it WORKS! now.
Remarks: By default it does not work when you build from source 2.1 +
temporary our sip-tools also have same issue if you install it from
package.
Solution: To workout/fix it follow this steps:
Step 1: install FreeSwitch (sip server for test simulation)
$ sudo yum install git autoconf automake libtool ncurses-devel libjpeg-devel
$ sudo yum install expat-devel openssl-devel libtiff-devel libX11-devel
unixODBC-devel libssl-devel python-devel \
zlib-devel libzrtpcpp-devel alsa-lib-devel libogg-devel
libvorbis-devel perl-libs gdbm-devel \
libdb-devel uuid-devel @development-tools
$ git clone git://git.freeswitch.org/freeswitch.git
$ ./bootstrap.sh
$ ./configure
$ make
$ make install
$ make all install cd-sounds-install cd-moh-install
$ /usr/local/freeswitch/bin/freeswitch
A) use Bria or webRTC sip phone to anything that already works as sip
client to verify/test if the username 1000 and password 1234 is working
B) make sure with other sip tools your sip server is registering and you
can hear audio IVR
Step 2: install Pjsip 2.1 (action)
$ cd /var/tmp
$ wget http://www.pjsip.org/release/2.1/pjproject-2.1.tar.bz2
$ tar xvfj pjproject-2.1.tar.bz2
$ cd pjproject-2.1
$ ./configure
$ make dep && make && make install
A) This wont give you audio playback and audio capture so do this fix
Step 3: make those changes as below shown.
to fix audio problem to ALSA instead of PortAudio driver (which was the
problem)
$ cat /var/tmp/pjproject-2.1.0/pjlib/include/pj/config_site.h
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 1
//this following line is for Windows Mobile, i do not need it because i am
in 32/64-bit desktop
//#include <pj/config_site_sample.h>
$ cat /var/tmp/pjproject-2.1.0/pjmedia/build/os-linux.mak
# Linux
# Define the desired sound device backend
# Valid values are:
# - pa_unix: PortAudio on Unix (OSS or ALSA)
# - pa_darwinos: PortAudio on MacOSX (CoreAudio)
# - pa_old_darwinos: PortAudio on MacOSX (old CoreAudio, for OSX 10.2)
# - pa_win32: PortAudio on Win32 (WMME)
# - ds: Win32 DirectSound (dsound.c)
# - null: Null sound device (nullsound.c)
AC_PJMEDIA_SND=alsa
# leave all the rest as it is, only change that AC_PJMEDIA_SND from pa_unix
to alsa
Step 4:
$ cd /var/tmp/pjproject-2.1.0
$ ./configure && make dep && make && make install
Step 5: eat the cake
$ /var/tmp/pjproject-2.1.0/pjsip-apps/bin/pjsua-x86_64-unknown-linux-gnu
follow this to place a call and have fun: http://paste.ubuntu.com/6518780/
10:25:52.339 os_core_unix.c !pjlib 2.1 for POSIX initialized
10:25:52.339 sip_endpoint.c .Creating endpoint instance...
10:25:52.339 pjlib .select() I/O Queue created (0x95b8a0)
10:25:52.339 sip_endpoint.c .Module "mod-msg-print" registered
10:25:52.339 sip_transport. .Transport manager created.
