Re: How to use pjsua to make a phone call?

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Ok thanks for your answer but it’s exactly what I’d like to understand :

 

Here is what I exactly did after starting pjsua : First I register, i’ts done successfully and than I make a call which fails and tells ‘not registered’ :

 

       $ ./pjsua-x86_64-unknown-linux-gnu

       16:43:53.666         os_core_unix.c !pjlib 2.8 for POSIX initialized

       16:43:53.668         sip_endpoint.c  .Creating endpoint instance...

       16:43:53.669                  pjlib  .select() I/O Queue created (0x55ab4f4d27f0)

       16:43:53.669         sip_endpoint.c  .Module "mod-msg-print" registered

       16:43:53.669        sip_transport.c  .Transport manager created.

       16:43:53.669           pjsua_core.c  .PJSUA state changed: NULL --> CREATED

       16:43:53.670         sip_endpoint.c  .Module "mod-pjsua-log" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-tsx-layer" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-stateful-util" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-ua" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-100rel" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-pjsua" registered

       16:43:53.670         sip_endpoint.c  .Module "mod-invite" registered

       16:43:53.754             alsa_dev.c  ..ALSA driver found 32 devices

       16:43:53.755             alsa_dev.c  ..ALSA initialized

       16:43:53.755                  pjlib  ..select() I/O Queue created (0x55ab4f55bc88)

       16:43:53.758         sip_endpoint.c  .Module "mod-evsub" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-presence" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-mwi" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-refer" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-pjsua-pres" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-pjsua-im" registered

       16:43:53.758         sip_endpoint.c  .Module "mod-pjsua-options" registered

       16:43:53.758           pjsua_core.c  .1 SIP worker threads created

       16:43:53.758           pjsua_core.c  .pjsua version 2.8 for Linux-4.15.0.39/x86_64/glibc-2.27 initialized

       16:43:53.758           pjsua_core.c  .PJSUA state changed: CREATED --> INIT

       16:43:53.758         sip_endpoint.c  Module "mod-default-handler" registered

       16:43:53.758           pjsua_core.c  SIP UDP socket reachable at 192.168.105.26:5060

       16:43:53.758      udp0x55ab4f53c450  SIP UDP transport started, published address is 192.168.105.26:5060

       16:43:53.758            pjsua_acc.c  Adding account: id=<sip:192.168.105.26:5060>

       16:43:53.758            pjsua_acc.c  .Account <sip:192.168.105.26:5060> added with id 0

       16:43:53.758            pjsua_acc.c  Modifying account 0

       16:43:53.758            pjsua_acc.c  Acc 0: setting online status to 1..

       16:43:53.758             tcptp:5060  SIP TCP listener ready for incoming connections at 192.168.105.26:5060

       16:43:53.758            pjsua_acc.c  Adding account: id=<sip:192.168.105.26:5060;transport=TCP>

       16:43:53.758            pjsua_acc.c  .Account <sip:192.168.105.26:5060;transport=TCP> added with id 1

       16:43:53.758            pjsua_acc.c  Modifying account 1

       16:43:53.758            pjsua_acc.c  Acc 1: setting online status to 1..

       16:43:53.758           pjsua_core.c  PJSUA state changed: INIT --> STARTING

       16:43:53.758         sip_endpoint.c  .Module "mod-unsolicited-mwi" registered

       16:43:53.758           pjsua_core.c  .PJSUA state changed: STARTING --> RUNNING

       16:43:53.758                 main.c  Ready: Success

       >>>>

       Account list:

         [ 0] <sip:192.168.105.26:5060>: does not register

              Online status: Online

       *[ 1] <sip:192.168.105.26:5060;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:0033972nnnnnn@xxxxxxxxxxx

       URL of the registrar: (empty to cancel): sip:sip3.ovh.fr

       Auth Realm: (empty to cancel): *

       Auth Username: (empty to cancel): 0033972nnnnnn

       Auth Password: (empty to cancel): mypasswd

       16:44:38.922            pjsua_acc.c  Adding account: id=sip:0033972nnnnnn@xxxxxxxxxxx

       16:44:38.922            pjsua_acc.c  .Account sip:0033972nnnnnn@xxxxxxxxxxx added with id 2

       16:44:38.922            pjsua_acc.c  .Acc 2: setting registration..

