ECT (Explicit call transfer )

Hi All,

About the ECT (Explicit call transfer) the senario is

A calls B call connected
A put B call on hold
A calls C call connected
Now in case of ECT A will exit and B and C connected.

this is depends on the Refer method to connect two call .But in my case there is some problem.

So i want to know is there any thing special to implement the same.I am trying to know where the problem is but get to fail.below is the sip trace for the same.Any hint will help me.

<------------>
Scheduling destruction of SIP dialog ‘3659945832-29929209’ in 32000 ms (Method: REGISTER)
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
INVITE sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660200832-48700157
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: 100rel,replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111
Call-ID: 3660195832-48700156
CSeq: 1 INVITE
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Type: application/sdp
Content-Length: 162

v=0
o=- 3660180832 3660180832 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51612 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

<------------->
— (13 headers 9 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)
Using INVITE request as basis request - 3660195832-48700156

<— Reliably Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660200832-48700157;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as6d460bc8
Call-ID: 3660195832-48700156
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0f98019d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3660195832-48700156’ in 32000 ms (Method: INVITE)
Found user '2101’
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
ACK sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660200832-48700157
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as6d460bc8
Call-ID: 3660195832-48700156
CSeq: 1 ACK
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
INVITE sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660225832-48700159
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: 100rel,replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111
Call-ID: 3660195832-48700156
CSeq: 2 INVITE
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Type: application/sdp
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“de93d4e2ca319f424bf1ad3c97a8dcee”,algorithm=MD5
Content-Length: 162

v=0
o=- 3660180832 3660180832 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51612 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

<------------->

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660225832-48700159;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111
Call-ID: 3660195832-48700156
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Length: 0

<------------>
Video is at 192.168.1.111 port 14652
Audio is at 192.168.1.111 port 13306
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x10000 (jpeg) to SDP
Adding codec 0x20000 (png) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.191:5060:
INVITE sip:2501@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK19c3cd09;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone
Contact: sip:2101@192.168.1.111
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Date: Fri, 18 Apr 2008 20:36:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 687

v=0
o=root 2190 2190 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
b=CT:384
t=0 0
m=audio 13306 RTP/AVP 8 3 0 112 5 10 7 110 97 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 14652 RTP/AVP 26 31 34 103 99
a=rtpmap:26 JPEG/90000
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv


trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK19c3cd09;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2501@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (8 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK19c3cd09;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2501@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660225832-48700159;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Length: 0

<------------>
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK19c3cd09;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2501@192.168.1.191:5060;user=phone
Supported: 100rel,replaces,histinfo
Content-Type: application/sdp
Content-Length: 207

v=0
o=- 3660265832 3660265833 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51616 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 26 31 34 103 99

<------------->
— (10 headers 11 lines) —
Found RTP audio format 8
Found RTP video format 26
Found RTP video format 31
Found RTP video format 34
Found RTP video format 103
Found RTP video format 99
Peer audio RTP is at port 192.168.1.191:51616
Found description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3d0008 (alaw|jpeg|h261|h263|h263p|h264)/video=0x3d0000 (jpeg|h261|h263|h263p|h264), combined - 0x3d0008 (alaw|jpeg|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.191:51616
list_route: hop: sip:2501@192.168.1.191:5060;user=phone
set_destination: Parsing sip:2501@192.168.1.191:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.191, port 5060
Transmitting (no NAT) to 192.168.1.191:5060:

ACK sip:2501@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK6989fbdd;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Contact: sip:2101@192.168.1.111
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Content-Length: 0


Audio is at 192.168.1.111 port 13532
Adding codec 0x8 (alaw) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3660225832-48700159;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 2190 2190 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 13532 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
ACK sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3662615832-48700164
Max-Forwards: 70
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“de93d4e2ca319f424bf1ad3c97a8dcee”,algorithm=MD5
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 2 ACK
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.107:2837 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
Really destroying SIP dialog ‘00E5CF69-7E0B-DD11-B9D4-0019D11009D7@192.168.1.107’ Method: REGISTER
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>