10:25:52.339 pjsua_core.c .PJSUA state changed: NULL --> CREATED
10:25:52.340 sip_endpoint.c .Module "mod-pjsua-log" registered
10:25:52.340 sip_endpoint.c .Module "mod-tsx-layer" registered
10:25:52.340 sip_endpoint.c .Module "mod-stateful-util" registered
10:25:52.340 sip_endpoint.c .Module "mod-ua" registered
10:25:52.340 sip_endpoint.c .Module "mod-100rel" registered
10:25:52.340 sip_endpoint.c .Module "mod-pjsua" registered
10:25:52.340 sip_endpoint.c .Module "mod-invite" registered
10:25:52.382 alsa_dev.c ..ALSA driver found 16 devices
10:25:52.383 alsa_dev.c ..ALSA initialized
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
bt_audio_service_open: connect() failed: Connection refused (111)
10:25:52.446 pa_dev.c ..PortAudio sound library initialized, status=0
10:25:52.446 pa_dev.c ..PortAudio host api count=2
10:25:52.446 pa_dev.c ..Sound device count=19
10:25:52.446 pjlib ..select() I/O Queue created (0x9d2c68)
10:25:52.454 sip_endpoint.c .Module "mod-evsub" registered
10:25:52.454 sip_endpoint.c .Module "mod-presence" registered
10:25:52.454 sip_endpoint.c .Module "mod-mwi" registered
10:25:52.454 sip_endpoint.c .Module "mod-refer" registered
10:25:52.454 sip_endpoint.c .Module "mod-pjsua-pres" registered
10:25:52.454 sip_endpoint.c .Module "mod-pjsua-im" registered
10:25:52.454 sip_endpoint.c .Module "mod-pjsua-options" registered
10:25:52.455 pjsua_core.c .1 SIP worker threads created
10:25:52.455 pjsua_core.c .pjsua version 2.1 for
Linux-3.11.0.12/x86_64/glibc-2.17 initialized
10:25:52.455 pjsua_core.c .PJSUA state changed: CREATED --> INIT
10:25:52.455 sip_endpoint.c Module "mod-default-handler" registered
10:25:52.455 pjsua_core.c SIP UDP socket reachable at 192.168.1.19:5060
10:25:52.455 udp0x9b0450 SIP UDP transport started, published address
is 192.168.1.19:5060
10:25:52.455 pjsua_acc.c Adding account: id=<sip:192.168.1.19:5060>
10:25:52.455 pjsua_acc.c .Account <sip:192.168.1.19:5060> added with id
0
10:25:52.455 pjsua_acc.c Acc 0: setting online status to 1..
10:25:52.455 tcplis:5060 SIP TCP listener ready for incoming
connections at 192.168.1.19:5060
10:25:52.455 pjsua_acc.c Adding account: id=<sip:192.168.1.19:5060
;transport=TCP>
10:25:52.455 pjsua_acc.c .Account <sip:192.168.1.19:5060;transport=TCP>
added with id 1
10:25:52.455 pjsua_acc.c Acc 1: setting online status to 1..
10:25:52.455 pjsua_core.c PJSUA state changed: INIT --> STARTING
10:25:52.455 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
10:25:52.455 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
>>>>
Account list:
[ 0] <sip:192.168.1.19:5060>: does not register
Online status: Online
*[ 1] <sip:192.168.1.19:5060;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 |
|
+-----------------------------------------------------------------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT type
|
+=============================================================================+
You have 0 active call
>>> +a
Your SIP URL: (empty to cancel): sip:1shamun at 192.168.1.12
URL of the registrar: (empty to cancel): sip:192.168.1.12
Auth Realm: (empty to cancel): *
Auth Username: (empty to cancel): 1shamun
Auth Password: (empty to cancel): admin2013
10:26:34.804 pjsua_acc.c Adding account: id=sip:1shamun at 192.168.1.12
10:26:34.805 pjsua_acc.c .Account sip:1shamun at 192.168.1.12 added with
id 2
10:26:34.805 pjsua_acc.c .Acc 2: setting registration..
