Hello:
I followed the step by step guide
(http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb)
that describe the realtime integration
between Kamailio and Asterisk. I have no problem with registration but when I
try a call from 106 to 107 I get the followng error :
1. in the asterisk console: " Unresolvable destination (478/SL)"
2. in the kamailio log:
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: <core>
[resolve.c:1540]: ERROR: sip_hostport2su: could not resolve hostname: "(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [ut.h:327]: failed
to resolve "(null)"
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: tm [t_fwd.c:1530]:
ERROR: t_forward_nonack: failure to add branches
May 6 17:01:38 WH-PC /usr/sbin/kamailio[4192]: ERROR: sl [sl_funcs.c:371]:
ERROR: sl_reply_error used: Unresolvable destination (478/SL)
Any idea about the cause of this problem?
ps:kamailio and asterisk are running in the same machine.
ps:attachment is Asterisk CLI log
Best Regards,
zhengyw
郑友闻
医疗终端部
东软熙康健康有限公司
沈阳市浑南新区新秀街2号，A1楼216
Postcode：110179
Tel：+86 24 83662278
Mobile：15242493836
Email : zhen...@neusoft.com
http://www.neusoft.com
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---------------------------------------------------------------------------------------------------
<--- SIP read from UDP:10.11.2.47:5060 --->
INVITE sip:107@10.11.2.47 SIP/2.0
Record-Route: <sip:10.11.2.47;lr=on;ftag=29997>
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158
From: <sip:106@10.11.2.47>;tag=29997
To: "107" <sip:107@10.11.2.47>
Call-ID: 18342
CSeq: 21 INVITE
Contact: <sip:106@(null)>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Max-Forwards: 69
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Subject: Phone call
Content-Length: 349
v=0
o=106 976 976 IN IP4 10.11.2.37
s=Talk
c=IN IP4 10.11.2.37
t=0 0
m=audio 7078 RTP/AVP 110 3 0 8 101
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
<------------->
--- (15 headers 16 lines) ---
Sending to 10.11.2.47:5060 (NAT)
Using INVITE request as basis request - 18342
Found peer '106' for '106' from 10.11.2.47:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 98
Found video description format H263-1998 for ID 98
Capabilities: us - (gsm|ulaw|alaw|h263|h263p|h264|testlaw), peer -
audio=(gsm|ulaw|alaw|speex)/video=(h263p)/text=(nothing), combined -
(gsm|ulaw|alaw|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.11.2.37:7078
Peer video RTP is at port 10.11.2.37:9078
Looking for 107 in from-sip (domain 10.11.2.47)
list_route: hop: <sip:10.11.2.47;lr=on;ftag=29997>
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158
Record-Route: <sip:10.11.2.47;lr=on;ftag=29997>
From: <sip:106@10.11.2.47>;tag=29997
To: "107" <sip:107@10.11.2.47>
Call-ID: 18342
CSeq: 21 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:107@10.11.2.47:5080>
Content-Length: 0
<------------>
-- Executing [107@from-sip:1] Dial("SIP/106-0000001d", "SIP/107,60") in new
stack
== Using SIP RTP CoS mark 5
Audio is at 15732
Video is at 10.11.2.47:16908
Adding codec 100003 (ulaw) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
INVITE sip:107@10.11.2.47:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.7.0
Date: Wed, 08 May 2013 09:09:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 434
v=0
o=root 476348733 476348733 IN IP4 10.11.2.47
s=Asterisk PBX 10.7.0
c=IN IP4 10.11.2.47
b=CT:384
t=0 0
m=audio 15732 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16908 RTP/AVP 34 98 99
a=rtpmap:34 H263/90000
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Called SIP/107
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Record-Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Contact: <sip:107@10.11.2.50:5060>
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Record-Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Contact: <sip:107@10.11.2.50:5060>
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
list_route: hop: <sip:10.11.2.47;lr=on;ftag=as149287b0>
-- SIP/107-0000001e is ringing
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158
Record-Route: <sip:10.11.2.47;lr=on;ftag=29997>
From: <sip:106@10.11.2.47>;tag=29997
To: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
Call-ID: 18342
CSeq: 21 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:107@10.11.2.47:5080>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.11.2.47:5060 --->
REGISTER sip:10.11.2.47:5080 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK9f36.d8026017.0
To: sip:106@10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-972d
CSeq: 10 REGISTER
Call-ID: 5eefa7c2-4330@127.0.0.1
Content-Length: 0
User-Agent: kamailio (3.3.1 (i386/linux))
Contact: <sip:106@10.11.2.47:5060>
Expires: 60
<------------->
--- (10 headers 0 lines) ---
Sending to 10.11.2.47:5060 (NAT)
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK9f36.d8026017.0;received=10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-972d
