 [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" <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
To: <sip:1127@10.10.11.210:5060>
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1127@10.10.11.210:5060>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 102 INVITE
From: "495XXXXXXX"<sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1127@10.10.11.210:5060>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 102 INVITE
From: "495XXXXXXX"<sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1127@10.10.11.210:5060>;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: <sip:1127@10.10.11.210:5060>
 set_destination: Parsing <sip:1127@10.10.11.210:5060> 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" <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
To: <sip:1127@10.10.11.210:5060>;tag=2208271871221
Contact: <sip:495XXXXXXX@10.10.11.61>
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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 1 INVITE
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210
Content-Type: application/sdp
CSeq: 1 INVITE
From: "unknown"<sip:1127@10.10.11.61>;tag=22084595325516
Max-Forwards: 70
To: <sip:7677@10.10.11.61>
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"<sip:1127@10.10.11.61>;tag=22084595325516
To: <sip:7677@10.10.11.61>;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: <sip:7677@10.10.11.61>
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"<sip:1127@10.10.11.61>;tag=22084595325516
Max-Forwards: 70
To: <sip:7677@10.10.11.61>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 5503978E-61AE-421E-AEFB-60D827950187@10.10.11.210
Content-Type: application/sdp
CSeq: 2 INVITE
From: "unknown"<sip:1127@10.10.11.61>;tag=22084595325516
Max-Forwards: 70
To: <sip:7677@10.10.11.61>
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: <sip:1127@10.10.11.210:5060>
 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"<sip:1127@10.10.11.61>;tag=22084595325516
To: <sip:7677@10.10.11.61>
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: <sip:7677@10.10.11.61>
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" <sip:1127@10.10.10.61>;tag=as678d6c6c
To: <sip:677@10.10.10.37>
Contact: <sip:1127@10.10.10.61>
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" <sip:1127@10.10.10.61>;tag=as678d6c6c
To: <sip:677@10.10.10.37>;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" <sip:1127@10.10.10.61>;tag=as678d6c6c
To: <sip:677@10.10.10.37>;tag=1c2030783507
Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61
CSeq: 102 INVITE
Contact: <sip:491@10.10.10.37>
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"<sip:1127@10.10.11.61>;tag=22084595325516
To: <sip:7677@10.10.11.61>;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: <sip:7677@10.10.11.61>
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" <sip:1127@10.10.10.61>;tag=as678d6c6c
To: <sip:677@10.10.10.37>;tag=1c2030783507
Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61
CSeq: 102 INVITE
Contact: <sip:491@10.10.10.37>
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: <sip:491@10.10.10.37>
 set_destination: Parsing <sip:491@10.10.10.37> 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" <sip:1127@10.10.10.61>;tag=as678d6c6c
To: <sip:677@10.10.10.37>;tag=1c2030783507
Contact: <sip:1127@10.10.10.61>
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"<sip:1127@10.10.11.61>;tag=22084595325516
To: <sip:7677@10.10.11.61>;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: <sip:7677@10.10.11.61>
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"<sip:1127@10.10.11.61>;tag=22084595325516
Max-Forwards: 70
To: <sip:7677@10.10.11.61>;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: <sip:677@10.10.10.37>;tag=1c2030783507
To: "operator name" <sip:1127@10.10.10.61>;tag=as678d6c6c
Call-ID: 11cbe07716079c24483d25975bdc9db0@10.10.10.61
CSeq: 1 BYE
Contact: <sip:491@10.10.10.37>
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: <sip:677@10.10.10.37>;tag=1c2030783507
To: "operator name" <sip:1127@10.10.10.61>;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: <sip:1127@10.10.10.61>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
 [Kphone2*CLI> set_destination: Parsing <sip:1127@10.10.11.210:5060> 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: <sip:7677@10.10.11.61>;tag=as2ff42a49
To: "unknown"<sip:1127@10.10.11.61>;tag=22084595325516
Contact: <sip:7677@10.10.11.61>
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: <sip:7677@10.10.11.61>;tag=as2ff42a49
Server: SJphone/1.60.289a (SJ Labs)
To: "unknown"<sip:1127@10.10.11.61>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 2 INVITE
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 3 INVITE
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
Referred-By: <sip:1127@10.10.11.210:5060>
Refer-To: <sip:495XXXXXXX@masterhost.ru?Replaces=13c380d7783e881372ebd560453e866c%40masterhost.ru%3Bto-tag%3Das3886ec4e%3Bfrom-tag%3D2208271871221>
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
[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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 5 INVITE
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 5 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 5 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 5 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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: <sip:1127@10.10.11.210:5060>
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
Content-Type: application/sdp
CSeq: 6 INVITE
From: "unknown"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 6 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 6 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 6 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
To: <sip:495XXXXXXX@masterhost.ru>;tag=as3886ec4e
Call-ID: 13c380d7783e881372ebd560453e866c@masterhost.ru
CSeq: 6 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:495XXXXXXX@10.10.11.61>
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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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"<sip:1127@10.10.11.210:5060>;tag=2208271871221
Max-Forwards: 70
To: <sip:495XXXXXXX@masterhost.ru>;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.