10:26:34.805 pjsua_core.c ...TX 549 bytes Request msg
REGISTER/cseq=61881 (tdta0x9b62e0) to UDP 192.168.1.12:5060:
REGISTER sip:192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjlkenSNEG8TnMI6f8HphjRH-POIzPaEOu
Max-Forwards: 70
From: <sip:1shamun@192.168.1.12>;tag=YvjITguv-c05VT86yXNb6UQVk-XlieAH
To: <sip:1shamun at 192.168.1.12>
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61881 REGISTER
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
10:26:34.805 pjsua_acc.c ..Acc 2: Registration sent
>>> 10:26:34.807 pjsua_core.c .RX 680 bytes Response msg
401/REGISTER/cseq=61881 (rdata0x9b1ab8) from UDP 192.168.1.12:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjlkenSNEG8TnMI6f8HphjRH-POIzPaEOu
From: <sip:1shamun@192.168.1.12>;tag=YvjITguv-c05VT86yXNb6UQVk-XlieAH
To: <sip:1shamun at 192.168.1.12>;tag=8U0B3jFUcQjZS
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61881 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="192.168.1.12",
nonce="fb290164-5cd9-11e3-a4b8-3586b66a1730", algorithm=MD5, qop="auth"
Content-Length: 0
--end msg--
10:26:34.807 pjsua_core.c ....TX 809 bytes Request msg
REGISTER/cseq=61882 (tdta0x9b62e0) to UDP 192.168.1.12:5060:
REGISTER sip:192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjcPq6l5sXk0r3XxOv08Irwq.aFJ51YZW7
Max-Forwards: 70
From: <sip:1shamun@192.168.1.12>;tag=YvjITguv-c05VT86yXNb6UQVk-XlieAH
To: <sip:1shamun at 192.168.1.12>
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61882 REGISTER
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Authorization: Digest username="1shamun", realm="192.168.1.12",
nonce="fb290164-5cd9-11e3-a4b8-3586b66a1730", uri="sip:192.168.1.12",
response="e11985e6271b3b88e3c94dfd11672df6", algorithm=MD5,
cnonce="FjVwPDrg7SatsNbTVFHg6ttEtXeUhzY1", qop=auth, nc=00000001
Content-Length: 0
--end msg--
10:26:34.809 pjsua_core.c .RX 644 bytes Response msg
200/REGISTER/cseq=61882 (rdata0x7fde0c002998) from UDP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjcPq6l5sXk0r3XxOv08Irwq.aFJ51YZW7
From: <sip:1shamun@192.168.1.12>;tag=YvjITguv-c05VT86yXNb6UQVk-XlieAH
To: <sip:1shamun at 192.168.1.12>;tag=94S44D0y9Z8HN
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61882 REGISTER
Contact: <sip:1shamun at 192.168.1.19:5060;ob>;expires=300
Date: Wed, 04 Dec 2013 11:48:24 GMT
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0
--end msg--
10:26:34.809 pjsua_acc.c ....SIP outbound status for acc 2 is not active
10:26:34.809 pjsua_acc.c ....sip:1shamun at 192.168.1.12: registration
success, status=200 (OK), will re-register in 300 seconds
10:26:34.809 pjsua_acc.c ....Keep-alive timer started for acc 2,
destination:192.168.1.12:5060, interval:15s
10:26:34.878 pjsua_core.c .RX 945 bytes Request msg NOTIFY/cseq=52742788
(rdata0x7fde0c002998) from UDP 192.168.1.12:5060:
NOTIFY sip:1shamun at 192.168.1.19:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12;rport;branch=z9hG4bKSNaNHc9Z36eND
Max-Forwards: 70
From: <sip:1shamun@192.168.1.12>;tag=aeKX68g268y4g
To: <sip:1shamun at 192.168.1.12>
Call-ID: d2923b4a-d77c-1231-f2ab-782bcb36fe3d
CSeq: 52742788 NOTIFY
Contact: <sip:mod_sofia at 192.168.1.12:5060>
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Event: message-summary
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Subscription-State: terminated;reason=noresource
Content-Type: application/simple-message-summary
Content-Length: 67
Messages-Waiting: no
Message-Account: sip:1shamun at 192.168.1.12
--end msg--
10:26:34.878 pjsua_pres.c .Got unsolicited NOTIFY from 192.168.1.12:5060
..