To: sip:106@10.11.2.47;tag=as282227a3
Call-ID: 5eefa7c2-4330@127.0.0.1
CSeq: 10 REGISTER
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:106@10.11.2.47:5060>;expires=60
Date: Wed, 08 May 2013 09:09:22 GMT
Content-Length: 0
<------------>
[May 8 17:09:22] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:09:22] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:09:22] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:09:27] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:09:27] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:09:27] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:09:27] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:09:27] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:09:32] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:09:32] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:09:32] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute
error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/106/INBOX']
Scheduling destruction of SIP dialog
'2b27053f3491a2371bef4b0c12ff7754@10.11.2.47' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
NOTIFY sip:106@10.11.2.47:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK5c0ba38d
Max-Forwards: 70
From: "asterisk" <sip:106@10.11.2.47>;tag=as063ef50c
To: <sip:106@10.11.2.47:5060>
Contact: <sip:106@10.11.2.47:5080>
Call-ID: 2b27053f3491a2371bef4b0c12ff7754@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.7.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.11.2.47
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '5eefa7c2-4330@127.0.0.1' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Record-Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Contact: <sip:107@10.11.2.50>
Content-Type: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 342
v=0
o=107 3544 3544 IN IP4 10.11.2.50
s=Talk
c=IN IP4 10.11.2.50
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98 99
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 98
Found RTP video format 99
Found video description format H263-1998 for ID 98
Found video description format H264 for ID 99
Capabilities: us - (gsm|ulaw|alaw|h263|h263p|h264|testlaw), peer -
audio=(ulaw|alaw)/video=(h263p|h264)/text=(nothing), combined -
(ulaw|alaw|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.11.2.50:7078
Peer video RTP is at port 10.11.2.50:9078
list_route: hop: <sip:10.11.2.47;lr=on;ftag=as149287b0>
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK78a9184a
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Record-Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Contact: <sip:107@10.11.2.50>
Content-Type: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 342
v=0
o=107 3544 3544 IN IP4 10.11.2.50
s=Talk
c=IN IP4 10.11.2.50
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98 99
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
<------------->
-- SIP/107-0000001e answered SIP/106-0000001d
Audio is at 16686
Video is at 10.11.2.47:16818
Adding video codec 200003 (h263p) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK357.81bdcce5.0;received=10.11.2.47
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK4158
Record-Route: <sip:10.11.2.47;lr=on;ftag=29997>
From: <sip:106@10.11.2.47>;tag=29997
To: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
Call-ID: 18342
CSeq: 21 INVITE
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:107@10.11.2.47:5080>
Content-Type: application/sdp
Content-Length: 380
v=0
o=root 742615961 742615961 IN IP4 10.11.2.47
s=Asterisk PBX 10.7.0
c=IN IP4 10.11.2.47
b=CT:384
t=0 0
m=audio 16686 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16818 RTP/AVP 98
a=rtpmap:98 h263-1998/90000
a=sendrecv
<------------>
-- Remotely bridging SIP/106-0000001d and SIP/107-0000001e
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Audio is at 15732
--- (11 headers 15 lines) ---
Video is at 10.11.2.37:9078
Adding codec 100003 (ulaw) to SDP
Adding video codec 200003 (h263p) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
INVITE sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 103 INVITE
User-Agent: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 476348733 476348734 IN IP4 10.11.2.37
s=Asterisk PBX 10.7.0
c=IN IP4 10.11.2.37
b=CT:384
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 9078 RTP/AVP 98
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK2c4b9ccb
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK24c2b440
Record-Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 INVITE
Contact: <sip:107@10.11.2.50>
Content-Type: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 342
v=0
o=107 3544 3544 IN IP4 10.11.2.50
s=Talk
c=IN IP4 10.11.2.50
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98 99
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
<------------->
