Closed GoogleCodeExporter closed 9 years ago
Log snips...
D/PjService( 3064): Found pjsua 2 searching 2
D/RegHandlerReceiver( 3064): We restore
<sip:darryl@25.87.105.246:44461;transport=TCP;ob>
D/libpjsip( 3064): 10:17:44.722 mobile_reg_han ....The register already has
old contact in it, ignore
D/libpjsip( 3064): 10:17:44.722 pjsua_core.c ....TX 936 bytes Request msg
REGISTER/cseq=60670 (tdta0x2a458490) to TCP 173.255.213.166:7770:
D/libpjsip( 3064): REGISTER sip:173.255.213.166:7770;transport=tcp;lr SIP/2.0
D/libpjsip( 3064): Via: SIP/2.0/TCP
25.87.105.246:38134;rport;branch=z9hG4bKPj7ODfwFUVLMnzvfEZfaFiUyEXON2nkjwM;alias
D/libpjsip( 3064): Max-Forwards: 70
D/libpjsip( 3064): From:
<sip:darryl@173.255.213.166>;tag=Ok5z.DW7KF-y4nhsDuhGER5eOrS6XY7T
D/libpjsip( 3064): To: <sip:darryl@173.255.213.166>
D/libpjsip( 3064): Call-ID: gnDiw6r94gbgrG9x3BX9PE.fO7XrRll3
D/libpjsip( 3064): CSeq: 60670 REGISTER
D/libpjsip( 3064): User-Agent: CSipSimple_mako-19/r2330
D/libpjsip( 3064): Supported: outbound, path
D/libpjsip( 3064): Contact: <sip:darryl@24.114.39.46:55260;transport=TCP;ob>
D/libpjsip( 3064): Contact:
<sip:darryl@25.87.105.246:44461;transport=TCP;ob>;expires=0;reg-id=1;+sip.instan
ce="<urn:uuid:00000000-0000-0000-0000-0000e922f243>"
D/libpjsip( 3064): Expires: 900
D/libpjsip( 3064): Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO,
SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
D/libpjsip( 3064): Authorization: Digest username="darryl", realm="asterisk",
nonce="232eab6b", uri="sip:173.255.213.166:7770;transport=tcp;lr",
response="2c17f5198b060817570bfa02f0b8f58b", algorithm=MD5
D/libpjsip( 3064): Content-Length: 0
D/libpjsip( 3064):
D/libpjsip( 3064): --end msg--
D/libpjsip( 3064): 10:17:44.843 pjsua_core.c .RX 613 bytes Response msg
200/REGISTER/cseq=60670 (rdata0x2a45d11c) from TCP 173.255.213.166:7770:
D/libpjsip( 3064): SIP/2.0 200 OK
D/libpjsip( 3064): Via: SIP/2.0/TCP
25.87.105.246:38134;branch=z9hG4bKPj7ODfwFUVLMnzvfEZfaFiUyEXON2nkjwM;alias;recei
ved=24.114.39.46;rport=55260
D/libpjsip( 3064): From:
<sip:darryl@173.255.213.166>;tag=Ok5z.DW7KF-y4nhsDuhGER5eOrS6XY7T
D/libpjsip( 3064): To: <sip:darryl@173.255.213.166>;tag=as6a0b3b6a
D/libpjsip( 3064): Call-ID: gnDiw6r94gbgrG9x3BX9PE.fO7XrRll3
D/libpjsip( 3064): CSeq: 60670 REGISTER
D/libpjsip( 3064): Server: Asterisk PBX 11.6.0
D/libpjsip( 3064): Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY, INFO, PUBLISH
D/libpjsip( 3064): Supported: replaces, timer
D/libpjsip( 3064): Expires: 900
D/libpjsip( 3064): Contact:
<sip:darryl@24.114.39.46:55260;transport=TCP;ob>;expires=900
D/libpjsip( 3064): Date: Sun, 12 Jan 2014 18:17:44 GMT
D/libpjsip( 3064): Content-Length: 0
D/libpjsip( 3064):
D/libpjsip( 3064): --end msg--
D/libpjsip( 3064): 10:17:44.843 mobile_reg_han .mod_reg_tracker_on_rx_response
D/libpjsip( 3064): 10:17:44.843 mobile_reg_han .mod_reg_tracker_on_rx_response
2
D/libpjsip( 3064): 10:17:44.843 mobile_reg_han .Hook a REGISTER RX response !!!
