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Created July 1, 2012 00:34
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pjsua transcript
Raw
gistfile1.txt
ubuntu@ip-10-166-205-27:~$ pjsua --null-audio --rec-file=ex1.wav --auto-rec
00:33:08.624 os_core_unix.c pjlib 1.12.0 for POSIX initialized
00:33:08.626 sip_endpoint.c Creating endpoint instance...
00:33:08.626 pjlib select() I/O Queue created (0x1ae3cc0)
00:33:08.626 sip_endpoint.c Module "mod-msg-print" registered
00:33:08.626 sip_transport. Transport manager created.
00:33:08.626 sip_endpoint.c Module "mod-pjsua-log" registered
00:33:08.626 sip_endpoint.c Module "mod-tsx-layer" registered
00:33:08.626 sip_endpoint.c Module "mod-stateful-util" registered
00:33:08.626 sip_endpoint.c Module "mod-ua" registered
00:33:08.626 sip_endpoint.c Module "mod-100rel" registered
00:33:08.626 sip_endpoint.c Module "mod-pjsua" registered
00:33:08.626 sip_endpoint.c Module "mod-invite" registered
00:33:08.629 pa_dev.c PortAudio sound library initialized, status=0
00:33:08.629 pa_dev.c PortAudio host api count=2
00:33:08.629 pa_dev.c Sound device count=0
00:33:08.629 pjlib select() I/O Queue created (0x1b0a248)
00:33:08.730 sip_endpoint.c Module "mod-evsub" registered
00:33:08.730 sip_endpoint.c Module "mod-presence" registered
00:33:08.730 sip_endpoint.c Module "mod-mwi" registered
00:33:08.730 sip_endpoint.c Module "mod-refer" registered
00:33:08.730 sip_endpoint.c Module "mod-pjsua-pres" registered
00:33:08.730 sip_endpoint.c Module "mod-pjsua-im" registered
00:33:08.730 sip_endpoint.c Module "mod-pjsua-options" registered
00:33:08.730 pjsua_core.c 1 SIP worker threads created
00:33:08.730 pjsua_core.c pjsua version 1.12.0 for Linux-2.6.38.13/x86_64/glibc-2.13 initialized
00:33:08.730 sip_endpoint.c Module "mod-default-handler" registered
00:33:08.730 wav_writer.c File writer 'ex1.wav' created: samp.rate=16000, bufsize=4KB
00:33:08.730 pjsua_core.c SIP UDP socket reachable at 10.166.205.27:5060
00:33:08.730 udp0x1b21b00 SIP UDP transport started, published address is 10.166.205.27:5060
00:33:08.730 pjsua_acc.c Account <sip:10.166.205.27:5060> added with id 0
00:33:08.731 tcplis:5060 SIP TCP listener ready for incoming connections at 10.166.205.27:5060
00:33:08.731 pjsua_acc.c Account <sip:10.166.205.27:5060;transport=TCP> added with id 1
00:33:08.731 pjsua_media.c RTP socket reachable at 10.166.205.27:4000
00:33:08.731 pjsua_media.c RTCP socket reachable at 10.166.205.27:4001
00:33:08.731 pjsua_media.c RTP socket reachable at 10.166.205.27:4002
00:33:08.731 pjsua_media.c RTCP socket reachable at 10.166.205.27:4003
00:33:08.732 pjsua_media.c RTP socket reachable at 10.166.205.27:4004
00:33:08.732 pjsua_media.c RTCP socket reachable at 10.166.205.27:4005
00:33:08.732 pjsua_media.c RTP socket reachable at 10.166.205.27:4006
00:33:08.732 pjsua_media.c RTCP socket reachable at 10.166.205.27:4007
00:33:08.732 pjsua_media.c Opening null sound device..
00:33:08.732 sip_endpoint.c Module "mod-unsolicited-mwi" registered
>>>>
Account list:
[ 0] <sip:10.166.205.27:5060>: does not register
Online status: Online
*[ 1] <sip:10.166.205.27:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 0 active call
>>> m
(You currently have 0 calls)
Buddy list:
-none-
Choices:
