Re: [pjsip] Basic PJSUA question (without sound card)
Hello,
I'm new to PJSIP library so please forgive this smaybe silly question.
I'm starting the development of a call center app and consider using pjsip
for the moment. Basically the goal is to make a simple server application
that connects 2 separate calls while staying "between" the two (one use case
would be one call arrives, app calls another phone, then liaise them
together while recording in the conf. The app should also be able to
redirect one of call if needed).
I'm at the simple point of making/receiving calls while playing a little
with the lib to understand it. For the moment on my debugging machine it's
working fine (windows machine/sound card up and running), I can receive and
make calls without problems. But when deploying on testing server (linux
machine/no sound card), PJSUA is sending 488 SIP Code after logging a no
active media after sdp neg (logs below).
The thing is, I don't need to play audio on the server (though I should be
able to record the conf to a wav file) so do I really need a sound card on
the server?
I tried using pjsua_set_null_snd_dev() but it's not affecting anything.
Can someone please tell me what are my options here? Do I need to get below
pjsua lib, or can I just instruct it to use a conf port without sound
device?
Thanx for your help.
08:30:46.935 os_core_unix.c !pjlib 2.1 for POSIX initialized
08:30:46.936 sip_endpoint.c .Creating endpoint instance...
08:30:46.937 pjlib .select() I/O Queue created (0x137d110)
08:30:46.937 sip_endpoint.c .Module "mod-msg-print" registered
08:30:46.937 sip_transport. .Transport manager created.
08:30:46.937 pjsua_core.c .PJSUA state changed: NULL --> CREATED
08:30:46.937 pjsua_core.c SIP UDP socket reachable at 192.168.0.72:5060
08:30:46.937 udp0x13716f0 SIP UDP transport started, published address is
192.168.0.72:5060
08:30:46.937 sip_endpoint.c .Module "mod-pjsua-log" registered
08:30:46.937 sip_endpoint.c .Module "mod-tsx-layer" registered
08:30:46.937 sip_endpoint.c .Module "mod-stateful-util" registered
08:30:46.937 sip_endpoint.c .Module "mod-ua" registered
08:30:46.937 sip_endpoint.c .Module "mod-100rel" registered
08:30:46.937 sip_endpoint.c .Module "mod-pjsua" registered
08:30:46.937 sip_endpoint.c .Module "mod-invite" registered
08:30:46.946 pa_dev.c ..PortAudio sound library initialized, status=0
08:30:46.946 pa_dev.c ..PortAudio host api count=2
08:30:46.946 pa_dev.c ..Sound device count=0
08:30:46.947 pjlib ..select() I/O Queue created (0x139fe68)
08:30:46.957 sip_endpoint.c .Module "mod-evsub" registered
08:30:46.957 sip_endpoint.c .Module "mod-presence" registered
08:30:46.957 sip_endpoint.c .Module "mod-mwi" registered
08:30:46.957 sip_endpoint.c .Module "mod-refer" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-pres" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-im" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-options" registered
08:30:46.957 pjsua_core.c .1 SIP worker threads created
08:30:46.957 pjsua_core.c .pjsua version 2.1 for
Linux-4.4.0.62/x86_64/glibc-2.17 initialized
08:30:46.957 pjsua_core.c .PJSUA state changed: CREATED --> INIT
08:30:46.957 pjsua_aud.c Setting null sound device..
08:30:46.957 pjsua_aud.c .Opening null sound device..
08:30:46.957 pjsua_core.c PJSUA state changed: INIT --> STARTING
08:30:46.957 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
08:30:46.957 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
08:30:46.957 pjsua_acc.c Adding account:
id=sip:2000@frvln001.sodiware.lan
08:30:46.958 pjsua_acc.c .Account sip:2000@frvln001.sodiware.lan added
with id 0
08:30:46.958 pjsua_acc.c Acc 0: setting registration..
