Public feed #894152

Someone

public text v1 · immutable
Raw Edit
·published 2008-02-06 18:56 UTC
rendered paste body
call:/var/spool/asterisk# asterisk -r
Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.13 currently running on call (pid = 16320)
call*CLI> sip debug
SIP Debugging enabled
call*CLI> quit
call:/var/spool/asterisk# ./makecall.sh
call:/var/spool/asterisk# asterisk -r
Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.13 currently running on call (pid = 16320)
Feb  6 20:25:04 WARNING[16326]: pbx_spool.c:251 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/aiure1a.call: Operation not permitted
We're at 80.86.113.155 port 18928
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
INVITE sip:0743049336@sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215694650@80.86.113.155>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 16320 16320 IN IP4 80.86.113.155
s=session
c=IN IP4 80.86.113.155
t=0 0
m=audio 18928 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport=5060
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines) ---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport=5060
Record-Route: <sip:193.16.148.244;ftag=as487b799e;lr>
From: asterisk <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="193.16.148.244",nonce="cf172c1f0030bf651f93f69cc8d8aa7347a9fb80"


--- (9 headers 0 lines) ---
Transmitting (no NAT) to 193.16.148.244:5060:
ACK sip:0743049336@sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5b69baad;rport
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215694650@80.86.113.155>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 102 ACK
User-Agent: asterisk
Max-Forwards: 70
Content-Length: 0


---
We're at 80.86.113.155 port 18928
Adding codec 0x4 (ulaw) to SDP
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
INVITE sip:0743049336@sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215694650@80.86.113.155>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 103 INVITE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215694650", realm="193.16.148.244", algorithm=MD5, uri="sip:0743049336@sip.netmaster.ro", nonce="cf172c1f0030bf651f93f69cc8d8aa7347a9fb80", response="0aac3635635f9e25cd38648e8486511d", opaque=""
Date: Wed, 06 Feb 2008 18:25:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 162

v=0
o=root 16320 16321 IN IP4 80.86.113.155
s=session
c=IN IP4 80.86.113.155
t=0 0
m=audio 18928 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -

---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 103 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


--- (8 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK65842f90;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as77bba60c
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK65842f90;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as77bba60c
To: <sip:sip.netmaster.ro>
Call-ID: 24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '24276c180c6ea1793e91ebb46b0a25fd@80.86.113.155'
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060
Record-Route: <sip:193.16.148.244;ftag=as487b799e;lr>
From: asterisk <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>;tag=b88cae893f45b54799c6de2842e6596c
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 103 INVITE
Server: Sippy


--- (8 headers 0 lines) ---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK12c1bb34;rport=5060
Record-Route: <sip:193.16.148.244;ftag=as487b799e;lr>
From: asterisk <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>;tag=b88cae893f45b54799c6de2842e6596c
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 103 INVITE
Server: Sippy
Contact: Anonymous <sip:193.16.148.244:5061>
Content-Length: 160
Content-Type: application/sdp

=0ll*CLI>
o=Sippy 139289996 1 IN IP4 193.16.148.244
s=SIP Call
t=0 0
m=audio 19030 RTP/AVP 0
c=IN IP4 193.16.148.226
a=rtpmap:0 PCMU/8000
a=direction:passive

--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 193.16.148.226:19030
Found description format PCMU
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
list_route: hop: <sip:193.16.148.244;ftag=as487b799e;lr>
set_destination: Parsing <sip:193.16.148.244;ftag=as487b799e;lr> for address/port to send to
set_destination: set destination to 193.16.148.244, port 5060
Transmitting (no NAT) to 193.16.148.244:5060:
ACK sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK62119997;rport
Route: <sip:193.16.148.244;ftag=as487b799e;lr>
From: "asterisk" <sip:40215694650@netmaster.ro>;tag=as487b799e
To: <sip:0743049336@sip.netmaster.ro>;tag=b88cae893f45b54799c6de2842e6596c
Contact: <sip:40215694650@80.86.113.155>
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 103 ACK
User-Agent: asterisk
Max-Forwards: 70
Content-Length: 0


---
Feb  6 20:25:13 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format
Feb  6 20:25:13 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory
Feb  6 20:25:13 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test
Feb  6 20:25:14 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format
Feb  6 20:25:14 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory
Feb  6 20:25:14 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test
!! The files are missing, this doesn't change the setup
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK07ed2d00;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as4f7286f4
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 47793f687d077bf056b07cb20ffc5eae@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK07ed2d00;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as4f7286f4
To: <sip:sip.netmaster.ro>
Call-ID: 47793f687d077bf056b07cb20ffc5eae@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '47793f687d077bf056b07cb20ffc5eae@80.86.113.155'
Feb  6 20:25:15 WARNING[23141]: file.c:512 ast_openstream_full: File /tmp/test does not exist in any format
Feb  6 20:25:15 WARNING[23141]: file.c:824 ast_streamfile: Unable to open /tmp/test (format ulaw): No such file or directory
Feb  6 20:25:15 WARNING[23141]: pbx.c:5799 pbx_builtin_background: ast_streamfile failed on SIP/smartcall0-081dae98 for /tmp/test
call*CLI>
<-- SIP read from 193.16.148.244:5060:
BYE sip:40215694650@80.86.113.155 SIP/2.0
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKb087.ed99b927cbab958fd385005c501f5534.0
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bKc4573e9d3872d522005ed7040940405d;rport=5061
Max-Forwards: 16
From: <sip:0743049336@sip.netmaster.ro>;tag=b88cae893f45b54799c6de2842e6596c
To: asterisk <sip:40215694650@netmaster.ro>;tag=as487b799e
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 100 BYE
Contact: Anonymous <sip:193.16.148.244:5061>
Expires: 300
User-Agent: Sippy
cisco-GUID: 2400891340-3571454428-2228224023-2760872502
h323-conf-id: 2400891340-3571454428-2228224023-2760872502