       16:44:38.950           pjsua_core.c  ...TX 568 bytes Request msg REGISTER/cseq=32229 (tdta0x55ab4f542bc8) to UDP 91.121.129.159:5060:

       REGISTER sip:sip3.ovh.fr SIP/2.0

       Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjIR29GPvvEm8AWC5s1qV4EKMviO-3wRim

       Max-Forwards: 70

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       CSeq: 32229 REGISTER

       User-Agent: PJSUA v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

       Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

       Expires: 300

       Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

       Content-Length:  0

 

 

       --end msg--

       16:44:38.950            pjsua_acc.c  ..Acc 2: Registration sent

       >>> 16:44:38.974           pjsua_core.c  .RX 341 bytes Response msg 100/REGISTER/cseq=32229 (rdata0x55ab4f53ded8) from UDP 91.121.129.159:5060:

       SIP/2.0 100 Trying

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       CSeq: 32229 REGISTER

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjIR29GPvvEm8AWC5s1qV4EKMviO-3wRim

       Content-Length: 0

 

 

       --end msg--

       16:44:38.996           pjsua_core.c  .RX 549 bytes Response msg 401/REGISTER/cseq=32229 (rdata0x7f9b84009988) from UDP 91.121.129.159:5060:

       SIP/2.0 401 Unauthorized

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       CSeq: 32229 REGISTER

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=00-08027-0ae69417-6950486d5

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjIR29GPvvEm8AWC5s1qV4EKMviO-3wRim

       WWW-Authenticate: Digest realm="sip3.ovh.fr",nonce="0ae6882d0ce42861705875534fdfc7d4",opaque="0ae410ed63e0dd3",stale=false,algorithm=MD5

       Server: Cirpack/v4.76 (gw_sip)

       Content-Length: 0

 

 

       --end msg--

       16:44:38.996           pjsua_core.c  ....TX 788 bytes Request msg REGISTER/cseq=32230 (tdta0x55ab4f542bc8) to UDP 91.121.129.159:5060:

       REGISTER sip:sip3.ovh.fr SIP/2.0

       Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPj-vQTwMT-eq-zheaUxY7-qZAnQdFKL7ed

       Max-Forwards: 70

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       CSeq: 32230 REGISTER

       User-Agent: PJSUA v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

       Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

       Expires: 300

       Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

       Authorization: Digest username="0033972nnnnnn", realm="sip3.ovh.fr", nonce="0ae6882d0ce42861705875534fdfc7d4", uri="sip:sip3.ovh.fr", response="e70170dacab9d6f6f4d079c3d2a7165a", algorithm=MD5, opaque="0ae410ed63e0dd3"

       Content-Length:  0

 

 

       --end msg--

       16:44:39.020           pjsua_core.c  .RX 341 bytes Response msg 100/REGISTER/cseq=32230 (rdata0x7f9b84009988) from UDP 91.121.129.159:5060:

       SIP/2.0 100 Trying

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       CSeq: 32230 REGISTER

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPj-vQTwMT-eq-zheaUxY7-qZAnQdFKL7ed

       Content-Length: 0

 

 

       --end msg--

       16:44:39.055           pjsua_core.c  .RX 514 bytes Response msg 200/REGISTER/cseq=32230 (rdata0x7f9b84009988) from UDP 91.121.129.159:5060:

       SIP/2.0 200 OK

       Call-ID: EZarPAaegxJWYDDaUFn8IMS.fT1ynipE

       Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>;expires=300

       CSeq: 32230 REGISTER

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V-2VX9ftcecN4RnSV7HZISLlDK5SSlsb

       To: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=00-07606-0ae69433-66effe766

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPj-vQTwMT-eq-zheaUxY7-qZAnQdFKL7ed

       P-Associated-URI: <sip:0972617378@xxxxxxxxxxx>

       Server: Cirpack/v4.76 (gw_sip)

       Content-Length: 0

 

 

       --end msg--

       16:44:39.055            pjsua_acc.c  ....SIP outbound status for acc 2 is not active

       16:44:39.056            pjsua_acc.c  ....sip:0033972nnnnnn@xxxxxxxxxxx: registration success, status=200 (OK), will re-register in 300 seconds

       16:44:39.056            pjsua_acc.c  ....Keep-alive timer started for acc 2, destination:91.121.129.159:5060:15, interval:0s

 

 

 

       +=============================================================================+

       |       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:0033661nnnnnn@xxxxxxxxxxx

       16:45:20.251           pjsua_call.c !Making call with acc #2 to sip:0033661nnnnnn@xxxxxxxxxxx

       16:45:20.251            pjsua_aud.c  .Set sound device: capture=-1, playback=-2

       16:45:20.251            pjsua_app.c  ..Turning sound device -1 -2 ON

       16:45:20.251            pjsua_aud.c  ..Opening sound device (speaker + mic) PCM@16000/1/20ms

       16:45:20.290       ec0x55ab4f4fdc80  ...AEC created, clock_rate=16000, channel=1, samples per frame=320, tail length=200 ms, latency=0 ms

       16:45:20.290          pjsua_media.c  .Call 0: initializing media..

