SRST Incoming digits

Hi,
I have an issue where i see the incoming digits show up little different than normal. My incoming calls from the duct to the IP phone registered to SRST gateway does not come properly. But when we turn off the srst the calls work normal.
I am sharing the Debud isdn q931 of the same. Please let me know if some one came across the similar issue.
The Debug when in SRST ON:
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0036
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18382
                Preferred, Channel 2
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A11802010202010080102020566F696365205365727669636573
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '8026'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C060981, '8026', 0x0F1282818E, ' Voice Services'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8036
        Channel ID i = 0xA98382
                Exclusive, Channel 2
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8036
        Cause i = 0x8081 - Unallocated/unassigned number
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0036
        Cause i = 0x8181 - Unallocated/unassigned number
        User-User i = 0x00FEB0
Aug 20 03:29:56: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x3F6A
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA1838D
                Preferred, Channel 13
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A113020102020100800B5A6F72696E612054657374
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '68152'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C070981, '68152', 0x0F0D828684, 'ZorinaTest'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> SETUP_ACK pd = 8 callref = 0xBF6A
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '0'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '1'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xBF6A
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0xBF6A
        Facility i = 0x9FAA06800100820100A11D020101060528EC2C000180115A6F72616E2053746566616E6F76736B69
        Progress Ind i = 0x8088 - In-band info or appropriate now available
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- FACILITY pd = 8 callref = 0x3F6A
        Facility i = 0x9FAA06800100820100A406020101810101
Aug 20 03:29:59: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x3F6A
        Cause i = 0x8190 - Normal call clearing
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0036
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA18382
                Preferred, Channel 2
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A11802010202010080102020566F696365205365727669636573
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '8026'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C060981, '8026', 0x0F1282818E, ' Voice Services'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8036
        Channel ID i = 0xA98382
                Exclusive, Channel 2
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8036
        Cause i = 0x8081 - Unallocated/unassigned number
Aug 20 02:34:17: ISDN Se0/2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0036
        Cause i = 0x8181 - Unallocated/unassigned number
        User-User i = 0x00FEB0
The below is when the SRST is OFF :
Aug 20 03:29:56: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x3F6A
        Bearer Capability i = 0x8090A3
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA1838D
                Preferred, Channel 13
        Facility i = 0x9FAA068001008201008B0100A1100201010201553008820301B850860101
        Facility i = 0x9FAA068001008201008B0100A113020102020100800B5A6F72696E612054657374
        Progress Ind i = 0x8183 - Origination address is non-ISDN
        Calling Party Number i = 0x0181, '68152'
                Plan:ISDN, Type:Unknown
        Called Party Number i = 0x89, '30123'
                Plan:Private, Type:Unknown
        User-User i = 0x00FE, 'U', 0x0100, 'Y', 0x0100B00C070981, '68152', 0x0F0D828684, 'ZorinaTest'
        Shift to Codeset 4
        Codeset 4 IE 0x31 i = 0x80
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> SETUP_ACK pd = 8 callref = 0xBF6A
        Channel ID i = 0xA9838D
                Exclusive, Channel 13
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '0'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- INFORMATION pd = 8 callref = 0x3F6A
        Called Party Number i = 0x89, '1'
                Plan:Private, Type:Unknown
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xBF6A
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0xBF6A
        Facility i = 0x9FAA06800100820100A11D020101060528EC2C000180115A6F72616E2053746566616E6F76736B69
        Progress Ind i = 0x8088 - In-band info or appropriate now available
Aug 20 03:29:57: ISDN Se0/2/0:15 Q931: RX <- FACILITY pd = 8 callref = 0x3F6A
        Facility i = 0x9FAA06800100820100A406020101810101
Aug 20 03:29:59: ISDN Se0/2/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x3F6A
        Cause i = 0x8190 - Normal call clearing

Hi,
This is an MGCP gateway. We are testing the SRST configs. When we apply access list to block the route to the ccm, then we are treating as SRST ON and when we remove the block commands to the CUCM in Gateway we assume that it is SRST OFF.
Now when the SRST is OFF, CUCM is waiting untill all the digits are received. However in SRST we are only receiving the first 5 digits and not the complete digits.
Not sure why we are receiving the other two digits seperately. when the gateway is registering with CCM. When the gateway is in SRST, it does not get the other two digits. In our example it is 0,1 that we are getting seperately when GW registered with CCM.