10:26:34.878 pjsua_core.c ...TX 309 bytes Response msg
200/NOTIFY/cseq=52742788 (tdta0x7fde0c0048c0) to UDP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.12;rport=5060;received=192.168.1.12;branch=z9hG4bKSNaNHc9Z36eND
Call-ID: d2923b4a-d77c-1231-f2ab-782bcb36fe3d
From: <sip:1shamun@192.168.1.12>;tag=aeKX68g268y4g
To: <sip:1shamun at 192.168.1.12>;tag=z9hG4bKSNaNHc9Z36eND
CSeq: 52742788 NOTIFY
Content-Length: 0
--end msg--
10:26:34.878 pjsua_app.c ..Received MWI for acc 2:
10:26:34.878 pjsua_app.c .. Content-Type:
application/simple-message-summary
10:26:34.878 pjsua_app.c .. Body:
Messages-Waiting: no
Message-Account: sip:1shamun at 192.168.1.12
>>>>
Account list:
[ 0] <sip:192.168.1.19:5060>: does not register
Online status: Online
[ 1] <sip:192.168.1.19:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:1shamun at 192.168.1.12: 200/OK (expires=292)
Online status: Offline
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 |
|
+-----------------------------------------------------------------------------+
| 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:9198 at 192.168.1.12
10:26:50.421 pjsua_call.c !Making call with acc #2 to
sip:9198 at 192.168.1.12
10:26:50.421 pjsua_aud.c .Set sound device: capture=-1, playback=-2
10:26:50.421 pjsua_app.c ..Turning sound device ON
10:26:50.421 pjsua_aud.c ..Opening sound device PCM at 16000/1/20ms
10:26:50.445 ec0x9e4110 ...AEC created, clock_rate=16000, channel=1,
samples per frame=320, tail length=200 ms, latency=100 ms
10:26:50.445 pjsua_media.c .Call 0: initializing media..
10:26:50.445 pjsua_media.c ..RTP socket reachable at 192.168.1.19:4000
10:26:50.445 pjsua_media.c ..RTCP socket reachable at 192.168.1.19:4001
10:26:50.445 pjsua_media.c ..Media index 0 selected for audio call 0
10:26:50.446 pjsua_core.c ....TX 1118 bytes Request msg INVITE/cseq=4733
(tdta0xa196b0) to UDP 192.168.1.12:5060:
INVITE sip:9198 at 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjpZh6Vi7ii8i0ss9iNEicu4g2zGUl5H87
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4733 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Content-Type: application/sdp
Content-Length: 473
v=0
o=- 3595138010 3595138010 IN IP4 192.168.1.19
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.1.19
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.1.19
a=sendrecv
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:96 telephone-event/8000
a=fmtp:96 0-15
--end msg--
10:26:50.446 pjsua_app.c .......Call 0 state changed to CALLING
>>> 10:26:50.447 pjsua_core.c .RX 372 bytes Response msg
100/INVITE/cseq=4733 (rdata0x7fde0c002998) from UDP 192.168.1.12:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjpZh6Vi7ii8i0ss9iNEicu4g2zGUl5H87
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4733 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Content-Length: 0
--end msg--
10:26:50.468 pjsua_core.c .RX 889 bytes Response msg
407/INVITE/cseq=4733 (rdata0x7fde0c002998) from UDP 192.168.1.12:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjpZh6Vi7ii8i0ss9iNEicu4g2zGUl5H87
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>;tag=BQcp83153HNQc
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4733 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Proxy-Authenticate: Digest realm="192.168.1.12",
nonce="047ebee8-5cda-11e3-a4ba-3586b66a1730", algorithm=MD5, qop="auth"
Content-Length: 0
--end msg--
10:26:50.468 pjsua_core.c ..TX 339 bytes Request msg ACK/cseq=4733
(tdta0x7fde0c007820) to UDP 192.168.1.12:5060:
ACK sip:9198 at 192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjpZh6Vi7ii8i0ss9iNEicu4g2zGUl5H87
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12;tag=BQcp83153HNQc
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4733 ACK
Content-Length: 0
--end msg--
10:26:50.468 tcpc0x7fde0c00 .......TCP client transport created
10:26:50.468 tcpc0x7fde0c00 .......TCP transport 192.168.1.19:56773 is
connecting to 192.168.1.12:5060...