INVITE sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3667780832-48700165
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: 100rel,replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 3 INVITE
Contact: sip:2101@192.168.1.191:5060;user=phone
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“de93d4e2ca319f424bf1ad3c97a8dcee”,algorithm=MD5
Content-Type: application/sdp
Content-Length: 172

v=0
o=- 3660180832 3660180833 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendonly
m=audio 51612 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly

<------------->
— (13 headers 10 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)
Found RTP audio format 8
Peer audio RTP is at port 192.168.1.191:51612
Found description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.191:51612
Peer video RTP is at port 192.168.1.191:33280
Audio is at 192.168.1.111 port 13532
Adding codec 0x8 (alaw) to SDP

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3667780832-48700165;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 2190 2191 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 13532 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=recvonly

<------------>
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
ACK sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3667810832-48700167
Max-Forwards: 70
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“de93d4e2ca319f424bf1ad3c97a8dcee”,algorithm=MD5
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 3 ACK
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
INVITE sip:2201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668005832-48700169
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: 100rel,replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111
Call-ID: 3668000832-48700168
CSeq: 1 INVITE
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Type: application/sdp
Content-Length: 162

v=0
o=- 3667980832 3667980832 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51620 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

<------------->
— (13 headers 9 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)
Using INVITE request as basis request - 3668000832-48700168

<— Reliably Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668005832-48700169;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as6fb80247
Call-ID: 3668000832-48700168
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5251ef6d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3668000832-48700168’ in 32000 ms (Method: INVITE)
Found user '2101’
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
ACK sip:2201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668005832-48700169
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as6fb80247
Call-ID: 3668000832-48700168
CSeq: 1 ACK
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
INVITE sip:2201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668035832-48700171
Route: sip:192.168.1.111:5060;lr
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: 100rel,replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111
Call-ID: 3668000832-48700168
CSeq: 2 INVITE
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Type: application/sdp
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“5251ef6d”,uri="sip:2201@192.168.1.111",response=“840d1623c6cb3e41ba2ec99c41cf0f46”,algorithm=MD5
Content-Length: 162

v=0
o=- 3667980832 3667980832 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51620 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1
a=ptime:20

<------------->
— (14 headers 9 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)
Using INVITE request as basis request - 3668000832-48700168
Found user '2101’
Found RTP audio format 8
Peer audio RTP is at port 192.168.1.191:51620
Found description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.191:51620
Peer video RTP is at port 192.168.1.191:33280
Looking for 2201 in default (domain 192.168.1.111)
list_route: hop: sip:2101@192.168.1.191:5060;user=phone

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668035832-48700171;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111
Call-ID: 3668000832-48700168
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2201@192.168.1.111
Content-Length: 0

<------------>
Video is at 192.168.1.111 port 19812
Audio is at 192.168.1.111 port 18066
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x10000 (jpeg) to SDP
Adding codec 0x20000 (png) to SDP
Adding codec 0x40000 (h261) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.191:5060:
INVITE sip:2201@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK32c63c06;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone
Contact: sip:2101@192.168.1.111
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Date: Fri, 18 Apr 2008 20:36:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 687

v=0
o=root 2190 2190 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
b=CT:384
t=0 0
m=audio 18066 RTP/AVP 8 3 0 112 5 10 7 110 97 111 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 19812 RTP/AVP 26 31 34 103 99
a=rtpmap:26 JPEG/90000
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv


trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK32c63c06;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2201@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (8 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK32c63c06;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2201@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668035832-48700171;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as00d720ed
Call-ID: 3668000832-48700168
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2201@192.168.1.111
Content-Length: 0

