XI Channel breaks SXI_CACHE
Hi experts,
We are working in the following scenario:
ERP 5.0 -> XI 3.0 (NW04) -> NW04s
We have created two communication channels, one Idoc channel (to communicate ERP and XI) and one XI channel (to communicate XI and NW04s)
After creating XI channel the cache on XI ABAP gets error (SXI_CACHE transaction shows UPDATE error), this is the error:
"Incorrect XML display for channel , AII, XI_Channel
Error in secure store
Error when writing to secure store: ERROR_UNKNOWN: Internal error in secure store (SECSTORE,021)"
We can´t see where is the error, we are stuck on this, please can anybody help us??? Thanks a lot.
Best regards
Hi Xavier,
I was reading the NW04s master guide, and it mentions in several places that the XI (or rather PI) component should be the component in your landscape with teh highest release. So your combination ERP 5.0 -> XI 3.0 -> NW04s may not work for this reason.
Hope this helps/Farshad
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Created a simple file to file scenario. While testing the scenario, i've got the following error while looking at the SXMB_MONI.
"com.sap.aii.af.ra.ms.api.ConfigException: ConfigException in XI protocol handler. Failed to get a receiver channel from the agreement for the given message. Root cause: com.sap.aii.af.service.cpa.impl.exception.CPALookupException: Channel object with Id Channel:ca8f9060afd235e7b77958ffc1bb2096 not available in CPA Cache"
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ThanksHi Jason,
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Hi,
The JMS sender communication channel throws the following error in Communication Channel monitoring in RWB.
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AmbiliHi Abhishek,
Thanks for ur reply.
1. Already tried it, but still the same status.
2. Yes, i can see the channel in sxi_cache.
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The issue is just for this channel and as I mentioned before, a copy object of the same channel works perfectly fine.
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Ambili
Edited by: Ambili Jose on Jan 19, 2010 7:30 AM -
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Apr 13 13:02:58 SPA112 daemon.notice system[100]: NTP update successfully, Year:4,Month:13,Day:13,Hour:2,Min:58,Sec:-1344375504
Apr 13 13:02:58 SPA112 syslog.notice syslog-ng[118]: STATS: dropped 0
Apr 13 13:03:04 SPA112 kern.info [17179593.752000] cordless: loading synergy-2011-09-15
Apr 13 13:03:04 SPA112 kern.info [17179593.784000] cordless: init successful
Apr 13 13:03:06 SPA112 user.notice msgswitchd: MSGSWITCH fd_rtp fifo created 11
Apr 13 13:03:06 SPA112 user.notice msgswitchd: MSGSWITCH fd_ch fifo created 13
Apr 13 13:03:06 SPA112 user.notice msgswitchd: MSGSWITCH fd_gmep fifo created 14
Apr 13 13:03:07 SPA112 daemon.notice msgswitchd[192]: new ap 00000001 (AP_SIP) at pid 00179
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:08:32 SPA112 kern.warning [17179593.784000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.780000] ###### sock_sendmsg return 172
Apr 13 13:08:34 SPA112 kern.warning [17179595.780000]
Apr 13 13:08:34 SPA112 kern.warning [17179595.780000] #### RTP STOP Flag set in this channel break ####
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:08:34 SPA112 kern.warning [17179595.788000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:08:54 SPA112 daemon.notice msgswitchd[192]: MSGSWD RTCP Reqt len 12 Data 2,2434896,7304,0
Apr 13 13:08:54 SPA112 kern.warning [17179615.780000] RTCP is running so calling rtcp stop
Apr 13 13:08:54 SPA112 kern.warning [17179615.780000] chan->kmode is present not null
Apr 13 13:08:54 SPA112 kern.warning [17179615.784000] ###### RTCP sock_sendmsg return 172
Apr 13 13:08:54 SPA112 kern.warning [17179615.788000] ###### sock_sendmsg return 172
Apr 13 13:08:54 SPA112 kern.warning [17179615.788000]
Apr 13 13:08:54 SPA112 kern.warning [17179615.788000] #### RTP STOP Flag set in this channel break ####
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:09:41 SPA112 kern.warning [17179662.584000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.228000] ###### sock_sendmsg return 172
Apr 13 13:09:43 SPA112 kern.warning [17179664.232000]
Apr 13 13:09:43 SPA112 kern.warning [17179664.232000] #### RTP STOP Flag set in this channel break ####
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:09:43 SPA112 kern.warning [17179664.236000] In cordless Driver Codec 255 and str chan 0
Apr 13 13:10:02 SPA112 daemon.notice msgswitchd[192]: MSGSWD RTCP Reqt len 12 Data 2,2435044,7304,0
Apr 13 13:10:03 SPA112 kern.warning [17179684.260000] RTCP is running so calling rtcp stop
Apr 13 13:10:03 SPA112 kern.warning [17179684.260000] chan->kmode is present not null
Apr 13 13:10:03 SPA112 kern.warning [17179684.264000] ###### RTCP sock_sendmsg return 172
Apr 13 13:10:03 SPA112 kern.warning [17179684.268000] ###### sock_sendmsg return 172
Apr 13 13:10:03 SPA112 kern.warning [17179684.268000]
Apr 13 13:10:03 SPA112 kern.warning [17179684.268000] #### RTP STOP Flag set in this channel break ####
Apr 13 13:10:47 SPA112 daemon.notice msgswitchd[192]: MSGSWD RTCP Reqt len 12 Data 2,2635368,0,0
Apr 13 13:11:46 SPA112 kern.info [17179787.236000] cordless: deinit
Apr 13 13:11:49 SPA112 kern.info [17179790.416000] cordless: loading synergy-2011-09-15
Apr 13 13:11:49 SPA112 kern.info [17179790.456000] cordless: init successful
Apr 13 13:11:51 SPA112 user.notice msgswitchd: MSGSWITCH fd_rtp fifo created 11
Apr 13 13:11:51 SPA112 user.notice msgswitchd: MSGSWITCH fd_ch fifo created 13
Apr 13 13:11:51 SPA112 user.notice msgswitchd: MSGSWITCH fd_gmep fifo created 14
Apr 13 13:11:52 SPA112 daemon.notice msgswitchd[308]: new ap 00000001 (AP_SIP) at pid 00287
Apr 13 13:11:53 SPA112 daemon.notice msgswitchd[308]: MSGSWD RTCP Reqt len 12 Data 2,2362808,0,287
Apr 13 13:12:58 SPA112 syslog.notice syslog-ng[118]: STATS: dropped 0
Apr 13 13:22:58 SPA112 syslog.notice syslog-ng[118]: STATS: dropped 0
Apr 13 13:32:59 SPA112 syslog.notice syslog-ng[118]: STATS: dropped 0
©Hi Nseto,
Thank you for the advise. I've attached the debug log as requested. Let me know if there's something that stands out to you.
syslog server(port:514) started on Tue Apr 16 11:23:23 2013
Firmware downgrade limit()
httpd_handle_request(), request method = 1
httpd_handle_request(), request path = /admin/voice/
httpd_handle_request(), pswlReq->ubType = 0
Requesting call statistics...
Call statistics updated.
httpd_handle_request(), request method = 2
*** show dtmf tx holdoff time 70 for line 0
*** show dtmf tx holdoff time 70 for line 1
get_dhcp_option66_67_info, voice interface is 'br0'
uch_syncParameter start
uchInitDTMFTbl(), dtmf level -160
uchEnableEchoCan(), lid 0 EP 2 enable
UCH sync parameter hold off time is 70
uch_syncParameter(), uch_syncDTMFHoldOffTime(0)=0
uchEnableEchoCan(), lid 1 EP 1 enable
UCH sync parameter hold off time is 70
uch_syncParameter(), uch_syncDTMFHoldOffTime(1)=0
Firmware downgrade limit()
httpd_handle_request(), request method = 1
httpd_handle_request(), request path = /admin/voice/
httpd_handle_request(), pswlReq->ubType = 0
Requesting call statistics...
Call statistics updated.
