Lync Calls to PSTN gets disonnected after few rings if user is outside the organisation.

We are able to call from Lync client to PSTN if user is inside office.
From outside office, it rings and the disconnects.

Hi,
Did you receive any error message from the Event Viewer of FE server and Mediation server?
Check if you received Event ID 11.
If yes, as the initiate a call to PSTN network requires Edge Server and Lync Pool to establish a connection, you should double check port configuration between Edge Server and FE server, Mediation server and FE server.
More details:
http://terenceluk.blogspot.com/2011/01/outbound-call-to-pstn-network-fails.html
Note: Microsoft is providing this information as a convenience to you. The sites are not controlled by Microsoft. Microsoft cannot make any representations regarding the quality, safety, or suitability of any software or information found there. Please make
sure that you completely understand the risk before retrieving any suggestions from the above link.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

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Analog line (FXO) Incoming calls getting connected after 3 rings

     HI,
we are having 4 Analog line (FXO)...Every time when callers call the number they hear 3 rings & after that call frwds to AA or any extension.
In show voice port summary, we can see that voice port is getting connect at the first ring but after 3 rings only phone rings.
here is the o/p of voice port.
Foreign Exchange Office 0/0/0 Slot is 0, Sub-unit is 0, Port is 0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar
Connection Number is 250
Initial Time Out is set to 15 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Companding Type is u-law
Region Tone is set for AE
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming): INCOMING_CallerID_PROFILE
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabled

hostname VGUAE001
no aaa new-model
clock timezone UAE 4 0
ip cef
ip domain name yourdomain.com
no ipv6 cef
multilink bundle-name authenticated
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
voice-card 0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
voice class cause-code 1
no-circuit
voice translation-rule 1112
rule 1 /^9/ //
voice translation-rule 3265
rule 1 // /9\1/
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 50
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
license udi pid CISCO2901/K9 sn FCZ173992Z8
hw-module pvdm 0/0
hw-module pvdm 0/1
username cisco privilege 15 secret 4 opjnnkXqCr4kCOa9DuALcNpBOMetBAc/usnpSWADsCI
username godiva privilege 15 secret 4 cH8b8z.ioYu/pMv/AKuEcBd/f6g9v/vm/s3aXeqUAd6
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address 192.168.31.2 255.255.255.0
ip helper-address 192.168.31.11
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 192.168.31.2
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http access-class 23
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
ip route 0.0.0.0 0.0.0.0 192.168.31.1
control-plane
voice-port 0/0/0
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
groundstart auto-tip
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/1
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/2
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 222
caller-id enable
voice-port 0/0/3
trunk-group ALL_FXO 64
translation-profile incoming INCOMING_CallerID_PROFILE
cptone AE
connection plar opx 250
caller-id enable
mgcp profile default
dial-peer voice 2000 voip
destination-pattern 2..
session target ipv4:192.168.31.11
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 10 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fire**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 997
forward-digits all
no sip-register
dial-peer voice 11 pots
trunkgroup ALL_FXO
description **CCA*UAE*International Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 900T
forward-digits all
no sip-register
dial-peer voice 12 pots
trunkgroup ALL_FXO
description **CCA*UAE*Eitisalat**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9101
forward-digits all
no sip-register
dial-peer voice 13 pots
trunkgroup ALL_FXO
description **CCA*UAE*Water or electrical emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 971
forward-digits all
no sip-register
dial-peer voice 14 pots
trunkgroup ALL_FXO
description **CCA*UAE*Police and emergencies**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 999
forward-digits all
no sip-register
dial-peer voice 15 pots
trunkgroup ALL_FXO
description **CCA*UAE*National area codes**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[1-579].......
forward-digits all
no sip-register
dial-peer voice 16 pots
trunkgroup ALL_FXO
description **CCA*UAE*Mobile Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 90[5-6][0-7].......
forward-digits all
no sip-register
dial-peer voice 17 pots
trunkgroup ALL_FXO
description **CCA*UAE*toll-free**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]00T
forward-digits all
no sip-register
dial-peer voice 18 pots
trunkgroup ALL_FXO
description **CCA*UAE*Fixed Line Numbers**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-8]T
forward-digits all
no sip-register
dial-peer voice 19 pots
trunkgroup ALL_FXO
description **CCA*UAE*808**
translation-profile outgoing OUTGOING_TRANSLATION_PROFILE
preference 5
destination-pattern 9808T
forward-digits all
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/0/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/0/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/0/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/0/2
no sip-register
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/0/3
no sip-register
Debug vpm signal:
Nov 23 19:31:31.556: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Nov 23 19:31:31.556: htsp_timer - 125 msec
Nov 23 19:31:31.684: htsp_process_event: [0/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Nov 23 19:31:31.684: htsp_timer - 10000 msec
Nov 23 19:31:31.684: htsp_timer3 - 5600 msec
Nov 23 19:31:31.684: [0/0/0] htsp_start_caller_id_rx:Mode BELLCORE. Alerting 0x1
Nov 23 19:31:31.684: htsp_start_caller_id_rx create dsp_stream_manager
Nov 23 19:31:31.684: [0/0/0] htsp_dsm_create_success returns 1
Nov 23 19:31:33.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:33.604: fxols_ringing_not
Nov 23 19:31:33.604: htsp_timer_stop
Nov 23 19:31:33.604: htsp_timer - 10000 msec
Nov 23 19:31:37.284: htsp_process_event: [0/0/0, FXOLS_RINGING, E_HTSP_EVENT_TIMER3]fxols_snoop_clid_stop
Nov 23 19:31:37.284: htsp_timer_stop3
Nov 23 19:31:37.516: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
Nov 23 19:31:39.604: htsp_process_event: [0/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
Nov 23 19:31:39.604: fxols_ringing_not
Nov 23 19:31:39.604: htsp_timer_stop
Nov 23 19:31:39.604: htsp_timer_stop3
Nov 23 19:31:39.604: [0/0/0] htsp_stop_caller_id_rx. message length 0htsp_setup_ind
Nov 23 19:31:39.604: [0/0/0] get_fxo_caller_id:Caller ID receive failed. parseCallerIDString:no data.
Nov 23 19:31:39.604: [0/0/0] get_local_station_id calling num= calling name= calling time=11/23 23:31 orig called=
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   cc_api_call_setup_ind_common:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Interface=0x3CE27724, Call Info(
   Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,
   Incoming Dial-peer=50, Progress Indication=ORIGINATING SIDE IS NON ISDN(3), Calling IE Present=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: cc_get_feature_vsa count is 1
Nov 23 19:31:39.604: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.604: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218944,feature_id:83
Nov 23 19:31:39.604: //83/B583C95F8093/CCAPI/cc_api_call_setup_ind_common:
   Set Up Event Sent;
   Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown))
Nov 23 19:31:39.608: [0/0/0] htsp_dsm_close_done
Nov 23 19:31:39.608: htsp_process_event: [0/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Nov 23 19:31:39.608: fxols_wait_setup_ack:
Nov 23 19:31:39.608: [0/0/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   Event=0x22ACD828
Nov 23 19:31:39.608: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search:
   Try with the demoted called number 250
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230F9C10
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/cc_process_call_setup_ind:
   >>>>CCAPI handed cid 83 with tag 50 to app "_ManagedAppProcess_Default"
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallProceeding:
   Progress Indication=NULL(0)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination=, Calling IE Present=FALSE, Mode=0,
   Outgoing Dial-peer=2000, Params=0x230FB0D0, Progress Indication=ORIGINATING SIDE IS NON ISDN(3)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   In: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCheckClipClir:
   Out: Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed)
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Destination Pattern=2.., Called Number=250, Digit Strip=FALSE
Nov 23 19:31:39.608: //83/B583C95F8093/CCAPI/ccCallSetupRequest:
   Calling Number=(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown),
   Redirect Number=, Display Info=
   Account Number=, Final Destination Flag=TRUE,
   Guid=B583C95F-53AC-11E3-8093-C8EEBDE4256A, Outgoing Dial-peer=2000
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/cc_api_display_ie_subfields:
   ccCallSetupRequest:
   cisco-username=
   ----- ccCallInfo IE subfields -----
   cisco-ani=
   cisco-anitype=0
   cisco-aniplan=0
   cisco-anipi=0
   cisco-anisi=0
   dest=250
   cisco-desttype=0
   cisco-destplan=0
   cisco-rdie=FFFFFFFF
   cisco-rdn=
   cisco-rdntype=0
   cisco-rdnplan=0
   cisco-rdnpi=0
   cisco-rdnsi=0
   cisco-redirectreason=0   fwd_final_type =0
   final_redirectNumber =
   hunt_group_timeout =0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   Interface=0x22847B14, Interface Type=1, Destination=, Mode=0x0,
   Call Params(Calling Number=,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=250(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
   Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=2000, Call Count On=FALSE,
   Source Trkgrp Route Label=ALL_FXO, Target Trkgrp Route Label=, tg_label_flag=1, Application Call Id=)
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :cc_get_feature_vsa malloc success
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: cc_get_feature_vsa count is 2
Nov 23 19:31:39.612: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Nov 23 19:31:39.612: :FEATURE_VSA attributes are: feature_name:0,feature_time:1025218720,feature_id:84
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccCallSetContext:
   Context=0x230FB080
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSaveDialpeerTag:
   Outgoing Dial-peer=2000
Nov 23 19:31:39.612: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Nov 23 19:31:39.612: htsp_timer - 120000 msec
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGetMediaClassTag:
   media class tag 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccSetMediaclassIp2ipTags:
   media class tags set: NR 0, ASP 0
Nov 23 19:31:39.612: //84/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.612: //83/B583C95F8093/CCAPI/ccGet_xc_nr_asp_info:
   media class tags: NR 0, ASP 0
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.620: //84/B583C95F8093/CCAPI/cc_api_call_proceeding:
   Interface=0x22847B14, Progress Indication=NULL(0)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_delay_xport:
   CallInfo(delay xport=TRUE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Interface=0x22847B14, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_alert:
   Call Entry(Retry Count=0, Responsed=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallAlert:
   Call Entry(Responsed=TRUE, Alert Sent=TRUE)htsp_alert_notify
Nov 23 19:31:39.628: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x5, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.628: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.628: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x5, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.672: //84/B583C95F8093/CCAPI/ccIsInfoRingback:
   Returning dpRingBack=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Interface=0x22847B14, Data Bitmask=0x1, Progress Indication=NULL(0),
   Connection Handle=0
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_connected:
   Call Entry(Connected=TRUE, Responsed=TRUE, Retry Count=0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_set_called_ccm_detected:
   CallInfo(called ccm detected=TRUE ccmVersion 3)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_call_notify:
   Data Bitmask=0x7, Interface=0x22847B14, Call Id=84
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Network, Params=0x0, Call Id=83
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID1=0x53, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   (confID=0xFFFFFFFF, callID2=0x54, gcid=B583C95F-53AC11E3-8093C8EE-BDE4256A, tag=0x0)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Conference Id=0xFFFFFFFF, Call Id1=83, Call Id2=84, Tag=0x0
Nov 23 19:31:39.700: htsp_call_bridged invoked
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0xFFFFFFFF
Nov 23 19:31:39.700: //84/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:
Nov 23 19:31:39.700: cc_api_get_xcode_stream : 4819
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=84)
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/ccConferenceCreate:
   Call Entry(Conference Id=0x21, Destination Call Id=83)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/ccConferenceCreate:
Nov 23 19:31:39.700: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=0x1, Fax Rate=0x1, Fax Version:=0, Vad=0x1,
   Modem=0x2, Codec Bytes=20, Signal Type=3)
Nov 23 19:31:39.700: //83/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.700: //84/B583C95F8093/CCAPI/cc_api_get_ssCTreRoutingNotSupported:
   CallInfo(ssCTreRoutingNotSupported=FALSE)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_get_ccm_detected:
   CallInfo(ccm detected=TRUE)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallNotify:
   Data Bitmask=0x7, Call Id=83htsp_call_service_msghtsp_call_service_msg not EFXS (2)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_process_notify_bridge_done:
   Conference Id=0x21, Call Id1=83, Call Id2=84
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=0x1, Fax Rate=0x2, Fax Version:=0, Vad=0x1,
   Modem=0x0, Codec Bytes=160, Signal Type=2)
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ind:
   Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
   Playout Max=1000(ms), Fax Nom=300(ms))
Nov 23 19:31:39.704: //84/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x3CE27724, Destination Call Id=83, Source Call Id=84,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_caps_ack:
   Destination Interface=0x22847B14, Destination Call Id=84, Source Call Id=83,
   Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_VOICE(0x2), Fax Version:=0, Vad=OFF(0x1),
   Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=9438)
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Progress Indication=NULL(0), Data Bitmask=0x1
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/ccCallConnect:
   Call Entry(Connected=TRUE, Responsed=TRUE)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Nov 23 19:31:39.704: htsp_timer_stop
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Id=83
Nov 23 19:31:39.704: //83/B583C95F8093/CCAPI/cc_api_voice_mode_event:
   Call Entry(Context=0x230F9C10)
Nov 23 19:31:39.704: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Nov 23 19:31:39.932: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:39.932: htsp_timer_stop2
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Cause Value=16, Interface=0x22847B14, Call Id=84
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_call_disconnected:
   Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
   Conference Id=0x21, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/ccConferenceDestroy:
Nov 23 19:31:48.860: confID:0x21; callEntry1 callID1:0x53, type:6; callEntry2 callID2:0x54, type:1
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x3CE27724, Source Call Id=83,
   Destination Call Id=84, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //84/B583C95F8093/CCAPI/cc_api_bridge_drop_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.860: //83/B583C95F8093/CCAPI/cc_generic_bridge_done:
   Conference Id=0x21, Source Interface=0x22847B14, Source Call Id=84,
   Destination Call Id=83, Disposition=0x0, Tag=0x0
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //83/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/ccCallDisconnect:
   Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
Nov 23 19:31:48.864: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.864: htsp_timer_stop3
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_get_transfer_info:
   Transfer Number=NULL
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x22847B14, Tag=0x0, Call Id=84,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.872: //84/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: :cc_free_feature_vsa freeing 3D1B9898
Nov 23 19:31:48.876: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.876: vsacount in free is 1
Nov 23 19:31:48.884: htsp_process_event: [0/0/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Nov 23 19:31:48.884: htsp_timer_stop
Nov 23 19:31:48.884: htsp_timer_stop2
Nov 23 19:31:48.884: htsp_timer_stop3
Nov 23 19:31:48.884: [0/0/0] set signal state = 0x4 timestamp = 0
Nov 23 19:31:48.884: htsp_timer - 2000 msec
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Disposition=0, Interface=0x3CE27724, Tag=0x0, Call Id=83,
   Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
Nov 23 19:31:48.884: //83/B583C95F8093/CCAPI/cc_api_call_disconnect_done:
   Call Disconnect Event Sent
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: :cc_free_feature_vsa freeing 3D1B9978
Nov 23 19:31:48.884: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Nov 23 19:31:48.884: vsacount in free is 0
Nov 23 19:31:49.156: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Nov 23 19:31:50.884: htsp_process_event: [0/0/0, FXOLS_ONHOOK, E_DSP_SIG_0100]

