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.
E
Miket67m 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.
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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
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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
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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
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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
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Apr 13 13:10:03 SPA112 kern.warning [17179684.260000] chan->kmode is present not null
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Apr 13 13:10:03 SPA112 kern.warning [17179684.268000] ###### sock_sendmsg return 172
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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
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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 -
Trying to call landlines with Android phone, but the call repeatedly drops after 5:02 (according to the call log). This happened with the latest Android version, so I tried downgrading but that didn't help.
The internet connection is stable, i.e. I don't experience any other drop in connectivity, especially every 5 minutes.
Any thoughts? What can I do to prevent this from happening?
LG 4X HD
Android 4.1.2
Skype 4.9.0.45564
Thanks!Hello and welcome to the Skype Community.
From your account it appears that a percentage of calls are dropped while others behave normally.
Checking for you.
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. -
Call dropped in 10 minutes - 3 times in a row!!
I was on the phone with my cousin and precisely in 10 minutes my call was dropped. This happened 3 times during our 30 minute conversation.
Did anyone else face this issue? I hope this is because of the strain on AT&T's network today with all the new activations.I'm having drops, not with any certain times. My first call dropped twice.
I'm also having issues with the data side of the network. None of the apps that require accessing the internet work. They all are timing out.
And my phone also has the issue with covering up the black strips on the sides causes signal to fade.
Safe to say I'm ****** about this whole situation. -
UC320 - Phone Calls Drop After Transferred
Good morning all -
I have my UC320 setup for a small law firm. I have all calls in normal business hours transfer to an automatic call service (help desk/reception). Once they receive the call they then forward it to the attorney in which they want to speak to. Once the call is forwarded, after approximately 9 minutes and 30 seconds the call drops. Does anyone know what this can be caused by?
Any and all help is greatly appreciated.Hello,
I personally make calls longer than 60 min, with CCME to some South American friends (trying to keep it vague to not release any personal info :)
Anyway, if you use ISDN, it is easy to see which side is dropping the call. Enter "debug isdn q931" and "term mon". This is relatively low output and can be left on for days. You can also log to router memory or syslog server.
If you don't have isdn you can still see many things with "debug vpm signal". It is a little more verbose but still not overwhelming.
Hope this helps, if so please rate post! -
Lenovo A--730 wifi drops after five minutes, reappears on disconnct/reconnect
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
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