[0]<<31.6.78.124:5060(499)
[0]<<31.6.78.124:5060(499)
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK100c9cd0;rport
From: "Unknown" ;tag=as0871a40b
To:
Contact:
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 16 Apr 2013 01:27:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[0]->31.6.78.124:5060(400)
[0]->31.6.78.124:5060(400)
SIP/2.0 200 OK
To: ;tag=e86c9e9a830a9c1ei0
From: "Unknown" ;tag=as0871a40b
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK100c9cd0
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[0]<<31.6.78.124:5060(499)
[0]<<31.6.78.124:5060(499)
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK6f447a66;rport
From: "Unknown" ;tag=as6203a281
To:
Contact:
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 16 Apr 2013 01:28:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[0]->31.6.78.124:5060(400)
[0]->31.6.78.124:5060(400)
SIP/2.0 200 OK
To: ;tag=e86c9e9a830a9c1ei0
From: "Unknown" ;tag=as6203a281
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK6f447a66
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[0]<<31.6.78.124:5060(849)
[0]<<31.6.78.124:5060(849)
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc;rport
From: "Frankston South, Australia" ;tag=as5e956228
To:
Contact:
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 16 Apr 2013 01:28:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 2809 2809 IN IP4 31.6.78.124
s=session
c=IN IP4 31.6.78.124
t=0 0
m=audio 10582 RTP/AVP 97 0 8 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[0]->31.6.78.124:5060(312)
[0]->31.6.78.124:5060(312)
SIP/2.0 100 Trying
To:
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
CC_eventProc(), event: CC_EV_SIG_CALL_ARRIVED(0x2F), lid: 0, par: 0, par2: 0x40324a28
AUD_ccEventProc: event 47 vid 0 par 0x0 par2 0x40324a28
pconly: 0
CC:pc(0)=130 not in codec list
CC:pc(1)=18 not in codec list
clRemote: 0x40324ab8, clLocal->ucNumAudioCodec: 2
[AUD]Get UCH node for AUD_LINE 0.
uchAllocateNode(), Node 0 allocated
[AUD]UCH node 0 allocated to AUD_LINE 0.
uchConnectEpToNode(), connecting EP FXS 1 to node 0
uchEnableNode(), Node 0 enbaled
CC_eventProc(), inf.strName = Frankston South, Australia
CC_eventProc(), inf.strPhone = 6004
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c652c, event=18(CC_EV_USR_ACCEPTCALL), par=0, par2=0x40324a28)
cepIdleProc(), lid=0
cepIdleProc(), line->sigProc(CC_CMD_ACCEPT)
cepIdleProc(), call->cinf.bAutoAnswer = 0
NEW_CALL_STATE(), call 0: old state = 0, new state 5
CC_eventProc(), msg CC_EV_USR_ACCEPTCALL(92) sent to CC
[0]CID:CID_initGen() >>> offhook 0 delay 2200 phone 6004 name Frankston South, Australia
SLIC_startRing state 0 ts 0x1ed790on 2000 off 4000 len 60000
[0]Ring cad event 0 pol 0
RTP_nextMediaPort(), port = 16392
RTP_nextMediaPort(), rc=16390
AUD_allocCallObj() call(0x1ef7c0)
[0:0]AUD ALLOC CALL (port=16390)
[AUD]AUD_startRtpRx(0x1ef7c0)
Local loopback mode: None. Type: None.
Remote loopback mode: None. Type None.
UCH sync parameter hold off time is 70
RTP channel setup: udp_no_checksum 0, sysmmetric_rtp 0, tos 0xb8, cos 6, mlb 0.
uchConnectEpToNode(), connecting EP VoIP 0 to node 0
cordless_start_rtp(), chan:0 remote ip:(null) port:0 local:16390 rx:1 ipt:0 ptime:0
Starting Rx only RTP.
Socket 19 bound to port 16390.
Remote IP/port: 0.0.0.0:0
Codec list from SDP (internal pt):Codec list from SDP (internal pt): 0 0 8 8 134 134 136 136
Rx payload list: Rx payload list: PCMU/8000(0) PCMU/8000(0) PCMA/8000(8) PCMA/8000(8) NSE/8000(100) NSE/8000(100) encaprtp/8000(112) encaprtp/8000(112)
set RTP_SESSION_OPT_DTMF
uchEnableEchoCan(), lid 0 EP 2 enable
RTP configuration:
audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 0, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 PCMU/8000, rx[1] 8 PCMA/8000, rx[2] 100 NSE/8000
rx[3] 112 encaprtp/8000, rx[4] -1 , rx[5] -1
Jib: max 180ms, min 60ms, adapt 1
RTP Channel 0 is virgin: 1.
RTP session 0 started
uchSetDTMFMute(), ENABLE
[AUD]RTP Rx Up
[0]->31.6.78.124:5060(400)
[0]->31.6.78.124:5060(400)
SIP/2.0 180 Ringing
To: ;tag=93d9d48c852dff3ei0
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc
Contact: "Brahma Kumari Canberra"
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Set QoS succeed
[0]Ring cad event 1 pol 0
CID:OnHookTx Pol
[0]CID CID_ST_POLREV_POST_DELAY
uchDisplayCIDFSK(), EP 2 lid 0 buflen 107 overhead 60 SZ_MAX_USERDATA 200
[0]CID Start DTMF/FSK, CID_ST_ACTIVE
[0]CID CID_ST_FSK_COMP_DELAY
[0]CID CID:DONE
[0]CID CID_ST_ACTIVE_POST_DELAY
[0]CID CID_ST_IDLE
UCH sync parameter hold off time is 70
[0]Ring cad event 0 pol 0
[0]Ring cad event 1 pol 0
[0]Off Hook
CC_eventProc(), event: CC_EV_USR_OFFHOOK(0x2), lid: 0, par: 0, par2: (nil)
AUD_ccEventProc: event 2 vid 0 par 0x0 par2 0x0
sysstatus_set_led_status_payton(), led_id: 1, statusCode:7, systemEvent: 0x100095
callEventProcTable[5] is cepRingingProc
NEW_CALL_STATE(), call 0: old state = 5, new state 7
SLIC_stopRing
[0]Ring cad event 2 pol 0
SLIC_stopRing
SLIC_stopTone
[0]->31.6.78.124:5060(763)
[0]->31.6.78.124:5060(763)
SIP/2.0 200 OK
To: ;tag=93d9d48c852dff3ei0
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc
Contact: "Brahma Kumari Canberra"
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 251
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 96219 96219 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(763)
[0]->31.6.78.124:5060(763)
SIP/2.0 200 OK
To: ;tag=93d9d48c852dff3ei0
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc
Contact: "Brahma Kumari Canberra"
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 251
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 96219 96219 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(763)
[0]->31.6.78.124:5060(763)
SIP/2.0 200 OK
To: ;tag=93d9d48c852dff3ei0
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK276264bc
Contact: "Brahma Kumari Canberra"
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 251
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 96219 96219 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16390 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]<<31.6.78.124:5060(401)
[0]<<31.6.78.124:5060(401)
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK683598f6;rport
From: "Frankston South, Australia" ;tag=as5e956228
To: ;tag=93d9d48c852dff3ei0
Contact:
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
SIP_sessDlgEventProc: event: 42, ucState: 0
CC_eventProc(), event: CC_EV_SIG_CALL_CONNECTED(0x2A), lid: 0, par: 2, par2: (nil)
AUD_ccEventProc: event 42 vid 0 par 0x2 par2 0x0
callEventProcTable[7] is cepAnsweringProc
cepAnsweringProc(line=0x1c6528, call=0x1c652c, event=42(CC_EV_SIG_CALL_CONNECTED), par=2, par2=(nil))
CC:Connected
NEW_CALL_STATE(), call 0: old state = 7, new state 8
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
[AUD]AUD_startRtpTx(0x1ef7c0, 0, 31.6.78.124, 10582, 20)
Local loopback mode: None. Type: None.
Remote loopback mode: None. Type None.
UCH sync parameter hold off time is 70
Already has a RTP channel.
Already has a RTP channel.
uchConnectEpToNode(), connecting EP VoIP 0 to node 0
cordless_start_rtp(), chan:0 remote ip:31.6.78.124 port:10582 local:16390 rx:0 ipt:0 ptime:20
Going from Rx only to bi-directional.
Old remote IP/port: 0.0.0.0:0
Remote IP/port: 31.6.78.124:10582
Codec list from SDP (internal pt):Codec list from SDP (internal pt): 0 0 8 8 134 134 136 136
Rx payload list: Rx payload list: PCMU/8000(0) PCMU/8000(0) PCMA/8000(8) PCMA/8000(8) NSE/8000(100) NSE/8000(100) encaprtp/8000(112) encaprtp/8000(112)
set RTP_SESSION_OPT_DTMF
uchEnableEchoCan(), lid 0 EP 2 enable
RTP configuration:
audio_mode RTP_MODE_ACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 PCMU/8000, rx[1] 8 PCMA/8000, rx[2] 100 NSE/8000
rx[3] 112 encaprtp/8000, rx[4] -1 , rx[5] -1
Jib: max 180ms, min 60ms, adapt 1
RTP Channel 0 is virgin: 0.
Need to stop RTP session then restart.
RTP session 0 started
uchSetDTMFMute(), ENABLE
[AUD]RTP Tx Up
[AUD]AUD_startRtcpTx(0x1ef7c0)
cordless_start_rtcp(), chan:0 remote ip:31.6.78.124 port:10583 intvl:0
Socket 26 bound to RTCP port 16391.
CNAME [email protected]
NAME "Brahma Kumari Canberra"
TOOL Cisco/SPA112-1.0.2(006)
Starting RTCP session on channel 0. Interval 0. Rx only.
RTCP session started on RTP channel 0.