<------------>
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK32c63c06;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 102 INVITE
Contact: sip:2201@192.168.1.191:5060;user=phone
Supported: 100rel,replaces,histinfo
Content-Type: application/sdp
Content-Length: 207

v=0
o=- 3668075832 3668075833 IN IP4 192.168.1.191
s=-
c=IN IP4 192.168.1.191
t=0 0
a=sendrecv
m=audio 51624 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 26 31 34 103 99

<------------->
— (10 headers 11 lines) —
Found RTP audio format 8
Found RTP video format 26
Found RTP video format 31
Found RTP video format 34
Found RTP video format 103
Found RTP video format 99
Peer audio RTP is at port 192.168.1.191:51624
Found description format PCMA for ID 8
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x3d0008 (alaw|jpeg|h261|h263|h263p|h264)/video=0x3d0000 (jpeg|h261|h263|h263p|h264), combined - 0x3d0008 (alaw|jpeg|h261|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.191:51624
list_route: hop: sip:2201@192.168.1.191:5060;user=phone
set_destination: Parsing sip:2201@192.168.1.191:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.191, port 5060
Transmitting (no NAT) to 192.168.1.191:5060:

ACK sip:2201@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK668623b5;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Contact: sip:2101@192.168.1.111
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Content-Length: 0


Audio is at 192.168.1.111 port 18630
Adding codec 0x8 (alaw) to SDP

<— Reliably Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3668035832-48700171;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as00d720ed
Call-ID: 3668000832-48700168
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2201@192.168.1.111
Content-Type: application/sdp
Content-Length: 184

v=0
o=root 2190 2190 IN IP4 192.168.1.111
s=session
c=IN IP4 192.168.1.111
t=0 0
m=audio 18630 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
ACK sip:2201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3670615832-48700176
Max-Forwards: 70
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“5251ef6d”,uri="sip:2201@192.168.1.111",response=“840d1623c6cb3e41ba2ec99c41cf0f46”,algorithm=MD5
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as00d720ed
Call-ID: 3668000832-48700168
CSeq: 2 ACK
Contact: sip:2101@192.168.1.191:5060;user=phone
Content-Length: 0

<------------->
— (10 headers 0 lines) —
trixbox1*CLI>
<— SIP read from 192.168.1.107:5095 —>

<------------->
— (0 headers 0 lines) Nat keepalive —
Really destroying SIP dialog ‘3639915832-29929199’ Method: REGISTER
Really destroying SIP dialog ‘3639925832-29929202’ Method: REGISTER
trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>

REFER sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3675785832-48700177
Max-Forwards: 70
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,UPDATE,PRACK,REFER,NOTIFY,INFO
Supported: replaces,histinfo
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 4 REFER
Contact: sip:2101@192.168.1.191:5060;user=phone
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“f1d70a260d7e2e51ade8c973159dd21b”,algorithm=MD5
Refer-To: sip:2201@192.168.1.111?Replaces=3668000832-48700168%3Bfrom-tag%3DKEYMILE_3668005832-48700170%3Bto-tag%3Das00d720ed
Referred-By: sip:2101@192.168.1.111:5060;user=phone
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Call 3660195832-48700156 got a SIP call transfer from caller: (REFER)!
Failed SIP Transfer to non-existing extension 2201 in context from-internal-xfer
n
<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3675785832-48700177;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Length: 0

<------------>
set_destination: Parsing sip:2101@192.168.1.191:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.191, port 5060
Reliably Transmitting (no NAT) to 192.168.1.191:5060:
NOTIFY sip:2101@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK056977f3;rport
From: sip:2501@192.168.1.111;tag=as2030e535
To: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
Contact: sip:2501@192.168.1.111
Call-ID: 3660195832-48700156
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 23

