[Kphone2*CLI> -- Accepting call from '495XXXXXXX' to '7729720' on channel 0/12, span 2 [Kphone2*CLI> -- Executing Answer("Zap/43-1", "") in new stack [Kphone2*CLI> -- Executing Monitor("Zap/43-1", "|in-20060619-103421-495XXXXXXX|mb") in new stack -- Executing Wait("Zap/43-1", "0.5") in new stack [Kphone2*CLI> -- Executing Queue("Zap/43-1", "callcenterq|tw") in new stack [Kphone2*CLI> -- Started music on hold, class 'default', on Zap/43-1 [.. A lot of lines, while Zap/43 waits for it's turn in queue are skipped ..] -- Executing NoOp("Local/1127@toagent-9c2a,2", "0") in new stack -- Executing GotoIf("Local/1127@toagent-9c2a,2", "0?unavail") in new stack -- Executing GotoIf("Local/1127@toagent-9c2a,2", "0?busy") in new stack -- Executing Dial("Local/1127@toagent-9c2a,2", "SIP/1127|30|tw") in new stack -- Called Local/1127@toagent We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (NAT) to 10.10.11.210:5060: INVITE sip:1127@10.10.11.210:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.61:5060;branch=z9hG4bK07fc3bfd;rport From: "495XXXXXXX" ;tag=as3886ec4e To: Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 19 Jun 2006 06:39:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 309 v=0 o=root 4032 4032 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 17984 RTP/AVP 34 a=rtpmap:34 H263/90000 --- -- Called 1127 <-- SIP read from 10.10.11.210:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.11.61:5060;rport=5060;received=10.10.11.61;branch=z9hG4bK07fc3bfd Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 102 INVITE From: "495XXXXXXX";tag=as3886ec4e Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=2208271871221 --- (8 headers 0 lines)--- <-- SIP read from 10.10.11.210:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.11.61:5060;rport=5060;received=10.10.11.61;branch=z9hG4bK07fc3bfd Content-Length: 0 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 102 INVITE From: "495XXXXXXX";tag=as3886ec4e Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=2208271871221 --- (9 headers 0 lines)--- [Kphone2*CLI> -- SIP/1127-05e2 is ringing -- Local/1127@toagent-9c2a,1 is ringing <-- SIP read from 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.61:5060;rport=5060;received=10.10.11.61;branch=z9hG4bK07fc3bfd Content-Length: 219 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 102 INVITE From: "495XXXXXXX";tag=as3886ec4e Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=2208271871221 v=0 o=- 3359687954 3359687954 IN IP4 10.10.11.210 s=SJphone c=IN IP4 10.10.11.210 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (10 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.11.210:49286 Peer video RTP is at port 10.10.11.210:65535 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.11.210, port 5060 Transmitting (NAT) to 10.10.11.210:5060: ACK sip:1127@10.10.11.210:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.61:5060;branch=z9hG4bK10b29a3e;rport From: "495XXXXXXX" ;tag=as3886ec4e To: ;tag=2208271871221 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/1127-05e2 answered Local/1127@toagent-9c2a,2 -- Local/1127@toagent-9c2a,1 answered Zap/43-1 -- Stopped music on hold on Zap/43-1 [.. here a talk starts, and a client asks to transfer her to fax machine. but strange behavior starts ..] <-- SIP read from 10.10.11.210:5060: INVITE sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a0000013230000057f Content-Length: 214 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 1 INVITE From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> v=0 o=- 3359687954 3359687955 IN IP4 10.10.11.210 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 10 lines)--- [Kphone2*CLI> Using INVITE request as basis request - 13c380d7783e881372ebd560453e866c@masterhost.ru [Kphone2*CLI> Sending to 10.10.11.210 : 5060 (NAT) [Kphone2*CLI> Found RTP audio format 0 [Kphone2*CLI> Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49286 [Kphone2*CLI> Peer video RTP is at port 0.0.0.0:65535 [Kphone2*CLI> Found description format PCMU Found description format telephone-event [Kphone2*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [Kphone2*CLI> -- Started music on hold, class 'default', on Zap/43-1 [Kphone2*CLI> We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 [Kphone2*CLI> Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Kphone2*CLI> Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a0000013230000057f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4033 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a0000013230000057f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4033 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a0000013230000057f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4033 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a000007a9600000581 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a00000518800000584 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a00000160200000585 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 1 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- <-- SIP read from 10.10.11.210:5060: INVITE sip:7677@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000177449646a5000016b300000586 Content-Length: 337 Contact: Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 Content-Type: application/sdp CSeq: 1 INVITE From: "unknown";tag=22084595325516 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) v=0 o=- 3359687973 3359687973 IN IP4 10.10.11.210 s=SJphone c=IN IP4 10.10.11.210 t=0 0 a=direction:active m=audio 49288 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 15 lines)--- Using INVITE request as basis request - 