10:26:50.468 pjsua_core.c .......TX 1390 bytes Request msg
INVITE/cseq=4734 (tdta0xa196b0) to TCP 192.168.1.12:5060:
INVITE sip:9198 at 192.168.1.12 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.19:56773
;rport;branch=z9hG4bKPjTBJaCQaIxWhH.yGXAXr0TVjyZbN0CWU9
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4734 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Proxy-Authorization: Digest username="1shamun", realm="192.168.1.12",
nonce="047ebee8-5cda-11e3-a4ba-3586b66a1730", uri="sip:9198 at 192.168.1.12",
response="ec96f0715b9a8d27c5d907816568ca0e", algorithm=MD5,
cnonce="d426wCIf5LnU6S1DArLg0KjYRtUdkhoJ", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 473
v=0
o=- 3595138010 3595138010 IN IP4 192.168.1.19
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
c=IN IP4 192.168.1.19
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.1.19
a=sendrecv
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:96 telephone-event/8000
a=fmtp:96 0-15
--end msg--
10:26:50.469 tcpc0x7fde0c00 TCP transport 192.168.1.19:56773 is connected
to 192.168.1.12:5060
10:26:50.469 pjsua_app.c SIP TCP transport is connected to [
192.168.1.12:5060]
10:26:50.469 pjsua_core.c .RX 374 bytes Response msg
100/INVITE/cseq=4734 (rdata0x7fde0c00aa98) from TCP 192.168.1.12:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.19:56773
;rport=56773;branch=z9hG4bKPjTBJaCQaIxWhH.yGXAXr0TVjyZbN0CWU9
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4734 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Content-Length: 0
--end msg--
10:26:50.494 pjsua_core.c .RX 1252 bytes Response msg
200/INVITE/cseq=4734 (rdata0x7fde0c00aa98) from TCP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.19:56773
;rport=56773;branch=z9hG4bKPjTBJaCQaIxWhH.yGXAXr0TVjyZbN0CWU9
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4734 INVITE
Contact: <sip:9198 at 192.168.1.12:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Require: timer
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Session-Expires: 1800;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 243
Remote-Party-ID: "9198" <sip:9198 at 192.168.1.12
>;party=calling;privacy=off;screen=no
v=0
o=FreeSWITCH 1386129693 1386129694 IN IP4 192.168.1.12
s=FreeSWITCH
c=IN IP4 192.168.1.12
t=0 0
m=audio 28026 RTP/AVP 3 96
a=rtpmap:3 GSM/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
a=ptime:20
--end msg--
10:26:50.494 pjsua_app.c .....Call 0 state changed to CONNECTING
10:26:50.494 pjsua_media.c .....Call 0: updating media..
10:26:50.494 pjsua_aud.c ......Audio channel update..
10:26:50.494 strm0x7fde0c00 .......VAD temporarily disabled
10:26:50.494 strm0x7fde0c00 .......Encoder stream started
10:26:50.494 strm0x7fde0c00 .......Decoder stream started
10:26:50.494 pjsua_media.c ......Audio updated, stream #0: GSM (sendrecv)
10:26:50.494 pjsua_app.c .....Call 0 media 0 [type=audio], status is
Active
10:26:50.494 pjsua_aud.c .....Conf connect: 3 --> 0
10:26:50.495 conference.c ......Port 3 (sip:9198 at 192.168.1.12)
transmitting to port 0 (default)
10:26:50.495 pjsua_aud.c .....Conf connect: 0 --> 3
10:26:50.495 conference.c ......Port 0 (default) transmitting to port 3 (
sip:9198 at 192.168.1.12)
10:26:50.495 pjsua_core.c .....TX 358 bytes Request msg ACK/cseq=4734
(tdta0x7fde0c012fa0) to UDP 192.168.1.12:5060:
ACK sip:9198 at 192.168.1.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPj3KZp-pacRr499c-X9FrnRAL5w.TapRe2
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4734 ACK
Content-Length: 0
--end msg--
10:26:50.495 pjsua_app.c .....Call 0 state changed to CONFIRMED
10:26:50.510 Master/sound Underflow, buf_cnt=0, will generate 1 frame
10:26:50.906 Master/sound Underflow, buf_cnt=0, will generate 1 frame
10:26:51.124 strm0x7fde0c00 VAD re-enabled
>>>>
Account list:
[ 0] <sip:192.168.1.19:5060>: does not register
Online status: Online
[ 1] <sip:192.168.1.19:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:1shamun at 192.168.1.12: 200/OK (expires=276)
Online status: Offline
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 |
|
+-----------------------------------------------------------------------------+
| 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:9198 at 192.168.1.12 [CONFIRMED]
>>> q
10:26:55.939 pjsua_core.c !Shutting down, flags=0...