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Show Run
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.02.27 15:14:52 =~=~=~=~=~=~=~=~=~=~=~=
sh run
Building configuration...
Current configuration : 12139 bytes
! Last configuration change at 06:35:59 UTC Tue Feb 25 2014
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
! NVRAM config last updated at 11:16:38 UTC Mon Feb 24 2014 by administrator
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname eucamvgw01
boot-start-marker
boot system flash:c2900-universalk9-mz.SPA.151-4.M5.bin
boot-end-marker
card type e1 0 0
logging buffered 51200 warnings
no logging console
no aaa new-model
no network-clock-participate wic 0
no ipv6 cef
ip source-route
ip traffic-export profile cuecapture mode capture
bidirectional
ip cef
ip multicast-routing
ip domain name drreddys.eu
ip name-server 10.197.20.1
ip name-server 10.197.20.2
multilink bundle-name authenticated
stcapp ccm-group 2
stcapp
stcapp feature access-code
stcapp feature speed-dial
stcapp supplementary-services
port 0/1/0
fallback-dn 5428025
port 0/1/1
fallback-dn 5428008
port 0/1/2
fallback-dn 5421462
port 0/1/3
fallback-dn 5421463
isdn switch-type primary-net5
crypto pki token default removal timeout 0
voice-card 0
dsp services dspfarm
voice call send-alert
voice call disc-pi-off
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 10.198.0.0 255.255.255.0
ipv4 152.63.1.0 255.255.255.0
address-hiding
allow-connections sip to sip
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
fax-relay ans-disable
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privacy pstn
no update-callerid
early-offer forced
call-route p-called-party-id
voice class uri 100 sip
host 41.206.187.71
voice class codec 10
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 ilbc
codec preference 4 g729r8
codec preference 5 g729br8
voice class codec 20
codec preference 1 g729br8
codec preference 2 g729r8
voice moh-group 1
moh flash:moh/Panjo.alaw.wav
description MOH G711 alaw
multicast moh 239.1.1.2 port 16384 route 10.198.2.9
voice translation-rule 1
rule 1 /^012237280$..$/ /54280\1/
rule 2 /^012236514$..$/ /54214\1/
rule 3 /^01223651081/ /5428010/
rule 4 /^01223506701/ /5428010/
voice translation-rule 2
rule 1 /^00$.+$/ /+\1/
rule 2 /^0$.+$/ /+44\1/
rule 3 /^$[0-9].+$/ /+\1/
voice translation-rule 3
rule 1 /^9$.+$/ /\1/
rule 2 /^\+44$.+$/ /0\1/
rule 3 /^\+$.+$/ /00\1/
voice translation-rule 4
rule 1 /^54280$..$/ /12237280\1/
rule 2 /^54214$..$/ /12236514\1/
rule 3 /^\+44$.+$/ /\1/
rule 4 /^.54280$..$/ /12237280\1/
rule 5 /^.54214$..$/ /12236514\1/
voice translation-rule 9
rule 1 /^$....$/ /542\1/
voice translation-rule 10
voice translation-rule 11
rule 1 /^\+44122372$....$/ /542\1/
rule 2 /^\+44122365$....$/ /542\1/
voice translation-rule 12
voice translation-rule 13
rule 1 /^$[18]...$/ /542\1/
voice translation-rule 14
voice translation-profile MPLS-incoming
translate calling 10
translate called 9
voice translation-profile MPLS-outgoing
translate calling 11
translate called 12
voice translation-profile PSTN-incoming
translate calling 2
translate called 1
voice translation-profile PSTN-outgoing
translate calling 4
translate called 3
voice translation-profile SRST-incoming
translate calling 14
translate called 13
license udi pid CISCO2921/K9 sn FGL145110RE
hw-module ism 0
hw-module pvdm 0/0
username administrator privilege 15 secret 5 $1$syu5$DsxdOgfS7Wltx78o4PV.60
redundancy
controller E1 0/0/0
ip tcp path-mtu-discovery
ip scp server enable
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description internal LAN
ip address 10.198.2.9 255.255.255.0
duplex auto
speed auto
interface ISM0/0
ip unnumbered GigabitEthernet0/0
service-module ip address 10.198.2.8 255.255.255.0
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service-module ip default-gateway 10.198.2.9
interface GigabitEthernet0/1
description to TATA NGN
ip address 115.114.225.122 255.255.255.252
duplex auto
speed auto
interface GigabitEthernet0/2
description SIP Trunks external
ip address 79.121.254.83 255.255.255.248
ip access-group SIP-InBound in
ip traffic-export apply cuecapture size 8000000
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speed auto
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no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
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ip route 10.198.2.8 255.255.255.255 ISM0/0
ip route 41.206.187.0 255.255.255.0 115.114.225.121
ip route 77.37.25.46 255.255.255.255 79.121.254.81
ip route 83.245.6.81 255.255.255.255 79.121.254.81
ip route 83.245.6.82 255.255.255.255 79.121.254.81
ip route 95.223.1.107 255.255.255.255 79.121.254.81
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permit ip host 83.245.6.81 any
permit ip host 83.245.6.82 any
permit ip 192.54.47.0 0.0.0.255 any
permit icmp any any
permit ip host 95.223.1.107 any
deny ip any any log
control-plane
voice-port 0/1/0
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/1
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
voice-port 0/1/2
compand-type a-law
timeouts initial 60
timeouts interdigit 60
timeouts ringing infinity
caller-id enable
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compand-type a-law
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timeouts interdigit 60
timeouts ringing infinity
caller-id enable
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ccm-manager music-on-hold bind GigabitEthernet0/0
ccm-manager config server 152.63.1.19 152.63.1.100 172.27.210.5
ccm-manager sccp local GigabitEthernet0/0
ccm-manager sccp
mgcp profile default
sccp local GigabitEthernet0/0
sccp ccm 10.198.2.9 identifier 3 priority 3 version 7.0
sccp ccm 152.63.1.19 identifier 4 version 7.0
sccp ccm 152.63.1.100 identifier 5 version 7.0
sccp ccm 172.27.210.5 identifier 6 version 7.0
sccp
sccp ccm group 2
bind interface GigabitEthernet0/0
associate ccm 4 priority 1
associate ccm 5 priority 2
associate ccm 6 priority 3
associate ccm 3 priority 4
associate profile 1002 register CFB_UK_CAM_02
associate profile 1001 register XCODE_UK_CAM_02
associate profile 1000 register MTP_UK_CAM_02
dspfarm profile 1001 transcode
codec ilbc
codec g722-64
codec g729br8
codec g729r8
codec gsmamr-nb
codec pass-through
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 18
associate application SCCP
dspfarm profile 1002 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dspfarm profile 1000 mtp