[AUD]RTCP Up
[0]<<31.6.78.124:5060(401)
[0]<<31.6.78.124:5060(401)
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK22a27136;rport
From: "Frankston South, Australia" ;tag=as5e956228
To: ;tag=93d9d48c852dff3ei0
Contact:
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[0]<<31.6.78.124:5060(401)
[0]<<31.6.78.124:5060(401)
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK015e71ac;rport
From: "Frankston South, Australia" ;tag=as5e956228
To: ;tag=93d9d48c852dff3ei0
Contact:
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[0]<<31.6.78.124:5060(368)
[0]<<31.6.78.124:5060(368)
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK2acff592;rport
From: "Frankston South, Australia" ;tag=as5e956228
To: ;tag=93d9d48c852dff3ei0
Call-ID: [email protected]
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
[0]->31.6.78.124:5060(328)
[0]->31.6.78.124:5060(328)
SIP/2.0 200 OK
To: ;tag=93d9d48c852dff3ei0
From: "Frankston South, Australia" ;tag=as5e956228
Call-ID: [email protected]
CSeq: 103 BYE
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK2acff592
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
SIP_sessDlgEventProc: event: 44, ucState: 3
CC_eventProc(), event: CC_EV_SIG_CALL_ENDED(0x32), lid: 0, par: 2, par2: (nil)
AUD_ccEventProc: event 50 vid 0 par 0x2 par2 0x0
callEventProcTable[8] is cepConnectedProc
cepConnectedProc(line=0x1c6528, call=0x1c652c, event=50(CC_EV_SIG_CALL_ENDED), par=2, par2=(nil))
CC:Ended
NEW_CALL_STATE(), call 0: old state = 8, new state 6
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
Requesting call statistics...
RTP TX stats updated for channel 0
RTP TX stats updated for channel 0
RTP RX stats updated for channel 0
RTP RX stats updated for channel 0
Call statistics updated.
AUD_releaseCallObj() call(0x1ef7c0)
[AUD]AUD_stopRtpTx(0x1ef7c0)
cordless_stop_rtp_tx(), Channel 0.
RTP channel 0 going from Bi-dir to Rx.
RTP configuration:
audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 PCMU/8000, rx[1] 8 PCMA/8000, rx[2] 100 NSE/8000
rx[3] 112 encaprtp/8000, rx[4] -1 , rx[5] -1
Jib: max 180ms, min 60ms, adapt 1
RTP channel 0 is now Rx.
[AUD]RTP Tx Down
[AUD]AUD_stopRtpRx(0x1ef7c0)
cordless_stop_rtp_rx(), Channel 0.
RTP channel 0 going from Rx to Idle.
RTP configuration:
audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 PCMU/8000, rx[1] 8 PCMA/8000, rx[2] 100 NSE/8000
rx[3] 112 encaprtp/8000, rx[4] -1 , rx[5] -1
Jib: max 180ms, min 60ms, adapt 1
RTP channel 0 is now Idle.
[AUD]RTP Down
[AUD]AUD_releaseRtp(0x1ef7c0)
cordless_stop_rtp(), releasing RTP channel:0
cordless_stop_rtp(), RTP session 0 stopped succussfully
uchDisconnectEpFromNode(), disconnecting EP VoIP 0 from node 0
[AUD]RTP channel released
[0:0]AUD Rel Call
[AUD]Release UCH node for AUD_LINE 0.
uchDisableNode(), Node 0 released
[AUD]UCH node 0 freed.
Set QoS succeed
callEventProcTable[6] is cepInvalidProc
cepInvalidProc(line=0x1c6528, call=0x1c652c, event=30(CC_EV_TMR_INVALID), par=0, par2=(nil))
SLIC_stopRing
SLIC_startTone 8
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
DLG Terminated 255aa4
SIP_sessDlgEventProc: event: 40, ucState: 4
Sess Terminated
[AUD]Release UCH node for AUD_LINE 0.
AUD_LINE 0 has no associated UCH node.
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
uchSetMute(), ENABLE
uchSetMute(), DISABLE
[0]On Hook
CC_eventProc(), event: CC_EV_USR_ONHOOK(0x1), lid: 0, par: 0, par2: (nil)
AUD_ccEventProc: event 1 vid 0 par 0x0 par2 0x0
sysstatus_set_led_status_payton(), led_id: 1, statusCode:1, systemEvent: 0x100093
callEventProcTable[6] is cepInvalidProc
cepInvalidProc(line=0x1c6528, call=0x1c652c, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil))
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil))
cepIdleProc(), lid=0
[IVR_eventProc] evt 1 lid 0
callEventProcTable[6] is cepInvalidProc
cepInvalidProc(line=0x1c6528, call=0x1c652c, event=10(CC_EV_USR_ENDCALL), par=0, par2=(nil))
NEW_CALL_STATE(), call 0: old state = 6, new state 0
[AUD]Release UCH node for AUD_LINE 0.
AUD_LINE 0 has no associated UCH node.
SLIC_stopRing
SLIC_stopTone
[0]<<31.6.78.124:5060(499)
[0]<<31.6.78.124:5060(499)
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK51f45b89;rport
From: "Unknown" ;tag=as57087f59
To:
Contact:
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 16 Apr 2013 01:29:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[0]->31.6.78.124:5060(400)
[0]->31.6.78.124:5060(400)
SIP/2.0 200 OK
To: ;tag=e86c9e9a830a9c1ei0
From: "Unknown" ;tag=as57087f59
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK51f45b89
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
CC:Clean Up
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
NEW_CALL_STATE(), call 0: old state = 0, new state 0
NEW_CALL_STATE(), call 1: old state = 0, new state 0
NEW_CALL_STATE(), call 2: old state = 0, new state 0
--- OBJ POOL STAT ---
OP:TIMEOU = 67 (120 52) OP:SIPCOR = 0 ( 1 28)
OP:SIPCTS = 32 ( 32 936) OP:SIPSTS = 32 ( 32 6408)
OP:SIPAUS = 6 ( 8 680) OP:SIPDLG = 10 ( 10 148)
OP:SIPSES = 12 ( 12 9124) OP:SIPREG = 3 ( 4 468)
OP:SIPLIN = 0 ( 13 140) OP:SUBDLG = 13 ( 13 6444)
OP:STUNTS = 16 ( 16 68)
[0]<<31.6.78.124:5060(499)
[0]<<31.6.78.124:5060(499)
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK12d518b2;rport
From: "Unknown" ;tag=as188c9ad9
To:
Contact:
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 16 Apr 2013 01:30:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[0]->31.6.78.124:5060(400)
[0]->31.6.78.124:5060(400)
SIP/2.0 200 OK
To: ;tag=e86c9e9a830a9c1ei0
From: "Unknown" ;tag=as188c9ad9
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 31.6.78.124:5060;branch=z9hG4bK12d518b2
Server: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
[0]Off Hook
CC_eventProc(), event: CC_EV_USR_OFFHOOK(0x2), lid: 0, par: 0, par2: (nil)
AUD_ccEventProc: event 2 vid 0 par 0x0 par2 0x0
sysstatus_set_led_status_payton(), led_id: 1, statusCode:7, systemEvent: 0x100095
[IVR_eventProc] evt 2 lid 0
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c652c, event=9(CC_EV_USR_SEIZURE), par=0, par2=(nil))
cepIdleProc(), lid=0
cepIdleProc(), pname=voip-asia.brahmakumaris.org
cepIdleProc(), SYS_NOREG_CALL(0)=0, SIP_REGISTER_OK(0)=1
[AUD]Get UCH node for AUD_LINE 0.
uchAllocateNode(), Node 0 allocated
[AUD]UCH node 0 allocated to AUD_LINE 0.