D/libpjsip( 3064): 10:17:44.843 mobile_reg_han . Hook should save contact :
<sip:darryl@24.114.39.46:55260;transport=TCP;ob> > 900
D/PjService( 3064): Found pjsua 2 searching 2
D/libpjsip( 3064): 10:17:44.867 mobile_reg_han .mod_reg_tracker_on_rx_response
done
D/libpjsip( 3064): 10:17:44.881 pjsua_acc.c ....SIP outbound status for acc
2 is not active
I/libpjsip( 3064): 10:17:44.881 pjsua_acc.c ....
<sip:darryl@173.255.213.166>: registration success, status=200 (OK), will
re-register in 900 seconds
D/libpjsip( 3064): 10:17:44.881 pjsua_pres.c ....Starting MWI subscription..
D/libpjsip( 3064): 10:17:44.884 pjsua_core.c .......TX 628 bytes Request msg
SUBSCRIBE/cseq=31457 (tdta0x2a1e4db8) to UDP 173.255.213.166:5060:
D/libpjsip( 3064): SUBSCRIBE sip:darryl@173.255.213.166 SIP/2.0
D/libpjsip( 3064): Via: SIP/2.0/UDP
25.87.105.246:34594;rport;branch=z9hG4bKPjQARxigp1UceHV3IPgpyZYkiRFX5qRDJW
D/libpjsip( 3064): Max-Forwards: 70
D/libpjsip( 3064): From:
<sip:darryl@173.255.213.166>;tag=T7QMXiQx9IWliTIj21dmqyUOW6F.Csex
D/libpjsip( 3064): To: <sip:darryl@173.255.213.166>
D/libpjsip( 3064): Contact: <sip:darryl@24.114.39.46:55260;transport=TCP;ob>
D/libpjsip( 3064): Call-ID: fa-zFlbZIStuQtIOQq1iAt77J1wP8fCe
D/libpjsip( 3064): CSeq: 31457 SUBSCRIBE
D/libpjsip( 3064): Event: message-summary
D/libpjsip( 3064): Expires: 3600
D/libpjsip( 3064): Supported: replaces, 100rel, timer, norefersub
D/libpjsip( 3064): Accept: application/simple-message-summary
D/libpjsip( 3064): Allow-Events: presence, message-summary, refer
D/libpjsip( 3064): User-Agent: CSipSimple_mako-19/r2330
D/libpjsip( 3064): Content-Length: 0
D/libpjsip( 3064):
D/libpjsip( 3064): --end msg--
D/libpjsip( 3064): 10:17:44.899 evsub0x2a1b7a0 .........Subscription state
changed NULL --> SENT
D/libpjsip( 3064): 10:17:44.899 pjsua_pres.c ..........MWI subscription for
<sip:darryl@173.255.213.166> is SENT
D/SIP UA Receiver( 3064): < LOCK CPU
D/SIP UA Receiver( 3064): > UNLOCK CPU 0
D/PjService( 3064): Found pjsua 2 searching 2
D/PjService( 3064): Update profile from service for 2 aka in db 1
D/DBProvider( 3064): Updated status_text=OK status_code=200 display_name=mb
expires=894 account_id=1 added_status=0 priority=100 active=true wizard=EXPERT
reg_uri=sip:173.255.213.166:7770 pjsua_id=2
D/AccountChooserButton( 5878): Accounts status.onChange( false)
D/PjService( 3064): Profile state UP : status_text=OK status_code=200
expires=894
D/SIP SRV ( 3064): Accounts status.onChange( false)
D/SIP SRV ( 3064): Update registration state
D/libpjsip( 3064): 10:31:16.845 evsub0x2a1a805 ...Subscription state changed
SENT --> TERMINATED
D/libpjsip( 3064): 10:31:16.845 pjsua_pres.c ....MWI subscription for
<sip:darryl@173.255.213.166> is TERMINATED
D/libpjsip( 3064): 10:31:16.845 evsub0x2a1a805 ...Subscription destroyed
D/libpjsip( 3064): 10:31:21.788 pjsua_core.c .TX 985 bytes Request msg
INVITE/cseq=8395 (tdta0x2a4231a0) to UDP 173.255.213.166:5060:
D/libpjsip( 3064): INVITE sip:6043679618@173.255.213.166 SIP/2.0
D/libpjsip( 3064): Via: SIP/2.0/UDP
25.87.105.246:52097;rport;branch=z9hG4bKPj9eFHz0LNOW1LoI0GMqV9p-FkcYmf146f
D/libpjsip( 3064): Max-Forwards: 70
D/libpjsip( 3064): From:
<sip:darryl@173.255.213.166>;tag=g3j7vbtpYtoJgSUua6Lkd5Vn49TKKS6F
D/libpjsip( 3064): To: <sip:6043679618@173.255.213.166>
D/libpjsip( 3064): Contact: <sip:darryl@24.114.39.46:55278;transport=TCP;ob>
D/libpjsip( 3064): Call-ID: OMIY8MtHWFJymCeJ8HsKGBxc0qbSjYct
D/libpjsip( 3064): CSeq: 8395 INVITE