       16:45:20.290          pjsua_media.c !..RTP socket reachable at 192.168.105.26:4000

       16:45:20.290          pjsua_media.c !..RTCP socket reachable at 192.168.105.26:4001

       16:45:20.290          pjsua_media.c  ..Media index 0 selected for audio call 0

       16:45:20.291           pjsua_core.c  ....TX 1197 bytes Request msg INVITE/cseq=12436 (tdta0x55ab4f597c58) to UDP 91.121.129.159:5060:

       INVITE sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

       Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjLV1XZ.poXUu0e6ret.J4DNdX74yIG7Qk

       Max-Forwards: 70

       From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=5k7dKtZTZPpoK56fFkhEYkfhFPWEO8gd

       To: sip:0033661nnnnnn@xxxxxxxxxxx

       Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

       Call-ID: cN.De5mv.0UqSrDLEYaDomHf498ySfMr

       CSeq: 12436 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 v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

       Content-Type: application/sdp

       Content-Length:   520

 

       v=0

       o=- 3752495120 3752495120 IN IP4 192.168.105.26

       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 192.168.105.26

       b=TIAS:64000

       a=rtcp:4001 IN IP4 192.168.105.26

       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 0-16

       a=ssrc:472429953 cname:6d08069e32850a6a

 

       --end msg--

       16:45:20.291            pjsua_app.c  .......Call 0 state changed to CALLING

       >>> 16:45:20.314           pjsua_core.c  .RX 339 bytes Response msg 100/INVITE/cseq=12436 (rdata0x7f9b84009988) from UDP 91.121.129.159:5060:

       SIP/2.0 100 Trying

       Call-ID: cN.De5mv.0UqSrDLEYaDomHf498ySfMr

       CSeq: 12436 INVITE

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=5k7dKtZTZPpoK56fFkhEYkfhFPWEO8gd

       To: <sip:0033661nnnnnn@xxxxxxxxxxx>

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjLV1XZ.poXUu0e6ret.J4DNdX74yIG7Qk

       Content-Length: 0

 

 

       --end msg--

       16:45:20.314           pjsua_core.c  .RX 379 bytes Response msg 403/INVITE/cseq=12436 (rdata0x7f9b84009988) from UDP 91.121.129.159:5060:

       SIP/2.0 403 not registered

       Call-ID: cN.De5mv.0UqSrDLEYaDomHf498ySfMr

       CSeq: 12436 INVITE

       From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=5k7dKtZTZPpoK56fFkhEYkfhFPWEO8gd

       To: <sip:0033661nnnnnn@xxxxxxxxxxx>;tag=02-21423-716b2aa4-5cfb38d43

       Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjLV1XZ.poXUu0e6ret.J4DNdX74yIG7Qk

       Content-Length: 0

 

 

       --end msg--

       16:45:20.314           pjsua_core.c  ..TX 377 bytes Request msg ACK/cseq=12436 (tdta0x7f9b84007268) to UDP 91.121.129.159:5060:

       ACK sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

       Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjLV1XZ.poXUu0e6ret.J4DNdX74yIG7Qk

       Max-Forwards: 70

       From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=5k7dKtZTZPpoK56fFkhEYkfhFPWEO8gd

       To: sip:0033661nnnnnn@xxxxxxxxxxx;tag=02-21423-716b2aa4-5cfb38d43

       Call-ID: cN.De5mv.0UqSrDLEYaDomHf498ySfMr

       CSeq: 12436 ACK

       Content-Length:  0

 

 

       --end msg--

       16:45:20.314            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=403 (not registered)]

       16:45:20.315     pjsua_app_common.c  .....

         [DISCONNCTD] To: sip:0033661nnnnnn@xxxxxxxxxxx

           Call time: 00h:00m:00s, 1st res in 24 ms, conn in 0ms

       16:45:20.315          pjsua_media.c  .....Call 0: deinitializing media..

       16:45:20.315          pjsua_media.c  ......Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0

       16:45:21.315            pjsua_aud.c !Closing sound device after idle for 1 second(s)

       16:45:21.315            pjsua_app.c  .Turning sound device -1 -2 OFF

       16:45:21.315            pjsua_aud.c  .Closing default sound playback device and default sound capture device

 

 

-----------------------------------------------------------------------------------------------------------------------------------------------------------------------

 

 

,

 

may be because  sip client is not registered ?