10:26:55.939 pjsua_core.c PJSUA state changed: RUNNING --> CLOSING
10:26:55.944 pjsua_call.c .Hangup all calls..
10:26:55.944 pjsua_call.c ..Call 0 hanging up: code=0..
10:26:55.945 pjsua_core.c ......TX 416 bytes Request msg BYE/cseq=4735
(tdta0x9ae0e0) to UDP 192.168.1.12:5060:
BYE sip:9198 at 192.168.1.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjkH-C6QzNlNQIkEF9DnpoDI7gIly9LIjS
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4735 BYE
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Content-Length: 0
--end msg--
10:26:55.945 pjsua_pres.c .Shutting down presence..
10:26:55.945 pjsua_media.c .Shutting down media..
10:26:55.945 pjsua_app.c ...Turning sound device OFF
10:26:55.945 pjsua_aud.c ...Closing default sound playback device and
default sound capture device
10:26:56.113 pjsua_media.c ..Call 0: deinitializing media..
10:26:56.113 pjsua_media.c ....Media stream call00:0 is destroyed
10:26:56.113 pjsua_media.c ..Call 1: deinitializing media..
10:26:56.113 pjsua_media.c ..Call 2: deinitializing media..
10:26:56.113 pjsua_media.c ..Call 3: deinitializing media..
10:26:56.445 pa_dev.c ..PortAudio sound library shutting down..
10:26:56.445 pjsua_acc.c .Acc 2: setting unregistration..
10:26:56.445 pjsua_core.c ...TX 449 bytes Request msg
REGISTER/cseq=61883 (tdta0x99d690) to UDP 192.168.1.12:5060:
REGISTER sip:192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjOB8d1guTBEcZ9.KkvTrfgrZWta.vWdPZ
Max-Forwards: 70
From: <sip:1shamun@192.168.1.12>;tag=GgYXSp6rmQgQHu4tn-lNqSbd00BD1een
To: <sip:1shamun at 192.168.1.12>
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61883 REGISTER
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Expires: 0
Content-Length: 0
--end msg--
10:26:56.445 pjsua_acc.c ..Acc 2: Unregistration sent
10:26:56.445 pjsua_core.c ..TX 416 bytes Request msg BYE/cseq=4735
(tdta0x9ae0e0) to UDP 192.168.1.12:5060:
BYE sip:9198 at 192.168.1.12:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjkH-C6QzNlNQIkEF9DnpoDI7gIly9LIjS
Max-Forwards: 70
From: sip:1shamun@192.168.1.12;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: sip:9198 at 192.168.1.12;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4735 BYE
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Content-Length: 0
--end msg--
10:26:56.445 pjsua_core.c ..RX 541 bytes Response msg 200/BYE/cseq=4735
(rdata0x7fde0c002998) from UDP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjkH-C6QzNlNQIkEF9DnpoDI7gIly9LIjS
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4735 BYE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0
--end msg--
10:26:56.445 pjsua_app.c ......Call 0 is DISCONNECTED [reason=200
(Normal call clearing)]
10:26:56.445 pjsua_app.c ......
[DISCONNCTD] To: sip:9198 at 192.168.1.12;tag=c05eaZj90tBar
Call time: 00h:00m:05s, 1st res in 49 ms, conn in 50ms
#0 audio deactivated
10:26:56.445 pjsua_media.c ......Call 0: deinitializing media..