codec g711alaw
maximum sessions software 200
associate application SCCP
dial-peer cor custom
name SRSTMode
dial-peer cor list SRST
member SRSTMode
dial-peer voice 100 voip
description *** Inbound CUCM ***
translation-profile incoming PSTN-incoming
incoming called-number .
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 500 voip
description *** Inbound TATA MPLS ***
translation-profile incoming MPLS-incoming
session protocol sipv2
session target sip-server
incoming called-number ....
incoming uri from 100
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 510 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 54[013-9]....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 520 voip
description *** Outbound TATA MPLS ***
translation-profile outgoing MPLS-outgoing
destination-pattern 5[0-35-9].....
session protocol sipv2
session target ipv4:41.206.187.71
session transport udp
voice-class codec 20
dtmf-relay rtp-nte
no vad
dial-peer voice 200 voip
description *** Inbound M12 *** 01223651081, 01223651440 - 01223651489
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 0122365....
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 201 voip
description *** Inbound M12 *** 012237280XX
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 012237280..
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 202 voip
description *** Inbound M12 *** 01223506701
translation-profile incoming PSTN-incoming
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 01223506701
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 210 voip
description *** Outbound M12 ***
translation-profile outgoing PSTN-outgoing
destination-pattern +...T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 211 voip
description *** Outbound ISDN for SRST and emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 9.T
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 212 voip
description *** Outbound ISDN for emergency ***
translation-profile outgoing PSTN-outgoing
destination-pattern 11[02]
session protocol sipv2
session target ipv4:83.245.6.81
session transport udp
dtmf-relay rtp-nte
codec g711alaw
no vad
dial-peer voice 2000 voip
description *** Outbound to CUCM Primary ***
preference 1
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.19
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2001 voip
description *** Outbound to CUCM Secondary ***
preference 2
destination-pattern 542....
session protocol sipv2
session target ipv4:152.63.1.100
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 2002 voip
description *** Outbound to CUCM Teritiary ***
preference 3
destination-pattern 542....
session protocol sipv2
session target ipv4:172.27.210.5
voice-class codec 10
voice-class sip call-route p-called-party-id
dtmf-relay rtp-nte
no vad
dial-peer voice 999010 pots
service stcapp
port 0/1/0
dial-peer voice 999011 pots
service stcapp
port 0/1/1
dial-peer voice 999012 pots
service stcapp
port 0/1/2
dial-peer voice 999013 pots
service stcapp
port 0/1/3
sip-ua
no remote-party-id
gatekeeper
shutdown
call-manager-fallback
secondary-dialtone 9
max-conferences 4 gain -6
transfer-system full-consult
ip source-address 10.198.2.9 port 2000
max-ephones 110
max-dn 400 dual-line no-reg
translation-profile incoming SRST-incoming
moh flash:/moh/Panjo.ulaw.wav
multicast moh 239.1.1.1 port 16384 route 10.198.2.9
time-zone 22
time-format 24
date-format dd-mm-yy
line con 0
login local
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 131
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
line vty 5 15
session-timeout 60
exec-timeout 60 0
privilege level 15
login local
transport input all
scheduler allocate 20000 1000
ntp server 10.1.30.1
end
eucamvgw01#
Sh SCCP
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2014.03.03 17:57:44 =~=~=~=~=~=~=~=~=~=~=~=
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
IPv4 Address: 10.198.2.9
Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.198.2.9, Port Number: 2000
Priority: 3, Version: 7.0, Identifier: 3
Call Manager: 152.63.1.19, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 4
Trustpoint: N/A
Call Manager: 152.63.1.100, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 5
Trustpoint: N/A
Call Manager: 172.27.210.5, Port Number: 2000
Priority: N/A, Version: 7.0, Identifier: 6
Trustpoint: N/A
MTP Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1000
Reported Max Streams: 400, Reported Max OOS Streams: 0
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1001
Reported Max Streams: 36, Reported Max OOS Streams: 0
Supported Codec: ilbc, Maximum Packetization Period: 120
Supported Codec: g722r64, Maximum Packetization Period: 30
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: gsmamr-nb, Maximum Packetization Period: 60
Supported Codec: pass-thru, Maximum Packetization Period: N/A
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
Conferencing Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 1002
Reported Max Streams: 16, Reported Max OOS Streams: 0
Supported Codec: g711ulaw, Maximum Packetization Period: 30
Supported Codec: g711alaw, Maximum Packetization Period: 30
Supported Codec: g729ar8, Maximum Packetization Period: 60
Supported Codec: g729abr8, Maximum Packetization Period: 60
Supported Codec: g729r8, Maximum Packetization Period: 60
Supported Codec: g729br8, Maximum Packetization Period: 60
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30
Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30
TLS : ENABLED
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070080
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070081
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070082
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
Alg_Phone Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 152.63.1.19, Port Number: 2000
TCP Link Status: CONNECTED, Device Name: AN71FEF7F070083
Reported Max Streams: 1, Reported Max OOS Streams: 0
Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30
Supported Codec: g711ulaw, Maximum Packetization Period: 20
Supported Codec: g711alaw, Maximum Packetization Period: 20
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: g729ar8, Maximum Packetization Period: 220
Supported Codec: g729br8, Maximum Packetization Period: 220
Supported Codec: g729r8, Maximum Packetization Period: 220
Supported Codec: ilbc, Maximum Packetization Period: 120
eucamvgw01#