This is from you log:

"....

    16:00:26.966           pjsua_core.c  .RX 379 bytes Response msg 403/INVITE/cseq=13359 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 403 not registered

 

......"

 


From: pjsip <pjsip-bounces@xxxxxxxxxxxxxxx> on behalf of alain.bonnefoy@xxxxxxxxxxxxxxxxxxxxxxx <alain.bonnefoy@xxxxxxxxxxxxxxxxxxxxxxx>
Sent: Thursday, November 29, 2018 17:23
To:
pjsip@xxxxxxxxxxxxxxx
Subject: How to use pjsua to make a phone call?

 

Hello,

 

I try to use pjsua to call a phone number a play a wav file.

I looked at this tuto(http://aminastaneh.net/2012-05-06/fun-with-command-line-sip.html) which seems to do it successfully, but with a quite old version.

 

 

If I try to do like this tuto, I cannot register to my sip account.

Here is the command I use:

 

    ./pjsua --config-file ./pjsua_config sip:0033661nnnnnn@xxxxxxxxxxx

 

The config file is:

 

                --id sip:0033972nnnnnnnn@xxxxxxxxxxx

                --registrar sip:sip3.ovh.fr

                --username 0033972nnnnnnnn

                --password myPasswd

                --realm *

                --null-audio

                --auto-play

                --play-file ./test.wav

 

and the result is:

 

                16:00:26.844         os_core_unix.c !pjlib 2.8 for POSIX initialized

                16:00:26.845         sip_endpoint.c  .Creating endpoint instance...

                16:00:26.845                  pjlib  .select() I/O Queue created (0x55ffcb0d37f0)

                16:00:26.845         sip_endpoint.c  .Module "mod-msg-print" registered

                16:00:26.845        sip_transport.c  .Transport manager created.

                16:00:26.845           pjsua_core.c  .PJSUA state changed: NULL --> CREATED

                16:00:26.845         sip_endpoint.c  .Module "mod-pjsua-log" registered

                16:00:26.845         sip_endpoint.c  .Module "mod-tsx-layer" registered

                16:00:26.845         sip_endpoint.c  .Module "mod-stateful-util" registered

                16:00:26.845         sip_endpoint.c  .Module "mod-ua" registered

                16:00:26.845         sip_endpoint.c  .Module "mod-100rel" registered

                16:00:26.845         sip_endpoint.c  .Module "mod-pjsua" registered

                16:00:26.846         sip_endpoint.c  .Module "mod-invite" registered

                16:00:26.938             alsa_dev.c  ..ALSA driver found 32 devices

                16:00:26.938             alsa_dev.c  ..ALSA initialized

                16:00:26.939                  pjlib  ..select() I/O Queue created (0x55ffcb0ed158)

                16:00:26.941         sip_endpoint.c  .Module "mod-evsub" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-presence" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-mwi" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-refer" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-pjsua-pres" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-pjsua-im" registered

                16:00:26.941         sip_endpoint.c  .Module "mod-pjsua-options" registered

                16:00:26.941           pjsua_core.c  .1 SIP worker threads created

                16:00:26.941           pjsua_core.c  .pjsua version 2.8 for Linux-4.15.0.39/x86_64/glibc-2.27 initialized

                16:00:26.941           pjsua_core.c  .PJSUA state changed: CREATED --> INIT

                16:00:26.941         sip_endpoint.c  Module "mod-default-handler" registered

                16:00:26.941            pjsua_aud.c  Creating file player: ./test.wav..

                16:00:26.941           wav_player.c  .File player './test.wav' created: samp.rate=16000, ch=1, bufsize=4KB, filesize=163KB

                16:00:26.941            pjsua_aud.c  .Player created, id=0, slot=1

                16:00:26.941           pjsua_core.c  SIP UDP socket reachable at 192.168.105.26:5060

                16:00:26.941      udp0x55ffcb14bd90  SIP UDP transport started, published address is 192.168.105.26:5060

                16:00:26.942            pjsua_acc.c  Adding account: id=<sip:192.168.105.26:5060>

                16:00:26.942            pjsua_acc.c  .Account <sip:192.168.105.26:5060> added with id 0

                16:00:26.942            pjsua_acc.c  Modifying account 0

                16:00:26.942            pjsua_acc.c  Acc 0: setting online status to 1..

                16:00:26.942            pjsua_acc.c  Adding account: id=sip:0033972617378@xxxxxxxxxxx

                16:00:26.942            pjsua_acc.c  .Account sip:0033972nnnnnn@xxxxxxxxxxx added with id 1

                16:00:26.942            pjsua_acc.c  .Acc 1: setting registration..

