obelix:~# asterisk -r -vvvv -dddd
Parsing /etc/asterisk/asterisk.conf
Parsing /etc/asterisk/extconfig.conf
  == Binding extensions to odbc/asterisk/realtime_extensions
  == Binding voicemail-extensions to odbc/asterisk/realtime_voicemail_extensions
  == Binding voicemail to odbc/asterisk/voicemail_users
Asterisk CVS-D2005.06.14.22.00.00-07/13/05-15:37:33, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <markster@digium.com>
=========================================================================
Connected to Asterisk CVS-D2005.06.14.22.00.00-07/13/05-15:37:33 currently running on obelix (pid = 12229)
Verbosity is at least 4
    -- Remote UNIX connection
obelix*CLI> sip debug
SIP Debugging enabled
obelix*CLI> set verbose 4
Verbosity is at least 4
obelix*CLI> debug level 4
Debugging level set to 4, file '<any>'
    -- B-channel 0/1 successfully restarted on span 1
    -- B-channel 0/2 successfully restarted on span 1
    -- B-channel 0/3 successfully restarted on span 1
    -- B-channel 0/4 successfully restarted on span 1
    -- B-channel 0/5 successfully restarted on span 1
    -- B-channel 0/6 successfully restarted on span 1
    -- B-channel 0/7 successfully restarted on span 1
    -- B-channel 0/8 successfully restarted on span 1
    -- B-channel 0/9 successfully restarted on span 1
    -- B-channel 0/10 successfully restarted on span 1
    -- B-channel 0/11 successfully restarted on span 1
    -- B-channel 0/12 successfully restarted on span 1
    -- B-channel 0/13 successfully restarted on span 1
    -- B-channel 0/14 successfully restarted on span 1
    -- B-channel 0/15 successfully restarted on span 1
    -- B-channel 0/17 successfully restarted on span 1
    -- B-channel 0/18 successfully restarted on span 1
    -- B-channel 0/19 successfully restarted on span 1
    -- B-channel 0/20 successfully restarted on span 1
    -- B-channel 0/21 successfully restarted on span 1
    -- B-channel 0/22 successfully restarted on span 1
    -- B-channel 0/23 successfully restarted on span 1
    -- B-channel 0/24 successfully restarted on span 1
    -- B-channel 0/25 successfully restarted on span 1
    -- B-channel 0/26 successfully restarted on span 1
    -- B-channel 0/27 successfully restarted on span 1
    -- B-channel 0/28 successfully restarted on span 1
    -- B-channel 0/29 successfully restarted on span 1
    -- B-channel 0/30 successfully restarted on span 1
    -- B-channel 0/31 successfully restarted on span 1
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:84@voip.sysfrog.org SIP/2.0
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=910cf8b324a3aa7co0;lr=on>
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec
From: u6 <sip:u6@voip.sysfrog.org>;tag=910cf8b324a3aa7co0
To: <sip:84@voip.sysfrog.org>
Call-ID: be6c723f-5b799e60@10.122.32.77
CSeq: 102 INVITE
Max-Forwards: 16
Proxy-Authorization: Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@voip.sysfrog.org",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"
Contact: u6 <sip:u6@XXX.XXX.XXX.XXX:1409>
Expires: 240
User-Agent: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 426
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 42717460 42717460 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32358 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (17 headers 19 lines)---
Using INVITE request as basis request - be6c723f-5b799e60@10.122.32.77
Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port XXX.XXX.XXX.XXX:32358
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G729a
Found description format G726-40
Found description format G726-24
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 84 in incoming-sip
list_route: hop: <sip:YYY.YYY.YYY.YYY;ftag=910cf8b324a3aa7co0;lr=on>
list_route: hop: <sip:u6@XXX.XXX.XXX.XXX:1409>
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec
From: u6 <sip:u6@voip.sysfrog.org>;tag=910cf8b324a3aa7co0
To: <sip:84@voip.sysfrog.org>
Call-ID: be6c723f-5b799e60@10.122.32.77
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:84@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
    -- Executing Set("SIP/voip.sysfrog.org-081c3000", "foo=Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@voip.sysfrog.org",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"") in new stack
    -- Executing Dial("SIP/voip.sysfrog.org-081c3000", "SIP/u4@voip.sysfrog.org") in new stack
We're at ZZZ.ZZZ.ZZZ.ZZZ port 17366
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
INVITE sip:u4@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 20 Jul 2005 14:20:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 12249 12249 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 17366 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called u4@voip.sysfrog.org
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells:  pid=12007 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:u4@voip.sysfrog.org out_uri=sip:u4@XXX.XXX.XXX.XXX:37701 via_cnt==1"


--- (9 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Record-Route: <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de>
Content-Length: 0


--- (12 headers 0 lines)---
    -- SIP/voip.sysfrog.org-590d is ringing
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec
From: u6 <sip:u6@voip.sysfrog.org>;tag=910cf8b324a3aa7co0
To: <sip:84@voip.sysfrog.org>;tag=as43a7adb7
Call-ID: be6c723f-5b799e60@10.122.32.77
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:84@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Record-Route: <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de>
Content-Length: 0


