Calls to phones dropping after 1 minute
Hi I am using an Iphone 5 and when i am making a call to mobile/landline after 1 minute call just hang up and I can start hearing a recording of conversation whats been talked within that 1 minute. Any help on this please??
I am also facing the same issue, I have a Skype subscription to make calls to India and I also have Skype credit.
I have just installed the skype on my new Iphone 5, When I am making call to land landlines or mobile in India and when the recipient picks up the phone Skype the call dropping immediately every time. This has been happening for the last six weeks. An occasional call gets through but my experience is that 99% of them don't. Please help me to resolve this issue.
Similar Messages
-
The inbound calls to our call center is drop after putting it on hold or transfer
Dear All;
Good day
The inbound calls to our call center is drop after putting it on hold or transfer the call to another agent. The MOH file is playing till 21 sec only then call drop . the agent cant resume the call again. The MOH file is running from Gateway (multicast).
No problem in outbound calls.
I urgent need you help
Should you require any more information , please do not hesitate to contact me.
Thanks & Best Regards,
Muhammad Fathy,
IT Network Manager
ALEXBANK
A subsidiary of Intesa Sanpaolo Group
Head office: B210-F1, Smart Village , KM 28 Cairo-Alex Desert Road, Egypt.
Cell: +201017288844.
Office: +202-35311300 Ext: 8090.
eMail: [email protected]
i To maintain a paperless environment, please don't print this e-mail unless you really need to.Typically you have a codec or media resource issue to track down. IE, MTP, region, location, gateway trunk to trunk to call or something in that area. Bypass UCCX and do the same call without this app... does it happen with a normal call?
-
Dial tone continues after call is answered then call drops after 1 minutes
This morning I have had problems with calling skype contacts. After I initiate the call the person on the other end picks up but the dail tone is still in the background. After a minute or so the dial tone go to busy and the entire call drops.
I downloaded new version of Skype and it works. Strange that it didn't auto update since 6.21 even thought auto update was on. Now it is 6.22.
-
SPA112 line drops after 1 minute
Hi All,
I have an SPA112 and its all configured to work as per my network settings. However, it keeps cutting out the line after every minute. Not sure what the real problem is. I've updated the firmware to the latest..still the same thing...Any thoughts, anyone?
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 lenovo A-730 tab and I have been facing issues with my wifi.
Here are the details of my tab:-
Model number - LenovoA3300-GV
For the first five minutes, it works like a charm, then the connectivity goes away. Though, the wifi signal strength shows full.
Again, if I turn wifi off and again on, it works for another five minutes again and then stops working.
This is a consistent issue.
Somehow, it seems I am missing some settings or some software issue.
Please guide me in this regard to get this issue of mine fixed.Hi,
I tired the steps given in this link.
http://www.xda-developers.com/kitkat-wifi-drop-fix/
The steps are:-
1- goto Settings > WiFi settings
2- long-press on Connection
3- Select "Modify network config"
4- check the "Show advanced options" tick.
5- select proxy settings to Manual
6- proxy host name , input the value from 192.168.1.1 to 192.168.1.225
7- Proxy port = input one of them "8080" , "3128" , "80"
8- press the "Save".
9- turn wifi off and the On, you will not see the message ""Sign into Wi-Fi network"
10-now repeat the steps from 1 ~ 4
11- and revert back the proxy settings to "None", its Done.
12- you will be fine with this network.
But no luck, after five minutes, the wifi again goes off after five mins and again comes up on doing disconnect/reconnect.
One more thign did not understand here is as per step 6 to 8, we are changing the proxy and doing the required setting and saving it.
Then, in Step 10, we are reverting it back to the original state as we are setting proxy to none again.
Will it solve the purpose or am I missing something.
My issue is still not fixed -
Calling India will disconnect after one minute
dear Customer service,
when we are calling to india and it will disconnect call after 1 minute and then again we have to call
please solve this issue immediately and we cannot go like that.
awaiting your reply soon
This post was transferred from its previous location to create its own new topic here; its subject and/or title has been edited to differentiate the post from other inquiries and to reflect the post's content.callfailure wrote:
but I am sorry to write the service is becoming worst.
Hello
We have answered this query many times. The problem has nothing to do with Skype or Microsoft. The local telco authorities in Saudi Arabia where you are located frequently disconnect outbound VoIP traffic to protect their own state run monopoly. Try accessing the internet via a VPN.
