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 = 12171)
Verbosity is at least 4
    -- Remote UNIX connection
obelix*CLI> sip debug
SIP Debugging enabled
obelix*CLI> set verobse 4
No such command 'set verobse' (type 'help' for help)
    -- 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
obelix*CLI> set verbose 4
Verbosity is at least 4
    -- 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> debug level 4
Debugging level set to 4, file '<any>'
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=45641578bbe2bfb0o0;lr=on>
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc8c8.c4be8e97.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-342dacec
From: u6 <sip:u6@voip.sysfrog.org>;tag=45641578bbe2bfb0o0
To: <sip:84@voip.sysfrog.org>
Call-ID: 17137f18-aa916290@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=- 42689609 42689609 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32344 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 - 17137f18-aa916290@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:32344
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=45641578bbe2bfb0o0;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=z9hG4bKc8c8.c4be8e97.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-342dacec
From: u6 <sip:u6@voip.sysfrog.org>;tag=45641578bbe2bfb0o0
To: <sip:84@voip.sysfrog.org>
Call-ID: 17137f18-aa916290@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 Python("SIP/voip.sysfrog.org-0821a198", "incoming_sip_dial|84|voip.sysfrog.org|Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@voip.sysfrog.org",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"|proxy") in new stack
We're at ZZZ.ZZZ.ZZZ.ZZZ port 18946
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=z9hG4bK35cacdd3
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
To: <sip:u4@voip.sysfrog.org>
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 20 Jul 2005 14:15:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 12191 12191 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 18946 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
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc8c8.c4be8e97.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-342dacec
From: u6 <sip:u6@voip.sysfrog.org>;tag=45641578bbe2bfb0o0
To: <sip:84@voip.sysfrog.org>;tag=as637ef855
Call-ID: 17137f18-aa916290@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/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK35cacdd3
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
To: <sip:u4@voip.sysfrog.org>
Call-ID: 08fc66202784e8c91281ba1134066cd0@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=12008 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=as7299d0a2
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK35cacdd3
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=as7299d0a2>
Content-Length: 0


--- (12 headers 0 lines)---
    -- SIP/voip.sysfrog.org-0c9b 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=as7299d0a2
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK35cacdd3
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=as7299d0a2>
Content-Type: application/sdp
Content-Length: 218

v=0
o=u4 1121868916 1121868916 IN IP4 XXX.XXX.XXX.XXX
s=_
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32335 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:32335
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=as7299d0a2>
list_route: hop: <sip:u4@XXX.XXX.XXX.XXX:37701>
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;lr;ftag=as7299d0a2> 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=z9hG4bK3c4dfa82
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-0c9b answered SIP/voip.sysfrog.org-0821a198
We're at ZZZ.ZZZ.ZZZ.ZZZ port 19378
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=z9hG4bKc8c8.c4be8e97.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-342dacec
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=45641578bbe2bfb0o0;lr=on>
From: u6 <sip:u6@voip.sysfrog.org>;tag=45641578bbe2bfb0o0
To: <sip:84@voip.sysfrog.org>;tag=as637ef855
Call-ID: 17137f18-aa916290@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 12191 12191 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 19378 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-0821a198 and SIP/voip.sysfrog.org-0c9b
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-d40eef71
From: u6 <sip:u6@voip.sysfrog.org>;tag=45641578bbe2bfb0o0
To: <sip:84@voip.sysfrog.org>;tag=as637ef855
Call-ID: 17137f18-aa916290@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-13c442de5c74
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 1 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKb1c5.2c6e06a3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c76-5bc5b60-344b
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 1121868916 1121868917 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 - 08fc66202784e8c91281ba1134066cd0@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-0821a198
We're at ZZZ.ZZZ.ZZZ.ZZZ port 18946
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=z9hG4bKb1c5.2c6e06a3.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c76-5bc5b60-344b
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
Call-ID: 08fc66202784e8c91281ba1134066cd0@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>
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 12191 12192 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 18946 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 16:15:26 NOTICE[12191]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-0821a198 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-13c442de5c74
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
Call-ID: 08fc66202784e8c91281ba1134066cd0@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-42de5c77-5bc5bce-4e28
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-13c442de5c79;lr=on>
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
To: <sip:85@voip.sysfrog.org>
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@10.122.32.100
CSeq: 2 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8259.7db9a564.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c79-5bc65ce-61c3
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 1121868921 1121868922 IN IP4 XXX.XXX.XXX.XXX
s=_
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32312 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 - 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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:32312
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-13c442de5c79;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=z9hG4bK8259.7db9a564.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c79-5bc65ce-61c3
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
To: <sip:85@voip.sysfrog.org>
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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 Python("SIP/voip.sysfrog.org-08248d98", "incoming_sip_dial|85|voip.sysfrog.org|Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5|proxy") in new stack
We're at ZZZ.ZZZ.ZZZ.ZZZ port 19994
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=z9hG4bK70e04b75
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as4280b4f1
To: <sip:u5@voip.sysfrog.org>
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 0de1c90e0c72b93f0188fec02bd857bf@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
User-Agent: Asterisk
Date: Wed, 20 Jul 2005 14:15:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 261