0 For current dialog.
-1 All 0 buddies in buddy list
[1 - 0] Select from buddy list
URL An URL
<Enter> Empty input (or 'q') to cancel
Make call: sip:9996194333@sip.tropo.com
00:33:23.348 pjsua_call.c Making call with acc #1 to sip:9996194333@sip.tropo.com
00:33:23.349 pjsua_media.c Media index 0 selected for call 0
00:33:23.490 pjsua_core.c TX 1126 bytes Request msg INVITE/cseq=29817 (tdta0x1b370a0) to UDP 199.230.57.186:5060:
INVITE sip:9996194333@sip.tropo.com SIP/2.0
Via: SIP/2.0/UDP 10.166.205.27:5060;rport;branch=z9hG4bKPjgWlhc0eZZHyVLkoZApw4cdXUkaOaPhJQ
Max-Forwards: 70
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: sip:9996194333@sip.tropo.com
Contact: <sip:10.166.205.27:5060;ob>
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29817 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v1.12.0 Linux-2.6.38.13/x86_64/glibc-2.13
Content-Type: application/sdp
Content-Length: 480
v=0
o=- 3550091603 3550091603 IN IP4 10.166.205.27
s=pjmedia
c=IN IP4 10.166.205.27
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 105 96
a=rtcp:4001 IN IP4 10.166.205.27
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:105 AMR/8000
a=sendrecv
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
--end msg--
00:33:23.490 pjsua_app.c Call 0 state changed to CALLING
>>> 00:33:23.570 pjsua_core.c RX 372 bytes Response msg 100/INVITE/cseq=29817 (rdata0x1b22f58) from UDP 199.230.57.186:5060:
SIP/2.0 100 Trying
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: <sip:9996194333@sip.tropo.com>
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29817 INVITE
Via: SIP/2.0/UDP 10.166.205.27:5060;received=184.169.241.26;rport=5060;branch=z9hG4bKPjgWlhc0eZZHyVLkoZApw4cdXUkaOaPhJQ
Contact: <sip:9996194333@199.230.57.186:5060>
Content-Length: 0
--end msg--
00:33:23.610 pjsua_core.c RX 425 bytes Response msg 180/INVITE/cseq=29817 (rdata0x1b22f58) from UDP 199.230.57.186:5060:
SIP/2.0 180 Ringing
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: <sip:9996194333@sip.tropo.com>;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29817 INVITE
Via: SIP/2.0/UDP 10.166.205.27:5060;received=184.169.241.26;rport=5060;branch=z9hG4bKPjgWlhc0eZZHyVLkoZApw4cdXUkaOaPhJQ
Contact: <sip:9996194333@199.230.57.186:5060>
Content-Length: 0
--end msg--
00:33:23.610 conference.c Port 2 (ringback) transmitting to port 0 (Master/sound)
00:33:23.610 pjsua_app.c Call 0 state changed to EARLY (180 Ringing)
00:33:24.658 pjsua_core.c RX 732 bytes Response msg 200/INVITE/cseq=29817 (rdata0x1b22f58) from UDP 199.230.57.186:5060:
SIP/2.0 200 OK
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: <sip:9996194333@sip.tropo.com>;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29817 INVITE
Via: SIP/2.0/UDP 10.166.205.27:5060;received=184.169.241.26;rport=5060;branch=z9hG4bKPjgWlhc0eZZHyVLkoZApw4cdXUkaOaPhJQ
Contact: <sip:9996194333@199.230.57.186:5060>
x-accountid: 2
x-appid: 24601
x-sid: e2d46928423b6813670938814526c25e
User-Agent: VCS11.5.55126.0
Content-Type: application/sdp
Content-Length: 177
v=0
o=- 1088 1088 IN IP4 10.6.63.186
s=voxeo
c=IN IP4 10.6.63.186
t=0 0
m=audio 10454 RTP/AVP 98 96
a=rtpmap:98 SPEEX/16000
a=rtpmap:96 telephone-event/8000
a=ptime:20
--end msg--
00:33:24.658 pjsua_app.c Call 0 state changed to CONNECTING
00:33:24.658 strm0x1b3ddf8 VAD temporarily disabled
00:33:24.658 strm0x1b3ddf8 Encoder stream started
00:33:24.658 strm0x1b3ddf8 Decoder stream started
00:33:24.659 pjsua_media.c Media updates, stream #0: speex (sendrecv)
00:33:24.659 conference.c Port 2 (ringback) stop transmitting to port 0 (Master/sound)
00:33:24.659 conference.c Port 4 (sip:9996194333@sip.tropo.com) transmitting to port 1 (ex1.wav)