Monitoring via external TV monitor

hey, i know i've done this before but I cannot for the life of me remember how to monitor my editing through an external TV connected through a deck to my computer via firewire. i.e. I am trying to get the TV to function as the 'canvas' window in FCP.
i tried messing around in Audio/Video settings, but I just can't figure it out.
any advice?
thanks in advance

Its covered pretty well in the manual, but here's a few tips:
Make sure the DV device (VTR, camcorder or converter) is set to Firewire in and to output the incoming digital signal to its analog outputs.
Connect the analog outputs of the DV device to the analog inputs (AV or line) of your TV monitor, then make sure the TV is set to the proper input.
Power up the DV device and TV, then launch FCP. If you do not see FCP's output on the TV, go to View->External Video and make sure its set to ALL FRAMES.
If that doesn't work, post back.
-DH

Frequency divider + narrow pulse output

Hello all,
I need to divide the frequency of an incoming digital pulse by a factor of N (typically 10-50) but I would like the output signal to be 200ns wide. I'm able to divide the incoming pulse by using CO pulse ticks with the incoming pulse as the tick source and specifying the low and high ticks (low + high ticks = N). The problem is that the narrowest pulse I can generate is 2/f_in (or in my case 2/75kHz~ 26 us).
Is there any other way to divide a pulse and control the width of the output pulse?
I can use the "wide" pulse to trigger a narrow pulse on a different channel but I rather not use so many channels for this application. Is it possible to use just 1 input channel and 1 output channel?
Any advise will be greatly appreciated.
I'm using Labview 8 and pci 6251
Eyal

Hello Eyal,
Let me rephrase what it is I think you want to do:
1. You have a digital input with a frequency at or approximately 75 kHz.
2. You want to divide down this input frequency by N, where N is between 10 and 50 or so.
3. On every Nth pulse you want to generate a pulse with a 200ns high time and then return to a low state until the next 200ns pulse is generated.
If this is what you want to do then you would need 1 CO task to generate a pulse ever Nth rising edge of your input. Then you would use this pulse to trigger a retriggerable pulse train to output your 200ns pulse. All of this routing can be done internally so you would only physically connect one input and one output, however this setup would require three counters. One counter for the CO task and two counters for retriggerable pulse generation. Unfortunately your PCI-6251 only has two counters so to do this you would need to get a board with at least 3 counters.
If I didn't describe what you are trying to do accurately please reply back with further clarification incase what you are actually trying to do can be accomplished on your board.
If you would like to contact National Instruments directly to speak with a technical representative about getting a counter board you can find contact information at www.ni.com/contact.
Have a good weekend!
Brooks

How to make detection and acquasition processes faster on 6602 board ?