08:30:46.959 pjsua_core.c ..TX 521 bytes Request msg REGISTER/cseq=10512
(tdta0x13be650) to UDP 192.168.0.71:5060:
REGISTER sip:frvln001.sodiware.lan SIP/2.0
Via: SIP/2.0/UDP
192.168.0.72:5060;rport;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040
Max-Forwards: 70
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10512 REGISTER
Contact: sip:2000@192.168.0.72:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
08:30:46.959 pjsua_acc.c .Acc 0: Registration sent
08:30:46.960 pjsua_core.c .RX 608 bytes Response msg
401/REGISTER/cseq=10512 (rdata0x13867a8) from UDP 192.168.0.71:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.72:5060;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040;recei
ved=192.168.0.72;rport=5060
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan;tag=as291f3de5
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10512 REGISTER
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sodiware.lan",
nonce="0562418a"
Content-Length: 0
--end msg--
08:30:46.960 pjsua_core.c ....TX 693 bytes Request msg
REGISTER/cseq=10513 (tdta0x13be650) to UDP 192.168.0.71:5060:
REGISTER sip:frvln001.sodiware.lan SIP/2.0
Via: SIP/2.0/UDP
192.168.0.72:5060;rport;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4
Max-Forwards: 70
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10513 REGISTER
Contact: sip:2000@192.168.0.72:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Authorization: Digest username="2000", realm="sodiware.lan",
nonce="0562418a", uri="sip:frvln001.sodiware.lan",
response="ca2dd328bfa588ca93b23599d3f71551", algorithm=MD5
Content-Length: 0
--end msg--
08:30:46.961 pjsua_core.c .RX 623 bytes Response msg
200/REGISTER/cseq=10513 (rdata0x13867a8) from UDP 192.168.0.71:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.72:5060;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4;recei
ved=192.168.0.72;rport=5060
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan;tag=as291f3de5
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10513 REGISTER
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 300
Contact: sip:2000@192.168.0.72:5060;ob;expires=300
Date: Sat, 11 Feb 2017 01:57:47 GMT
Content-Length: 0
--end msg--
08:30:46.961 pjsua_acc.c ....SIP outbound status for acc 0 is not active
08:30:46.961 pjsua_acc.c ....sip:2000@frvln001.sodiware.lan:
registration success, status=200 (OK), will re-register in 300 seconds
08:30:46.961 pjsua_acc.c ....Keep-alive timer started for acc 0,
destination:192.168.0.71:5060, interval:15s
08:31:04.791 pjsua_core.c .RX 882 bytes Request msg INVITE/cseq=102
(rdata0x13867a8) from UDP 192.168.0.71:5060:
INVITE sip:2000@192.168.0.72:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.71:5060;branch=z9hG4bK751edb1f
Max-Forwards: 70
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
To: sip:2000@192.168.0.72:5060;ob
Contact: sip:5000@192.168.0.71:5060
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Date: Sat, 11 Feb 2017 01:58:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296
v=0
o=root 595539155 595539155 IN IP4 192.168.0.71
s=Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
c=IN IP4 192.168.0.71
t=0 0
m=audio 10616 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=ptime:20
a=sendrecv
--end msg--
08:31:04.791 pjsua_call.c .Incoming Request msg INVITE/cseq=102
(rdata0x13867a8)
08:31:04.791 pjsua_media.c ..Call 0: initializing media..
08:31:04.791 pjsua_media.c ...RTP socket reachable at 192.168.0.72:40000
08:31:04.791 pjsua_media.c ...RTCP socket reachable at 192.168.0.72:40001
08:31:04.791 pjsua_media.c ...Media index 0 selected for audio call 0
08:31:04.792 /home/SODIWARE ..>>>>> Remote session from sdp form <<<<
08:31:04.792 pjsua_core.c .....TX 291 bytes Response msg
100/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
CSeq: 102 INVITE
Content-Length: 0
--end msg--
08:31:04.792 /home/SODIWARE ..Waiting 2 seconds : 1
08:31:06.792 pjsua_call.c ..Answering call 0: code=200
08:31:06.792 pjsua_media.c .....Call 0: updating media..
08:31:06.792 pjsua_aud.c ......Audio channel update..