00:33:24.659 conference.c Port 4 (sip:9996194333@sip.tropo.com) transmitting to port 0 (Master/sound)
00:33:24.659 conference.c Port 0 (Master/sound) transmitting to port 4 (sip:9996194333@sip.tropo.com)
00:33:24.659 conference.c Port 0 (Master/sound) transmitting to port 1 (ex1.wav)
00:33:24.659 pjsua_app.c Media for call 0 is active
00:33:24.659 pjsua_core.c TX 390 bytes Request msg ACK/cseq=29817 (tdta0x1b4f840) to UDP 199.230.57.186:5060:
ACK sip:9996194333@199.230.57.186:5060 SIP/2.0
Via: SIP/2.0/UDP 10.166.205.27:5060;rport;branch=z9hG4bKPjrxBFPIMQP9-KVfy518J1gWyRPmDzDZOX
Max-Forwards: 70
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: sip:9996194333@sip.tropo.com;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29817 ACK
Content-Length: 0
--end msg--
00:33:24.659 pjsua_app.c Call 0 state changed to CONFIRMED
00:33:25.292 strm0x1b3ddf8 VAD re-enabled
>>>>
Account list:
[ 0] <sip:10.166.205.27:5060>: does not register
Online status: Online
*[ 1] <sip:10.166.205.27:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 1 active call
Current call id=0 to sip:9996194333@sip.tropo.com [CONFIRMED]
>>> *
DTMF strings to send (0-9*#A-B) (empty to cancel): 1
00:33:38.858 pjsua_core.c TX 516 bytes Request msg INFO/cseq=29818 (tdta0x1b527a0) to UDP 199.230.57.186:5060:
INFO sip:9996194333@199.230.57.186:5060 SIP/2.0
Via: SIP/2.0/UDP 10.166.205.27:5060;rport;branch=z9hG4bKPjrsI5WMMycgLsBB232Ai-joILrGPBQnWq
Max-Forwards: 70
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: sip:9996194333@sip.tropo.com;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29818 INFO
User-Agent: PJSUA v1.12.0 Linux-2.6.38.13/x86_64/glibc-2.13
Content-Type: application/dtmf-relay
Content-Length: 22
Signal=1
Duration=160
--end msg--
>>> 00:33:38.936 pjsua_core.c RX 491 bytes Response msg 200/INFO/cseq=29818 (rdata0x1b22f58) from UDP 199.230.57.186:5060:
SIP/2.0 200 OK
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: <sip:9996194333@sip.tropo.com>;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29818 INFO
Via: SIP/2.0/UDP 10.166.205.27:5060;received=184.169.241.26;rport=5060;branch=z9hG4bKPjrsI5WMMycgLsBB232Ai-joILrGPBQnWq
Contact: <sip:9996194333@199.230.57.186:5060>
x-accountid: 2
x-appid: 24601
x-sid: e2d46928423b6813670938814526c25e
Content-Length: 0
--end msg--
00:33:38.936 pjsua_app.c Call 0: DTMF sent successfully with INFO
>>>>
Account list:
[ 0] <sip:10.166.205.27:5060>: does not register
Online status: Online
*[ 1] <sip:10.166.205.27:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 1 active call
Current call id=0 to sip:9996194333@sip.tropo.com [CONFIRMED]
>>> h
00:33:47.808 pjsua_core.c TX 451 bytes Request msg BYE/cseq=29819 (tdta0x1b527a0) to UDP 199.230.57.186:5060:
BYE sip:9996194333@199.230.57.186:5060 SIP/2.0
Via: SIP/2.0/UDP 10.166.205.27:5060;rport;branch=z9hG4bKPjaH1FaCNnGlgca9Ok1LcWFb2kMm4VookU
Max-Forwards: 70
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: sip:9996194333@sip.tropo.com;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29819 BYE
User-Agent: PJSUA v1.12.0 Linux-2.6.38.13/x86_64/glibc-2.13
Content-Length: 0
--end msg--
>>> 00:33:47.886 pjsua_core.c RX 443 bytes Response msg 200/BYE/cseq=29819 (rdata0x1b22f58) from UDP 199.230.57.186:5060:
SIP/2.0 200 OK
From: <sip:10.166.205.27>;tag=iPWBRUNql1ruqqCSd.t-pDJDciOAqHKi
To: <sip:9996194333@sip.tropo.com>;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call-ID: iHX4az3FPgkw8jrthryFrszcvcGwdv9W
CSeq: 29819 BYE
Via: SIP/2.0/UDP 10.166.205.27:5060;received=184.169.241.26;rport=5060;branch=z9hG4bKPjaH1FaCNnGlgca9Ok1LcWFb2kMm4VookU
x-accountid: 2
x-appid: 24601
x-sid: e2d46928423b6813670938814526c25e
Content-Length: 0
--end msg--