Hi to everybody,
Currently I am using a NI 6602 board. And I am designing my project using VC++ in .NET framework(1.1) .
In my application, I have to wait a digital signal(triggering signal) to go high level to start my acquasition.
At first, I tried to check continuously that incoming digital signal in a seperate thread using ReadSingleSampleSingleLine() fxn (polling). In my application there aren't so much threads , just 3-4 threads, and this thread has the highest priority.
But I recognised that I am missing some acquasition data after I detect the trigger signal.
In more detail the case is: when the trigger signal arrives, I have to capture an incoming byte(which is available on another 8 digital input lines for just 5-6 microseconds).
And the problem is : I can detect the trigger signal by polling but I am missing the incoming byte. I am trying to detect the incoming byte using ReadSingleSamplePortByte() function again by polling. The incoming byte disappears before I manage to read it.
My fist question:
1. Is there anything that you can suggest to make my detection and acquasition process faster?
I tried to use digital edge triggering and change detection events hoping to faster my process. But I got an exception about my board does not support that kind of triggering.
My second question:
2. Is there any way to use digital edge triggering or change detection events in 6602 board?
This is the first time I am using an NI board.
I noticed that 6602 is specialized on timer counter concepts, not on digital I/O concepts. But I don't have a chance to use any other board right now and I have to manage my application using 6602.
any help is welcomed.
thanks in advance.
koray.

In a more readable form:
Hi to everybody,
Currently I am using a NI 6602 board. And I am designing my project using VC++ in .NET framework(1.1) .
In my application, I have to wait a digital signal(triggering signal) to go high level to start my acquasition.
At first, I tried to check continuously that incoming digital signal in a seperate thread using ReadSingleSampleSingleLine() fxn (polling). In my application there aren't so much threads , just 3-4 threads, and this thread has the highest priority. But I recognised that I am missing some acquasition data after I detect the trigger signal.
In more detail the case is: when the trigger signal arrives, I have to capture an incoming byte(which is available on another 8 digital input lines for just 5-6 microseconds).
And the problem is : I can detect the trigger signal by polling but I am missing the incoming byte. I am trying to detect the incoming byte using ReadSingleSamplePortByte() function again by polling. The incoming byte disappears before I manage to read it.
My fist question:
1. Is there anything that you can suggest to make my detection and acquasition process faster?
I tried to use digital edge triggering and change detection events hoping to faster my process. But I got an exception about my board does not support that kind of triggering.
My second question:
2. Is there any way to use digital edge triggering or change detection events in 6602 board?
This is the first time I am using an NI board. I noticed that 6602 is specialized on timer counter concepts, not on digital I/O concepts. But I don't have a chance to use any other board right now and I have to manage my application using 6602.
any help is welcomed.
thanks in advance.
koray.

Analog Levels vs SPDIF Levels Input and Output in Logic Pro

Hello,
I ran a test last night for recording input and output levels from my Yamaha Motif XS8 through an Apogee Ensemble to compare Analog to SPDIF
I connected two TRS cables from the L and R outputs on the Motif XS into the Analog Inputs on the Apogee and also have the SPDIF connection from the Motif to the Apogee.
I put the master fader all the way up on the XS for the volume for analog.
The ensemble in Maestro has a +4 and -10 reference notional level option for analog inputs. i had it at +4 but changed to -10 and the analog got louder (i figured it would get louder for +10, confusing).
anyways, why is it that I can record louder levels for analog than the digital transfer?
I tracked both options at the same time then recorded vocals over it. the digital sound is too low. what's up with that?
and when I tried to bounce the recording to listen to it in ITunes, the volume levels were way lower than my cd playing through iTunes. Please enligthen me on these?
I use to record in a Roland 2480, and had similar results with loudness, but got a little louder through mastering but still....pro cds are way louder and still clear.

