1834b82609787d1525800e6874fa85@122.254.93.32' Method: OPTIONS pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- [Feb 19 09:05:37] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 7983ddb90cc737fb3dc049ff2891bfc6@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- [Feb 19 09:05:59] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 501c96365ce3ce311aa418304335d7f6@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- Reliably Transmitting (NAT) to 122.254.x.y:56197: OPTIONS sip:1100057@192.168.0.173;user=phone SIP/2.0 Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK5e5aed4e;rport From: "asterisk" ;tag=as47aa2b5a To: Contact: Call-ID: 390070b2257147061bf32ac71c23f154@122.254.93.32 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 19 Feb 2008 01:06:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 122.254.93.32:5060;branch=z9hG4bK5e5aed4e;rport From: "asterisk" ;tag=as47aa2b5a To: ;tag=98a36f7712072652 Call-ID: 390070b2257147061bf32ac71c23f154@122.254.93.32 CSeq: 102 OPTIONS User-Agent: Grandstream HT487 1.0.8.33 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE Supported: replaces, timer Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '390070b2257147061bf32ac71c23f154@122.254.93.32' Method: OPTIONS pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- [Feb 19 09:06:32] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 50be79c25b2e9e0a2cba15eb6a08846b@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI> <--- SIP read from 122.254.x.y:56197 ---> <-------------> --- (0 headers 0 lines) Nat keepalive --- [Feb 19 09:06:54] WARNING[11619]: chan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 705569117b82fff80472db306a67ae04@122.254.93.32 for seqno 102 (Non-critical Request) pbx*CLI>