--- (11 headers 15 lines) ---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK14e1df5b
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 103 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK49de740f
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 103 INVITE
Contact: <sip:107@10.11.2.50>
Content-Type: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 280
v=0
o=107 3544 3544 IN IP4 10.11.2.50
s=Talk
c=IN IP4 10.11.2.50
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
<------------->
--- (10 headers 13 lines) ---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK0cee6d4d
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 103 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK5c0ba38d
From: "asterisk" <sip:106@10.11.2.47>;tag=as063ef50c
To: <sip:106@10.11.2.47:5060>;tag=22663
Call-ID: 2b27053f3491a2371bef4b0c12ff7754@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.11.2.47:5060 --->
ACK sip:107@10.11.2.47:5080 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 10.11.2.37:5060;rport=5060;branch=z9hG4bK31853
From: <sip:106@10.11.2.47>;tag=29997
To: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
Call-ID: 18342
CSeq: 21 ACK
Contact: <sip:106@10.11.2.37>
Proxy-Authorization: Digest username="106", realm="10.11.2.47",
nonce="UYoXa1GKFj+vG/6Ro/04MgF5oIIWA37e", uri="sip:107@10.11.2.47",
response="ab8737041d9609bddd05bc5956a6604c", algorithm=MD5
Max-Forwards: 69
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=29997> for address/port to
send to
set_destination: set destination to 10.11.2.47:5060
Audio is at 16686
Video is at 10.11.2.50:9078
Adding video codec 200003 (h263p) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
INVITE sip:106@(null) SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
Route: <sip:10.11.2.47;lr=on;ftag=29997>
Max-Forwards: 70
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 18342
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 355
v=0
o=root 742615961 742615962 IN IP4 10.11.2.50
s=Asterisk PBX 10.7.0
c=IN IP4 10.11.2.50
b=CT:384
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 9078 RTP/AVP 98
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Call-ID: 18342
CSeq: 102 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 478 Unresolvable destination (478/SL)
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Call-ID: 18342
CSeq: 102 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 478 "Unresolvable destination (478/SL)" back from
10.11.2.47:5060
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=29997> for address/port to
send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:106@(null) SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
Route: <sip:10.11.2.47;lr=on;ftag=29997>
Max-Forwards: 70
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 18342
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 478 Unresolvable destination (478/TM)
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Call-ID: 18342
CSeq: 102 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Audio is at 15732
Video is at 10.11.2.47:16908
Adding codec 100003 (ulaw) to SDP
Adding video codec 200003 (h263p) to SDP
Adding video codec 200004 (h264) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
INVITE sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 104 INVITE
User-Agent: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 384
v=0
o=root 476348733 476348735 IN IP4 10.11.2.47
s=Asterisk PBX 10.7.0
c=IN IP4 10.11.2.47
b=CT:384
t=0 0
m=audio 15732 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 16908 RTP/AVP 98 99
a=rtpmap:98 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv
---
Scheduling destruction of SIP dialog
'08e663336578468a35cd30953f5ae115@10.11.2.47' in 32000 ms (Method: INVITE)
== Spawn extension (from-sip, 107, 1) exited non-zero on 'SIP/106-0000001d'
--- (8 headers 0 lines) ---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=29997> for address/port to
send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:106@(null) SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4e44a5a4
Route: <sip:10.11.2.47;lr=on;ftag=29997>
Max-Forwards: 70
From: "107" <sip:107@10.11.2.47>;tag=as6e6ec5ef
To: <sip:106@10.11.2.47>;tag=29997
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 18342
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
Really destroying SIP dialog '18342' Method: ACK
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 104 INVITE
Server: kamailio (3.3.1 (i386/linux))
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK6f188933
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 104 INVITE
Contact: <sip:107@10.11.2.50>
Content-Type: application/sdp
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 342
v=0
o=107 3544 3544 IN IP4 10.11.2.50
s=Talk
c=IN IP4 10.11.2.50
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 98 99
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014
<------------->