uchConnectEpToNode(), connecting EP FXS 1 to node 0
uchEnableNode(), Node 0 enbaled
NEW_CALL_STATE(), call 0: old state = 0, new state 1
SLIC_stopRing
SLIC_startTone 1
uchSetMute(), ENABLE
Set QoS succeed
uchAppCb(), Event 37 received EP 2 lid 0
line:0 DTMFON event received digit 0
EVQ_read: 1 0 in 2024092
EVENT_DTMFON 0
CC_eventProc(), event: CC_EV_USR_DTMFON(0x4), lid: 0, par: 48, par2: (nil)
AUD_ccEventProc: event 4 vid 0 par 0x30 par2 0x0
uchAppCb(), Event 38 received EP 2 lid 0
line:0 DTMFOFF event received digit 0
EVQ_read: 0 0 in 2024092
EVENT_OOB_DTMF 0
CC_eventProc(), event: CC_EV_USR_OOB_DTMF(0x8), lid: 0, par: 48, par2: 0x64
AUD_ccEventProc: event 8 vid 0 par 0x30 par2 0x64
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=8(CC_EV_USR_OOB_DTMF), par=48, par2=0x64)
cepDialingProc(), event = 8(CC_EV_USR_OOB_DTMF)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=8(CC_EV_USR_OOB_DTMF), par=48, par2=0x64)
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_startTone 1
EVENT_DTMFOFF 0
CC_eventProc(), event: CC_EV_USR_DTMFOFF(0x5), lid: 0, par: 48, par2: (nil)
AUD_ccEventProc: event 5 vid 0 par 0x30 par2 0x0
[IVR_eventProc] evt 5 lid 0
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=5(CC_EV_USR_DTMFOFF), par=48, par2=(nil))
cepDialingProc(), event = 5(CC_EV_USR_DTMFOFF)
cepDialingProc(), digit = 0
cepDialingProc(), dn = (null)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=5(CC_EV_USR_DTMFOFF), par=48, par2=(nil))
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
uchSetMute(), DISABLE
uchAppCb(), Event 37 received EP 2 lid 0
line:0 DTMFON event received digit 0
EVQ_read: 1 0 in 2024092
EVENT_DTMFON 0
CC_eventProc(), event: CC_EV_USR_DTMFON(0x4), lid: 0, par: 48, par2: (nil)
AUD_ccEventProc: event 4 vid 0 par 0x30 par2 0x0
uchAppCb(), Event 38 received EP 2 lid 0
line:0 DTMFOFF event received digit 0
EVQ_read: 0 0 in 2024092
EVENT_OOB_DTMF 0
CC_eventProc(), event: CC_EV_USR_OOB_DTMF(0x8), lid: 0, par: 48, par2: 0x64
AUD_ccEventProc: event 8 vid 0 par 0x30 par2 0x64
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=8(CC_EV_USR_OOB_DTMF), par=48, par2=0x64)
cepDialingProc(), event = 8(CC_EV_USR_OOB_DTMF)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=8(CC_EV_USR_OOB_DTMF), par=48, par2=0x64)
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
EVENT_DTMFOFF 0
CC_eventProc(), event: CC_EV_USR_DTMFOFF(0x5), lid: 0, par: 48, par2: (nil)
AUD_ccEventProc: event 5 vid 0 par 0x30 par2 0x0
[IVR_eventProc] evt 5 lid 0
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=5(CC_EV_USR_DTMFOFF), par=48, par2=(nil))
cepDialingProc(), event = 5(CC_EV_USR_DTMFOFF)
cepDialingProc(), digit = 0
cepDialingProc(), dn = (null)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=5(CC_EV_USR_DTMFOFF), par=48, par2=(nil))
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
uchAppCb(), Event 37 received EP 2 lid 0
line:0 DTMFON event received digit 4
EVQ_read: 1 4 in 2024092
EVENT_DTMFON 4
CC_eventProc(), event: CC_EV_USR_DTMFON(0x4), lid: 0, par: 52, par2: (nil)
AUD_ccEventProc: event 4 vid 0 par 0x34 par2 0x0
uchAppCb(), Event 38 received EP 2 lid 0
line:0 DTMFOFF event received digit 4
EVQ_read: 0 4 in 2024092
EVENT_OOB_DTMF 4
CC_eventProc(), event: CC_EV_USR_OOB_DTMF(0x8), lid: 0, par: 52, par2: 0x64
AUD_ccEventProc: event 8 vid 0 par 0x34 par2 0x64
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=8(CC_EV_USR_OOB_DTMF), par=52, par2=0x64)
cepDialingProc(), event = 8(CC_EV_USR_OOB_DTMF)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=8(CC_EV_USR_OOB_DTMF), par=52, par2=0x64)
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
EVENT_DTMFOFF 4
CC_eventProc(), event: CC_EV_USR_DTMFOFF(0x5), lid: 0, par: 52, par2: (nil)
AUD_ccEventProc: event 5 vid 0 par 0x34 par2 0x0
[IVR_eventProc] evt 5 lid 0
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=5(CC_EV_USR_DTMFOFF), par=52, par2=(nil))
cepDialingProc(), event = 5(CC_EV_USR_DTMFOFF)
cepDialingProc(), digit = 4
cepDialingProc(), dn = (null)
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=5(CC_EV_USR_DTMFOFF), par=52, par2=(nil))
cepIdleProc(), lid=0
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
callEventProcTable[1] is cepDialingProc
cepDialingProc(line=0x1c6528, call=0x1c652c, event=32(CC_EV_TMR_DIALPLAN), par=0, par2=(nil))
cepDialingProc(), event = 32(CC_EV_TMR_DIALPLAN)
cepDialingProc(), digit =
cepDialingProc(), dn = 004
[checkSuppFeatActCode] lid 0 vid 0
pconly: 0
clRemote: (nil), clLocal->ucNumAudioCodec: 4
[AUD]Get UCH node for AUD_LINE 0.
AUD_LINE 0 already has UCH node 0.
NEW_CALL_STATE(), call 0: old state = 1, new state 3
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
Calling:[email protected]:0, rc=0
RTP_nextMediaPort(), port = 16394
RTP_nextMediaPort(), rc=16392
AUD_allocCallObj() call(0x1ef7c0)
[0:0]AUD ALLOC CALL (port=16392)
[AUD]AUD_startRtpRx(0x1ef7c0)
Local loopback mode: None. Type: None.
Remote loopback mode: None. Type None.
UCH sync parameter hold off time is 70
RTP channel setup: udp_no_checksum 0, sysmmetric_rtp 0, tos 0xb8, cos 6, mlb 0.
uchConnectEpToNode(), connecting EP VoIP 0 to node 0
cordless_start_rtp(), chan:0 remote ip:(null) port:0 local:16392 rx:1 ipt:0 ptime:0
Starting Rx only RTP.
Socket 19 bound to port 16392.
Remote IP/port: 0.0.0.0:0
Codec list from SDP (internal pt):Codec list from SDP (internal pt): 130 130 18 18 0 0 8 8 134 134 136 136
Rx payload list: Rx payload list: G.726/8000(2) G.726/8000(2) G.729/8000(18) G.729/8000(18) PCMU/8000(0) PCMU/8000(0) PCMA/8000(8) PCMA/8000(8) NSE/8000(100) NSE/8000(100) encaprtp/8000(112) encaprtp/8000(112)
set RTP_SESSION_OPT_DTMF
uchEnableEchoCan(), lid 0 EP 2 enable
RTP configuration:
audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 0, rx_pt_event 101, tx_pt_event -1, tx_pt 0
rx[0] 2 G.726/8000, rx[1] 18 G.729/8000, rx[2] 0 PCMU/8000
rx[3] 8 PCMA/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
Jib: max 180ms, min 60ms, adapt 1
RTP Channel 0 is virgin: 1.
RTP session 0 started
uchSetDTMFMute(), ENABLE
[AUD]RTP Rx Up
[rse_refresh_addr_list] query voip-asia.brahmakumaris.org block 0
[0]->31.6.78.124:5060(910)
[0]->31.6.78.124:5060(910)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-11d04ccf
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(910)
[0]->31.6.78.124:5060(910)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-11d04ccf
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(910)
[0]->31.6.78.124:5060(910)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-11d04ccf
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]<<31.6.78.124:5060(549)
[0]<<31.6.78.124:5060(549)
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-11d04ccf;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>;tag=as7d844705
Call-ID: [email protected]
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="458c760d"
Content-Length: 0
[0]->31.6.78.124:5060(457)
[0]->31.6.78.124:5060(457)
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-11d04ccf
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>;tag=as7d844705
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Brahma Kumari Canberra"
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 0
[0]->31.6.78.124:5060(1089)
[0]->31.6.78.124:5060(1089)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="080626f0d87dbcd5b124b72491369bcb"
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(1089)
[0]->31.6.78.124:5060(1089)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="080626f0d87dbcd5b124b72491369bcb"
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]->31.6.78.124:5060(1089)
[0]->31.6.78.124:5060(1089)
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="080626f0d87dbcd5b124b72491369bcb"
Contact: "Brahma Kumari Canberra"
Expires: 240
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 331
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 109351 109351 IN IP4 192.168.1.41
s=-
c=IN IP4 192.168.1.41
t=0 0
m=audio 16392 RTP/AVP 2 18 0 8 100 101
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
[0]<<31.6.78.124:5060(471)
[0]<<31.6.78.124:5060(471)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
[0]<<31.6.78.124:5060(471)
[0]<<31.6.78.124:5060(471)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
[0]<<31.6.78.124:5060(471)
[0]<<31.6.78.124:5060(471)
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Length: 0
[0]<<31.6.78.124:5060(789)
[0]<<31.6.78.124:5060(789)
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>;tag=as54d5f390
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 260
v=0
o=root 2809 2809 IN IP4 31.6.78.124
s=session
c=IN IP4 31.6.78.124
t=0 0
m=audio 12518 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[AUD]AUD_stopRtpTx(0x1ef7c0)
cordless_stop_rtp_tx(), Channel 0.
*** RTP channel not in Tx. Nothing to stop!
*** RTP channel not in Tx. Nothing to stop!
[AUD]RTP Tx Down
[AUD]AUD_startRtpTx(0x1ef7c0, 0, 31.6.78.124, 12518, 20)
Local loopback mode: None. Type: None.
Remote loopback mode: None. Type None.
UCH sync parameter hold off time is 70
Already has a RTP channel.
Already has a RTP channel.
uchConnectEpToNode(), connecting EP VoIP 0 to node 0
cordless_start_rtp(), chan:0 remote ip:31.6.78.124 port:12518 local:16392 rx:0 ipt:0 ptime:20
Going from Rx only to bi-directional.