I think there is much confusion here!
In summary, you wont be able to control the recording level of S/PDIF.
The reason is that you don't want to!
You need to think of the SPDIF connection as being more like a file transfer method. You are copying the digital data at an output to your harddrive in effect. If I send you an MP3 via email you'd never imagine that your email software is capable of changing the gain of the MP3 I send you. This might sound daft but its a useful analogy. If you need to increase the "volume" of that MP3 then you'd need to ask the sender. Its the same with your set-up.
There could well be somewhere on your synth that adjusts the instruments level, other than the master (analogue) output control. For example, make sure the midi volume of the intrument being played is set to full - ie midi vol 128. Perhaps there is somesort of virtual mixer onboard to control all the muti-timbral parts so make sure your part has its virtual fader turned up.
This is what (basically) is going on in the chain...
Your synth creates sound in its "digital brain". This sound is sent to an "output stage" which will distribute the sound to various outputs. In the case of the S/PDIF it will just send the raw digital data untouched. For the analogue side the digital signal (same as the one sent to S/PDIF) will be converted to analogue and then sent to a amplifier to get it to an appropriate "line level". This final level could well be controlled by an anogue volume control which could be adding more gain (than you think) too.
When things go to your sound card/ daw...
The purpose of a analogue gain control is to set the i/p signal so that it suitably loud to beat any noise that exists in your input circuits - so that a good signal to noise ratio is achieved. Analogue signals need to work in the right loudness zone (so to speak) as the analogue electronics will be designed to handle signal levels of a particular range. the gain control is there to make sure the signal is in that range.
Digital signals are far more predictable though and there is no advantage to your recordings if the incoming digital signal gets an increase of level at the input stage. All you are doing here is effectively adding a few zeros to the binary digital data!
Lets face it the point of recording is to get a copy of the original sound, that is as similar to the original as possible. With S/PDIF you get a perfect copy of what's coming out of your synth - so job's a good un!
If, when you come to mix in logic, you find the level of the digital recording is indeed too low for mixing/mastering purposes then just boost it in logic via a fader or via the gain plugin.
Those referrence values of -10dBV and +4dBu refer to analogue voltage levels only. they have nothing to do with the digital domain. The -10/+4 switch will be only relevant to analogue inputs and outputs. Using an analogue VU meter you should find that a sine wave that peaks at 0dBVU (totally different to 0dBFS BTW) is the equivant of a digital sine wave peaking at -18dBFS.
The analogue headroom (how loud you can go before things distort) depends on the analogue electronics and varies with different design. Analogue stuff, like mixers) often has headroom of 24dB or more. So that digital stuff can interface with analogue properly we allow for that analogue headroom to be around 18dB (usually enough in practise!)... hence -18dBFS(digital)=0dBVU(analogue).
To make your digital and analogue input signals sound similar in level you will probably have to reduce the gain of the analogue input. If you set the incoming analogue signal to peak around -14dB (or less!) or so you will probably find things more equal. If you are working in 24 bit your analogue levels can be seemingly very low before sound quality is affected. Its quite safe to record at -20 or even -30dB as shown on logic's meters for eg.
I hope all this waffle helps LOL!

JFreeChart

Hi,
I have a line chart that iv created using jfreechart. The graph has six lines.when the graph comes up, there are no lines on it.This is delibrate. Iv also made six check boxes. The problem i have is that when the check boxes are checked, the lines are supposed to apear, each line corresponding to the boxes that has been checked. But the lines are'nt appearing when the boxes are checked. Please can someone help. Thank you for your help.

