我尝试使用pjsua（PJLIB 2.8）通过VoIP服务提供商（OVH）向我的手机打电话。
我可以成功注册它，但是当我尝试拨打电话时，出现403 not registered错误，我不明白这是什么意思。
这是完整的日志：
Account list:
[ 0] <sip:192.168.105.22:5060>: does not register
Online status: Online
[ 1] <sip:192.168.105.22:5060;transport=TCP>: does not register
Online status: Online
*[ 2] sip:0033972nnnnnn@sip3.ovh.fr: 200/OK (expires=292)
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
>>> 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@sip3.ovh.fr
16:25:35.727 pjsua_call.c !Making call with acc #2 to sip:0033661nnnnnn@sip3.ovh.fr
16:25:35.728 pjsua_aud.c .Set sound device: capture=-99, playback=-99
16:25:35.728 pjsua_aud.c ..Setting null sound device..
16:25:35.728 pjsua_app.c ...Turning sound device -99 -99 ON
16:25:35.728 pjsua_aud.c ...Opening null sound device..
16:25:35.728 pjsua_media.c .Call 0: initializing media..
16:25:35.728 pjsua_media.c ..RTP socket reachable at 192.168.105.22:4000
16:25:35.728 pjsua_media.c ..RTCP socket reachable at 192.168.105.22:4001
16:25:35.728 pjsua_media.c ..Media index 0 selected for audio call 0
16:25:35.729 pjsua_core.c ....TX 1197 bytes Request msg INVITE/cseq=27460 (tdta0x147b0a8) to UDP 91.121.129.159:5060:
INVITE sip:0033661nnnnnn@sip3.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.105.22:5060;rport;branch=z9hG4bKPjoKRxGB-lH36JSYhcyTLZiKj8ig24yc8R
Max-Forwards: 70
From: sip:0033972nnnnnn@sip3.ovh.fr;tag=NOWJhAuXqRLgtG5r8zWl29F9N-.cZ4mO
To: sip:0033661nnnnnn@sip3.ovh.fr
Contact: <sip:0033972nnnnnn@192.168.105.22:5060;ob>
Call-ID: ScoXEVAN0wWg5r-iM7PHr3jAEkimxnS4
CSeq: 27460 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-2.23
Content-Type: application/sdp
Content-Length: 520
v=0
o=- 3752580335 3752580335 IN IP4 192.168.105.22
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 192.168.105.22
b=TIAS:64000
a=rtcp:4001 IN IP4 192.168.105.22
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:874369552 cname:19f7007e0c6f13ba
--end msg--
16:25:35.729 pjsua_app.c .......Call 0 state changed to CALLING
>>> 16:25:35.752 pjsua_core.c .RX 339 bytes Response msg 100/INVITE/cseq=27460 (rdata0x7f4f68005fb8) from UDP 91.121.129.159:5060:
SIP/2.0 100 Trying
Call-ID: ScoXEVAN0wWg5r-iM7PHr3jAEkimxnS4
CSeq: 27460 INVITE
From: <sip:0033972nnnnnn@sip3.ovh.fr>;tag=NOWJhAuXqRLgtG5r8zWl29F9N-.cZ4mO
To: <sip:0033661nnnnnn@sip3.ovh.fr>
Via: SIP/2.0/UDP 192.168.105.22:5060;received=192.168.105.22;rport=5060;branch=z9hG4bKPjoKRxGB-lH36JSYhcyTLZiKj8ig24yc8R
Content-Length: 0
--end msg--
16:25:35.752 pjsua_core.c .RX 379 bytes Response msg 403/INVITE/cseq=27460 (rdata0x7f4f68005fb8) from UDP 91.121.129.159:5060:
SIP/2.0 403 not registered
Call-ID: ScoXEVAN0wWg5r-iM7PHr3jAEkimxnS4
CSeq: 27460 INVITE
From: <sip:0033972nnnnnn@sip3.ovh.fr>;tag=NOWJhAuXqRLgtG5r8zWl29F9N-.cZ4mO
To: <sip:0033661nnnnnn@sip3.ovh.fr>;tag=02-22412-78589635-1a8786244
Via: SIP/2.0/UDP 192.168.105.22:5060;received=192.168.105.22;rport=5060;branch=z9hG4bKPjoKRxGB-lH36JSYhcyTLZiKj8ig24yc8R
Content-Length: 0
--end msg--
16:25:35.752 pjsua_core.c ..TX 377 bytes Request msg ACK/cseq=27460 (tdta0x7f4f68007f88) to UDP 91.121.129.159:5060:
ACK sip:0033661nnnnnn@sip3.ovh.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.105.22:5060;rport;branch=z9hG4bKPjoKRxGB-lH36JSYhcyTLZiKj8ig24yc8R
Max-Forwards: 70
From: sip:0033972nnnnnn@sip3.ovh.fr;tag=NOWJhAuXqRLgtG5r8zWl29F9N-.cZ4mO
To: sip:0033661nnnnnn@sip3.ovh.fr;tag=02-22412-78589635-1a8786244
Call-ID: ScoXEVAN0wWg5r-iM7PHr3jAEkimxnS4
CSeq: 27460 ACK
Content-Length: 0
--end msg--
16:25:35.752 pjsua_app.c .....Call 0 is DISCONNECTED [reason=403 (not registered)]
16:25:35.752 pjsua_app_common.c .....
[DISCONNCTD] To: sip:0033661nnnnnn@sip3.ovh.fr
Call time: 00h:00m:00s, 1st res in 24 ms, conn in 0ms
16:25:35.752 pjsua_media.c .....Call 0: deinitializing media..
16:25:35.752 pjsua_media.c ......Call 0: cleaning up provisional media, prov_med_cnt=1, med_cnt=0
16:25:36.756 pjsua_aud.c Closing sound device after idle for 1 second(s)
16:25:36.756 pjsua_app.c .Turning sound device -99 -99 OFF
16:25:36.756 pjsua_aud.c .Closing null sound device..
我在OVH论坛2013存档中看到了一些有关SIP兼容性问题的类似问题，但我认为在2018年不再如此。...
我也看到了非常相似的post，但不幸的是它没有得到回答。
有什么主意吗？