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axonite-ivr:/var/lib/asterisk# asterisk -r
Asterisk 1.4.17~dfsg-2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core 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 'core show license' for details.
=========================================================================
This package has been modified for the Debian GNU/Linux distribution
Please report all bugs to http://bugs.debian.org/asterisk
=========================================================================
Connected to Asterisk 1.4.17~dfsg-2 currently running on axonite-ivr (pid = 1354)
axonite-ivr*CLI> sip debug
SIP Debugging enabled
The 'sip debug' command is deprecated and will be removed in a future release. Please use 'sip set debug' instead.
[Feb  7 02:49:14] WARNING[1366]: pbx_spool.c:276 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/aiurea.call: Operation not permitted
Audio is at 10.12.0.7 port 18910
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 10.12.0.7:5060;branch=z9hG4bK574a2efa;rport
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215699522@10.12.0.7>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 102 INVITE
User-Agent: asterisk
Max-Forwards: 70
Date: Thu, 07 Feb 2008 00:49:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 1354 1354 IN IP4 10.12.0.7
s=session
c=IN IP4 10.12.0.7
t=0 0
m=audio 18910 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
axonite-ivr*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 10.12.0.7:5060;branch=z9hG4bK574a2efa
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
axonite-ivr*CLI>
<--- SIP read from 193.16.148.244:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK574a2efa
Record-Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: asterisk <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="193.16.148.244",nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b"


<------------->
--- (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 10.12.0.7:5060;branch=z9hG4bK574a2efa;rport
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215699522@10.12.0.7>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 102 ACK
User-Agent: asterisk
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.12.0.7 port 18910
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 10.12.0.7:5060;branch=z9hG4bK23f1e03a;rport
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Contact: <sip:40215699522@10.12.0.7>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 103 INVITE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:0743049336@sip.netmaster.ro", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="cce1f1eb8b6a9b8f7ee8e30b926e75dd", opaque=""
Date: Thu, 07 Feb 2008 00:49:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 176

v=0
o=root 1354 1355 IN IP4 10.12.0.7
s=session
c=IN IP4 10.12.0.7
t=0 0
m=audio 18910 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
axonite-ivr*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 10.12.0.7:5060;branch=z9hG4bK23f1e03a
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 103 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
axonite-ivr*CLI>
<--- SIP read from 193.16.148.244:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK23f1e03a
Record-Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: asterisk <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 103 INVITE
Server: Sippy


<------------->
--- (8 headers 0 lines) ---
axonite-ivr*CLI>
<--- SIP read from 193.16.148.244:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK23f1e03a
Record-Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: asterisk <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 103 INVITE
Server: Sippy
Contact: Anonymous <sip:193.16.148.244:5061>
Content-Length: 160
Content-Type: application/sdp

v=0
o=Sippy 148624940 1 IN IP4 193.16.148.226
s=SIP Call
t=0 0
m=audio 19064 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 193.16.148.226:19064
Found audio description format PCMU for ID 0
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 193.16.148.226:19064
list_route: hop: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
set_destination: Parsing <sip:193.16.148.244:5060;ftag=as7570d174;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 10.12.0.7:5060;branch=z9hG4bK0d2310a8;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Contact: <sip:40215699522@10.12.0.7>
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 103 ACK
User-Agent: asterisk
Max-Forwards: 70
Content-Length: 0

!!At this point I hang up. The processing continues 15 seconds in the wait and then exits

---
Really destroying SIP dialog '3632eb4131842ab94bc5a67714453540@10.12.0.7' Method: REGISTER
Really destroying SIP dialog '1f66166f7d2c09db22dfbf9139088334@10.12.0.7' Method: REGISTER
Really destroying SIP dialog '63778e6c3f3199955cb7a4110f33197a@10.12.0.7' Method: REGISTER
Really destroying SIP dialog '737ffc1f3f992c6f7a7e12e0025dc967@10.12.0.7' Method: REGISTER
Really destroying SIP dialog '26f0c3586ab7a4465035378c4c2fccf0@10.12.0.7' Method: REGISTER
Scheduling destruction of SIP dialog '07ceba056ddfefe13576fc8906c2b180@netmaster.ro' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:193.16.148.244:5060;ftag=as7570d174;lr> for address/port to send to
set_destination: set destination to 193.16.148.244, port 5060
Reliably Transmitting (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
[Feb  7 02:49:40] NOTICE[1394]: pbx_spool.c:371 attempt_thread: Call completed to SIP/0743049336@smartcall0
Retransmitting #1 (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
Retransmitting #2 (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
Retransmitting #3 (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
Retransmitting #4 (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
Retransmitting #5 (no NAT) to 193.16.148.244:5060:
BYE sip:193.16.148.244:5061 SIP/2.0
Via: SIP/2.0/UDP 10.12.0.7:5060;branch=z9hG4bK1150e06d;rport
Route: <sip:193.16.148.244:5060;ftag=as7570d174;lr>
From: "asterisk" <sip:40215699522@netmaster.ro>;tag=as7570d174
To: <sip:0743049336@sip.netmaster.ro>;tag=2c687455bcba68aa0eb62af92f79ab3e
Call-ID: 07ceba056ddfefe13576fc8906c2b180@netmaster.ro
CSeq: 104 BYE
User-Agent: asterisk
Max-Forwards: 70
Authorization: Digest username="40215699522", realm="193.16.148.244", algorithm=MD5, uri="sip:193.16.148.244:5061", nonce="eac4726f4e161bdd56e43e8137ebcbb847aa022b", response="a8d16aad08d33cc44f47a61690aeca0c", opaque=""
Content-Length: 0


---
axonite-ivr*CLI>