--- (10 headers 15 lines) ---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Transmitting (no NAT) to 10.11.2.47:5060:
ACK sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4ac1db0a
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Contact: <sip:107@10.11.2.47:5080>
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 104 ACK
User-Agent: Asterisk PBX 10.7.0
Content-Length: 0
---
set_destination: Parsing <sip:10.11.2.47;lr=on;ftag=as149287b0> for
address/port to send to
set_destination: set destination to 10.11.2.47:5060
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
BYE sip:107@10.11.2.50 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK37e4c951
Route: <sip:10.11.2.47;lr=on;ftag=as149287b0>
Max-Forwards: 70
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 105 BYE
User-Agent: Asterisk PBX 10.7.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
Scheduling destruction of SIP dialog
'08e663336578468a35cd30953f5ae115@10.11.2.47' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK37e4c951
From: "106" <sip:107@10.11.2.47>;tag=as149287b0
To: <sip:107@10.11.2.47:5060>;tag=1777248976
Call-ID: 08e663336578468a35cd30953f5ae115@10.11.2.47
CSeq: 105 BYE
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '08e663336578468a35cd30953f5ae115@10.11.2.47'
Method: INVITE
Really destroying SIP dialog '2b27053f3491a2371bef4b0c12ff7754@10.11.2.47'
Method: NOTIFY
Really destroying SIP dialog '5eefa7c2-4330@127.0.0.1' Method: REGISTER
<--- SIP read from UDP:10.11.2.47:5060 --->
REGISTER sip:10.11.2.47:5080 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKbf09.b6677ad3.0
To: sip:106@10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-8e78
CSeq: 10 REGISTER
Call-ID: 5eefa7c2-4331@127.0.0.1
Content-Length: 0
User-Agent: kamailio (3.3.1 (i386/linux))
Contact: <sip:106@10.11.2.47:5060>
Expires: 60
<------------->
--- (10 headers 0 lines) ---
Sending to 10.11.2.47:5060 (NAT)
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bKbf09.b6677ad3.0;received=10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-8e78
To: sip:106@10.11.2.47;tag=as7930b677
Call-ID: 5eefa7c2-4331@127.0.0.1
CSeq: 10 REGISTER
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:106@10.11.2.47:5060>;expires=60
Date: Wed, 08 May 2013 09:10:17 GMT
Content-Length: 0
<------------>
[May 8 17:10:17] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:10:17] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:10:17] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:10:22] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:10:22] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:10:22] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:10:22] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:10:22] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:10:27] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:10:27] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:10:27] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute
error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/106/INBOX']
Scheduling destruction of SIP dialog
'1e79945f12aff0b43b9a2ed03c072152@10.11.2.47' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
NOTIFY sip:106@10.11.2.47:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4c0421f3
Max-Forwards: 70
From: "asterisk" <sip:106@10.11.2.47>;tag=as1b9afa13
To: <sip:106@10.11.2.47:5060>
Contact: <sip:106@10.11.2.47:5080>
Call-ID: 1e79945f12aff0b43b9a2ed03c072152@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.7.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.11.2.47
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '5eefa7c2-4331@127.0.0.1' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK4c0421f3
From: "asterisk" <sip:106@10.11.2.47>;tag=as1b9afa13
To: <sip:106@10.11.2.47:5060>;tag=12759
Call-ID: 1e79945f12aff0b43b9a2ed03c072152@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1e79945f12aff0b43b9a2ed03c072152@10.11.2.47'
Method: NOTIFY
Really destroying SIP dialog '5eefa7c2-4331@127.0.0.1' Method: REGISTER
<--- SIP read from UDP:10.11.2.47:5060 --->
REGISTER sip:10.11.2.47:5080 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK6f9.3171097.0
To: sip:106@10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a6d6
CSeq: 10 REGISTER
Call-ID: 5eefa7c0-4332@127.0.0.1
Content-Length: 0
User-Agent: kamailio (3.3.1 (i386/linux))
Contact: <sip:106@10.11.2.47:5060>
Expires: 60
<------------->
--- (10 headers 0 lines) ---
Sending to 10.11.2.47:5060 (NAT)
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK6f9.3171097.0;received=10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a6d6
To: sip:106@10.11.2.47;tag=as7b360b52
Call-ID: 5eefa7c0-4332@127.0.0.1
CSeq: 10 REGISTER
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:106@10.11.2.47:5060>;expires=60
Date: Wed, 08 May 2013 09:11:12 GMT
Content-Length: 0
<------------>