Sound gets off after few hours Satellite L745

Hi,
I have Satellite L745 model, bought 4 months back.
Two days before, I was using internal speakers for watching movie and after 2 or more hours there was no sound from the internal speakers of the laptop. So I tried other sound tracks, youtube etc but there was no sound from the internal speakers nor from headphones. I thought sound card is burnt as I was feeling hot right above the audio jack input. So I switched off the laptop for 30 minutes and switched on again, and found that it was working fine, there was sound from both internal and external headphones.
Yesterday night same problem happened again, there was sound, working fine but suddenly after few hours there is no sound. I switched off the sound track and waited 5 minutes and again switched on the sound track then there was sound for few seconds but then again no sound. So I did same switched off the computer for 30 minutes and checked again then there was sound from the interanl and external both.
I dont know what is the problem I did call service centre for technical assistance and he told me to re-install the drivers I did so. To check whether it is working fine now or not I need to switch on my laptop and sound track on for more than 3 hours.
Any one experiencing such problem or any assistance from Toshiba technicians will be welcome.
Thank you.
Dileep

I had exactly the same problem during the reproduction of a youtube video or any other sound from a music file or a video played in Windows Media Player, if i don't put the level of the volume at the top, the problem doesn't exist. I updated the drivers of the sound card Conexant Cx20585 @ Intel Cougar Point PCH, as well as the drivers of the BIOS, but nothing improved. Previously I did the system restore, but then I started to update the drivers since the restoration did not solve the problem.
I can not think what else to do, at least I'm relieved not to be the only one with the same model and the same problem notebook, Toshiba staff stay tuned!
Thank you!
PS: sorry for my English, it should not be the best, I'm from Argentina.

SBO Windows clear after few seconds and users has to re-type

After any window is opened, you can start typing. However, after few seconds the form clears and the users has to start typing again. It is quite annoying and affects productivity. It only happens on SBO to a couple of users though.
Thanks for your suggestion,
Andres
Edited by: Andres Castrillon on Sep 24, 2009 4:47 PM

Hi.
Which Add-on was the cause of the issue?.
Regards.

