Calls from Etisalat PSTN to FXO to voicemail do not disconnect

I have a tricky issue where outside caller calls in and when the call is forwareded to voicemail because of CFNA, the FXO do not disconnect. I have a setup where a Etisalat Analog lines are directly connected to UC560 FXO ports using RJ11.
When a call comes in over the PSTN to an FXO port on my UC560 and the call is answered by the user and after that user goes on-hook, FXO disconnects or gets released normally . When the user does not answer the call and becuse of CFNA timeout the call is forwarded to users voicemail box , then CUE answers, a voicemail is recorded, but when the calling party hangs up FXO doesnot disconnect instead it stays in OFFHOOK state (HANGS). Because of this no more calls are possible on that FXO line. I have to issue shut and no shut command on the FXO to get it released.
The IOS version as follows
uc500-advipservicesk9-mz.151-2.T4
and CME and CUE version are as follows 8.0.2
The follwing is the configuration on my UC560
voice class dualtone-detect-params 1
freq-max-deviation 25
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 4
voice class custom-cptone UAE-CUSTOM-SIEMENS
dualtone disconnect
frequency 425
cadence 425 325 250 500
voice-port 0/1/0
trunk-group ALL_FXO 61
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM-SIEMENS
supervisory dualtone-detect-params 1
no battery-reversal
input gain 14
cptone AE
timeouts call-disconnect 2
timeouts wait-release 2
timing min-ring 62
connection plar opx 202
description Configured by CCA 4 FXO-0/1/0-Custom-BG
caller-id enable
This the dial peer for viocemail
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 399
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad

HI David,
Here is the debug vpm signal information that i have taken for two scenarios
the configuration on voice-port is as follows
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 425
cadence 400 350 225 525
voice-port 0/1/1
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone AE
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 202
description Configured by CCA 4 FXO-0/1/1-Custom-OP
caller-id enable
the came same configuration above with battery reversal answer but no use sitll same issue.
The other tricky thing that is happening is when the call is forwarded to voicemail of the user and after the external caller disconnects the FXO on UC540 does not disconnect immediately, instead it disconnects after the default messgae size is reached. ie the default message size of voicemail box is 240 sec so after 240 sec the FXO port is released or disconnects and a large amount of silence is being recorded in the users mailbox for about 240 seconds.
the following is the debug capture taken
========================================================================================================================
When the call comes in and call is forwarded to voicemail because of CFNA on the user phone
UC_540#
000691: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
000692: htsp_timer - 125 msec
000693: htsp_process_event: [0/1/1, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
000694: htsp_timer - 10000 msec
000695: htsp_timer3 - 5600 msec
000696: [0/1/1] htsp_start_caller_id_rx:BELLCORE
000697: htsp_start_caller_id_rx create dsp_stream_manager
000698: [0/1/1] htsp_dsm_create_success returns 1
UC_540#
000699: htsp_process_event: [0/1/1, FXOLS_RINGING, E_DSP_SIG_0100]
000700: fxols_ringing_not
000701: htsp_timer_stop
000702: htsp_timer - 10000 msec
000703: [0/1/1] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX
000704: htsp_process_event: [0/1/1, FXOLS_RINGING, E_HTSP_CALLERID_RX_DONE]
000705: htsp_timer_stop
000706: htsp_timer_stop3
000707: [0/1/1] htsp_stop_caller_id_rx. message length 25htsp_setup_ind
000708: [0/1/1] get_fxo_caller_id:Caller ID received. Message type=128 length=25 checksum=B1
000709: [0/1/1] Caller ID String 80 16 01 08 30 34 31 39 31 33 34 30 02 0A 30 35 30 39 35 37 38 33 30 39 B1
000710: [0/1/1] get_fxo_caller_id calling num=0509856909 calling name= calling time=04/19 13:40
000711: fxols_callerid_done: call being answered
000712: [0/1/1] htsp_dsm_close_done
000713: htsp_process_event: [0/1/1, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
000714: fxols_wait_setup_ack:
000715: htsp_timer - 6000 msec
000716: htsp_timer_stop
UC_540#3
000717: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_prochtsp_setup_req
000718: htsp_process_event: [50/0/30.1, EFXS_ONHOOK, E_HTSP_SETUP_REQ]efxs_onhook_setup
000719: htsp_ephone_start_caller_id_tx calling num=90509578309 calling name = called num=201 orig called num=
000720: [50/0/30.1] set signal state = 0x0 timestamp = 0
000721: efxs_onhook_setup: local target is available
htsp_alerthtsp_alert_notify
000722: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
UC_540#
000723: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000724: htsp_timer_stop
000725: htsp_timer_stop2
UC_540#
000726: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000727: htsp_timer_stop2
000728: htsp_timer - 6000 msec
UC_540#
000729: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000730: htsp_timer_stop
000731: htsp_timer_stop2
UC_540#
000732: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000733: htsp_timer_stop2
000734: htsp_timer - 6000 msec
UC_540#
000735: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000736: htsp_timer_stop
000737: htsp_timer_stop2
UC_540#
000738: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000739: htsp_timer_stop2
000740: htsp_timer - 6000 msec
UC_540#
000741: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000742: htsp_timer_stop
000743: htsp_timer_stop2
UC_540#
000744: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000745: htsp_timer_stop2
000746: htsp_timer - 6000 msec
UC_540#
000747: htsp_timer_stop3
000748: htsp_process_event: [50/0/30.1, EFXS_WAIT_OFFHOOK, E_HTSP_RELEASE_REQ]efxs_waitoff_release
000749: [50/0/30.1] set signal state = 0x4 timestamp = 0
000750: htsp_call_bridged invoked
000751: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
000752: [0/1/1] set signal state = 0xC timestamp = 0
000753: htsp_timer_stop
000754: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
000755: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
000756: htsp_timer_stop2
000757: htsp_timer_stop2
UC_540#
After the voicemail box, default message size is reached ie after 240 seconds the FXO port disconnects and following is the continuation of debug vpm signal cmd.
UC_540#
000758: htsp_timer_stop3 htsp_setup_req
000759: htsp_process_event: [50/0/300.1, EFXS_ONHOOK, E_HTSP_SETUP_REQ]efxs_onhook_setup
000760: htsp_ephone_start_caller_id_tx calling num=399 calling name = called num=A800201 orig called num=
000761: [50/0/300.1] set signal state = 0x0 timestamp = 0
000762: efxs_onhook_setup: local target is available
htsp_alerthtsp_call_feature:feature 25
htsp_call_feature: caller id enable 0x3 call_connected 0
000763: htsp_process_event: [50/0/300.1, EFXS_WAIT_OFFHOOK, E_HTSP_CALLERID_WAITING]
000764: efxs_callerid_update
000765: efxs_callerid_update process caller_id_string
000766: efxs_callerid_update process caller_id_string OK
UC_540#
000767: efxs_callerid_update number= [399] name= []
UC_540#
000768: htsp_timer_stop3
000769: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
000770: htsp_timer_stop
000771: htsp_timer_stop2
000772: htsp_timer_stop3
000773: [0/1/1] set signal state = 0x4 timestamp = 0
000774: htsp_timer - 2000 msec
000775: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
UC_540#
000776: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
000777: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
000778: htsp_timer_stop3
000779: htsp_process_event: [50/0/300.1, EFXS_WAIT_OFFHOOK, E_HTSP_RELEASE_REQ]efxs_waitoff_release
000780: [50/0/300.1] set signal state = 0x4 timestamp = 0
UC_540#
Can any one let me know what is happing here. when the call is forwarded to voicemail of the user, why FXO Port on UC540 not getting disconnected soon after the external caller disconnects the call.and insted it disconnects approximately after 240 seconds of call forwarded to voicemail.
I tried the same cofiguration as above with battery reversal answer in voice-port configuration but no use sitll same issue.