00:33:47.886 pjsua_app.c Call 0 is DISCONNECTED [reason=200 (Normal call clearing)]
00:33:47.886 pjsua_app.c
[DISCONNCTD] To: sip:9996194333@sip.tropo.com;tag=2aaac1022ff0-0-13c4-6009-1ca109-7efde7e3-1ca109
Call time: 00h:00m:23s, 1st res in 262 ms, conn in 1311ms
SRTP status: Not active Crypto-suite: (null)
#0 speex @16KHz, sendrecv, peer=-
RX pt=98, stat last update: 00h:00m:00.414s ago
total 1pkt 0B (40B +IP hdr) @avg=0bps/13bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
TX pt=98, ptime=20ms, stat last update: never
total 85pkt 2.7KB (6.1KB +IP hdr) @avg 933bps/2.1Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
00:33:47.886 pjsua_media.c Media session for call 0 is destroyed
>>>>
Account list:
[ 0] <sip:10.166.205.27:5060>: does not register
Online status: Online
*[ 1] <sip:10.166.205.27:5060;transport=TCP>: does not register
Online status: Online
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: | Account: |
| | | |
| m Make new call | +b Add new buddy .| +a Add new accnt |
| M Make multiple calls | -b Delete buddy | -a Delete accnt. |
| a Answer call | i Send IM | !a Modify accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr (Re-)register |
| H Hold call | u Unsubscribe presence | ru Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next ac.|
| U send UPDATE | T Set online status | < Cycle prev ac.|
| ],[ Select next/prev call +--------------------------+-------------------+
| x Xfer call | Media Commands: | Status & Config: |
| X Xfer with Replaces | | |
| # Send RFC 2833 DTMF | cl List ports | d Dump status |
| * Send DTMF with INFO | cc Connect port | dd Dump detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump config |
| | V Adjust audio Volume | f Save config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type |
+=============================================================================+
You have 0 active call
>>> q
00:33:50.518 pjsua_core.c Shutting down, flags=0...
00:33:50.528 pjsua_pres.c Shutting down presence..
00:33:50.528 pjsua_media.c Shutting down media..
00:33:50.528 pjsua_media.c Closing null sound device..
00:33:50.852 pa_dev.c PortAudio sound library shutting down..
00:33:51.859 pjsua_core.c Destroying...
00:33:51.859 sip_transactio Stopping transaction layer module
00:33:51.859 sip_endpoint.c Module "mod-default-handler" unregistered
00:33:51.859 sip_endpoint.c Module "mod-unsolicited-mwi" unregistered
00:33:51.859 sip_endpoint.c Module "mod-pjsua-options" unregistered
00:33:51.859 sip_endpoint.c Module "mod-pjsua-im" unregistered
00:33:51.859 sip_endpoint.c Module "mod-pjsua-pres" unregistered
00:33:51.859 sip_endpoint.c Module "mod-pjsua" unregistered
00:33:51.859 sip_endpoint.c Module "mod-stateful-util" unregistered
00:33:51.859 sip_endpoint.c Module "mod-refer" unregistered
00:33:51.859 sip_endpoint.c Module "mod-mwi" unregistered
00:33:51.859 sip_endpoint.c Module "mod-presence" unregistered
00:33:51.859 sip_endpoint.c Module "mod-evsub" unregistered
00:33:51.859 sip_endpoint.c Module "mod-invite" unregistered
00:33:51.859 sip_endpoint.c Module "mod-100rel" unregistered
00:33:51.859 sip_endpoint.c Module "mod-ua" unregistered
00:33:51.859 sip_transactio Transaction layer module destroyed
00:33:51.859 sip_endpoint.c Module "mod-tsx-layer" unregistered
00:33:51.859 sip_endpoint.c Module "mod-msg-print" unregistered
00:33:51.859 sip_endpoint.c Module "mod-pjsua-log" unregistered
00:33:51.860 tcplis:5060 SIP TCP listener destroyed
00:33:51.861 sip_endpoint.c Endpoint 0x1ad8f28 destroyed
00:33:51.861 ex1.wav Pool is not released by application, releasing now
00:33:51.861 pjsua_core.c PJSUA destroyed...
ubuntu@ip-10-166-205-27:~$
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