Old remote IP/port: 0.0.0.0:0
Remote IP/port: 31.6.78.124:12518
Codec list from SDP (internal pt):Codec list from SDP (internal pt): 130 130 18 18 0 0 8 8 134 134 136 136
Rx payload list: Rx payload list: G.726/8000(2) G.726/8000(2) G.729/8000(18) G.729/8000(18) PCMU/8000(0) PCMU/8000(0) PCMA/8000(8) PCMA/8000(8) NSE/8000(100) NSE/8000(100) encaprtp/8000(112) encaprtp/8000(112)
set RTP_SESSION_OPT_DTMF
uchEnableEchoCan(), lid 0 EP 2 enable
RTP configuration:
audio_mode RTP_MODE_ACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 2 G.726/8000, rx[1] 18 G.729/8000, rx[2] 0 PCMU/8000
rx[3] 8 PCMA/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
Jib: max 180ms, min 60ms, adapt 1
RTP Channel 0 is virgin: 0.
Need to stop RTP session then restart.
RTP session 0 started
uchSetDTMFMute(), ENABLE
[AUD]RTP Tx Up
[AUD]AUD_startRtcpTx(0x1ef7c0)
cordless_start_rtcp(), chan:0 remote ip:31.6.78.124 port:12519 intvl:0
Socket 26 bound to RTCP port 16393.
CNAME [email protected]
NAME "Brahma Kumari Canberra"
TOOL Cisco/SPA112-1.0.2(006)
Starting RTCP session on channel 0. Interval 0. Rx only.
RTCP session started on RTP channel 0.
[AUD]RTCP Up
CC_eventProc(), event: CC_EV_SIG_CALL_PROGRESS(0x35), lid: 0, par: 3, par2: (nil)
AUD_ccEventProc: event 53 vid 0 par 0x3 par2 0x0
callEventProcTable[3] is cepCallingProc
cepCallingProc(line=0x1c6528, call=0x1c652c, event=53(CC_EV_SIG_CALL_PROGRESS), par=3, par2=(nil))
CC:CallProgress
NEW_CALL_STATE(), call 0: old state = 3, new state 4
SLIC_stopRing
SLIC_stopRing
SLIC_stopTone
[AUD]AUD_startRtpRx(0x1ef7c0)
Local loopback mode: None. Type: None.
Remote loopback mode: None. Type None.
UCH sync parameter hold off time is 70
Already has a RTP channel.
Already has a RTP channel.
uchConnectEpToNode(), connecting EP VoIP 0 to node 0
cordless_start_rtp(), chan:0 remote ip:(null) port:0 local:16392 rx:1 ipt:0 ptime:0
Going from Tx only to bi-directional.
Remote IP/port: 31.6.78.124:12518
Codec list from SDP (internal pt):Codec list from SDP (internal pt): 130 130 18 18 0 0 8 8 134 134 136 136
Rx payload list: Rx payload list: G.726/8000(2) G.726/8000(2) G.729/8000(18) G.729/8000(18) PCMU/8000(0) PCMU/8000(0) PCMA/8000(8) PCMA/8000(8) NSE/8000(100) NSE/8000(100) encaprtp/8000(112) encaprtp/8000(112)
set RTP_SESSION_OPT_DTMF
uchEnableEchoCan(), lid 0 EP 2 enable
RTP configuration:
audio_mode RTP_MODE_ACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 2 G.726/8000, rx[1] 18 G.729/8000, rx[2] 0 PCMU/8000
rx[3] 8 PCMA/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
Jib: max 180ms, min 60ms, adapt 1
RTP Channel 0 is virgin: 0.
Going from Tx to bi-dir just need updating.
RTP session 0 updated
uchSetDTMFMute(), ENABLE
[AUD]RTP Rx Up
[0]On Hook
CC_eventProc(), event: CC_EV_USR_ONHOOK(0x1), lid: 0, par: 0, par2: (nil)
AUD_ccEventProc: event 1 vid 0 par 0x0 par2 0x0
sysstatus_set_led_status_payton(), led_id: 1, statusCode:1, systemEvent: 0x100093
callEventProcTable[4] is cepCallingProc
cepCallingProc(line=0x1c6528, call=0x1c652c, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil))
callEventProcTable[0] is cepIdleProc
cepIdleProc(line=0x1c6528, call=0x1c6734, event=1(CC_EV_USR_ONHOOK), par=0, par2=(nil))
cepIdleProc(), lid=0
callEventProcTable[4] is cepCallingProc
cepCallingProc(line=0x1c6528, call=0x1c652c, event=10(CC_EV_USR_ENDCALL), par=0, par2=(nil))
NEW_CALL_STATE(), call 0: old state = 4, new state 0
SLIC_stopRing
SLIC_stopTone
SIP_lineCcCmdProc(), cmd=CC_CMD_ENDCALL
Requesting call statistics...
RTP TX stats updated for channel 0
RTP TX stats updated for channel 0
RTP RX stats updated for channel 0
RTP RX stats updated for channel 0
Call statistics updated.
AUD_releaseCallObj() call(0x1ef7c0)
[AUD]AUD_stopRtpTx(0x1ef7c0)
cordless_stop_rtp_tx(), Channel 0.
RTP channel 0 going from Bi-dir to Rx.
RTP configuration:
audio_mode RTP_MODE_REC_ONLY, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 G.726/8000, rx[1] 18 G.729/8000, rx[2] 0 PCMU/8000
rx[3] 8 PCMA/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
Jib: max 180ms, min 60ms, adapt 1
RTP channel 0 is now Rx.
[AUD]RTP Tx Down
[AUD]AUD_stopRtpRx(0x1ef7c0)
cordless_stop_rtp_rx(), Channel 0.
RTP channel 0 going from Rx to Idle.
RTP configuration:
audio_mode RTP_MODE_INACTIVE, media_loop_level RTP_LOOP_LEVEL_NONE, dtmf2833numEndPakcets 3, opts 0x0
Codec: duration 20, rx_pt_event 101, tx_pt_event 101, tx_pt 0
rx[0] 0 G.726/8000, rx[1] 18 G.729/8000, rx[2] 0 PCMU/8000
rx[3] 8 PCMA/8000, rx[4] 100 NSE/8000, rx[5] 112 encaprtp/8000
Jib: max 180ms, min 60ms, adapt 1
RTP channel 0 is now Idle.
[AUD]RTP Down
[AUD]AUD_releaseRtp(0x1ef7c0)
cordless_stop_rtp(), releasing RTP channel:0
cordless_stop_rtp(), RTP session 0 stopped succussfully
uchDisconnectEpFromNode(), disconnecting EP VoIP 0 from node 0
[AUD]RTP channel released
[0:0]AUD Rel Call
[AUD]Release UCH node for AUD_LINE 0.
uchDisableNode(), Node 0 released
[AUD]UCH node 0 freed.
[0]->31.6.78.124:5060(563)
[0]->31.6.78.124:5060(563)
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="9e73603bd3ec65e6abc330572f446d0d"
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 0
Set QoS succeed
[0]->31.6.78.124:5060(563)
[0]->31.6.78.124:5060(563)
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="9e73603bd3ec65e6abc330572f446d0d"
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 0
[0]->31.6.78.124:5060(563)
[0]->31.6.78.124:5060(563)
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
Max-Forwards: 70
Proxy-Authorization: Digest username="6010",realm="asterisk",nonce="458c760d",uri="sip:[email protected]",algorithm=MD5,response="9e73603bd3ec65e6abc330572f446d0d"
User-Agent: Cisco/SPA112-1.0.2(006)
Content-Length: 0
[0]<<31.6.78.124:5060(460)
[0]<<31.6.78.124:5060(460)
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97;received=60.240.77.7
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>;tag=as54d5f390
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
[0]->31.6.78.124:5060(636)
[0]->31.6.78.124:5060(636)
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.41:5060;branch=z9hG4bK-2f9add97
From: "Brahma Kumari Canberra" [email protected]>;tag=b99a4fb8adee114eo0
To: [email protected]>;tag=as54d5f390
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="60 -
I have a 3D model created in PROE that I have imported to Acrobat 3D. I have made an exploded view and then imported it in Word. The picture that looks great in Adobe, looks like ¤%!##& in word. I have "Publishing" in the Import Settings Profile while importing it into word. That is the best as I understood from the books I read about Acrobat 3D. Are there any other settings that can make the picture look better in word (both on screen and on a paper copy).
BR,
AnnaOverall the picture quality on FiOS should be superior to comcast or cox or any other provider with the exception of DTV. they have a pretty close picture, if not equal. Some basics are important, You should have an HD tv set, you should have an HD Set top box, and they should be connected to one another using a single HDMI cable.
Make sure to check that.
Then the set top box needs to be set properly. it's unusual for it not to be, but double check just in case.
click menu>settings>video settings.
There are two settings that are important here
Video Format should be set to 1080i
and
SD OVERRIDE should be set to either OFF (preferred) or 480i
If those things are all ok, or done, then go to channel 600 for CNN HD or Try to find discovery HD, and then make the comparison.
If you flip between channel 100 and 600, can you tell the difference?