--- (12 headers 0 lines)---
    -- SIP/voip.sysfrog.org-590d is ringing
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Record-Route: <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de>
Content-Type: application/sdp
Content-Length: 218

v=0
o=u4 1121869194 1121869194 IN IP4 XXX.XXX.XXX.XXX
s=_
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32362 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
a=ptime:20
a=sendrecv

--- (13 headers 11 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port XXX.XXX.XXX.XXX:32362
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de>
list_route: hop: <sip:u4@XXX.XXX.XXX.XXX:37701>
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de> for address/port to send to
set_destination: set destination to YYY.YYY.YYY.YYY, port 5060
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
ACK sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6a0b1dcc
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-590d answered SIP/voip.sysfrog.org-081c3000
We're at ZZZ.ZZZ.ZZZ.ZZZ port 11478
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=910cf8b324a3aa7co0;lr=on>
From: u6 <sip:u6@voip.sysfrog.org>;tag=910cf8b324a3aa7co0
To: <sip:84@voip.sysfrog.org>;tag=as43a7adb7
Call-ID: be6c723f-5b799e60@10.122.32.77
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:84@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 12249 12249 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 11478 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/voip.sysfrog.org-081c3000 and SIP/voip.sysfrog.org-590d
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
ACK sip:84@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-596204f
From: u6 <sip:u6@voip.sysfrog.org>;tag=910cf8b324a3aa7co0
To: <sip:84@voip.sysfrog.org>;tag=as43a7adb7
Call-ID: be6c723f-5b799e60@10.122.32.77
CSeq: 102 ACK
Max-Forwards: 16
Proxy-Authorization: Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@ZZZ.ZZZ.ZZZ.ZZZ",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"
Contact: u6 <sip:u6@XXX.XXX.XXX.XXX:1409>
User-Agent: Sipura/SPA1001-2.0.13(SEg)
Content-Length: 0


--- (12 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 1 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKac44.0fb4e8b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09b94-12fc
Max-Forwards: 16
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Content-Type: application/sdp
Content-Length: 213

v=0
o=u4 1121869194 1121869195 IN IP4 10.122.32.100
s=_
c=IN IP4 0.0.0.0
t=0 0
m=audio 8006 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
a=ptime:20
a=sendonly

--- (14 headers 11 lines)---
Using INVITE request as basis request - 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:8006
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-081c3000
We're at ZZZ.ZZZ.ZZZ.ZZZ port 17366
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKac44.0fb4e8b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09b94-12fc
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 1 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
ontent-Type: application/sdp
Content-Length: 261

v=0
o=root 12249 12250 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 17366 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jul 20 16:20:04 NOTICE[12249]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-081c3000 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik'
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
ACK sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 1 ACK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09c0c-d85
Max-Forwards: 16
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Content-Length: 0


--- (12 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:85@voip.sysfrog.org SIP/2.0
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5d90;lr=on>
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 2 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594
Max-Forwards: 16
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Proxy-Authorization: Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5
Content-Type: application/sdp
Content-Length: 242

v=0
o=u4 1121869200 1121869201 IN IP4 XXX.XXX.XXX.XXX
s=_
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32370 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
a=ptime:20
a=sendrecv

--- (16 headers 12 lines)---
Using INVITE request as basis request - 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT)
Found peer 'sipsepp'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port XXX.XXX.XXX.XXX:32370
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 85 in incoming-sip
list_route: hop: <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5d90;lr=on>
list_route: hop: <sip:u4@XXX.XXX.XXX.XXX:37701>
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
    -- Executing Set("SIP/voip.sysfrog.org-081eb528", "foo=Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5") in new stack
    -- Executing Dial("SIP/voip.sysfrog.org-081eb528", "SIP/u5@voip.sysfrog.org") in new stack
We're at ZZZ.ZZZ.ZZZ.ZZZ port 10524
Answering/Requesting with root capability 0x8 (alaw)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
INVITE sip:u5@voip.sysfrog.org SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as5647a7a8
To: <sip:u5@voip.sysfrog.org>
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 20 Jul 2005 14:20:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 12252 12252 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 10524 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called u5@voip.sysfrog.org
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as5647a7a8
To: <sip:u5@voip.sysfrog.org>
Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Server: OpenSer (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells:  pid=12012 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:u5@voip.sysfrog.org out_uri=sip:u5@XXX.XXX.XXX.XXX:1265 via_cnt==1"


--- (9 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
To: <sip:u5@voip.sysfrog.org>;tag=37d443a25925fd55i0
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as5647a7a8
Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=as5647a7a8;lr=on>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 0