Facetime- How do I get it to work? I've tried calling my iPhone, but it doesn't ring? I see that the facetime connected somewhat to my phone, but it doesn't ring or allow me to answer.

I recently purchased Facetime for my iMac. I tried to test it out by calling my iPhone, but it doesn't ring at all. When looking at my call history on my cell, it shows that I missed Facetime calls, but no vibration, no ring, nothing. I then tried to call from my iPhone and still no real connection. Any ideas?

Is the iPhone connected to a WiFi network? FaceTime only works if the iPhone is connected to a WiFi network. Also, it must be enabled in Settings -> Phone

Can't install Exchange 2013 after previous uninstall (insufficient user privileges with the same user)

I have installed Exchange 2013 on Windows Server 2012, which ist a member of a Windows Server 2012 R2 Domain. All prerequisites and AD modifications were successfully completed and I could install Exchange 2013. Unfortunately I made a mistake with
the target directory and had to uninstal Exchange 2013. If I start the Setup again, it fails in prerequisite check - the current user should not be a member of Enterprise and Schema admin Group, but it's the same user, which comleted the previous Installation
of Exchange 2013! Do you have any idea how to solve this Problem - I'm running out of ideas.

Inhave tried to run setup /PrepareSchema again and get the following error in exchange setup log:
[03.07.2014 19:25:53.0305] [0] Setup encountered a problem while validating the state of Active Directory: Couldn't find the Enterprise Organization container.
[03.07.2014 19:25:53.0337] [0] Validating options for the 0 requested roles
[03.07.2014 19:25:53.0383] [0] [ERROR] Setup encountered a problem while validating the state of Active Directory: Couldn't find the Enterprise Organization container.
[03.07.2014 19:25:53.0399] [0] The Exchange Server setup operation didn't complete. More details can be found in ExchangeSetup.log located in the <SystemDrive>:\ExchangeSetupLogs folder.
After removing all Exchange Parts by ADUC (advanced View) and ADSI Edit (Services) I'm able to install Exchnge 2013 again - the uninstall procedure seems not to work in a clean manner.

XBMC: HDMI audio starts out OK, then gets distorted after few minutes

Hi,
I have posted about this on the XBMC forums, but one of the developers kindly informed me that they do not support XBMC on Arch Linux in any way, so I should take my queries elsewhere. I am hoping that someone here might be able to shed some light on my issue.
I am trying to use XBMC 13.1 (xbmc-13.1-2) with Samsung EH5300 32" Smart TV (2013 model) over HDMI connection. The HTPC computer is AMD-based (AMD A4-5300 APU, which includes Radeon HD 7480D GPU), and runs an up-to-date Arch Linux (last full system upgrade on 2014-06-26). I am using the open source "radeon" driver (xf86-video-ati-1:7.3.0-1). Everything has been great except the sound:
The sound starts out fine, but a few minutes later it begins to get distorted. At first it sounds like reverb affecting voice only, but with time the distortion increases until it sounds kind of cued TV noise. If I restart the video, the sound works again for a while. If I change options in Settings -> System -> Audio Output, like the number of channels or pass-through, the sound is again good for a little while. If I enable or disable "Stereo Upmix" or change audio stream from the "Audio - Settings" dialogue window available during video playback, it too seems to "reset" the sound to sounding normal for a while.
Has anyone seen something like this?
I have tried different videos, and it seems like HD videos are affected more than SD videos, starting with distortions only a few minutes in (except the one Xvid video, which was fine for 15 minutes). Some SD videos are completely fine, some start having distortion issues only after 20 minutes or so. Judging from video information, if the video uses 5.1 channels (all HD videos fall in this category), it definitely gets distorted; if a video only uses 2.0 channels (like the SD videos), some are fine while others get distortion later on. I have "Number of channels" in Audio Output settings set to 2.0. Changing this (or any other audio settings) has not affected the distortion.
Following Arch Wiki: Radeon note about HDMI audio, I tried adding radeon.audio=1 to the kernel line; this did not solve the issue, just added some visual glitches.
In the XBMC debug log I posted, the distortions happen at around 17:26, though there is absolutely nothing logged at that time, even though I enabled "Verbose logging of AUDIO component" in the Debugging settings. The video I was playing there showed the following in the information window: 720 H.264 dts 5.1 16:9.
Any suggestions, ideas, or experiences would be appreciated. It definitely seems to be XBMC issue, since I was able to play the same HD video through mplayer without any problems (for at least half an hour, which would have already showed distortions several times through XBMC).