If not, then there is something wrong.
Hope that helps a little.
Also let us know if it's just overall picture quality, or if you are getting channel break up or pixelization. it looks like the picture is digitizing or is blocky, or choppy.
If it's tiling or pixelizing, then take a look at this page, and see if these suggestions help.
Tiling and Pixelation issues -
Hi,
I have created file to idoc scenario.
I have done all the Design and Configuration parts.
But when executed the same scenario by placing input file on input folder.
Its not picking the file and in sxmb_moni also it wont give any messages?!!
What would be the problem....Can you help me out??!!
Cheers.,
StallinHi Stallin,
Do you see any channels in the Adapter monitoring?
It's look to me like a cache problem,
Do you see the channel in SXI_CACHE?
Have you try to perform CPA CACHE REFRESH?
Try to read the following guide "How to Handle Caches in SAP XI 3.0" https://www.sdn.sap.com/irj/sdn/go/portal/prtroot/docs/library/uuid/1a69ea11-0d01-0010-fa80-b47a79301290
I think that one of the cache refresh in http://SERVER:PORT/rep -> Administartion will solve yor problem
Best Regards
Guza -
SPA112 - Can receive but not make calls
I have just setup an SPA112 at home behind a NAT router and have configured it to a state where I can receive but not make calls.
It registers fine and remains registered.
The Line 1 Status indicates a State of "invalid" and a Tone of "reorder" when the call fails.
I believe I have fiddled with the dial plan sufficient to ensure that the number is not being rejected based on it. My current dial plan is as follows:
(*xx|000S0|0[2-9][69]xxxxxxxS0|[69]xxxxxxxS0|04xxxxxxxxS0|xxxxxxxxxxxx.)
I have tried with and without an outbound proxy.
I have attempted to debug the issue, enabling full DEBUG level logging and setting "SIP debug option" to "full", but all I get in the log when making a failed call is the following:
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 18 and str G.729/8000 chan 0
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 8 and str PCMA/8000 chan 0
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 0 and str PCMU/8000 chan 0
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 2 and str G.726/8000 chan 0
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 100 and str NSE/8000 chan 0
Jul 23 21:35:29 SPA112 kern.warning [17339229.112000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
Jul 23 21:35:29 SPA112 daemon.notice msgswitchd[310]: MSGSWD RTCP Reqt len 12 Data 2,17,7304,0
Jul 23 21:35:29 SPA112 kern.warning [17339229.268000]
Jul 23 21:35:29 SPA112 kern.warning [17339229.268000] #### RTP STOP Flag set in this channel break ####
Jul 23 21:35:29 SPA112 kern.warning [17339229.272000] ###### sock_sendmsg return 172
I have been unable to find any information about these warnings on th net.
I can successfully make calls with soft phones configured with similar settings behind the same router.
What am I missing?
Any suggestions appreciated, thanks.Thanks for the post.
To clarify, I have been testing the connection by attempting calls to and from my mobile phone. If I use my mobile to call the number linked to the VOIP account that the SPA112 connects to, the call is received fine and the call works perfectly; I can hear voice at both ends fine.
If I attempt to call the mobile from the phone behind the SPA112 I immediately get a reorder tone as soon as the number is dialed by the SPA112 (i.e. immediately if it matches a number in the dial plan followed by S0 or after the short wait period otherwise etc.)
On the other hand, if I use a softphone (I've tried MicroSIP and Blink) I can place calls to the same mobile through the same VOIP account from the same LAN behind the same NAT as the SPA112, and the calls are received and bidirectional voice comms works just fine.
I'm not sure if I can make calls internal to my LAN with this setup... I only have the one VOIP account, and have no SIP server on the LAN (if that's the right terminology, I'm new to this).
Unfortunately I'm not sure what you're refering to when you talk about ACL and COR. The topology is very straight forward: the SPA112 (along with a couple of PCs) on a LAN behind a NAT router with an internet connection on the other side. The SPA112 is configured as a DMZ, though I don't believed this had any impact for better or worse. In no test for either SPA112 or softphone have I forwarded any ports or changed any router configuration at all (with the exception of the DMZ).
So the question is why the softphone can place calls fine, but the SPA112 can only receive calls. The dial plan is an obvious answer, but I'm fairly certain I've ruled this out.
If someone can tell me how to get more detailed logging out of the SPA112 that would be very useful as well. -
SPA112 Crashes - No dialtone, restart needed
Yesterday morning, we picked up the phone to make a call via the SPA112 and there was no dialtone. I disconnected/reconnected the power and then we were able to get a dialtone and make a call. Unfortunately I didn't have the opportunity to see if the web-interface was up. Am running the latest GA firmware, 1.1.0 (011) Feb 10 2012. It had probably been running for only about 4 days or so since the last reboot. Anyone else seeing stability issues? (I thought I'd seen another older posting about someone having had to reboot their device regularly. In addition, I saw the report about caller id's longer than 26 chars causing issues, but I haven't publicized my new number yet so it is unlikely that was the cause.)
Cisco, any suggestions on a way to capture additional debug info, other than keeping a logging server running? Didn't see any unusual activity in the device logs.
DanHello,
One thing I know crashes the SPA 122 on FW 1.1.0(011) is a "484 Address Incomplete". If the box is running 1.0.1(022), it responds appropriately, and just sends reorder to the phone. Of course, that brings back all the issues that were fixed in 1.1.0.
I opened a case asking about trying the beta to see if it fixes the issues such as the DTMF problem, Caller ID problem, and this issue, but the tech. wanted configs, fresh packet caps, and all kinds of stuff that shouldn't be necessary given that some of the issues I mentioned are supposedly fixed.
He was real helpful with regard to the config migration from SPA 2102 in that the old config to xml migation only works with a specific version of the SPC tool, which isn't mentioned in the "we're killing SPA2102, but it's OK 'cause you can migate" document.
Not sure if the 484 box crash is related to the issue discussed here, as it recovers, but one never knows if it causes some progressive issue with the box that after 10 or 20, it burns up all available memory, or something similar. Packet cap and syslog output of the event that crashes the box below.
23:06:37.954693 IP (tos 0x0, ttl 64, id 22296, offset 0, flags [none], proto UDP (17), length 830)
10.33.90.60.sip > SPA122.8060: SIP, length: 802
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.33.90.111:8060;branch=z9hG4bK-2817ba48
From: ;tag=1481fc25925927f6o0
To: ;tag=NNmBD3S5vZSyK
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120624T003326Z~86df8b338e+unclean~20120624T044059Z
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
Remote-Party-ID: "4669" ;party=calling;privacy=off;screen=no
23:06:37.973366 IP (tos 0x68, ttl 64, id 0, offset 0, flags [DF], proto UDP (17), length 635)
SPA122.8060 > 10.33.90.60.sip: SIP, length: 607
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.33.90.111:8060;branch=z9hG4bK-2817ba48
From: ;tag=1481fc25925927f6o0
To: ;tag=NNmBD3S5vZSyK
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="4331",realm="10.33.90.60",nonce="455f6ec0-c0ce-11e1-bdff-c9d1f242411e",uri="sip:[email protected]",algorithm=MD5,response="e59568b7cab23ca5081d697b612fef0e",qop=auth,nc=00000001,cnonce="be94babd"
Contact:
User-Agent: Cisco/SPA122-1.1.0(011)
Content-Length: 0
Clock in ATA is slightly diferent than NTP locked server, but death begins at 23:06:37
<4>Jun 27 23:06:35 SPA122 [17179877.700000] In cordless Driver Codec 100 and str NSE/8000 chan 0
10.50.50.1 27/06 23:06:29.268
<4>Jun 27 23:06:35 SPA122 [17179877.700000] In cordless Driver Codec 112 and str encaprtp/8000 chan 0
10.50.50.1 27/06 23:06:29.268
<29>Jun 27 23:06:37 SPA122 msgswitchd[380]: MSGSWD RTCP Reqt len 12 Data 2,35,7304,0
10.50.50.1 27/06 23:06:31.359
<4>Jun 27 23:06:37 SPA122 [17179879.824000] RTCP is running so calling rtcp stop
10.50.50.1 27/06 23:06:31.390
<4>Jun 27 23:06:37 SPA122 [17179879.824000] chan->kmode is present not null
10.50.50.1 27/06 23:06:31.390
<4>Jun 27 23:06:37 SPA122 [17179879.828000] ###### RTCP sock_sendmsg return 172
10.50.50.1 27/06 23:06:31.390
<4>Jun 27 23:06:37 SPA122 [17179879.836000] ###### sock_sendmsg return 172
10.50.50.1 27/06 23:06:31.390
<4>Jun 27 23:06:37 SPA122 [17179879.836000]
10.50.50.1 27/06 23:06:31.452
<4>Jun 27 23:06:37 SPA122 [17179879.836000] #### RTP STOP Flag set in this channel break ####
10.50.50.1 27/06 23:06:31.452
<6>Jun 27 23:06:37 SPA122 [17179879.888000] cordless: deinit
10.50.50.1 27/06 23:06:31.452
<6>Jun 27 23:06:52 SPA122 [17179894.068000] cordless: loading synergy-2012-01-18
10.50.50.1 27/06 23:06:45.757
<6>Jun 27 23:06:52 SPA122 [17179894.108000] cordless: init successful
10.50.50.1 27/06 23:06:45.757
<13>Jun 27 23:06:54 SPA122 msgswitchd: MSGSWITCH fd_rtp fifo created 12
10.50.50.1 27/06 23:06:47.817
<13>Jun 27 23:06:54 SPA122 msgswitchd: MSGSWITCH fd_ch fifo created 14
10.50.50.1 27/06 23:06:47.817
<13>Jun 27 23:06:54 SPA122 msgswitchd: MSGSWITCH fd_gmep fifo created 15
10.50.50.1 27/06 23:06:47.817 -
Getting following exception on TCP ssl at server side
Hi,
I am trying to setup an ssl client and ssl server.