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
Calls dropped after free minutes run out
Hello,
There is already a post on this subject, but it's back in 2013, so I figured re-raising it would be worthwhile, particularly given how annoying it is. Here is the original post:
http://community.skype.com/t5/Rates-and-subscriptions/After-Office-365-Minutes-Run-Out-Call-Drops-Bu...
Basically, I have some free minutes given to me every month thanks to my subscription with Office 365. When these minutes run out, my call drops - even though I have cash credit on my account. The fact it does not do a seamless transfer from my account of "free minutes" to my account of "cash" is extremely annoying. Given that most of my calls are to companies (e.g. to make payments, transfers, talk with the bank, etc) it is extremely annoying to have to reconnect with the caller, through the machine, give address/card details again, etc... And this is something that happens once every single month!! It's very annoying, and really stupid that Skype won't fix this. It's hardly rocket science to make both accounts refer to a single consolidated account.
With increasing amounts of competition out there this is a good enough reason to take my business elsewhere. Even if it's slightly more expensive, whilst I can understand that issues sometimes take a while to fix - this seems to me like a deliberate decision not to do anything about a customer complaint, and I can't support such an attitude from a company.
Skype - please tell me that this will post will change your mind about this issue, and that you will add this to your list of things to fix.
EMiket67m wrote:
1. It is clear that when a 60 mins subscription comes to and end the call drops rather than the call continuting and moving to Skpe credit. It is at best irritating and should be fixable.
2. If nothing else it would be helpful to have a warning flag up that the call is about to drop, particular if the call is important. Sort it out Skype!
Hello and welcome to the Skype Community.
1. Unfortunately, calls can't be continued seamlessly when the call charging protocol changes. In this case the change is from Subscription to Credit.
2. I don't think that's possible but I'll check and advise.
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
Cisco SX20 Call drops after 15 minutes Please help urgent !!
Hi All,
We have Below scenario
SX20----> MCU<----- Polycom
SX20 -----> Polycom
Issue: When we make a Call from SX20 to Polycom IP to IP call or even using MCU Cisco SX20 gets disconnected after fixed interval ( 15 - 20 minutes)
Calls from Other endpoints ---> Polycom to polycom with MCU and also 1:1 ,1: multiple over MCU works fine.
Only issue is when Cisco SX20 comes in to the picture.
Please sugest is it a known issue?
Or doi we have to update firmware/licenses or any other software in cisco SX20?
Or is it a bug?
Or do we need to check on firewall level
The SX20 has the latest firmware already.
Please help !!!!!!!!
Regards,
AjithAs Martin has said above, the 15 minute disconnection issues are usually a network issue.
Are all your devices on the same LAN, or are they at different locations? Are you using any call control (VCS/CUCM), or are the devices standalone? What protocol are you using (H.323/SIP)?
If you're using NAT, have you got the NAT settings set up correctly on your devices?
What model device is the Polycom? Is the software on the Polycom up to date?
There are many questions which we can ask, but need to understand your environment a little more.
Wayne
Please remember to rate responses and to mark your question as answered if appropriate. -
Upgraded to 5S calls drop after a minute or so
All calls to Miami effected - local calls (Orlando) don't seem impacted
By connecting to ethernet and wifi you're creating a network loop back on itself. I've experienced problems like this in the past ... nothing really to do with Apple TV. Try one or the other, WiFi or Ethernet ... assuming you have a wireless router that's hooked up to you high speed modem.
-
Internet connection drops after 1 minute
Using iPhone 4, version 4.3.3 - I have been using wi-fi connectivity without issue for several months at home and other public locations. Currently having issues at new work location with keeping a persistent internet connection using company wi-fi. After device wakes up, I'm able to browse the internet without problems for 1-2 minutes, then when loading a page, will get the error that Safari can't open the page because no internet connection is available. I still have connectivity to the wi-fi network, and can get email from company Exchange server, but can't get access to anything past the company network. I don't understand why it will work for a couple of minutes and then lose internet connection.
Any ideas?Hello and thank you for the response. The only way I can get these iMacs to stay connected to the internet is via Static IP. The previous post states that static doesn't work but I didn't fill out all the fields correctly, like DNS and search domain.
Both Ethernet and Wi-Fi pick up an IP address via DHCP and after a few minutes the IP is dropped then a self-assigned 169 address appear. With Static IP, we're good to go.
I've tried DHCP with manual address.
Editied the location on network preferences.
I've disabled "use passive FTP mode"
I connected Windows machines to the same network port as these iMacs and there are no issues.