08:31:06.793 strm0x13d1a98 .......VAD temporarily disabled
08:31:06.793 strm0x13d1a98 .......Encoder stream started
08:31:06.793 strm0x13d1a98 .......Decoder stream started
08:31:06.793 pjsua_media.c ......pjsua_aud_channel_update() failed
ftitanslaved.out:
/home/SODIWARE/kpamafre/projects/swsiprt/IncomingCall.cpp:31: virtual
HRESULT CIncomingCall::Answer(): Assertion `status == PJ_SUCCESS' failed.
or call_id 0 media 0: Invalid operation (PJ_EINVALIDOP)
08:31:06.793 pjsua_call.c .....Unable to create media session: No active
media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]
08:31:06.793 pjsua_core.c ........TX 491 bytes Response msg
488/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
To: sip:2000@192.168.0.72;ob;tag=baa5cf52-af70-42a0-a919-5d511bfcfb78
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0
--end msg--
08:31:06.793 /home/SODIWARE ...........Found agent 0 for call -781117144
08:31:06.793 /home/SODIWARE ...........Call state changed (id: -781115088)
08:31:06.793 pjsua_media.c ...........Call 0: deinitializing media..
08:31:06.793 pjsua_media.c .............Media stream call00:0 is destroyed
08:31:06.793 pjsua_call.c ...Error creating response: Not Acceptable Here
[status=170488]
Ok fixed,
Turned out it was a problem with the builtin (apt-get) distrib of pjsip/alsa
library from ubuntu.
I recompiled the sources and it works ok now.
Fred
From: Kpama Frederic [mailto:kpamafrederic@gmail.com]
Sent: dimanche 12 février 2017 00:53
To: pjsip@lists.pjsip.org
Subject: RE: Basic PJSUA question (without sound card)
Hello,
Im new to PJSIP library so please forgive this smaybe silly question.
Im starting the development of a call center app and consider using pjsip
for the moment. Basically the goal is to make a simple server application
that connects 2 separate calls while staying between the two (one use case
would be one call arrives, app calls another phone, then liaise them
together while recording in the conf. The app should also be able to
redirect one of call if needed).
Im at the simple point of making/receiving calls while playing a little
with the lib to understand it. For the moment on my debugging machine its
working fine (windows machine/sound card up and running), I can receive and
make calls without problems. But when deploying on testing server (linux
machine/no sound card), PJSUA is sending 488 SIP Code after logging a no
active media after sdp neg (logs below).
The thing is, I dont need to play audio on the server (though I should be
able to record the conf to a wav file) so do I really need a sound card on
the server?
I tried using pjsua_set_null_snd_dev() but its not affecting anything.
Can someone please tell me what are my options here? Do I need to get below
pjsua lib, or can I just instruct it to use a conf port without sound
device?
Thanx for your help.
08:30:46.935 os_core_unix.c !pjlib 2.1 for POSIX initialized
08:30:46.936 sip_endpoint.c .Creating endpoint instance...
08:30:46.937 pjlib .select() I/O Queue created (0x137d110)
08:30:46.937 sip_endpoint.c .Module "mod-msg-print" registered
08:30:46.937 sip_transport. .Transport manager created.
08:30:46.937 pjsua_core.c .PJSUA state changed: NULL --> CREATED
08:30:46.937 pjsua_core.c SIP UDP socket reachable at 192.168.0.72:5060
08:30:46.937 udp0x13716f0 SIP UDP transport started, published address is
192.168.0.72:5060
08:30:46.937 sip_endpoint.c .Module "mod-pjsua-log" registered
08:30:46.937 sip_endpoint.c .Module "mod-tsx-layer" registered
08:30:46.937 sip_endpoint.c .Module "mod-stateful-util" registered
08:30:46.937 sip_endpoint.c .Module "mod-ua" registered
08:30:46.937 sip_endpoint.c .Module "mod-100rel" registered
08:30:46.937 sip_endpoint.c .Module "mod-pjsua" registered
08:30:46.937 sip_endpoint.c .Module "mod-invite" registered
08:30:46.946 pa_dev.c ..PortAudio sound library initialized, status=0
08:30:46.946 pa_dev.c ..PortAudio host api count=2
08:30:46.946 pa_dev.c ..Sound device count=0
08:30:46.947 pjlib ..select() I/O Queue created (0x139fe68)
08:30:46.957 sip_endpoint.c .Module "mod-evsub" registered
08:30:46.957 sip_endpoint.c .Module "mod-presence" registered
08:30:46.957 sip_endpoint.c .Module "mod-mwi" registered
08:30:46.957 sip_endpoint.c .Module "mod-refer" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-pres" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-im" registered
08:30:46.957 sip_endpoint.c .Module "mod-pjsua-options" registered
08:30:46.957 pjsua_core.c .1 SIP worker threads created
08:30:46.957 pjsua_core.c .pjsua version 2.1 for
Linux-4.4.0.62/x86_64/glibc-2.17 initialized
08:30:46.957 pjsua_core.c .PJSUA state changed: CREATED --> INIT
08:30:46.957 pjsua_aud.c Setting null sound device..