v=0
o=root 12195 12195 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 19994 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
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8259.7db9a564.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c79-5bc65ce-61c3
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK70e04b75
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as4280b4f1
To: <sip:u5@voip.sysfrog.org>
Call-ID: 0de1c90e0c72b93f0188fec02bd857bf@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=12008 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=946e6f0656e0bc24i0
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as4280b4f1
Call-ID: 0de1c90e0c72b93f0188fec02bd857bf@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK70e04b75
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=as4280b4f1;lr=on>
Server: Sipura/SPA841-3.1.2(d)
Content-Length: 0


--- (9 headers 0 lines)---
    -- SIP/voip.sysfrog.org-45c1 is ringing
obelix*CLI>
<-- SIP read from YYY.YYY.YYY.YYY:5060:
SIP/2.0 200 OK
To: <sip:u5@voip.sysfrog.org>;tag=946e6f0656e0bc24i0
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as4280b4f1
Call-ID: 0de1c90e0c72b93f0188fec02bd857bf@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 INVITE
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK70e04b75
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=as4280b4f1;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=- 630003 630003 IN IP4 XXX.XXX.XXX.XXX
s=-
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 32348 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:32348
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=as4280b4f1;lr=on>
list_route: hop: <sip:u5@XXX.XXX.XXX.XXX:1265>
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;ftag=as4280b4f1;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=z9hG4bK7c8f2d91
Route: <sip:u5@XXX.XXX.XXX.XXX:1265>
From: "u4" <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as4280b4f1
To: <sip:u5@voip.sysfrog.org>;tag=946e6f0656e0bc24i0
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 0de1c90e0c72b93f0188fec02bd857bf@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0


---
    -- SIP/voip.sysfrog.org-45c1 answered SIP/voip.sysfrog.org-08248d98
We're at ZZZ.ZZZ.ZZZ.ZZZ port 13610
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=z9hG4bK8259.7db9a564.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c79-5bc65ce-61c3
Record-Route: <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5c79;lr=on>
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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 12195 12195 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 13610 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-08248d98 and SIP/voip.sysfrog.org-45c1
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-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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-42de5c7b-5bc6d08-3a40
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-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@10.122.32.100
CSeq: 3 INVITE
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK9259.bde5d863.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c7e-5bc796a-7abe
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 1121868921 1121868923 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 - 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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-45c1
We're at ZZZ.ZZZ.ZZZ.ZZZ port 13610
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=z9hG4bK9259.bde5d863.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c7e-5bc796a-7abe
From: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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 12195 12196 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=session
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=audio 13610 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 16:15:33 NOTICE[12195]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-45c1 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-13c442de5c79
To: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@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-42de5c7e-5bc79e2-571f
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-13c442de5c74
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 2 REFER
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK81c5.85f4b592.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c7e-5bc7a00-7835
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=1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6%4010.122.32.100%3Bto-tag%3Das32fe60c4%3Bfrom-tag%3D64207a0a-13c442de5c79>
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-0821a198
    -- Stopped music on hold on SIP/voip.sysfrog.org-45c1
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;ftag=64207a0a-13c442de5c79;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=z9hG4bK4a8cc961;rport
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: <sip:85@voip.sysfrog.org>;tag=as32fe60c4
To: "u4"<sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c79
Contact: <sip:85@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 1023cb48-64207a0a-13c4-42de5c79-5bc656a-34e6@10.122.32.100
CSeq: 102 BYE
User-Agent: Asterisk
Content-Length: 0


---
Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK81c5.85f4b592.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5c7e-5bc7a00-7835
From: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
To: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
Call-ID: 08fc66202784e8c91281ba1134066cd0@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=as7299d0a2> 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=z9hG4bK6a650bcf
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 103 NOTIFY
User-Agent: Asterisk
Event: refer;id=2
Subscription-state: terminated;reason=noresource
ontent-Type: message/sipfrag;version=2.0
Content-Length: 14

SIP/2.0 200 OK
---
set_destination: Parsing <sip:YYY.YYY.YYY.YYY;lr;ftag=as7299d0a2> 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=z9hG4bK7f0ec6cb
Route: <sip:u4@XXX.XXX.XXX.XXX:37701>
From: "u6"<sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>;tag=as7299d0a2
To: <sip:u4@voip.sysfrog.org>;tag=64207a0a-13c442de5c74
Contact: <sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ>
Call-ID: 08fc66202784e8c91281ba1134066cd0@ZZZ.ZZZ.ZZZ.ZZZ
CSeq: 104 BYE
User-Agent: Asterisk
Content-Length: 0


---
obelix*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
Asterisk ending (0).
obelix:~#