                16:00:26.942           pjsua_core.c  ...TX 568 bytes Request msg REGISTER/cseq=20718 (tdta0x55ffcb151698) to UDP 91.121.129.159:5060:

                REGISTER sip:sip3.ovh.fr SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjwZ0eQlyjNoDdLg6UG-Osr8gZFdkQ05JL

                Max-Forwards: 70

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnn@xxxxxxxxxxx>

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                CSeq: 20718 REGISTER

                User-Agent: PJSUA v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

                Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

                Expires: 300

                Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

                Content-Length:  0

 

 

                --end msg--

                16:00:26.942            pjsua_acc.c  ..Acc 1: Registration sent

                16:00:26.942            pjsua_acc.c  Acc 1: setting online status to 1..

                16:00:26.942           pjsua_pres.c  Adding buddy: sip:0033661nnnnnn@xxxxxxxxxxx

                16:00:26.942           pjsua_pres.c  .Buddy 0 added.

                16:00:26.942           pjsua_pres.c  ..Buddy 0: updating presence..

                16:00:26.942            pjsua_aud.c  Setting null sound device..

                16:00:26.942            pjsua_app.c  .Turning sound device -99 -99 ON

                16:00:26.942            pjsua_aud.c  .Opening null sound device..

                16:00:26.942           pjsua_core.c  PJSUA state changed: INIT --> STARTING

                16:00:26.942         sip_endpoint.c  .Module "mod-unsolicited-mwi" registered

                16:00:26.942           pjsua_core.c  .PJSUA state changed: STARTING --> RUNNING

                16:00:26.942                 main.c  Ready: Success

                16:00:26.942           pjsua_call.c  Making call with acc #1 to sip:0033661nnnnnn@xxxxxxxxxxx

                16:00:26.942          pjsua_media.c  .Call 0: initializing media..

                16:00:26.942          pjsua_media.c  ..RTP socket reachable at 192.168.105.26:4000

                16:00:26.942          pjsua_media.c  ..RTCP socket reachable at 192.168.105.26:4001

                16:00:26.942          pjsua_media.c  ..Media index 0 selected for audio call 0

                16:00:26.942           pjsua_core.c  ....TX 1198 bytes Request msg INVITE/cseq=13359 (tdta0x55ffcb16b4a8) to UDP 91.121.129.159:5060:

                INVITE sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjM2z7W8MLQLAIH9yQkxNQUMwAYogM9rwc

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=XqE9aDLlGlo3gE22AK4QBJoF1UzKt5FY

                To: sip:0033661nnnnnn@xxxxxxxxxxx

                Contact: <sip:003397nnnnnn@192.168.105.26:5060;ob>

                Call-ID: hTnBJcrx44AvHW1SnBZVLKqHHxQ0otOO

                CSeq: 13359 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 v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

                Content-Type: application/sdp

                Content-Length:   521

 

                v=0

                o=- 3752492426 3752492426 IN IP4 192.168.105.26

                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 192.168.105.26

                b=TIAS:64000

                a=rtcp:4001 IN IP4 192.168.105.26

                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 0-16

                a=ssrc:1011957511 cname:75cedd633ab5e97d

 

                --end msg--

                16:00:26.942            pjsua_app.c  .......Call 0 state changed to CALLING

                >>>>

                Account list:

                  [ 0] <sip:192.168.105.26:5060>: does not register

                       Online status: Online

                *[ 1] sip:0033972nnnnnn@xxxxxxxxxxx: 100/In Progress (expires=0)

                       Online status: Online

                Buddy list:

                [ 1] <?>  sip:0033661nnnnnn@xxxxxxxxxxx

 

                +=============================================================================+

                |       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:0033661nnnnnn@xxxxxxxxxxx [CALLING]

                >>> 16:00:26.966           pjsua_core.c  .RX 341 bytes Response msg 100/REGISTER/cseq=20718 (rdata0x55ffcb14d818) from UDP 91.121.129.159:5060:

                SIP/2.0 100 Trying

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                CSeq: 20718 REGISTER

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnn@xxxxxxxxxxx>

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjwZ0eQlyjNoDdLg6UG-Osr8gZFdkQ05JL

                Content-Length: 0

 

 

                --end msg--

                16:00:26.966           pjsua_core.c  .RX 339 bytes Response msg 100/INVITE/cseq=13359 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 100 Trying

                Call-ID: hTnBJcrx44AvHW1SnBZVLKqHHxQ0otOO

                CSeq: 13359 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=XqE9aDLlGlo3gE22AK4QBJoF1UzKt5FY