[May 8 17:11:12] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:11:12] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:11:12] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:11:17] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:11:17] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:11:17] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:11:17] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:11:17] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:11:22] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:11:22] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:11:22] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute
error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/106/INBOX']
Scheduling destruction of SIP dialog
'2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
NOTIFY sip:106@10.11.2.47:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK3f9b4370
Max-Forwards: 70
From: "asterisk" <sip:106@10.11.2.47>;tag=as776cae27
To: <sip:106@10.11.2.47:5060>
Contact: <sip:106@10.11.2.47:5080>
Call-ID: 2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.7.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.11.2.47
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '5eefa7c0-4332@127.0.0.1' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK3f9b4370
From: "asterisk" <sip:106@10.11.2.47>;tag=as776cae27
To: <sip:106@10.11.2.47:5060>;tag=19478
Call-ID: 2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '2d5c4bc9265ecbe21050c3007ba34d4c@10.11.2.47'
Method: NOTIFY
Really destroying SIP dialog '5eefa7c0-4332@127.0.0.1' Method: REGISTER
<--- SIP read from UDP:10.11.2.47:5060 --->
REGISTER sip:10.11.2.47:5080 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7f9.1dc36914.0
To: sip:106@10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a73e
CSeq: 10 REGISTER
Call-ID: 5eefa7c1-4332@127.0.0.1
Content-Length: 0
User-Agent: kamailio (3.3.1 (i386/linux))
Contact: <sip:106@10.11.2.47:5060>
Expires: 60
<------------->
--- (10 headers 0 lines) ---
Sending to 10.11.2.47:5060 (NAT)
<--- Transmitting (no NAT) to 10.11.2.47:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.2.47;branch=z9hG4bK7f9.1dc36914.0;received=10.11.2.47
From: sip:106@10.11.2.47;tag=533cb9e91f4b999cf76861cbb9ed54ed-a73e
To: sip:106@10.11.2.47;tag=as78794adf
Call-ID: 5eefa7c1-4332@127.0.0.1
CSeq: 10 REGISTER
Server: Asterisk PBX 10.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:106@10.11.2.47:5060>;expires=60
Date: Wed, 08 May 2013 09:12:07 GMT
Content-Length: 0
<------------>
[May 8 17:12:07] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:12:07] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:12:07] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:12:12] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:12:12] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:12:12] WARNING[4068]: res_odbc.c:646 ast_odbc_prepare_and_execute:
SQL Execute returned an error -1: 42S02: [MySQL][ODBC 5.1
Driver][mysqld-5.5.29-0ubuntu0.12.04.2]Table 'asterisk.voicemessages' doesn't
exist (100)
[May 8 17:12:12] WARNING[4068]: res_odbc.c:658 ast_odbc_prepare_and_execute:
SQL Execute error -1! Verifying connection to asterisk [asterisk]...
[May 8 17:12:12] WARNING[4068]: res_odbc.c:762 ast_odbc_sanity_check:
Connection is down attempting to reconnect...
[May 8 17:12:17] NOTICE[4068]: res_odbc.c:1531 odbc_obj_connect: Connecting
asterisk
[May 8 17:12:17] NOTICE[4068]: res_odbc.c:1563 odbc_obj_connect: res_odbc:
Connected to asterisk [asterisk]
[May 8 17:12:17] WARNING[4068]: app_voicemail.c:5142 inboxcount2: SQL Execute
error!
[SELECT COUNT(*) FROM voicemessages WHERE dir =
'/var/spool/asterisk/voicemail/default/106/INBOX']
Scheduling destruction of SIP dialog
'5aa0a13752a06608634fd1202102221a@10.11.2.47' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 10.11.2.47:5060:
NOTIFY sip:106@10.11.2.47:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK54115b1d
Max-Forwards: 70
From: "asterisk" <sip:106@10.11.2.47>;tag=as58704a46
To: <sip:106@10.11.2.47:5060>
Contact: <sip:106@10.11.2.47:5080>
Call-ID: 5aa0a13752a06608634fd1202102221a@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 10.7.0
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 90
Messages-Waiting: no
Message-Account: sip:asterisk@10.11.2.47
Voice-Message: 0/0 (0/0)
---
Scheduling destruction of SIP dialog '5eefa7c1-4332@127.0.0.1' in 32000 ms
(Method: REGISTER)
<--- SIP read from UDP:10.11.2.47:5060 --->
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.11.2.47:5080;branch=z9hG4bK54115b1d
From: "asterisk" <sip:106@10.11.2.47>;tag=as58704a46
To: <sip:106@10.11.2.47:5060>;tag=12490
Call-ID: 5aa0a13752a06608634fd1202102221a@10.11.2.47
CSeq: 102 NOTIFY
User-Agent: Linphone/3.5.2 (eXosip2/3.6.0)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
WH-PC*CLI>
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