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4FXS-DID configuration problem using BRI to receive/send calls from/to PSTN

Hi to all.
Follow is the partial configuration of my CME.
It has a 4FXS card to use analog phones and faxes, and 4 ISDN Basic Rate interfaces.
On each port of the 4FXS DID card happens the following:
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CME#sh ver
Cisco IOS Software, 2800 Software (C2800NM-IPVOICEK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2)
ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)
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System returned to ROM by power-on
System restarted at 18:54:45 CDT Fri Aug 2 2013
System image file is "flash:/c2800nm-ipvoicek9-mz.124-24.T3.bin"
Cisco 2811 (revision 53.50) with 249856K/12288K bytes of memory.
Processor board ID FTX1120A08L
2 FastEthernet interfaces
4 ISDN Basic Rate interfaces
16 terminal lines
4 Voice FXS interfaces
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
1948656K bytes of USB Flash usbflash0 (Read/Write)
497448K bytes of ATA CompactFlash (Read/Write)
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CME#sh telephony-service
CONFIG (Version=7.1)
=====================
Version 7.1
Cisco Unified Communications Manager Express
isdn switch-type basic-net3
voice translation-rule 1
rule 1 /^2929091$..$/ /\1/
rule 2 /^2929091$.$/ /\1/
rule 3 /^02929091$..$/ /\1/
rule 4 /^02929091$.$/ /\1/
voice translation-rule 2
rule 2 /$.*$/ /02929091\1/
voice translation-rule 10
rule 1 /$^......$$/ /0\1/ type national national plan isdn isdn
rule 8 /$^......$$/ /00\1/ type international international plan isdn isdn
voice translation-profile PSTN-IN
translate calling 10
translate called 1
voice translation-profile PSTN-OUT
translate calling 2
interface BRI0/0/0
description Isdn channels 1 & 2
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/0/1
description Isdn channels 3 & 4
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
isdn static-tei 0
interface BRI0/1/0
description Isdn channels 5 & 6
no ip address
isdn switch-type basic-net3
isdn timer T310 60000
isdn overlap-receiving T302 3000
isdn point-to-point-setup
isdn incoming-voice voice
isdn send-alerting
isdn sending-complete
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isdn timer T310 60000
isdn overlap-receiving T302 3000
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isdn incoming-voice voice
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station-id number 40
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station-id number 35
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station-id name FAX_1
station-id number 50
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station-id name FAX_1
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progress_ind progress enable 8
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dial-peer voice 1003 pots
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destination-pattern 0T
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progress_ind progress enable 8
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destination-pattern 2929091..
incoming called-number .T
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destination-pattern 0T
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progress_ind progress enable 8
progress_ind connect enable 8
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dial-peer voice 2005 pots
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destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
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destination-pattern 35$
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dial-peer voice 2040 pots
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incoming called-number 40
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direct-inward-dial
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dial-peer voice 2050 pots
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destination-pattern 50
incoming called-number 50
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direct-inward-dial
port 0/2/2
dial-peer voice 2060 pots
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destination-pattern 60
incoming called-number 60
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direct-inward-dial
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CME#

Hi Paolo.
Sorry for the delay. I was on holiday.
I would like to keep separate incoming calls from outgoing calls.
So I decided to keep two dial peers for every BRI interface.
I followed your suggestion on eliminate commands:
"incoming called-number" on FXSs,
various "progress_ind" on BRIs
Also I eliminated "direct-inward-dial" on FXSs,
Today I reconfigured and right tested the dial peers as following:
!   Bri Interfaces
dial-peer voice 1001 pots
description **Sends call to PSTN line 1-2**
destination-pattern 0T
port 0/0/0
dial-peer voice 2001 pots
description **Receives calls coming from PSTN line 1-2**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/0
dial-peer voice 1003 pots
description **Sends call to PSTN line 3-4**
destination-pattern 0T
port 0/0/1
dial-peer voice 2003 pots
description **Receives calls coming from PSTN line 3-4**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/0/1
dial-peer voice 1005 pots
description **Sends call to PSTN line 5-6**
destination-pattern 0T
port 0/1/0
dial-peer voice 2005 pots
description **Receives calls coming from PSTN line 5-6**
destination-pattern 2929091..
incoming called-number .T
direct-inward-dial
port 0/1/0
!   FXS interfaces
dial-peer voice 2035 pots
description **Receives calls coming from PSTN to Extension 35**
destination-pattern 35
no digit-strip
port 0/2/1
dial-peer voice 2040 pots
description **Receives calls coming from PSTN to Extension 40**
destination-pattern 40
no digit-strip
port 0/2/0
dial-peer voice 2050 pots
description **Receives calls coming from PSTN to Fax 50**
destination-pattern 50
no digit-strip
port 0/2/2
dial-peer voice 2060 pots
description **Receives calls coming from PSTN to Fax 60**
destination-pattern 60
no digit-strip
port 0/2/3
Thanks a lot Paolo. :-)
Giorgio.