But on ssl client side, My need_wrap going into loop(Buffer overflow)
On my server side, i am getting following error.
Please help me...I am not getting much discussion/articles to solve this problem.
javax.net.ssl.SSLProtocolException: Handshake message sequence violation, state = 1, type = 1
javax.net.ssl.SSLProtocolException: Handshake message sequence violation, state = 1, type = 1
at sun.security.ssl.Handshaker.checkThrown(Handshaker.java:1371)
at sun.security.ssl.SSLEngineImpl.checkTaskThrown(SSLEngineImpl.java:513)
at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:790)
at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:758)
at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624)
at com.ipay.ssl.SSLServerNio.doHandshake(SSLServerNio.java:55)
at com.ipay.ssl.SSLServerNio.main(SSLServerNio.java:215)
Caused by: javax.net.ssl.SSLProtocolException: Handshake message sequence violation, state = 1, type = 1
at sun.security.ssl.ServerHandshaker.processMessage(ServerHandshaker.java:156)
at sun.security.ssl.Handshaker.processLoop(Handshaker.java:868)
at sun.security.ssl.Handshaker$1.run(Handshaker.java:808)
at sun.security.ssl.Handshaker$1.run(Handshaker.java:806)
at java.security.AccessController.doPrivileged(Native Method)
at sun.security.ssl.Handshaker$DelegatedTask.run(Handshaker.java:1299)
at java.util.concurrent.ThreadPoolExecutor.runWorker(ThreadPoolExecutor.java:1145)
at java.util.concurrent.ThreadPoolExecutor$Worker.run(ThreadPoolExecutor.java:615)
at java.lang.Thread.run(Thread.java:744)
My server handshake code look like this.
void doHandshake(SocketChannel socketChannel, SSLEngine engine,
ByteBuffer myNetSendData, ByteBuffer myNetRecieveData) throws Exception {
// Create byte buffers to use for holding application data
ByteBuffer myAppDataSend = ByteBuffer.allocate(engine.getSession().getApplicationBufferSize());
ByteBuffer myAppDataRecieve = ByteBuffer.allocate(engine.getSession().getApplicationBufferSize());
System.out.println(engine.getPeerPort());
// Begin handshake
engine.beginHandshake();
SSLEngineResult.HandshakeStatus hs = engine.getHandshakeStatus();
// Process handshaking message
while (hs != SSLEngineResult.HandshakeStatus.FINISHED &&
hs != SSLEngineResult.HandshakeStatus.NOT_HANDSHAKING) {
switch (hs) {
case NEED_UNWRAP:
System.out.println("Reached NEED UNWRAP");
// Receive handshaking data from peer
if (socketChannel.read(myNetRecieveData) < 0) {
// Handle closed channel
System.out.println("not able toRead data from channel to buffer at client");
myNetRecieveData.flip();
// Process incoming handshaking data
SSLEngineResult res = engine.unwrap(myNetRecieveData, myAppDataRecieve);
// myNetRecieveData.compact();
// Getting handshake status
hs = res.getHandshakeStatus();
System.out.println("Debugging in NEED_UNWRAP-->"+hs);
// Check status
switch (res.getStatus()) {
case OK :
// Handle OK status
System.out.println("OK");
break;
case BUFFER_OVERFLOW:
System.out.println("BUFFER OVERFLOW");
break;
case BUFFER_UNDERFLOW:
System.out.println("BUFFER UNDERFLOW");
/* if (socketChannel.read(myNetRecieveData) < 0) {
// Handle closed channel
System.out.println("not able toRead data from channel to buffer at client");
hs=HandshakeStatus.NEED_UNWRAP;
System.out.println("Read data on underflow condition");
break;
case CLOSED:
System.out.println("CLOSED");
break;
// Handle other status: BUFFER_UNDERFLOW, BUFFER_OVERFLOW, CLOSED
break;
case NEED_WRAP :
System.out.println("Reached NEED WRAP");
// Empty the local network packet buffer.
// Generate handshaking data
//myAppDataSend.flip();
res = engine.wrap(myAppDataSend, myNetSendData);
// Getting handshake status
hs = res.getHandshakeStatus();
System.out.println("Debugging in NEED_WRAP-->"+hs);
// Check status
switch (res.getStatus()) {
case OK :
System.out.println("OK");
myNetSendData.flip();
// Send the handshaking data to peer
while (myNetSendData.hasRemaining()) {
if (socketChannel.write(myNetSendData) < 0) {
// closing socket channel
break;
case BUFFER_OVERFLOW:
System.out.println("BUFFER OVERFLOW");
//Writing network send buffer
myNetSendData.flip();
while (myNetSendData.hasRemaining()) {
if(socketChannel.write(myNetSendData) < 0)
System.out.println("Some thing wrong happened");
System.out.println("written data");
hs=HandshakeStatus.NEED_WRAP;
break;
case BUFFER_UNDERFLOW:
System.out.println("BUFFER UNDERFLOW");
break;
case CLOSED:
System.out.println("CLOSED");
break;
// Handle other status: BUFFER_OVERFLOW, BUFFER_UNDERFLOW, CLOSED
break;
case NEED_TASK :
System.out.println("NEED TASK");
System.out.println("Debugging in NEED_TASK-->"+hs);
Runnable task;
while((task=engine.getDelegatedTask()) != null)
System.out.println("Inside while loop");
ExecutorService executorService = Executors.newFixedThreadPool(1);
executorService.execute(task);
// Handle blocking tasks
hs=engine.getHandshakeStatus();
System.out.println("Printing"+engine.getHandshakeStatus());
break;
case FINISHED:
System.out.println("Debugging in FINISHED-->"+hs);
System.out.println("handshake done");
hs=HandshakeStatus.FINISHED;
break;
// Handle other status: // FINISHED or NOT_HANDSHAKING
// Processes after handshaking
brs,
vargheseHi Zia,
The error is...
Caused by: java.io.InvalidClassException: org.eclipse.persistence.indirection.IndirectList; local class incompatible: stream classdesc serialVersionUID = 4038061360325736360, local class serialVersionUID = -494763524358427112
...which means you have two different versions of org.eclipse.persistence.indirection.IndirectList, one on the server and one in the client (JDeveloper). As these classes have different serialVersionUID values then Java throws an exception when deserializing as they are probably not compatible.
I don't know much about the Eclipse stuff but from looking at your post I can only assume one version of the class is in D:\Coherence\toplink\jlib\eclipselink.jar on the Coherence server side and the other is in JDevloper in D:\OracleSOA\Middleware\modules\org.eclipse.persistence_1.1.0.0_2-1.jar as these are the only jar file containing "eclipse" that I can see on the classpaths.
JK -
I write a NIO Server program with SSL(JSSE).I encount some problem in here.
SSLEngineResult res = sslEngine.wrap(////);
HandshakeStatus hs = res.getHandshakeStatus();
HandshakeStatus hs2 = engine.getHandshakeStatus();
in my program ,hs = FINISH but hs2=NOT_HANDSHAKE,why it isn't same value? and why my program get a NOT_HANDSHAKE status?