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 Sending to 10.10.11.210 : 5060 (NAT) Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000177449646a5000016b300000586;received=10.10.11.210;rport=5060 From: "unknown";tag=22084595325516 To: ;tag=as3d0bf145 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest realm="phone1.masterhost.ru", nonce="56ea11fb" Content-Length: 0 --- Scheduling destruction of call '5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210' in 15000 ms Found user '1127' [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:7677@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000177449646a5000016b300000586 Content-Length: 0 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 1 ACK From: "unknown";tag=22084595325516 Max-Forwards: 70 To: ;tag=as3d0bf145 User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: INVITE sip:7677@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000177449646a500006ec000000589 Content-Length: 337 Contact: Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown";tag=22084595325516 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="1127",realm="phone1.masterhost.ru",nonce="***",uri="sip:7677@10.10.11.61",response="***" v=0 o=- 3359687973 3359687973 IN IP4 10.10.11.210 s=SJphone c=IN IP4 10.10.11.210 t=0 0 a=direction:active m=audio 49288 RTP/AVP 3 97 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (12 headers 15 lines)--- Using INVITE request as basis request - 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 [Kphone2*CLI> Sending to 10.10.11.210 : 5060 (NAT) [Kphone2*CLI> Found user '1127' Found RTP audio format 3 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.11.210:49288 Peer video RTP is at port 10.10.11.210:65535 [Kphone2*CLI> Found description format GSM Found description format iLBC Found description format iLBC Found description format PCMA Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 7677 in fromoffice (domain 10.10.11.61) [Kphone2*CLI> list_route: hop: Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000177449646a500006ec000000589;received=10.10.11.210;rport=5060 From: "unknown";tag=22084595325516 To: Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [.. 7667 is a fax. operator succ-ly dialed it. so asterisk begin to dial fax via Audiocodes gw ..] [Kphone2*CLI> -- Executing GotoIf("SIP/1127-c79e", "0?dial") in new stack [Kphone2*CLI> -- Executing Set("SIP/1127-c79e", "DIALOPT=TW") in new stack [Kphone2*CLI> -- Executing Dial("SIP/1127-c79e", "SIP/677@gatewayTDO|60|twTW") in new stack [Kphone2*CLI> We're at 10.10.10.61 port 10434 Video is at 10.10.10.61 port 12852 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 14 lines Reliably Transmitting (NAT) to 10.10.10.37:5060: INVITE sip:677@10.10.10.37 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK0cfec866;rport From: "operator name" ;tag=as678d6c6c To: Contact: Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 19 Jun 2006 06:39:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 309 v=0 o=root 4032 4032 IN IP4 10.10.10.61 s=session c=IN IP4 10.10.10.61 t=0 0 m=audio 10434 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 12852 RTP/AVP 34 a=rtpmap:34 H263/90000 --- -- Called 677@gatewayTDO [Kphone2*CLI> <-- SIP read from 10.10.10.37:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK0cfec866;rport From: "operator name" ;tag=as678d6c6c To: ;tag=1c2030783507 Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 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-104 FXO/v.4.40.215.400 Content-Length: 0 --- (10 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.10.37:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK0cfec866;rport From: "operator name" ;tag=as678d6c6c To: ;tag=1c2030783507 Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 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-104 FXO/v.4.40.215.400 Content-Length: 0 --- (11 headers 0 lines)--- -- SIP/gatewayTDO-c498 is ringing Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000177449646a500006ec000000589;received=10.10.11.210;rport=5060 From: "unknown";tag=22084595325516 To: ;tag=as2ff42a49 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- [Kphone2*CLI> <-- SIP read from 10.10.10.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK0cfec866;rport From: "operator name" ;tag=as678d6c6c To: ;tag=1c2030783507 Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 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-104 FXO/v.4.40.215.400 [Kphone2*CLI> Content-Type: application/sdp Content-Length: 253 v=0 o=AudiocodesGW 2030790589 2030790523 IN IP4 10.10.10.37 s=Phone-Call c=IN IP4 10.10.10.37 t=0 0 m=audio 4030 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 m=video 0 RTP/AVP --- (12 headers 13 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.10.37:4030 Found description format pcmu Found description format telephone-event [Kphone2*CLI> Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.10.37, port 5060 Transmitting (NAT) to 10.10.10.37:5060: ACK sip:491@10.10.10.37 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.61:5060;branch=z9hG4bK0089aa97;rport From: "operator name" ;tag=as678d6c6c To: ;tag=1c2030783507 Contact: Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/gatewayTDO-c498 answered SIP/1127-c79e We're at 10.10.11.61 port 10638 Video is at 10.10.11.61 port 16170 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000177449646a500006ec000000589;received=10.10.11.210;rport=5060 From: "unknown";tag=22084595325516 To: ;tag=as2ff42a49 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 4032 4032 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 10638 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1127-c79e and SIP/gatewayTDO-c498 [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:7677@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000177449646a7000026df0000058d Content-Length: 0 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 2 ACK From: "unknown";tag=22084595325516 Max-Forwards: 70 To: ;tag=as2ff42a49 User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.10.37:5060: BYE sip:1127@10.10.10.