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjM2z7W8MLQLAIH9yQkxNQUMwAYogM9rwc

                Content-Length: 0

 

 

                --end msg--

                16:00:26.966           pjsua_core.c  .RX 379 bytes Response msg 403/INVITE/cseq=13359 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 403 not registered

                Call-ID: hTnBJcrx44AvHW1SnBZVLKqHHxQ0otOO

                CSeq: 13359 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=XqE9aDLlGlo3gE22AK4QBJoF1UzKt5FY

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>;tag=02-22581-7130a9fb-6f8a0a441

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjM2z7W8MLQLAIH9yQkxNQUMwAYogM9rwc

                Content-Length: 0

 

 

                --end msg--

                16:00:26.966           pjsua_core.c  ..TX 377 bytes Request msg ACK/cseq=13359 (tdta0x7f414c00bab8) to UDP 91.121.129.159:5060:

                ACK sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjM2z7W8MLQLAIH9yQkxNQUMwAYogM9rwc

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=XqE9aDLlGlo3gE22AK4QBJoF1UzKt5FY

                To: sip:0033661nnnnnn@xxxxxxxxxxx;tag=02-22581-7130a9fb-6f8a0a441

                Call-ID: hTnBJcrx44AvHW1SnBZVLKqHHxQ0otOO

                CSeq: 13359 ACK

                Content-Length:  0

 

 

                --end msg--

                16:00:26.967            pjsua_app.c  .....Call 0 is DISCONNECTED [reason=403 (not registered)]

                16:00:26.967     pjsua_app_common.c  .....

                  [DISCONNCTD] To: sip:0033661nnnnnn@xxxxxxxxxxx

                    Call time: 00h:00m:00s, 1st res in 25 ms, conn in 0ms

                16:00:26.967          pjsua_media.c  .....Call 0: deinitializing media..

                16:00:26.967          pjsua_media.c  ......Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0

                16:00:26.969           pjsua_core.c  .RX 549 bytes Response msg 401/REGISTER/cseq=20718 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 401 Unauthorized

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                CSeq: 20718 REGISTER

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=00-07686-08f654bc-4d18b7a46

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjwZ0eQlyjNoDdLg6UG-Osr8gZFdkQ05JL

                WWW-Authenticate: Digest realm="sip3.ovh.fr",nonce="08f64a24258ffbd7790be25e506a8b45",opaque="08f232dd5578250",stale=false,algorithm=MD5

                Server: Cirpack/v4.76 (gw_sip)

                Content-Length: 0

 

 

                --end msg--

                16:00:26.970           pjsua_core.c  ....TX 788 bytes Request msg REGISTER/cseq=20719 (tdta0x55ffcb151698) to UDP 91.121.129.159:5060:

                REGISTER sip:sip3.ovh.fr SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjUpk6GULmUTzkGVjYXebYbqOKZo3mjF83

                Max-Forwards: 70

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnnn@xxxxxxxxxxx>

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                CSeq: 20719 REGISTER

                User-Agent: PJSUA v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

                Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

                Expires: 300

                Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS

                Authorization: Digest username="0033972nnnnnnnn", realm="sip3.ovh.fr", nonce="08f64a24258ffbd7790be25e506a8b45", uri="sip:sip3.ovh.fr", response="1aa4dc5667db77e63ab8d15817a95474", algorithm=MD5, opaque="08f232dd5578250"

                Content-Length:  0

 

 

                --end msg--

                16:00:26.993           pjsua_core.c  .RX 341 bytes Response msg 100/REGISTER/cseq=20719 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 100 Trying

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                CSeq: 20719 REGISTER

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnn@xxxxxxxxxxx>

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjUpk6GULmUTzkGVjYXebYbqOKZo3mjF83

                Content-Length: 0

 

 

                --end msg--

                16:00:27.045           pjsua_core.c  .RX 514 bytes Response msg 200/REGISTER/cseq=20719 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 200 OK

                Call-ID: 4By1vMQV8lap2d6zrgKf33cHhz0Gd2vV

                Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>;expires=300

                CSeq: 20719 REGISTER

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=-WvjTZZ7327XIVuooUeqiwt8Jpsx1cMa

                To: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=00-07755-08f654c7-0470b8783

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjUpk6GULmUTzkGVjYXebYbqOKZo3mjF83

                P-Associated-URI: <sip:097nnnnnn@xxxxxxxxxxx>

                Server: Cirpack/v4.76 (gw_sip)

                Content-Length: 0

 

 

                --end msg--

                16:00:27.045            pjsua_acc.c  ....SIP outbound status for acc 1 is not active

                16:00:27.045            pjsua_acc.c  ....sip:0033972nnnnnn@sip3.ovh.fr: registration success, status=200 (OK), will re-register in 300 seconds

                16:00:27.045            pjsua_acc.c  ....Keep-alive timer started for acc 1, destination:91.121.129.159:5060:15, interval:0s

                16:00:27.942            pjsua_aud.c  Closing sound device after idle for 1 second(s)

                16:00:27.942            pjsua_app.c  .Turning sound device -99 -99 OFF

                16:00:27.942            pjsua_aud.c  .Closing null sound device..