08:30:46.957 pjsua_aud.c .Opening null sound device..
08:30:46.957 pjsua_core.c PJSUA state changed: INIT --> STARTING
08:30:46.957 sip_endpoint.c .Module "mod-unsolicited-mwi" registered
08:30:46.957 pjsua_core.c .PJSUA state changed: STARTING --> RUNNING
08:30:46.957 pjsua_acc.c Adding account:
id=sip:2000@frvln001.sodiware.lan
08:30:46.958 pjsua_acc.c .Account sip:2000@frvln001.sodiware.lan added
with id 0
08:30:46.958 pjsua_acc.c Acc 0: setting registration..
08:30:46.959 pjsua_core.c ..TX 521 bytes Request msg REGISTER/cseq=10512
(tdta0x13be650) to UDP 192.168.0.71:5060:
REGISTER sip:frvln001.sodiware.lan SIP/2.0
Via: SIP/2.0/UDP
192.168.0.72:5060;rport;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040
Max-Forwards: 70
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10512 REGISTER
Contact: sip:2000@192.168.0.72:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Content-Length: 0
--end msg--
08:30:46.959 pjsua_acc.c .Acc 0: Registration sent
08:30:46.960 pjsua_core.c .RX 608 bytes Response msg
401/REGISTER/cseq=10512 (rdata0x13867a8) from UDP 192.168.0.71:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.0.72:5060;branch=z9hG4bKPj2c04dbf3-df29-447f-8fca-f71170074040;recei
ved=192.168.0.72;rport=5060
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan;tag=as291f3de5
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10512 REGISTER
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="sodiware.lan",
nonce="0562418a"
Content-Length: 0
--end msg--
08:30:46.960 pjsua_core.c ....TX 693 bytes Request msg
REGISTER/cseq=10513 (tdta0x13be650) to UDP 192.168.0.71:5060:
REGISTER sip:frvln001.sodiware.lan SIP/2.0
Via: SIP/2.0/UDP
192.168.0.72:5060;rport;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4
Max-Forwards: 70
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10513 REGISTER
Contact: sip:2000@192.168.0.72:5060;ob
Expires: 300
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Authorization: Digest username="2000", realm="sodiware.lan",
nonce="0562418a", uri="sip:frvln001.sodiware.lan",
response="ca2dd328bfa588ca93b23599d3f71551", algorithm=MD5
Content-Length: 0
--end msg--
08:30:46.961 pjsua_core.c .RX 623 bytes Response msg
200/REGISTER/cseq=10513 (rdata0x13867a8) from UDP 192.168.0.71:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.72:5060;branch=z9hG4bKPj98d7f725-60d5-4af2-91a0-9a8d96bc59a4;recei
ved=192.168.0.72;rport=5060
From:
sip:2000@frvln001.sodiware.lan;tag=ff67eaac-8b15-4757-b998-0465ba37aa45
To: sip:2000@frvln001.sodiware.lan;tag=as291f3de5
Call-ID: 42d958df-adb2-470f-9a13-e4cfe61ce487
CSeq: 10513 REGISTER
Server: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Expires: 300
Contact: sip:2000@192.168.0.72:5060;ob;expires=300
Date: Sat, 11 Feb 2017 01:57:47 GMT
Content-Length: 0
--end msg--
08:30:46.961 pjsua_acc.c ....SIP outbound status for acc 0 is not active
08:30:46.961 pjsua_acc.c ....sip:2000@frvln001.sodiware.lan:
registration success, status=200 (OK), will re-register in 300 seconds
08:30:46.961 pjsua_acc.c ....Keep-alive timer started for acc 0,
destination:192.168.0.71:5060, interval:15s