OSX 10.7
Thanks again,
Oz -
VPN connection drops after 1 minute or less
Hi All,
not sure if this is a server-side or client-side issue. Trying to establish a VPN connection from an MBP to my Mac Mini server (L2TP over IPSec, Shared Secret w/ Password authentication) I manage to connect, authenticate and establish a connection. However, after about 45-60 seconds I see no more traffic (from the remote LAN) coming through to the MBP. The status indicator still shows that the tunnel is up but I just don't get any further response from the server. I can still get to the web, however, the traceroute seems to indicate that this traffic is flowing outside the VPN tunnel (whether the "Send all traffic over VPN connection" option on the MBP is selected or not).
I've tried this with all combinations of 10.6.5 and 10.6.6 to no avail. It is possible to keep the tunnel alive just a little longer if I keep a ping to the server going on the MBP, but this eventually gets lost as well.
I've determined that there are some changes in the routing tables on the MBP whilst this is going on:
Last login: Thu Jan 20 09:45:37 on ttys001
guava:~ mark$ netstat -rn
Routing tables
## Normal Connection
Internet:
Destination Gateway Flags Refs Use Netif Expire
default 10.149.16.1 UGSc 13 0 en1
10.149.16/22 link#6 UCS 2 0 en1
10.149.16.1 0:0:5e:0:1:8e UHLWI 13 17 en1 1038
10.149.18.96 127.0.0.1 UHS 0 0 lo0
10.149.19.255 ff:ff:ff:ff:ff:ff UHLWbI 0 6 en1
127 127.0.0.1 UCS 0 0 lo0
127.0.0.1 127.0.0.1 UH 0 176 lo0
169.254 link#6 UCS 0 0 en1
192.168.0 link#8 UC 1 0 vmnet8
192.168.0.255 ff:ff:ff:ff:ff:ff UHLWbI 0 6 vmnet8
192.168.106 link#7 UC 1 0 vmnet1
192.168.106.255 ff:ff:ff:ff:ff:ff UHLWbI 0 6 vmnet1
guava:~ mark$ netstat -rn
Routing tables
## VPN Connection Alive
Internet:
Destination Gateway Flags Refs Use Netif Expire
default 10.149.16.1 UGSc 2 0 en1
default 192.168.1.180 UGScI 0 0 ppp0
10.149.16/22 link#6 UCS 3 0 en1
10.149.16.1 0:0:5e:0:1:8e UHLWI 3 0 en1 1198
10.149.18.96 127.0.0.1 UHS 0 0 lo0
10.149.19.255 ff:ff:ff:ff:ff:ff UHLWbI 1 8 en1
xx.xxx.xxx.xxx 10.149.16.1 UGHS 1 34 en1
127 127.0.0.1 UCS 0 0 lo0
127.0.0.1 127.0.0.1 UH 0 176 lo0
169.254 link#6 UCS 0 0 en1
192.168.0 link#8 UC 1 0 vmnet8
192.168.0.255 ff:ff:ff:ff:ff:ff UHLWbI 0 6 vmnet8
192.168.1 ppp0 USc 0 0 ppp0
192.168.1.180 192.168.1.160 UH 7 33 ppp0
192.168.106 link#7 UC 1 0 vmnet1
192.168.106.255 ff:ff:ff:ff:ff:ff UHLWbI 0 6 vmnet1
guava:~ mark$ netstat -rn
Routing tables
## VPN Connection but no net traffic coming back from server
Internet:
Destination Gateway Flags Refs Use Netif Expire
default 10.149.16.1 UGSc 11 0 en1
default 192.168.1.180 UGScI 0 0 ppp0
10.149.16/22 link#6 UCS 1 0 en1
10.149.16.1 0:0:5e:0:1:8e UHLWI 13 0 en1 1154
10.149.18.96 127.0.0.1 UHS 0 0 lo0
xx.xxx.xxx.xxx 10.149.16.1 UGHS 2 1644 en1
127 127.0.0.1 UCS 0 0 lo0
127.0.0.1 127.0.0.1 UH 0 176 lo0
169.254 link#6 UCS 0 0 en1
192.168.0 link#8 UC 0 0 vmnet8
192.168.1 ppp0 USc 1 0 ppp0
192.168.1.180 192.168.1.160 UH 21 1635 ppp0
192.168.106 link#7 UC 0 0 vmnet1
The MBP is connected via WiFi to a router which gives it 10.149.18.96. The remote network is on 192.168.1.0/24. The VPN server (as well as the DNS and DHCP server) are on machine 192.168.1.180 (the Mac Mini).
Neither the MBP nor the server logs really contain anything useful which would indicate why this is failing.