Here is the code iv written this is actually part of a bigger program
</code>
package seti;
public class Entropygraph extends JPanel implements ActionListener, ItemListener {
     JButton home;
     JButton back;
     JCheckBox Malay_Binary;
     JCheckBox English_Binary;
     JCheckBox Vizmusic_Binary;
     JCheckBox Polish_Binary;
     JCheckBox Welsh_Binary;
     JCheckBox Arecibo74b;
     boolean entdata = false;
     String entdata2;
     String entdata3 ;
     String entdata4 ;
     String entdata5 ;
     String entdata6 ;
     public Entropygraph() {
          super((new BorderLayout(40, 40)));
          JPanel topHandPanel = new JPanel();
          GridLayout gridLayout = new GridLayout(5, 100, 0, 10);
          topHandPanel.setLayout(gridLayout);
          topHandPanel.setBorder(BorderFactory.createEmptyBorder(0, 10, 10, 10));
          topHandPanel.setBackground(new Color(153, 205, 143));
          Malay_Binary = new JCheckBox("Malay");
     Malay_Binary.setMnemonic(KeyEvent.VK_C);
     Malay_Binary.setSelected(false);
     English_Binary = new JCheckBox("English");
     English_Binary.setMnemonic(KeyEvent.VK_G);
     English_Binary.setSelected(false);
     Vizmusic_Binary = new JCheckBox("Vismusic");
     Vizmusic_Binary.setMnemonic(KeyEvent.VK_M);
     Vizmusic_Binary.setSelected(false);
          Polish_Binary = new JCheckBox("Polish");
     Polish_Binary.setMnemonic(KeyEvent.VK_N);
     Polish_Binary.setSelected(false);
          Welsh_Binary = new JCheckBox("Welsh");
          Welsh_Binary.setMnemonic(KeyEvent.VK_B);
          Welsh_Binary.setSelected(false);
          Arecibo74b = new JCheckBox("Arecibo74b");
          Arecibo74b.setMnemonic(KeyEvent.VK_V);
          Arecibo74b.setSelected(false);
          home = new JButton("Home Page");
          back = new JButton("Back");
          topHandPanel.add(Malay_Binary);
          topHandPanel.add(English_Binary);
          topHandPanel.add(Vizmusic_Binary);
          topHandPanel.add(Polish_Binary);
          topHandPanel.add(Welsh_Binary);
          topHandPanel.add(Arecibo74b);
          topHandPanel.add(home);
          topHandPanel.add(back);
          add(topHandPanel, BorderLayout.NORTH);
          final CategoryDataset edata = createDataset();
          final JFreeChart echart = createChart(edata);
          final ChartPanel chartPanel = new ChartPanel(echart);
          chartPanel.setLayout(new BorderLayout());
          chartPanel.setPreferredSize(new Dimension(1000, 450));
          add(chartPanel);
          setBorder(BorderFactory.createEmptyBorder(80, 40, 40, 40));
               //Register a listener for the check boxes.
          Malay_Binary.addItemListener(this);
          English_Binary.addItemListener(this);
          Vizmusic_Binary.addItemListener(this);
          Polish_Binary.addItemListener(this);
          Welsh_Binary.addItemListener(this);
          Arecibo74b.addItemListener(this);
          home.addActionListener(this);
          back.addActionListener(this);
     public void actionPerformed(ActionEvent ae) {
          MainPanel panel = (MainPanel) getParent();
          if (home.equals(ae.getSource())) {
               panel.hideAll();
               panel.pageWelcome.setVisible(true);
          } else if (back.equals(ae.getSource())) {
               panel.hideAll();
               panel.pageRead.setVisible(true);
     public void itemStateChanged(ItemEvent e) {
Object source = e.getItemSelectable();
// if (source == Malay_Binary) {
     //entdata = ;
//Now that we know which button was pushed, find out
//whether it was selected or de-selected.
// if (e.getStateChange() == ItemEvent.SELECTED) {
     public CategoryDataset createDataset() {
          final String entdata = "Malay-Binary";
          final String entdata2 = "English-Binary";
          final String entdata3 = "Vizmusic-Binary";
          final String entdata4 = "Polish-Binary";
          final String entdata5 = "Welsh-Binary";
          final String entdata6 = "Arecibo74b";
          final String n0 = "0";
          final String n1 = "1";
          final String n2 = "2";
          final String n3 = "3";
          final String n4 = "4";
          final String n5 = "5";
          final String n6 = "6";
          final String n7 = "7";
          final String n8 = "8";
          final String n9 = "9";
          final String n10 = "10";
          final String n11 = "11";
          final String n12 = "12";
          final String n13 = "13";
          final String n14 = "14";
          final String n15 = "15";
          final DefaultCategoryDataset edata = new DefaultCategoryDataset();
          //This is all the data that is used to create
          //the Malay-Binary line chart.
// the data set has been removed so the code will fit on this post
          // This is all the data that is used to create
          // the English-Binary line chart
          // the data set has been removed so the code will fit on this post
          // This is all the data that is used to create
          // the Vizmusic-Binary line chart
     // the data set has been removed so the code will fit on this post
          // This is all the data that is used to create
          // the Polish-Binary line chart
// the data set has been removed so the code will fit on this post
          // This is all the data that is used to create
          // the Welsh-Binary line chart
     // the data set has been removed so the code will fit on this post
          // This is all the data that is used to create
          // the arecibo74b line chart
// the data set has been removed so the code will fit on this post
          return edata;
     private JFreeChart createChart(final CategoryDataset edata) {
          // create the chart...
          final JFreeChart chart = ChartFactory
                    .createLineChart(
                              "Visual Representation of the Entropy Calculation of the Incoming Digital signal",// chart
                              // title
                              "Bit Chunk Length", // x-axis label
                              "Entropic Values", // y-axis label
                              edata, // data
                              PlotOrientation.VERTICAL, // line graph position
                              true, // include legend
                              true, // this is for the tool tips
                              false
          chart.setBackgroundPaint(Color.white);
          final CategoryPlot plot = (CategoryPlot) chart.getPlot();
          plot.setBackgroundPaint(Color.black);
          plot.setRangeGridlinePaint(Color.green);
          final NumberAxis eaxis = (NumberAxis) plot.getRangeAxis();
          eaxis.setStandardTickUnits(NumberAxis.createIntegerTickUnits());
          eaxis.setAutoRangeIncludesZero(true);
          final LineAndShapeRenderer renderer = (LineAndShapeRenderer) plot
                    .getRenderer();
          renderer.setSeriesVisible(1, true);
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 3.0f,
          new float[] { 1.0f, 1.0f }, 0.0f));
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 5.0f,
                    new float[] { 1.0f, 1.0f }, 0.0f));
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 7.0f,
                    new float[] { 1.0f, 1.0f }, 0.0f));
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 9.0f,
                    new float[] { 1.0f, 1.0f }, 0.0f));
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 11.0f,
                    new float[] { 1.0f, 1.0f }, 0.0f));
          renderer.setSeriesStroke(0, new BasicStroke(3.0f,
                    BasicStroke.CAP_ROUND, BasicStroke.CAP_ROUND, 13.0f,
                    new float[] { 1.0f, 1.0f }, 0.0f));
          return chart;
<code>