 

 

I fact I wonder if pjsua takes care to register *before* calling the given URI.

 

So to make another test, once in pjsua I hit `m`, and enter `sip:0033661nnnnnn@xxxxxxxxxxx`, and I get the following result:

 

                16:03:31.129           pjsua_call.c !Making call with acc #1 to sip:0033661nnnnnn@xxxxxxxxxxx

                16:03:31.129            pjsua_aud.c  .Set sound device: capture=-99, playback=-99

                16:03:31.129            pjsua_aud.c  ..Setting null sound device..

                16:03:31.129            pjsua_app.c  ...Turning sound device -99 -99 ON

                16:03:31.129            pjsua_aud.c  ...Opening null sound device..

                16:03:31.129          pjsua_media.c  .Call 2: initializing media..

                16:03:31.129          pjsua_media.c  ..RTP socket reachable at 192.168.105.26:4002

                16:03:31.129          pjsua_media.c  ..RTCP socket reachable at 192.168.105.26:4003

                16:03:31.129          pjsua_media.c  ..Media index 0 selected for audio call 2

                16:03:31.130           pjsua_core.c  ....TX 1197 bytes Request msg INVITE/cseq=25951 (tdta0x55ffcb14daa8) to UDP 91.121.129.159:5060:

                INVITE sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: sip:0033661nnnnnn@xxxxxxxxxxx

                Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 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 v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

                Content-Type: application/sdp

                Content-Length:   520

 

                v=0

                o=- 3752492611 3752492611 IN IP4 192.168.105.26

                s=pjmedia

                b=AS:84

                t=0 0

                a=X-nat:0

                m=audio 4002 RTP/AVP 98 97 99 104 3 0 8 9 96

                c=IN IP4 192.168.105.26

                b=TIAS:64000

                a=rtcp:4003 IN IP4 192.168.105.26

                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 0-16

                a=ssrc:281817610 cname:6551173f0b0e5f0f

 

                --end msg--

                16:03:31.130            pjsua_app.c  .......Call 2 state changed to CALLING

                >>> 16:03:31.155           pjsua_core.c  .RX 339 bytes Response msg 100/INVITE/cseq=25951 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 100 Trying

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Content-Length: 0

 

 

                --end msg--

                16:03:31.155           pjsua_core.c  .RX 379 bytes Response msg 403/INVITE/cseq=25951 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 403 not registered

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>;tag=02-24827-7134b08a-07f4e44f1

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Content-Length: 0

 

 

                --end msg--

                16:03:31.155           pjsua_core.c  ..TX 377 bytes Request msg ACK/cseq=25951 (tdta0x7f414c00bab8) to UDP 91.121.129.159:5060:

                ACK sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: sip:0033661nnnnnn@xxxxxxxxxxx;tag=02-24827-7134b08a-07f4e44f1

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 ACK

                Content-Length:  0

 

 

                --end msg--

                16:03:31.155            pjsua_app.c  .....Call 2 is DISCONNECTED [reason=403 (not registered)]

                16:03:31.155     pjsua_app_common.c  .....

                  [DISCONNCTD] To: sip:0033661nnnnnn@xxxxxxxxxxx

                    Call time: 00h:00m:00s, 1st res in 26 ms, conn in 0ms

                16:03:31.155          pjsua_media.c  .....Call 2: deinitializing media..

                16:03:31.155          pjsua_media.c  ......Call 2: cleaning up provisional media, prov_med_cnt=1, med_cnt=0

                16:03:32.156            pjsua_aud.c  Closing sound device after idle for 1 second(s)

                16:03:32.156            pjsua_app.c  .Turning sound device -99 -99 OFF

                16:03:32.156            pjsua_aud.c  .Closing null sound device..

                               16:03:31.129           pjsua_call.c !Making call with acc #1 to sip:00336619nnnnnn@xxxxxxxxxxx

                16:03:31.129            pjsua_aud.c  .Set sound device: capture=-99, playback=-99

                16:03:31.129            pjsua_aud.c  ..Setting null sound device..