It looks as though this issue has been there for quite a long time in some form:
- http://discussions.apple.com/thread.jspa?threadID=1208715
- http://discussions.apple.com/message.jspa?messageID=10958263
- http://discussions.apple.com/thread.jspa?threadID=2462874
This is incredibly frustrating and I cannot understand while Apple has not done anything about this. Is there anyone here that is having the same problem and has possibly made some headway on this? Any suggestions would be highly appreciated!
Thanks.
MarkFirst guess would be a faulty Openreach VDSL modem (the white box).
Call 0800 111 4567, explain the problem. Be polite but firm and insist on an engineer visit. The engineer will test the line and probably replace the modem. -
Airplay drops after 1 minute with pioneer receiver
My iPad connects to my airplay enabled receiver pioneer. Connects and plays for about 1 minute then it drops and reverts back to the iPad. All firmware is up to date
The pioneer tab disappears from the airplay screen as a source then comes back.Do a reset (Hold Sleep/Wake and Home buttons about 10 secs or more till Apple logo appears, ignore the Slide to Power Off that appears)
Note: You will not lose any data. -
Re: Calling India will disconnect after one minute
Dear Norman,
It is disappointing to know that you dont reply to such kind of question. I am paid customer to SKYPE and using it from the time when it was not of microsoft. We as customer thought that with microsft it will be better but I am sorry to write the service is becoming worst. I sent many mails to customer support but not got any reply from them.
I am a paid and regular customer of SKYPE but your service is getting weaker and weaker day by day.
I wish Microsoft to address such issue with some authentic reply.
Wish you best luck and good time.
Sirfrazcallfailure wrote:
but I am sorry to write the service is becoming worst.
Hello
We have answered this query many times. The problem has nothing to do with Skype or Microsoft. The local telco authorities in Saudi Arabia where you are located frequently disconnect outbound VoIP traffic to protect their own state run monopoly. Try accessing the internet via a VPN.
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you. -
Hi All,
We have Below scenario
SX20----> MCU<----- Polycom
SX20 -----> Polycom
Issue: When we make a Call from SX20 to Polycom IP to IP call or even using MCU Cisco SX20 gets disconnected after fixed interval ( 15 - 20 minutes)
Calls from Other endpoints ---> Polycom to polycom with MCU and also 1:1 ,1: multiple over MCU works fine.
Only issue is when Cisco SX20 comes in to the picture.
Please sugest is it a known issue?
Or doi we have to update firmware/licenses or any other software in cisco SX20?
Or is it a bug?
Or do we need to check on firewall level
The SX20 has the latest firmware already.
Please help !!!!!!!!
Regards,
AjithPlease refer to your duplicate thread in the TelePresence section of the forums: https://supportforums.cisco.com/discussion/12383611/cisco-sx20-call-drops-after-15-minutes-please-help-urgent.
Wayne
Please remember to rate responses and to mark your question as answered if appropriate. -
Phone won't make outbound calls. can't call verizon support. this after being on hold for 17 minutes in an attempt to fix voicemail notification. this is a nightmare and Verizon is unreachable. Any way to get Verizon wireless to fix my service when they can't be called (because their phone service doesn't work)?
Well 1, I guess it's not that important to you, 2, customer support generally closes at 11pm in the time zone you are in, 3, toll free and local calls are free in hotels, 4 there is limited support after hours.
Maybe you are looking for
-
Can i use purchased sound effects off iTunes in an iOS app? - copyright issue
basically I'm making an iOS app, and i need a sound effect that i purchased off the iTunes store, aka from some company. Can i use this sound effect in my app without being sued a lot of money? If i payed for the file i should be okay correct?
-
I bought apple tv thinkng it would save me from my college life of not being able to have cable but it has cost more of a problem Be ause my campus is so large we have passwords to log into our network. When I called the tech people they siad if i ha
-
Right Click in Windows Vista (Bootcamp)
the two finger tapping works GREAT in OSX, but can I do something similar while running vista? The two finger gesturing seems to work nice for scrolling, what about the right click? Or Is there atleast a 3rd party program that will allow me to map th
-
New iOS7 update won't let me restrict iTunes radio/images for my kids. Help?
The new iOS7 update won't let me restrict iTunes radio/images for my kids. I've never had this problem with Apple before. I've always had the ability to restrict the iTunes store, and anything else I needed to restrict. I'm not a religous nut, I just
-
Airport extreme can't see any wireless networks.
Hi, I have an iMac that has always used an ethernet connection to the internet, but now I want to connect wirelessly. But it doesn't seem to "see" any networks, mine or otherwise, which work fine on my other apple devices. Does the fact that my wirel