!!! this is about where I hang up

--- (13 headers 0 lines) ---
Sending to 193.16.148.244 : 5060 (non-NAT)
Transmitting (no NAT) to 193.16.148.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.16.148.244;branch=z9hG4bKb087.ed99b927cbab958fd385005c501f5534.0;received=193.16.148.244
Via: SIP/2.0/UDP 193.16.148.244:5061;branch=z9hG4bKc4573e9d3872d522005ed7040940405d;rport=5061
From: <sip:0743049336@sip.netmaster.ro>;tag=b88cae893f45b54799c6de2842e6596c
To: asterisk <sip:40215694650@netmaster.ro>;tag=as487b799e
Call-ID: 1a3884401ab152fd3d3e18fc3f855152@netmaster.ro
CSeq: 100 BYE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:40215694650@80.86.113.155>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
Feb  6 20:25:16 NOTICE[23141]: pbx_spool.c:279 attempt_thread: Call completed to SIP/0743049336@smartcall0
Destroying call '1a3884401ab152fd3d3e18fc3f855152@netmaster.ro'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK1a5f44bd;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as0aed1590
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK1a5f44bd;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as0aed1590
To: <sip:sip.netmaster.ro>
Call-ID: 259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '259b0c8b21c68dd96c5640c1462b7af0@80.86.113.155'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK309ae7a0;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as3f3a0b05
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 6f93781a179da7dd7ab4d905272f5483@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK309ae7a0;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as3f3a0b05
To: <sip:sip.netmaster.ro>
Call-ID: 6f93781a179da7dd7ab4d905272f5483@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '6f93781a179da7dd7ab4d905272f5483@80.86.113.155'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.136:5060:
OPTIONS sip:*201@192.168.0.136:5060 SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5249fd38
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as641fa3f3
To: <sip:*201@192.168.0.136:5060>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 388125593aac8ebf43a85b921454e2f4@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 192.168.0.136:5060:
SIP/2.0 200 OK
To: <sip:*201@192.168.0.136:5060>;tag=81133be08fb82b0bi0
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as641fa3f3
Call-ID: 388125593aac8ebf43a85b921454e2f4@80.86.113.155
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5249fd38
Server: Linksys/SPA941-5.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


--- (10 headers 0 lines) ---
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.136:5061:
OPTIONS sip:*202@192.168.0.136:5061 SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK67951513
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as067cd32e
To: <sip:*202@192.168.0.136:5061>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 16db13571eee6f2d16aa15396506b26d@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '388125593aac8ebf43a85b921454e2f4@80.86.113.155'
call*CLI>
<-- SIP read from 192.168.0.136:5061:
SIP/2.0 200 OK
To: <sip:*202@192.168.0.136:5061>;tag=e1756d38f5cfe993i1
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as067cd32e
Call-ID: 16db13571eee6f2d16aa15396506b26d@80.86.113.155
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK67951513
Server: Linksys/SPA941-5.1.5
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces


--- (10 headers 0 lines) ---
Destroying call '16db13571eee6f2d16aa15396506b26d@80.86.113.155'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5d3322f2;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as0fc21d1b
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 578bb0a4485bd3483a24f57909977c6e@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK5d3322f2;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as0fc21d1b
To: <sip:sip.netmaster.ro>
Call-ID: 578bb0a4485bd3483a24f57909977c6e@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '578bb0a4485bd3483a24f57909977c6e@80.86.113.155'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
OPTIONS sip:sip.netmaster.ro SIP/2.0
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK3d1085f9;rport
From: "asterisk" <sip:asterisk@80.86.113.155>;tag=as5b319304
To: <sip:sip.netmaster.ro>
Contact: <sip:asterisk@80.86.113.155>
Call-ID: 5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155
CSeq: 102 OPTIONS
User-Agent: asterisk
Max-Forwards: 70
Date: Wed, 06 Feb 2008 18:25:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
call*CLI>
<-- SIP read from 193.16.148.244:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 80.86.113.155:5060;branch=z9hG4bK3d1085f9;rport=5060
From: asterisk <sip:asterisk@80.86.113.155>;tag=as5b319304
To: <sip:sip.netmaster.ro>
Call-ID: 5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155
CSeq: 102 OPTIONS
Server: Sippy


--- (7 headers 0 lines) ---
Destroying call '5a46434e506a527f4f3ebd235a2bd2b3@80.86.113.155'
call*CLI>