SIP/2.0 404 Not found


trixbox1*CLI>
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK056977f3;rport
From: sip:2501@192.168.1.111;tag=as2030e535
To: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
Call-ID: 3660195832-48700156
CSeq: 102 NOTIFY
Contact: sip:2101@192.168.1.191:5060;user=phone
Supported: 100rel,replaces,histinfo
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3679965832-29929237’ in 32000 ms (Method: REGISTER)
trixbox1*CLI> sip set debug o
<— SIP read from 192.168.1.191:5060 —>
BYE sip:2501@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3681285832-48700184
Max-Forwards: 70
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 5 BYE
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“0f98019d”,uri="sip:2501@192.168.1.111",response=“8e4f21c77bc526fd7cd9afe18184560d”,algorithm=MD5
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3681285832-48700184;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3660200832-48700158
To: sip:2501@192.168.1.111;tag=as2030e535
Call-ID: 3660195832-48700156
CSeq: 5 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2501@192.168.1.111
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘64e72a452f7862e31d0e747170d9da1a@192.168.1.111’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:2501@192.168.1.191:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.191, port 5060
Reliably Transmitting (no NAT) to 192.168.1.191:5060:
BYE sip:2501@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK7b32c787;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Content-Length: 0


trixbox1*CLI> sip set debug o
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK7b32c787;rport
From: “2101” sip:2101@192.168.1.111;tag=as0e49f1b3
To: sip:2501@192.168.1.191:5060;user=phone;tag=KEYMILE_3660265832-48700162
Call-ID: 64e72a452f7862e31d0e747170d9da1a@192.168.1.111
CSeq: 103 BYE
Contact: sip:2501@192.168.1.191:5060;user=phone
Supported: 100rel,replaces,histinfo
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘64e72a452f7862e31d0e747170d9da1a@192.168.1.111’ Method: INVITE
Really destroying SIP dialog ‘3660195832-48700156’ Method: BYE
trixbox1*CLI> sip set debug off
<— SIP read from 192.168.1.191:5060 —>
BYE sip:2201@192.168.1.111 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3681990832-48700185
Max-Forwards: 70
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as00d720ed
Call-ID: 3668000832-48700168
CSeq: 3 BYE
Proxy-Authorization: Digest username=“2101”,realm=“asterisk”,nonce=“5251ef6d”,uri="sip:2201@192.168.1.111",response=“6c49f327f30b49758977cd0b800cb3cd”,algorithm=MD5
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.1.191 : 5060 (no NAT)

<— Transmitting (no NAT) to 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.191:5060;branch=z9hG4bK3681990832-48700185;received=192.168.1.191
From: sip:2101@192.168.1.111:5060;user=phone;tag=KEYMILE_3668005832-48700170
To: sip:2201@192.168.1.111;tag=as00d720ed
Call-ID: 3668000832-48700168
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:2201@192.168.1.111
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘556a631e7ffa21cb498202651b142e77@192.168.1.111’ in 6400 ms (Method: INVITE)
set_destination: Parsing sip:2201@192.168.1.191:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.1.191, port 5060
Reliably Transmitting (no NAT) to 192.168.1.191:5060:
BYE sip:2201@192.168.1.191:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK13de0ded;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: “2101” sip:2101@192.168.1.111;privacy=off;screen=no
Content-Length: 0


trixbox1*CLI> sip set debug off
<— SIP read from 192.168.1.191:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.111:5060;branch=z9hG4bK13de0ded;rport
From: “2101” sip:2101@192.168.1.111;tag=as6b2485e5
To: sip:2201@192.168.1.191:5060;user=phone;tag=KEYMILE_3668070832-48700174
Call-ID: 556a631e7ffa21cb498202651b142e77@192.168.1.111
CSeq: 103 BYE
Contact: sip:2201@192.168.1.191:5060;user=phone
Supported: 100rel,replaces,histinfo
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘556a631e7ffa21cb498202651b142e77@192.168.1.111’ Method: INVITE
Really destroying SIP dialog ‘3668000832-48700168’ Method: BYE
trixbox1*CLI> sip set debug off
<— SIP read from 192.168.1.107:5108 —>

<------------->
— (0 headers 0 lines) Nat keepalive —