08:31:04.791 pjsua_core.c .RX 882 bytes Request msg INVITE/cseq=102
(rdata0x13867a8) from UDP 192.168.0.71:5060:
INVITE sip:2000@192.168.0.72:5060;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.0.71:5060;branch=z9hG4bK751edb1f
Max-Forwards: 70
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
To: sip:2000@192.168.0.72:5060;ob
Contact: sip:5000@192.168.0.71:5060
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
Date: Sat, 11 Feb 2017 01:58:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 296
v=0
o=root 595539155 595539155 IN IP4 192.168.0.71
s=Asterisk PBX 1.8.13.1~dfsg-3ubuntu3
c=IN IP4 192.168.0.71
t=0 0
m=audio 10616 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=ptime:20
a=sendrecv
--end msg--
08:31:04.791 pjsua_call.c .Incoming Request msg INVITE/cseq=102
(rdata0x13867a8)
08:31:04.791 pjsua_media.c ..Call 0: initializing media..
08:31:04.791 pjsua_media.c ...RTP socket reachable at 192.168.0.72:40000
08:31:04.791 pjsua_media.c ...RTCP socket reachable at 192.168.0.72:40001
08:31:04.791 pjsua_media.c ...Media index 0 selected for audio call 0
08:31:04.792 /home/SODIWARE ..>>>>> Remote session from sdp form <<<<
08:31:04.792 pjsua_core.c .....TX 291 bytes Response msg
100/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
CSeq: 102 INVITE
Content-Length: 0
--end msg--
08:31:04.792 /home/SODIWARE ..Waiting 2 seconds : 1
08:31:06.792 pjsua_call.c ..Answering call 0: code=200
08:31:06.792 pjsua_media.c .....Call 0: updating media..
08:31:06.792 pjsua_aud.c ......Audio channel update..
08:31:06.793 strm0x13d1a98 .......VAD temporarily disabled
08:31:06.793 strm0x13d1a98 .......Encoder stream started
08:31:06.793 strm0x13d1a98 .......Decoder stream started
08:31:06.793 pjsua_media.c ......pjsua_aud_channel_update() failed
ftitanslaved.out:
/home/SODIWARE/kpamafre/projects/swsiprt/IncomingCall.cpp:31: virtual
HRESULT CIncomingCall::Answer(): Assertion `status == PJ_SUCCESS' failed.
or call_id 0 media 0: Invalid operation (PJ_EINVALIDOP)
08:31:06.793 pjsua_call.c .....Unable to create media session: No active
media stream after negotiation (PJMEDIA_SDPNEG_ENOMEDIA) [status=220048]
08:31:06.793 pjsua_core.c ........TX 491 bytes Response msg
488/INVITE/cseq=102 (tdta0x13ccc50) to UDP 192.168.0.71:5060:
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.71:5060;received=192.168.0.71;branch=z9hG4bK751edb1f
Call-ID: 09302cad1054f32764b853443303030b@192.168.0.71:5060
mailto:09302cad1054f32764b853443303030b@192.168.0.71:5060
From: "New User" sip:5000@192.168.0.71;tag=as1e417004
To: sip:2000@192.168.0.72;ob;tag=baa5cf52-af70-42a0-a919-5d511bfcfb78
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Length: 0
--end msg--
08:31:06.793 /home/SODIWARE ...........Found agent 0 for call -781117144
08:31:06.793 /home/SODIWARE ...........Call state changed (id: -781115088)
08:31:06.793 pjsua_media.c ...........Call 0: deinitializing media..
08:31:06.793 pjsua_media.c .............Media stream call00:0 is destroyed
08:31:06.793 pjsua_call.c ...Error creating response: Not Acceptable Here
[status=170488]