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.37;branch=z9hG4bKac2033670330 Max-Forwards: 70 From: ;tag=1c2030783507 To: "operator name" ;tag=as678d6c6c Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 CSeq: 1 BYE 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-104 FXO/v.4.40.215.400 Content-Length: 0 --- (12 headers 0 lines)--- [Kphone2*CLI> Sending to 10.10.10.37 : 5060 (NAT) [Kphone2*CLI> Transmitting (NAT) to 10.10.10.37:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.37;branch=z9hG4bKac2033670330;received=10.10.10.37 From: ;tag=1c2030783507 To: "operator name" ;tag=as678d6c6c Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61 CSeq: 1 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- [Kphone2*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 10.10.11.210, port 5060 Reliably Transmitting (NAT) to 10.10.11.210:5060: BYE sip:1127@10.10.11.210:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.61:5060;branch=z9hG4bK35dc9e52;rport From: ;tag=as2ff42a49 To: "unknown";tag=22084595325516 Contact: Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.61:5060;rport=5060;received=10.10.11.61;branch=z9hG4bK35dc9e52 Content-Length: 0 Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210 CSeq: 102 BYE From: ;tag=as2ff42a49 Server: SJphone/1.60.289a (SJ Labs) To: "unknown";tag=22084595325516 --- (8 headers 0 lines)--- [Kphone2*CLI> Destroying call '11cbe07716079c24483d25975bdc9db0@10.10.10.61' [Kphone2*CLI> Destroying call '5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210' [.. leg to fax dropped, but stranges continue..] [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: INVITE sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800004e170000058f Content-Length: 219 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 2 INVITE From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> v=0 o=- 3359687954 3359687956 IN IP4 10.10.11.210 s=SJphone c=IN IP4 10.10.11.210 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 10 lines)--- Using INVITE request as basis request - 13c380d7783e881372ebd560453e866c@masterhost.ru Sending to 10.10.11.210 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.11.210:49286 Peer video RTP is at port 10.10.11.210:65535 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Stopped music on hold on Zap/43-1 We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Kphone2*CLI> Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004e170000058f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4034 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004e170000058f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4034 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004e170000058f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4034 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #3 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004e170000058f;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4034 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800001fa200000592 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a80000739c00000595 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a80000034e00000596 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 2 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: INVITE sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597 Content-Length: 214 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 3 INVITE From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> v=0 o=- 3359687954 3359687957 IN IP4 10.10.11.210 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 10 lines)--- Using INVITE request as basis request - 13c380d7783e881372ebd560453e866c@masterhost.ru Sending to 10.10.11.210 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49286 Peer video RTP is at port 0.0.0.0:65535 [Kphone2*CLI> Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on Zap/43-1 We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4035 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4035 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4035 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #3 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4035 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #4 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a80000410800000597;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4035 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a80000525a0000059a Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 3 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: REFER sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b Content-Length: 0 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER From: "unknown";tag=2208271871221 Max-Forwards: 70 Referred-By: Refer-To: To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (12 headers 0 lines)--- [Kphone2*CLI> Transfer to 495XXXXXXX in fromoffice Transfer from 1127 in fromoffice [Kphone2*CLI> Jun 19 10:39:36 NOTICE[4042]: chan_sip.c:6866 get_refer_info: Supervised transfer attempted to transfer into same call id (13c380d7783e881372ebd560453e866c@masterhost.ru == 13c380d7783e881372ebd560453e866c@masterhost.ru)! [Kphone2*CLI> Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: [Kphone2*CLI> Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800006e590000059e Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800004fe70000059f Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646a800002dfa000005a3 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- Retransmitting #3 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> Retransmitting #4 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> Retransmitting #5 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> Retransmitting #6 (NAT) to 10.10.11.210:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646a800004c110000059b;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 4 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- [Kphone2*CLI> Jun 19 10:39:36 WARNING[4042]: chan_sip.c:1217 retrans_pkt: Maximum retries exceeded on transmission 13c380d7783e881372ebd560453e866c@masterhost.ru for seqno 0 (Non-critical Response) [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: INVITE sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646ac000037b9000005a9 Content-Length: 219 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 5 INVITE From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> v=0 o=- 3359687954 3359687958 IN IP4 10.10.11.210 s=SJphone c=IN IP4 10.10.11.210 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 10 lines)--- Using INVITE request as basis request - 13c380d7783e881372ebd560453e866c@masterhost.ru Sending to 10.10.11.210 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 10.10.11.210:49286 Peer video RTP is at port 10.10.11.210:65535 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Stopped music on hold on Zap/43-1 We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP [Kphone2*CLI> Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646ac000037b9000005a9;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4036 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646ac000037b9000005a9;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4036 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646ac000037b9000005a9;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 5 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4036 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646ac00006a04000005ac Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 5 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [Kphone2*CLI> <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646ac0000723d000005ae Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 5 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- <-- SIP read from 10.10.11.210:5060: INVITE sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646af00005c07000005b4 Content-Length: 214 Contact: Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru Content-Type: application/sdp CSeq: 6 INVITE From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) [Kphone2*CLI> v=0 o=- 3359687954 3359687959 IN IP4 10.10.11.210 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 49286 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11,16 --- (11 headers 10 lines)--- Using INVITE request as basis request - 13c380d7783e881372ebd560453e866c@masterhost.ru Sending to 10.10.11.210 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:49286 Peer video RTP is at port 0.0.0.0:65535 [Kphone2*CLI> Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on Zap/43-1 We're at 10.10.11.61 port 17738 Video is at 10.10.11.61 port 17984 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646af00005c07000005b4;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4037 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Retransmitting #1 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646af00005c07000005b4;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4037 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #2 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646af00005c07000005b4;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4037 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- [Kphone2*CLI> Retransmitting #3 (NAT) to 10.10.11.210:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.11.210;branch=z9hG4bK0a0a0bd200000175449646af00005c07000005b4;received=10.10.11.210;rport=5060 From: "unknown";tag=2208271871221 To: ;tag=as3886ec4e Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 4032 4037 IN IP4 10.10.11.61 s=session c=IN IP4 10.10.11.61 t=0 0 m=audio 17738 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646af000021ac000005b9 Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- <-- SIP read from 10.10.11.210:5060: ACK sip:495XXXXXXX@10.10.11.61 SIP/2.0 Via: SIP/2.0/UDP 10.10.11.210;rport;branch=z9hG4bK0a0a0bd200000175449646b00000205c000005bb Content-Length: 0 Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru CSeq: 6 ACK From: "unknown";tag=2208271871221 Max-Forwards: 70 To: ;tag=as3886ec4e User-Agent: SJphone/1.60.289a (SJ Labs) --- (9 headers 0 lines)--- [.. some minutes later, far end drop the line, but my end still holds the line, so: ..] [Kphone2*CLI> -- Channel 0/12, span 2 got hangup [Kphone2*CLI> Jun 19 10:48:08 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:20 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:21 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:22 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:30 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:38 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner. [Kphone2*CLI> Jun 19 10:48:40 WARNING[4046]: chan_zap.c:8396 pri_dchannel: Ring requested on channel 0/12 already in use on span 2. Hanging up owner.