Hi GDykes,
OK Imagine that Connection speed from your ISP is like an empty freeway. You can go full tilt (upto that road's speed limit)
Bandwidth is like a busy freeway where you only get to go as the bit you are in allows.
Reducing the Bandwidth in iChat or in the System Preferences > Quicktime > Streaming tab is like a vehicle that has a maximum speed that is slower than the speed possible on the freeway.
10:02 PM Friday; July 6, 2007

Canvas playback gets stuck after few seconds

Hi Guys, I'm posting this as my last chance to find some help. Recently I read tons of threads about choppy playback in FCP canvas but couldn't find anything useful.
I'm using unibody MBP 2.6 GHz, 4GB RAM. Latest Mac OS X version, latest FCP 7 version.
I'm trying to edit a footage captured by Canon 5D Mark II Camera. Using Compressor I downscaled the footage from Full HD to 1280x720 using Apple ProRes Codec. After importing into the FCP timeline it plays smoothly but if I do any change to the clip (speed change), canvas plays only 2 or 3 seconds of the clip, gets stuck while playhead is still moving and then when playhead reaches the end canvas shows the last frame of the clip. So when I wanna play the clip, canvas plays first few seconds then gets stuck and then shows me final frame.
You would probably need some more information about my sequence...if you do so, just ask.
Please, help me as I really need to finish this clip for my client.
Thanks

Yep, it is...I wanted to make my editing portable...I'm gonna move the clip to my FW800 HDD and let you know if that cause the problem.
Thanks

Skypeout calls continue to go silent after few seconds

i make calls to numbers using skypeout (US calls mostly) and hear the other person fine for about 10 seconds then it just goes silent. Although we are still connected. sometimes calls work just fine, other times it does this. any ideas?

I know the law, but thanks for the legal advice. Wouldn't it be curious to ask this question if I already possessed software that allowed keyboard commands for transcription purposes?
Right now, I'm using MPlayer X. ( A lousy, often unstable program). What appears to be keyboard command functionality -- under AVS at the top of the screen -- does not function that way and, even if it did function, is not fully what I need. I'll look further into VLC, but a preliminary look into it shows me a program more adapted for video applications. My experience has been that such software is not designed to meet my particular needs. The Adobe program is to EDIT audio. It is not designed to have the keyboard functionality one uses for transcription. This is exactly the problem I keep running up against with several programs. I can do sophisticated sound editing, but I cannot do simple transcription functions -- keyboard commands to speed up, slow down, move back, move forward. It is an incorrect assumption that an audio editing program would have those controls, in my experience.

Issue When you open a book in Adobe Digital Editions, you get an error stating that another user account has the book locked: "E_LIC_ALREADY_FULFILLED_BY_ANOTHER_USER"

Need Help re-install

I would like you to de-authorize ADE and re-authorize ADE using your adobe ID using the below method :
• Launch ADE.
• Press Control+Shift+D (Windows) or Command+Shift+D (Mac)
• You will get an option to De-authorize or Erase Authorization(depending on your ADE version), click it and you are done.
• Exit and re-launch ADE.
• Go To Help-> Authorize Computer.
• Enter your credentials and authorize ADE.
Check if you still get the same error. If so, then the below article should help.
http://helpx.adobe.com/digital-editions/kb/e-lic-already-fulfilled.html
Also, make sure you have the latest ADE 3.0 version installed. If not, you can download the same from the below link.
http://www.adobe.com/products/digital-editions/download.html
Let me know in case you still face the same issue.
Cheers!
~ Arpit

HT5569 I can't seem to connect to wi-fi in my home.......I called a tech who helped me but once I took it outside the home & entered another connection where I was ......the home connection didn 't pick up again...Shouldn't my home connection exist still?

I just brought the ipad air......but have trouble connecting to my home wi-fi. Can anyone help me?