10:26:56.445 pjsua_media.c ........Media stream call00:0 is destroyed
10:26:56.446 pjsua_core.c ..RX 541 bytes Response msg 200/BYE/cseq=4735
(rdata0xa1ce58) from UDP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjkH-C6QzNlNQIkEF9DnpoDI7gIly9LIjS
From: <sip:1shamun@192.168.1.12>;tag=O1T.EQ56ts0cmygAHKTzcRyszBmnX-3L
To: <sip:9198 at 192.168.1.12>;tag=c05eaZj90tBar
Call-ID: 4-78f5GPGyoMTG6uzlfELcBF7vYPNu0P
CSeq: 4735 BYE
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0
--end msg--
10:26:56.446 pjsua_core.c ..RX 680 bytes Response msg
401/REGISTER/cseq=61883 (rdata0xa1ce58) from UDP 192.168.1.12:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjOB8d1guTBEcZ9.KkvTrfgrZWta.vWdPZ
From: <sip:1shamun@192.168.1.12>;tag=GgYXSp6rmQgQHu4tn-lNqSbd00BD1een
To: <sip:1shamun at 192.168.1.12>;tag=D9y7Bt3cy31vK
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61883 REGISTER
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
WWW-Authenticate: Digest realm="192.168.1.12",
nonce="080efe06-5cda-11e3-a4c3-3586b66a1730", algorithm=MD5, qop="auth"
Content-Length: 0
--end msg--
10:26:56.446 pjsua_core.c .....TX 709 bytes Request msg
REGISTER/cseq=61884 (tdta0x99d690) to UDP 192.168.1.12:5060:
REGISTER sip:192.168.1.12 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport;branch=z9hG4bKPjDxXDDVUPpMy9VwiWYn.yPM8iW8RImu8A
Max-Forwards: 70
From: <sip:1shamun@192.168.1.12>;tag=GgYXSp6rmQgQHu4tn-lNqSbd00BD1een
To: <sip:1shamun at 192.168.1.12>
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61884 REGISTER
User-Agent: PJSUA v2.1 Linux-3.11.0.12/x86_64/glibc-2.17
Contact: <sip:1shamun at 192.168.1.19:5060;ob>
Expires: 0
Authorization: Digest username="1shamun", realm="192.168.1.12",
nonce="080efe06-5cda-11e3-a4c3-3586b66a1730", uri="sip:192.168.1.12",
response="10f7cf097f2e14670d3aab4b30365f94", algorithm=MD5,
cnonce="FjVwPDrg7SatsNbTVFHg6ttEtXeUhzY1", qop=auth, nc=00000001
Content-Length: 0
--end msg--
10:26:56.448 pjsua_core.c ..RX 587 bytes Response msg
200/REGISTER/cseq=61884 (rdata0xa1ce58) from UDP 192.168.1.12:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.19:5060
;rport=5060;branch=z9hG4bKPjDxXDDVUPpMy9VwiWYn.yPM8iW8RImu8A
From: <sip:1shamun@192.168.1.12>;tag=GgYXSp6rmQgQHu4tn-lNqSbd00BD1een
To: <sip:1shamun at 192.168.1.12>;tag=ejr0DNmgUcrFF
Call-ID: 2E62MoLQ8xXutAI664t1QCZRK3Ruhqe7
CSeq: 61884 REGISTER
Date: Wed, 04 Dec 2013 11:48:45 GMT
User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130823T195449Z~863e6cfa3f
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Content-Length: 0
--end msg--
10:26:56.448 pjsua_acc.c .....sip:1shamun at 192.168.1.12: unregistration
success
10:26:57.456 pjsua_core.c .Destroying...
10:26:57.456 sip_transactio .Stopping transaction layer module
10:26:57.456 sip_transactio .Stopped transaction layer module
10:26:57.456 sip_endpoint.c .Module "mod-default-handler" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-unsolicited-mwi" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-pjsua-options" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-pjsua-im" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-pjsua-pres" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-pjsua" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-stateful-util" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-refer" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-mwi" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-presence" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-evsub" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-invite" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-100rel" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-ua" unregistered
10:26:57.456 sip_transactio .Transaction layer module destroyed
10:26:57.456 sip_endpoint.c .Module "mod-tsx-layer" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-msg-print" unregistered
10:26:57.456 sip_endpoint.c .Module "mod-pjsua-log" unregistered
10:26:57.457 tcpc0x7fde0c00 .TCP transport destroyed normally
10:26:57.457 tcplis:5060 .SIP TCP listener destroyed
10:26:57.457 sip_endpoint.c .Endpoint 0x950b08 destroyed
10:26:57.457 pjsua_core.c .PJSUA state changed: CLOSING --> NULL
10:26:57.457 pjsua_core.c .PJSUA destroyed...
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