                16:03:31.129            pjsua_app.c  ...Turning sound device -99 -99 ON

                16:03:31.129            pjsua_aud.c  ...Opening null sound device..

                16:03:31.129          pjsua_media.c  .Call 2: initializing media..

                16:03:31.129          pjsua_media.c  ..RTP socket reachable at 192.168.105.26:4002

                16:03:31.129          pjsua_media.c  ..RTCP socket reachable at 192.168.105.26:4003

                16:03:31.129          pjsua_media.c  ..Media index 0 selected for audio call 2

                16:03:31.130           pjsua_core.c  ....TX 1197 bytes Request msg INVITE/cseq=25951 (tdta0x55ffcb14daa8) to UDP 91.121.129.159:5060:

                INVITE sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: sip:0033661nnnnnn@xxxxxxxxxxx

                Contact: <sip:0033972nnnnnn@192.168.105.26:5060;ob>

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 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 v2.8 Linux-4.15.0.39/x86_64/glibc-2.27

                Content-Type: application/sdp

                Content-Length:   520

 

                v=0

                o=- 3752492611 3752492611 IN IP4 192.168.105.26

                s=pjmedia

                b=AS:84

                t=0 0

                a=X-nat:0

                m=audio 4002 RTP/AVP 98 97 99 104 3 0 8 9 96

                c=IN IP4 192.168.105.26

                b=TIAS:64000

                a=rtcp:4003 IN IP4 192.168.105.26

                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 0-16

                a=ssrc:281817610 cname:6551173f0b0e5f0f

 

                --end msg--

                16:03:31.130            pjsua_app.c  .......Call 2 state changed to CALLING

                >>> 16:03:31.155           pjsua_core.c  .RX 339 bytes Response msg 100/INVITE/cseq=25951 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 100 Trying

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Content-Length: 0

 

 

                --end msg--

                16:03:31.155           pjsua_core.c  .RX 379 bytes Response msg 403/INVITE/cseq=25951 (rdata0x7f414c009ae8) from UDP 91.121.129.159:5060:

                SIP/2.0 403 not registered

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 INVITE

                From: <sip:0033972nnnnnn@xxxxxxxxxxx>;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: <sip:0033661nnnnnn@xxxxxxxxxxx>;tag=02-24827-7134b08a-07f4e44f1

                Via: SIP/2.0/UDP 192.168.105.26:5060;received=192.168.105.26;rport=5060;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Content-Length: 0

 

 

                --end msg--

                16:03:31.155           pjsua_core.c  ..TX 377 bytes Request msg ACK/cseq=25951 (tdta0x7f414c00bab8) to UDP 91.121.129.159:5060:

                ACK sip:0033661nnnnnn@xxxxxxxxxxx SIP/2.0

                Via: SIP/2.0/UDP 192.168.105.26:5060;rport;branch=z9hG4bKPjnPUSaSErQ5cB3NwXli7DykVXR4Yf.qa8

                Max-Forwards: 70

                From: sip:0033972nnnnnn@xxxxxxxxxxx;tag=V56CnDzA6IA17NcOZzndfBnUeGMyLDvv

                To: sip:0033661nnnnnn@xxxxxxxxxxx;tag=02-24827-7134b08a-07f4e44f1

                Call-ID: Xl0e9hCHf5QIF7lhEajqnDIWLVHt8ECw

                CSeq: 25951 ACK

                Content-Length:  0

 

 

                --end msg--

                16:03:31.155            pjsua_app.c  .....Call 2 is DISCONNECTED [reason=403 (not registered)]

                16:03:31.155     pjsua_app_common.c  .....

                  [DISCONNCTD] To: sip:0033661nnnnnn@xxxxxxxxxxx

                    Call time: 00h:00m:00s, 1st res in 26 ms, conn in 0ms

                16:03:31.155          pjsua_media.c  .....Call 2: deinitializing media..

                16:03:31.155          pjsua_media.c  ......Call 2: cleaning up provisional media, prov_med_cnt=1, med_cnt=0

                16:03:32.156            pjsua_aud.c  Closing sound device after idle for 1 second(s)

                16:03:32.156            pjsua_app.c  .Turning sound device -99 -99 OFF

                16:03:32.156            pjsua_aud.c  .Closing null sound device..

 

               

What is this `Call 2 is DISCONNECTED [reason=403 (not registered)]` message?

Could you tell me what's wrong?

 

 

 

 

Regards ,

Alain

 

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