D/libpjsip( 3064): Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO,
SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
D/libpjsip( 3064): Supported: replaces, 100rel, timer, norefersub
D/libpjsip( 3064): Session-Expires: 1800
D/libpjsip( 3064): Min-SE: 90
D/libpjsip( 3064): User-Agent: CSipSimple_mako-19/r2330
D/libpjsip( 3064): Content-Type: application/sdp
D/libpjsip( 3064): Content-Length: 320
D/libpjsip( 3064):
D/libpjsip( 3064): v=0
D/libpjsip( 3064): o=- 3598540250 3598540250 IN IP4 25.87.105.246
D/libpjsip( 3064): s=pjmedia
D/libpjsip( 3064): c=IN IP4 25.87.105.246
D/libpjsip( 3064): t=0 0
D/libpjsip( 3064): m=audio 10000 RTP/AVP 3 96 101
D/libpjsip( 3064): c=IN IP4 25.87.105.246
D/libpjsip( 3064): a=rtcp:10001 IN IP4 25.87.105.246
D/libpjsip( 3064): a=sendrecv
D/libpjsip( 3064): a=rtpmap:3 GSM/8000
D/libpjsip( 3064): a=rtpmap:96 SILK/8000
D/libpjsip( 3064): a=fmtp:96 useinbandfec=0
D/libpjsip( 3064): a=rtpmap:101 telephone-event/8000
D/libpjsip( 3064): a=fmtp:101 0-15
D/libpjsip( 3064): --end msg--
D/SIP UA Receiver( 3064): < LOCK CPU
D/SIP UA Receiver( 3064): Call TSX state <<
D/SIP UA Receiver( 3064): Updating call infos from the stack
D/PjSipCalls( 3064): Update call 0
D/PjService( 3064): Found pjsua 2 searching 2
D/PjSipCalls( 3064): Last status code is 408
D/SIP UA Receiver( 3064): Call TSX state >>
D/SIP UA Receiver( 3064): > UNLOCK CPU 0
I/libpjsip( 3064): 10:31:22.254 pjsua_jni_addo ....Call 0 is DISCONNECTED
[reason=408 (Request Timeout)]
Original comment by darryl....@gmail.com
on 12 Jan 2014 at 6:45
Thanks for the report
Can you send logs using the HowToCollectLogs wiki page instructions?
Also when you reproduce, do not forget to include the previous call scenario.
It will include the remote from send by the other.
What you observe can be totally expected if remote send sip uri without port
and/or if you didn't configured your account with the "proxy field" set.
Indeed, the account should *not* use the registration port as you said in your
description. This port, as it's name suggest is for registration. For calls, if
proxy is set, it will send on your sip proxy configured, else it will resolved
based on the sip uri it has to call. This applies to domain, port and
transport. This is the way any sip client should behaves if it allows to
address all possible topologies.
Note that I recommend to use proxy value always set and that what's done by the
wizard "basic" designed for mainstream users.
Original comment by r3gis...@gmail.com
on 12 Jan 2014 at 8:09
Wow, thank you for the fast response and you're completely correct. My concern
was based on the sip invite being sent to port 5060, but as you said this is by
design per the sip uri for maximum flexibility.
I had originally configured csip via the Advanced wizard to specify the TCP
preference and using the proxy setting had not even occurred to me. Your advice
has resolved my use case issue.
There's always so much to learn and I appreciate your time. Thank you and keep
up the great work!
Original comment by darryl....@gmail.com
on 12 Jan 2014 at 8:42
I think I have the same problem outlined here. All the numbers in my call
history are recorded as number@server and they don't go through when I try to
re-dial.
What exactly should I do? I use Anveo, do I need put sip.anveo.com in the proxy
server?Should I add a port number?
Many thanks!
Original comment by the...@gmail.com
on 16 Nov 2014 at 9:41
Original issue reported on code.google.com by
darryl....@gmail.com
on 12 Jan 2014 at 6:23