my program :
package cn.com.infosec.isfw2.sfw;
import java.io.EOFException;
import java.io.FileOutputStream;
import java.io.IOException;
import java.net.Socket;
import java.nio.ByteBuffer;
import java.nio.channels.SelectionKey;
import java.nio.channels.Selector;
import java.nio.channels.SocketChannel;
import javax.net.ssl.SSLContext;
import javax.net.ssl.SSLEngine;
import javax.net.ssl.SSLEngineResult;
import javax.net.ssl.SSLException;
import javax.net.ssl.SSLEngineResult.HandshakeStatus;
import javax.net.ssl.SSLEngineResult.Status;
import com.sun.corba.se.pept.transport.OutboundConnectionCache;
import cn.com.infosec.isfw2.impl.DefaultProtocolHandler;
public class SSLSession extends SocketSession {
private SocketChannel channel;
private IOFilterChain filters;
private ByteBuffer buffer = null;
private Selector selector;
// private HashMap<String, Object> attributes;
private SSLEngine sslEngine;
private ProtocolHandler protocolHandler = null;
private ByteBuffer inNetBuffer;
private ByteBuffer outNetBuffer;
// * Applicaton cleartext data to be read by application
// private IoBuffer appBuffer;
* Empty buffer used during initial handshake and close operations
private final ByteBuffer emptyBuffer = ByteBuffer.allocate(0);
private SSLEngineResult.HandshakeStatus handshakeStatus;
private boolean initialHandshakeComplete;
private boolean handshakeComplete;
public SSLSession(SocketChannel channel, Selector sel, IOFilterChain chain) {
super(channel, sel, chain);
this.channel = channel;
this.filters = chain;
this.selector = sel;
buffer = null;
// attributes = null;
filters.SessionCreate(this);
System.out.println("----create ssl session--------");
System.out.println("create session,channel=" + this.channel);
public void read() throws IOException {
if (buffer == null) {
buffer = ByteBuffer.allocate(8192);
if (buffer.position() == buffer.capacity()) {
adjustBuffer(2 * buffer.capacity());
int readbytes = 0;
int ret = 0;
while (true) {
ret = read0();
System.out.println("buffer=" + buffer);
if (ret > 0) {
readbytes += ret;
if (buffer.position() == buffer.capacity()) {
adjustBuffer(2 * buffer.capacity());
System.out.println(Thread.currentThread().getName()
+ " session " + this + " channel " + channel
+ " buffer=" + buffer + " buffer is full " + " "
+ System.nanoTime());
} else if (ret == 0) {
break;
} else {
throw new EOFException("peer connection close.");
if (protocolHandler == null) {
protocolHandler = new DefaultProtocolHandler();
if (protocolHandler.isAllDataRecved(buffer)) {
final SocketSession s = this;
try {
Runnable r = new Runnable() {
public void run() {
filters.DataReceived(s);
System.out.println(channel.socket().getInetAddress());
//if(channel.socket().getInetAddress().isAnyLocalAddress()){
Socket s1 = channel.socket();
if (s1.getInetAddress().isLoopbackAddress()) {
System.out.println("---is local----");
new Thread(r).start();
} else {
ExcuterManager.parserExe.execute(r);
} catch (Throwable e) {
this.close();
} else {
continueRead();
public SocketChannel getChannel() {
return channel;
public ByteBuffer getBuffer() {
return buffer;
public void continueRead() throws IOException {
channel.register(selector, SelectionKey.OP_READ, this);
public void adjustBuffer(int len) {
if (len > buffer.position()) {
ByteBuffer newBuffer = ByteBuffer.allocate(len);
buffer.flip();
newBuffer.put(buffer);
buffer = newBuffer;
// public void sendData(ByteBuffer buff) throws IOException{
// buffer.clear();
// continueRead();
// OutputWriter.flushChannel(channel, buff);
public ProtocolHandler getProtocolHandler() {
return protocolHandler;
public void setProtocolHandler(ProtocolHandler protocolHandler) {
this.protocolHandler = protocolHandler;
public void close() {
buffer = null;
filters = null;
try {
channel.close();
} catch (Throwable e) {
channel = null;
selector = null;
protocolHandler = null;
// attributes = null;
public void setSslc(SSLContext sslc) {
this.sslEngine = sslc.createSSLEngine();
System.out.println("setsslc,sslengine=" + sslEngine);
sslEngine.setUseClientMode(false);
sslEngine.setNeedClientAuth(false);
sslEngine.setWantClientAuth(false);
// SSLSession session1 = sslEngine.getSession();
//ByteBuffer myAppData = ByteBuffer.allocate(sslEngine.getSession().getApplicationBufferSize());
ByteBuffer myNetData = ByteBuffer.allocate(sslEngine.getSession().getPacketBufferSize());
ByteBuffer peerNetData = ByteBuffer.allocate(sslEngine.getSession().getPacketBufferSize());
try {
doHandshake(channel,sslEngine,myNetData,peerNetData);
} catch (Exception e) {
// TODO Auto-generated catch block
e.printStackTrace();
HandshakeStatus hs = sslEngine.getHandshakeStatus();
System.out.println("in setsslcontext,shakehand status=" + hs);
//writingEncryptedData = false;
public SSLEngine getSslEngine() {
return sslEngine;
public int read0() throws IOException {
int dd = channel.read(inNetBuffer);
SSLEngineResult res = sslEngine.unwrap(inNetBuffer, buffer);
System.out.println("in read0,status=" + res);
return dd;
public void sendData(ByteBuffer buff) throws IOException {
//sslEngine.wrap(buff, outNetBuffer);
OutputWriter.flushChannel(channel, outNetBuffer);
void doHandshake(SocketChannel socketChannel, SSLEngine engine,
ByteBuffer myNetData, ByteBuffer peerNetData) throws Exception {
int appBufferSize = engine.getSession().getApplicationBufferSize();
ByteBuffer myAppData = ByteBuffer.allocate(appBufferSize);
ByteBuffer peerAppData = ByteBuffer.allocate(appBufferSize);
// Begin handshake
engine.beginHandshake();
SSLEngineResult.HandshakeStatus hs = engine.getHandshakeStatus();
// Process handshaking message
while ((hs = engine.getHandshakeStatus()) != SSLEngineResult.HandshakeStatus.FINISHED &&
(hs = engine.getHandshakeStatus()) != SSLEngineResult.HandshakeStatus.NOT_HANDSHAKING) {
switch (hs) {
case NEED_UNWRAP:
// Receive handshaking data from peer
if (socketChannel.read(peerNetData) < 0) {
// Handle closed channel
// Process incoming handshaking data
peerNetData.flip();
SSLEngineResult res = engine.unwrap(peerNetData, peerAppData);
peerNetData.compact();
hs = res.getHandshakeStatus();
// Check status
switch (res.getStatus()) {
case OK :
// if( hs == SSLEngineResult.HandshakeStatus.FINISHED ){
// break myio;
break;
case BUFFER_OVERFLOW:
System.out.println("---------overflow-------");
break;
case BUFFER_UNDERFLOW:
System.out.println("---------underflow-------");
break;
// Handle other status: BUFFER_UNDERFLOW, BUFFER_OVERFLOW, CLOSED
break;
case NEED_WRAP :
// Empty the local network packet buffer.
myNetData.clear();
// Generate handshaking data
res = engine.wrap(myAppData, myNetData);
hs = res.getHandshakeStatus();
//HandshakeStatus hs2 = engine.getHandshakeStatus();
// Check status
switch (res.getStatus()) {
case OK :
// if( hs == SSLEngineResult.HandshakeStatus.FINISHED ){
// break myio;
myNetData.flip();
// Send the handshaking data to peer
while (myNetData.hasRemaining()) {
if (socketChannel.write(myNetData) < 0) {
// Handle closed channel
break;
// Handle other status: BUFFER_OVERFLOW, BUFFER_UNDERFLOW, CLOSED
break;
case NEED_TASK :
new Thread( sslEngine.getDelegatedTask() ).start();
// Handle blocking tasks
break;
// Handle other status: // FINISHED or NOT_HANDSHAKING
// Processes after handshaking
}I write a NIO Server program with SSL(JSSE).You have tackled a seriously difficult problem.
in my program ,hs = FINISH but hs2=NOT_HANDSHAKEYou mean NOT_HANDSHAKING.
why it isn't same value?Because FINISHED is only delivered once. See the Javadoc. -
According to digitalTV I do not get a btvision / freeview signal. Yet at present I get 37 TV stations with my vision box. The confusion seems to stem from digital saying I get my signal from the Kiddderminster/Sutton Coldfield transmitter which living in Highley near Bridgnorth is incorrect. I get my signal from Wrekin. But according to Freeview I should get 70+ stations from Wrekin (Inc. Radio) yet as previously stated I only get 37.
I have a wideband - hi gain rooftop aerial tuned to Wrekin yet my neighbour (20ft away) is tuned into Sutton Coldfield and gets the 70 freeview odd stations.......I cannot get Sutton Coldfield....... Am I missing a trick?
Thanks in advance for any sdvice............Freeview can be very flaky during certain weather conditions -in my opinion- I have a portable TV running freeview upstairs from an indoor arial which most of the time is fine but in certain weather conditions some channels are unwatachable or just dissapear all together only to return once the weather changes.
Some freeview receivers require a much stronger signal than others and I suspect the one in the vision boxes likes signal that is both strong and good quality. the weather I think affects the quality of the signal rather than the strength. I can pick up a digital signal from 2 transmitters on the main TV but get more channels on the TV then I do on the vision box and have done since the switchover around here which was quite a while ago. I live right on the border of North Wales and until DSO we couldnt get Welsh Channels but now we can on the TV I get all the welsh channels but on the Vision box only BBC Wales
If you have already been through DSO then probably the direction your arial is pointing in could do with tweaking the signal pre DSO is quite often not as strong as post DSO. If you lose channels altogether after being able to get them, if you can check with a different freeview receiver it could be that once the weather condition changes and the signal quality improves they will return. Getting some but not all of the channels could be an indication that the signal is not high enough quality as does channels breaking up once DSO is completed the signal is stronger and better quality
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