PS3 Sound through Soundcard to Surround-System?

Hey guys! I'm new here, my name is Daniel and I'm from Germany (so my English might not be as good as you wish ). I have the following problem:
I want to connect my Playstation 3 to my PC-Speakers and use the soundcard as "decoder" for incoming digital signal.
Then I want to pass the encoded signal through to my speakers (they are connected via Cinch, so it's an analog Speaker System) so that I'm able to enjoy 5. sound.
Here is what I found out:
I tried to connect my PS3 with an optical cable to my SoundBlaster X-Fi Titanium Fatalty Professional Series.
It worked but the only thing I got was Stereo Sound. When I set the PS3 outgoing audio signal to DD/ DTS 5. encoding, I got no more sound.
Is there any way to get 5. (DTS/ DD decoding) over the soundcard and give the sound out to my Teufel Concept G THX 7.? Is there maybe a software-decoder, which can be used to decode (as I know, the soundcard has no integrated decoder) the signal?
I would be glad, if I get some help.
Greetz, Daniel

I had to connect my Apple TV directly to my home
theater system with an optical cable to get past the
problem. It looks like these Sharp TVs simply will
not pass sound input from an HDMI cable through to
the optical audio output.
I have one of the new Sharp LC-32GP1U 1080p LCD sets, and I have exactly the same problem:
Apple TV goes out via HDMI to TV; TV audio goes out via optical to Samsung receiver. The TV itself can play the audio, but the receiver cannot.
However, my HD cable box also goes out to the TV via HDMI (and then out via optical to the receiver also), and the receiver plays that audio just fine. So, at least for me, it is not an issue of the Sharp TV refusing to pass along HDMI audio to the optical output.
Is it possible that the Apple TV is the source of the problem? Maybe my receiver doesn't like the audio format (PCM? Dolby Digital?) the Apple TV sends?

Clicks in audio

I hear clicks in the audio whenever I watch a video on youtube or such video sites in Firefox but not in Internet Explorer or Google Chrome. The headset I have is a Steelseries Siberia V2 and with that I use the Steelseries USB Soundcard.
== This happened ==
Every time Firefox opened
== I got my new USB headset

Gorwel Owen1 wrote:
This problem arose whilst transferring files recorded in Logic into a Tascam HDP2 though I don't think thatthe latter is the problem. Essentially, when I play back files in the Tascam that were recorded in Logic, I get several clicks in the audio.
Any ideas? Thanks.
Gorwel
Was this an Analog or Digital transfer (s/pdif)?
If digital you will need to lock the Tascam to the incoming digital clock embedded in the s/pdif signal.
Also, if digital, only a single cable is used from the MOTU s/pdif out to the Tascam s/pdif in.
pancenter-

Clicks in audio on transfer

This problem arose whilst transferring files recorded in Logic into a Tascam HDP2 though I don't think thatthe latter is the problem. Essentially, when I play back files in the Tascam that were recorded in Logic, I get several clicks in the audio. Files that I've received from elsewhere recorded using ProTools seem fine when transferred. Also, the files I've recorded in Logic actually play back fine if I move them to a PC and play back using Audition.
Could this be related to the way that different DAWs interpret time-stamping?
These are 24-bit 44.1 WAVs though I note that the icon created by Logic for the files is its own rather than the usual WAV icon.
Any ideas? Thanks.
Gorwel

Gorwel Owen1 wrote:
This problem arose whilst transferring files recorded in Logic into a Tascam HDP2 though I don't think thatthe latter is the problem. Essentially, when I play back files in the Tascam that were recorded in Logic, I get several clicks in the audio.
Any ideas? Thanks.
Gorwel
Was this an Analog or Digital transfer (s/pdif)?
If digital you will need to lock the Tascam to the incoming digital clock embedded in the s/pdif signal.
Also, if digital, only a single cable is used from the MOTU s/pdif out to the Tascam s/pdif in.
pancenter-

SRST Incoming digits

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