Calls from the PSTN disconnect before ringing

hi,
i need some urgent assistance please guys.
i have a 2851 SRST router with an ISDN30 (E1) to the PSTN. i can make outgoing calls but i cannot receive incoming calls. the SP is cable and wireless. attached is a copy of my config. am i missing something here? do i need to change my incoming dial peer to match the digits presented by the SP?
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Hi Aaron,
yes at about 22:30 last night it was resolved and it was exactly as we had said, the SP. i will definitely be updating the last discussion as soon as i get a chance. thanks for all the help by the way.
here is the debug.
GPO-VoiceGW-11#
*Jun 14 11:26:23.418: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x000A
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98382
                Exclusive, Channel 2
        Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have in-band info
        Called Party Number i = 0x80, '300'
                Plan:Unknown, Type:Unknown
*Jun 14 11:26:23.426: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x800A
        Channel ID i = 0xA98382
                Exclusive, Channel 2
*Jun 14 11:26:23.430: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x800A
        Cause i = 0x8081 - Unallocated/unassigned number
*Jun 14 11:26:23.446: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x000A
        Cause i = 0x8295 - Call rejected
*Jun 14 11:26:23.450: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x800A
i can change the dial-peers to match 73xx as the main number is 0141 568 7300 - 7349.
many thanks
Dimitri

Incoming calls from one contact goes directly to voicemail

We recently updated my mother-in-law's phone and had the contact list transferred at a Verizon store. We later learned that one of her contacts was being sent directly to her voicemail even though her phone was on she wasn't talking to anyone else. We noticed when that contact called, her phone would briefly display her name and then would display the Verizon logo like it had reset. Tech support couldn't help me and tried to forward my support call to someone else, but all the support lines were busy and I got tired of waiting on hold.
The solution we came up with was to delete that contact info and then re-enter it. That solved the problem. Evidently there was some glitch in the data when the contact list was transferred.

Thanks for sharing your solution - it may help someone else! And it looks like it was simple to do, and effective. Have a great week!

I have a new IPhone5c and do not receive texts or calls from my daughter's phone. We did not have this problem with my old Envy Touch. Verizon is our provider. Can you troubleshoot?

I have a new IPhone5c and do not receive texts or calls sent from my daughter's phone; I'll call this phone 2. We did not have this problem with my old Envy Touch. Verizon is our provider. Can you troubleshoot? My other daughter has a new IPhone5c too. She receives calls and texts from sent from phone 2. Any tips for remedying this problem?
Thank you.

Turn off iMessage in your phones settings. Apple devices can only use iMessage to other Apple devices.
Turning off iMessage will allow the messages to go as regular text messages.

Calling ID to PSTN with CME vs Ericsson MD110 and E1

Hi all,
We are having problems with calling party identification in the following scenario: some IP phones managed by a 2800 CME router. This is connected to an Ericsson MD110 PBX through a E1 with QSIG signalling. There is absolutely no problem when identifying both called and calling parties in both directions ingoing and outgoing from the router to the PBX private extensions. It also works fine when calling from the PSTN to direct public numbers redirected from the PBX to the IP phones extensions. However, when calling from these particular extensions, we want the calling party to identify with its particular public number, but instead it shows at the PSTN end with the main number of the ISDN jump group.
Anyone has experienced the same?
Thanks and regards,
Jose Soriano