Try this:
1. Turn router off for 30 seconds and on again
2. Settings>General>Reset>Reset Network Settings

Call is getting disconnect after 3 min when the caller is in queue

Hi,
I am using icm 7.0 with ip ivr 7.0. when all the agents are busy and caller is in queue,call is automatically getting disconnected after 3 min with the message " We are experiancing system problem and unable to process the call. please try again later" .
Icm and ivr script are attached
Regards,
Dinesh Joshi

Hi Chrish,
you are right. the timer was for 180 sec only. now things are working fine after increasing the timer.
Thanks a lot for your reply.
Regards,
Dinesh Joshi

Lync calls routing wrongly

Hi All,
I'm sorted of stumped at the moment.
The Scenario :
Lync 2013 deployment
Client has 4 sites - the sites are connected via vpn tunnels and has firewalls at each site.
User calls form site D to Site C and they experience one way audio. User calls from site C to site D and the call is fine.
User calls from site A to site C and the call is fine. User calls from site C to site A and one way audio might occur. User calls from site a to site D and there is one way audio. User calls from site D to site A and call is fine.
Devices uses - vvx410 phones.
Problem - can't see QoE on reporting as the vvx phones don't provide it.
So over to logging it seems that calls from site D is trying to come form the internal ip of the edge server's private ip to the user in site C. So it seems that during the call setup that UDP to UDP peer to peer is not happening and then uses the edge server
to route the call through. No at site D hairpinning on the edge server is not allowed. From sites A-C it seems that the call is hairpinning via the edge's public ip as that is allowed.
There are firewall restrictions in place between the sites and what the client tried was to open full ip to ip on the phones between the sites as that should in theory allow UDP to UDP, but that does not seem to happen.
Could this be a routing issue between the sites? We have booked a network engineer and firewall engineer to troubleshoot this issue. On the lync side I've not restricted any ports.
Will be greay if someone had this issue before.
Regards
Danie

Hi Greg/Anthony
Thanks for your replies so far.
The Lync setup consists only out of 1 Lync Edge server and 4 Lync Standard Edition server that is spread across the WAN. The lync static route to the inside nic does cover 172.18.0.0, in which all the subnets between all the sites are covered. The vvx phones
are located in a voice vlan (we recently had a issue with vvx phones losing their ip addresses and got the network engineer to reconfig the one site to create smaller subnets and then the issue was traced to 4 3COM switches). Their used to be only one voice
vlan, which is still the case at 3 sites, but the one site has quite a few voice vlans now to cater for all the subnets that were created, but I don't think this is the root of the problem. The one unfortunate thing is that on the lync reporting server you
can't see any QoE info apart from when a call is made from a vvx to a cx600 phone and usually when it is cross site the report will show that the vvx phone is coming from the external private ip of the lync edge servers AV nic. Today I did ask the client to
test by doing a lync to lync call, but since they are remote I can't foresure say that they did it correctly as the one user had a cx600 connected to his desktop and I wanted them just to use headsets and make the lync calls - I will get to try that out next
week when we will be at the client with a FW and network engineer to troubleshoot.
I also used the centralized logging tool in lync, but I don't seem to be able to locate the section where it shows the final candidate paring for a particular call, so I can't verify what the final candidate paring is. I can however see the initial candidate.
From the vvx410 phone is always uses a 22** port on UDP as the 1st candidate - I'm not sure if that is a fault as I would expect it to be in the 50,000 - 59,999 range as that is what the firewall engineer is seeing on his side.
I will also be able to do a wireshark trace - we have a port mirrored permanently on the one switch for the one user.
The main issue here is that there does not seem to be a peer to peer udp connection between the clients, so the edge server should never come into play here as the clients are inside the same WAN, alltough they are separated by 2 firewalls between 2 sites,
but connected via a vpn tunnel between the 2 firewalls.
Lync calls from the internet is working without any issues by the way.
Regards
Danie

Anyone have a problem sending text messages being delivered after few minutes up to five minutes?

Has anyone have a problem sending text messages and the messages won't get delivered after few minutes? From time to time I would have this problem and it's annoying. I would expect someone to receive it by then and response to my text right away. But when I look up the text it hasn't been delivered and it's in the process. I would hear that the message get delivered after few minutes? What gives? I never had a problem with my iphone 4 i think? Now with iOS 5 it let's me know whether it is delivered or not. This new feature is awesome but disappointing when I know my text hasn't been delivered right away. I'm not sure if it's the software or my iphone 4s or my carrier? anyone suggestions? Thanks!!

I have the same problem when sending iMessages.. Wasnt sure if it was apple or Sprint? But it happens VERY often and I end up having to send it as a text message instead

Lync Calls to PSTN gets disonnected after few rings if user is outside the organisation.

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