--- (9 headers 0 lines)---
    -- SIP/voip.sysfrog.org-fde7 is ringing
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=37d443a25925fd55i0
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as5647a7a8
Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=as5647a7a8;lr=on>
Contact: u5 <sip:u5@XXX.XXX.XXX.XXX:1265>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 657936 657936 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32374 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port XXX.XXX.XXX.XXX:32374
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:YYY.YYY.YYY.YYY;ftag=as5647a7a8;lr=on>
list_route: hop: <sip:u5@XXX.XXX.XXX.XXX:1265>
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;ftag=as5647a7a8;lr=on> for address/port to send to
set_destination: set destination to YYY.YYY.YYY.YYY, port 5060
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
ACK sip:u5@XXX.XXX.XXX.XXX:1265 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2a1706c5
Route: <sip:u5@XXX.XXX.XXX.XXX:1265>
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as5647a7a8
To: <sip:u5@voip.sysfrog.org>;tag=37d443a25925fd55i0
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-fde7 answered SIP/voip.sysfrog.org-081eb528
We're at ZZZ.ZZZ.ZZZ.ZZZ port 14242
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5d90;lr=on>
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 2 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 12252 12252 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 14242 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/voip.sysfrog.org-081eb528 and SIP/voip.sysfrog.org-fde7
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
ACK sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 2 ACK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d92-5c0afd0-683d
Max-Forwards: 16
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Proxy-Authorization: Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5
Content-Length: 0


--- (12 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
INVITE sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 3 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKd9f4.3a305df.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0b9e4-983
Max-Forwards: 16
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Content-Type: application/sdp
Content-Length: 237

v=0
o=u4 1121869200 1121869202 IN IP4 10.122.32.100
s=_
c=IN IP4 0.0.0.0
t=0 0
m=audio 8008 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15,16
a=ptime:20
a=sendonly

--- (14 headers 12 lines)---
Using INVITE request as basis request - 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 0.0.0.0:8008
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-fde7
We're at ZZZ.ZZZ.ZZZ.ZZZ port 14242
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKd9f4.3a305df.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0b9e4-983
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 3 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Type: application/sdp
Content-Length: 214

v=0
o=root 12252 12253 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 14242 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jul 20 16:20:12 NOTICE[12252]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-fde7 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik'
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
ACK sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
To: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 3 ACK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba5c-2700
Max-Forwards: 16
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Content-Length: 0


--- (12 headers 0 lines)---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
REFER sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 2 REFER
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK7c44.12f9be5.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba7a-157e
Max-Forwards: 16
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Referred-By: <sip:u4@voip.sysfrog.org>
Refer-To: <sip:85@voip.sysfrog.org?Replaces=1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b%4010.122.32.100%3Bto-tag%3Das2e10b33f%3Bfrom-tag%3D64207a0a-13c442de5d90>
Content-Length: 0


--- (15 headers 0 lines)---
Transfer to 85 in incoming-sip-default
Transfer from u4 in incoming-sip-default
    -- Stopped music on hold on SIP/voip.sysfrog.org-081c3000
    -- Stopped music on hold on SIP/voip.sysfrog.org-fde7
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK7c44.12f9be5.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba7a-157e
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 2 REFER
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de> for address/port to send to
set_destination: set destination to YYY.YYY.YYY.YYY, port 5060
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
NOTIFY sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6fd9a84b
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=2
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;lr;ftag=as07b5d5de> for address/port to send to
set_destination: set destination to YYY.YYY.YYY.YYY, port 5060
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
BYE sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK363d538b
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 104 BYE
User-Agent: Asterisk
Content-Length: 0


---
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5d90;lr=on> for address/port to send to
set_destination: set destination to YYY.YYY.YYY.YYY, port 5060
Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
BYE sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK5bee0904;rport
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
To: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 102 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 103 NOTIFY
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6fd9a84b
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Contact: <sip:u4@XXX.XXX.XXX.XXX:37701>
Content-Length: 0


--- (11 headers 0 lines)---
!!!!!!!---------------************* Why are we here with this packet???? NOTIFY
Response message is NOTIFY
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
BYE sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 3 BYE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8c44.764533a3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0bb10-61f5
Max-Forwards: 16
Supported: 100rel,replaces
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
User-Agent: SIP Phone-1.0.57
Content-Length: 0


--- (12 headers 0 lines)---
Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT)
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8c44.764533a3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0bb10-61f5
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 3 BYE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Content-Length: 0


---
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as07b5d5de
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d8a
Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 104 BYE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK363d538b
Supported: 100rel,replaces
User-Agent: SIP Phone-1.0.57
Content-Length: 0


--- (9 headers 0 lines)---
!!!!!!!---------------************* Why are we here with this packet???? BYE
Response message is BYE
Destroying call '07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ'
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
From: <sip:85@voip.sysfrog.org>;tag=as2e10b33f
To: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5d90
Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100
CSeq: 102 BYE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;rport=5060;branch=z9hG4bK5bee0904
Supported: 100rel,replaces
User-Agent: SIP Phone-1.0.57
Content-Length: 0


--- (9 headers 0 lines)---
!!!!!!!---------------************* Why are we here with this packet???? BYE
Response message is BYE
Destroying call '1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100'
obelix*CLI> quit
Executing last minute cleanups
Asterisk ending (0).
obelix:~#
obelix:~#