Hi,
I'll try to be more clear. As I said, we have a CCME connected to an Ericsson MD100 via an E1 Qsig link. Then the MD110 goes to the PSTN also with an E1. The customer has several ISDN public numbers associated with that E1, one of them is the main ISDN group number, with which the originator of all the outgoig calls from the private numbers to the PSTN identify, except for some private extensions associated with one of the other direct public numbers: these can be reached directly through these numbers and also are identified with those numbers when calling to the PSTN.
Let's say we have 30 public numbers, from 555100 to 555129, beig 555100 the main number. Ext 101 is associated to public number 555101, and 102 with 555102. In the situation with no CCME, the call flow is as follows.
Private Ext. 101 (dials a public number) --> PBX EMD110 --> PSTN (displays 555101 as the caller ID)
PSTN dials 555101 --> PBX EMD100 --> Private Ext. 101
Now, let's include the CCME system. Let's say Ext. 102 is an IP Phone extension correctly routed between CCME and MD110. Also, ext. 102 should be identified to the PSTN as 555102.
Case A)
IP Phone 102 (dials a public number) --> CCME --> PBX EMD110 --> PSTN (displays 555100, and NOT 555102 as the caller ID)
Case B)
PSTN dials 555102 --> PBX EMD100 --> CCME --> IP Phone 102
The PRIVATE flow between CCME and MD110 is fine:
IP Phone 102 dials 101 --> CCME --> PBX EMD110 --> Private Ext 101 displays 102 as the CLID
Private Ext 101 dials 102--> PBX EMD100 --> CCME --> IP Phone 102, with 101 as the CLID
So, the problem is in what I called CASE A), calls do not idetify with the especific public number, but with the main group number when they are routed through the PSTN.
Regards,
Jose Soriano

Call into an MGCP controlled FXO port not connecting

my setup is
PSTN --- GW --- CUCM --- 6921 phone
when we call from the PSTN then call rings and shows connected on the PSTN phone after about 10 sec the call disconnects with fast busy but the 4 digit plar does not ring at all, outgoing calls are working fine. Need help in resolving this issue
Shabbar

FXO is configured as loop start as below, cannot run the debug isdn
R_SCI_KQ#debug isdn q931
^
% Invalid input detected at '^' marker.
Registration
Registered with Cisco Unified Communications Manager 192.168.0.26
IP Address
192.168.100.6
End-Point Name
AALN/S0/SU0/0@R_SCI_KQ.sciafrica.net
Description
Device Pool
DP_KQ -- Not Selected -- Cisco Unity voice Mail Ports DP_DAR DP_DRC DP_MOZ DP_ZAM UCCX
Common Device Configuration
< None >
Media Resource Group List
MRGL < None >
Packet Capture Mode
None Batch Processing Mode
Packet Capture Duration
Calling Search Space
< None > CSS_AAR CSS_Block-Line-Tz-Restricted CSS_Block-Line-Tz-UnRestricted CSS_CdPTP-HQ-GW CSS_CngPTP-HQ-GW CSS_CngPTP-HQ-Phones CSS_Dial-Device-Tz-PSTN CSS_HQ_Calling CSS_ITSP CSS_PSTN CSS_SAB_Callpark CSS_SUBSCRIBE_HQ_PHONE KQ_CSS LOGISTICS_CSS MANAGERS_CSS MOZ_CSS RECEPTION_CSS SALES_CSS STAFF_CSS TECHNICAL_CSS VMRestrictedCSS
AAR Calling Search Space
< None > CSS_AAR CSS_Block-Line-Tz-Restricted CSS_Block-Line-Tz-UnRestricted CSS_CdPTP-HQ-GW CSS_CngPTP-HQ-GW CSS_CngPTP-HQ-Phones CSS_Dial-Device-Tz-PSTN CSS_HQ_Calling CSS_ITSP CSS_PSTN CSS_SAB_Callpark CSS_SUBSCRIBE_HQ_PHONE KQ_CSS LOGISTICS_CSS MANAGERS_CSS MOZ_CSS RECEPTION_CSS SALES_CSS STAFF_CSS TECHNICAL_CSS VMRestrictedCSS
Location
Hub_None Kenya Mozambique Phantom Shadow Zambia
AAR Group
< None > AAR_GRoup
Network Locale
< None > Argentina Australia Austria Belgium Brazil Canada China Colombia Cyprus Czech Republic Denmark Egypt Finland France Germany Ghana Greece Hong Kong Hungary Iceland India Indonesia Ireland Israel Italy Japan Jordan Kenya Korea Republic Lebanon Luxembourg Malaysia Mexico Nepal Netherlands New Zealand Nigeria Norway Pakistan Panama Peru Philippines Poland Portugal Russian Federation Saudi Arabia Singapore Slovakia Slovenia South Africa Spain Sweden Switzerland Taiwan Thailand Turkey United Kingdom United States Venezuela Zimbabwe Itu Chile Bulgaria Croatia Romania Serbia and Montenegro United Arab Emirates Oman Kuwait Algeria Bahrain Iraq Mauritania Republic of Montenegro Morocco Qatar Republic of Serbia Sudan Tunisia Vietnam Yemen Lithuania Latvia Estonia
Use Trusted Relay Point
Off On Default
Transmit UTF-8 for Calling Party Name
Called Party Transformation CSS
< None > CSS_AAR CSS_Block-Line-Tz-Restricted CSS_Block-Line-Tz-UnRestricted CSS_CdPTP-HQ-GW CSS_CngPTP-HQ-GW CSS_CngPTP-HQ-Phones CSS_Dial-Device-Tz-PSTN CSS_HQ_Calling CSS_ITSP CSS_PSTN CSS_SAB_Callpark CSS_SUBSCRIBE_HQ_PHONE KQ_CSS LOGISTICS_CSS MANAGERS_CSS MOZ_CSS RECEPTION_CSS SALES_CSS STAFF_CSS TECHNICAL_CSS VMRestrictedCSS
Use Device Pool Called Party Transformation CSS
Enable Caller ID

System Condition is not picked from TL while generating calls from Plan

Hi,
I've marked System condition field at work order header as required.
However, the system doesn't pick up that field from Task List when generating call from maint plan.. that would still be not populated therefore preventing e.g. from bulk release..
Do you know why is that? looks like a system bug or so..

Hi Paul,
I have to revive that topic..
defining system condition as decribed before seems to be working only for work order being called from maintenance plan
on the other hand, for notification generated from plan and then transformed to work orders defining system condition at task list details at item level does not work..
also ,defining system condition at TL header doesn't help (like in order example)..
at the same time, defining system condition at operation level are getting transferred to order with no problem..
do you have an idea?
Thanks in advance

IPhone calls from my Mac

I have an iPhone 5 running 8.1.2 and my Mac is running Yosemite. I also have an iPad Mini running 8.1.2.
I think all of my settings are correct across all three devices and phone calls to my iPhone are appropriately being forwarded to my iPad. When I attempt to originate a call from my Mac, I get the error "iPhone not available - your phone and Mac must be on the same Wifi network." They ARE on the same network.
Anyone else encounter this issue? Is there a setting I am missing?

Handoff System preferences.
Handoff Continuity Troubleshooting

Keep getting "FaceTime failed" when trying to make calls from my iMac

I am trying to make FaceTime calls from my iMac.
So far it has not worked at all. No matter what number I try to call (I am calling people who have FaceTime enabled iPhones), it rings for a while and then I get "FaceTime failed".
Any ideas what it could be?

what happens when you use Image Capture (in your applications folder) and see what happens
LN

Ipod stuck in DO NOT Disconnect mode, will not eject from itunes

I just got this thing replaced, my Ipod can't disconnect from my computer! It is in DO NOT disconnect mode, I try to eject it from itunes and it says that files are being used by another application, I log out of my computer and it still won't let it go

So the iPod remains in My Computer and in iTunes and won't disconnect? You should try shutting down or restarting your computer.
Or does it disappear from My Computer and iTunes but still says Do Not Disconnect? » http://docs.info.apple.com/article.html?artnum=93492
When you try logging out » http://docs.info.apple.com/article.html?artnum=61014

Old library from mini wont transfer to nano :( - just says dont disconnect!

hi. i copied my 'old library' from my ipod mini to itunes library. i now have received an ipod nano. itunes has beeen installed and uninstalled and reinstalled. i've tried the 5 R's, however, i plug in my nano and it goes immediately to 'do not disconnect' - doesnt transfer any songs from my library and never stops saying 'do not disconnect'. HELP!?!!? this is very frustrating.
thanks in advance

Check the settings. How are you attempting to transfer songs, auto sync or manual.
If you are using iTunes from version 7 onwards, then when you connect the iPod the summary screen should automatically open.
On the 'summary' page you will see the options to
"open iTunes when this iPod is attached".
"only sync checked items".
"manually manage music".
"enable disk use". This will be automatically enabled if you have the iPod set to 'manual' and be 'greyed out'.
If you click on the 'music' tab, you will see the options to
"sync music" with either "all songs and playlists" or "selected playlists".
What do you have selected?

Calls from Etisalat PSTN to FXO to voicemail do not disconnect

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