CUCMBE dtmf doesn't work via SIP Trunk to PSTN

The dtmf events packets send by CUCMBE have incorrect UDP Length. Based on RFC 2833 a correct UDP length value for dtmf event packets is 24 (decimal), but this CUCMBE sends its events packets with UDP length set to 36 (decimal). Since the events packets have an incorrect udp length field, it is causing the ITSP to discard the packet. The CUCMBE version is 7.1.3.30000-1.
The attachment fiel TestNo1.bmp is a capture with wireshark that show the udp length with 36. It was taken in CUCMBE version 7.1.3
However, with a CUCMBE version 8.6.2 it works well and it formats the events packets with udp length with 24.
The attachment fiel GoodTestNo.bmp is a capture with wireshark that show the udp length with 24. It was taken in CUCMBE version 8.6.2
I'll appreciate any help.
BR
Juan

Does DTMF work on *any* outbound call? If it's only broken to this destination it's possible that the length is too short for that conference bridge to recognize.
Please remember to rate helpful responses and identify helpful or correct answers.

Similar Messages

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

ILBC calls via SIP Trunk with CUBE and CUCM7

hi there,
our SIP Provider offers the IBLC codec which promises to provide better quality compard to G.729.
I'm using this scenario:
IP-Phone(G711) --- CUCM7 --- (SIP-Trunk1) --- CUBE --- (SIP-Trunk2) --- Provider
Everything workes unless I'm configuring IBLC at the provider and on trunk2.
I have the CUBE router acting as a trancoding device and also specified IBLC as codec to be handled.
SIP trunk 2 was placed in a region with IBLC as codec.
On the trunk configuration in CUCM the media ressource group with XCODE capability is configured
Transcoding workes between two IP Phones in different regions with different codecs within the intranet.
Unfortunately the CUBE router doesn't seem to use the transcoder to change internal G711u calls into IBLC codec
so calls are blocked by the CUBE device:
deb ccsip calls
for incoming call:
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4AE7AC98
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0237892992
Called Number            : 036677725231
Source IP Address (Sig ): 10.100.100.50
Destn SIP Req Addr:Port : <IP SIP Provicer>
Destn SIP Resp Addr:Port : <IP SIP Provicer>:5060
Destination Name         : <IP SIP Provicer>
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : ilbc
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): <IP CUBE>
Source IP Port    (Media): 0
Destn IP Address (Media): <IP SIP Provicer>
Destn IP Port    (Media): 22022
Orig Destn IP Address:Port (Media): [ - ]:0
.Mar 13 17:43:17.231: //145/BDB403DD8134/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 65
Disconnect Cause (SIP)   : 488
(Output lookes similar to outgoing calls)
I set up ccm on cube and assigned dsp ressources without success:
Here are the relevant configuration parts:
voice class codec 1
codec preference 1 iblc
voice service voip
address-hiding
allow-connections sip to sip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
h323
sip
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
voice-card 0
dspfarm
dsp services dspfarm
dial-peer voice 40991 voip
description *** Incoming from SIP-Provider
destination-pattern 03667772523.%
session protocol sipv2
session target ipv4:<IP_of_CUCM>
voice-class codec 1
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
ip qos dscp cs5 media
ip qos dscp cs5 signaling
sccp local GigabitEthernet0/0
sccp ccm 10.100.100.50 identifier 11 version 4.1
sccp
sccp ccm group 11
description *** lokaler CCM fuer Codec-Konvertierung von SIP/DUS.NET
associate ccm 11 priority 1
associate profile 21 register DE_WGT_MTP02
dspfarm profile 21 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec ilbc
maximum sessions 10
associate application SCCP
telephony-service
sdspfarm units 1
sdspfarm transcode sessions 10
sdspfarm tag 1 DE_WGT_MTP02
max-ephones 30
max-dn 30
ip source-address 10.100.100.50 port 2000
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 14 2010 02:10:34
sh sccp
SCCP Admin State: UP
Gateway Local Interface: GigabitEthernet0/0
        IPv4 Address: 10.100.100.50
        Port Number: 2000
IP Precedence: 5
User Masked Codec list: None
Call Manager: 10.100.100.50, Port Number: 2000
                Priority: N/A, Version: 4.1, Identifier: 11
                Trustpoint: N/A
Call Manager: 10.1.1.55, Port Number: 2000
                Priority: N/A, Version: 7.0, Identifier: 10
                Trustpoint: N/A
Transcoding Oper State: ACTIVE - Cause Code: NONE
Active Call Manager: 10.100.100.50, Port Number: 2000
TCP Link Status: CONNECTED, Profile Identifier: 21
Reported Max Streams: 20, 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: ilbc, Maximum Packetization Period: 120
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
sh dspfarm dsp all
SLOT DSP VERSION STATUS CHNL USE   TYPE    RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
0    2   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 1      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
1    1   26.3.4   UP     N/A FREE xcode 2      -         -         -
Thanks in advance,
David

Hi there,
Just wondering whether you ever got this resolved? I seem to have a very similiar problem.
Regards
Karen

2901 CME: Problem with incoming call via SIP Trunk

Dear All,
I have seen some others posted similar question regarding this but mine still doesn't work by using the reference solution.
Mine is quite standard setup too -> CME setup on my 2901 router, analog phone attach to my FXS port my outgoing calls are working fine via SIP but my incoming calls are not. Caller only listen to engage tone and analog phone is not ringing at all. Attached with my config and trace log of ccsip messages. Kindly assist. Thank you so much.

Hi Carlo,
Here it is
CME_2901#show sip-ua timers
SIP UA Timer Values (millisecs unless noted)
trying 500, expires 180000, connect 500, disconnect 500
prack 500, rel1xx 500, notify 500, update 500
refer 500, register 500, info 500, options 500, hold 2880 minutes
, registrar-dns-cache 3600 seconds
tcp/udp aging 5 minutes
CME_2901#show sip-ua retry
SIP UA Retry Values
invite retry count = 6   response retry count = 6
bye retry count    = 10 cancel retry count   = 10
prack retry count = 10 update retry count    = 6
reliable 1xx count = 6   notify retry count   = 10
refer retry count = 10 register retry count = 6
info retry count   = 6   subscribe retry count = 6
options retry count = 6
CME_2901#show sip-ua min-se
SIP UA MIN-SE Value (seconds)
Min-SE: 1800

DTMF doesn't work for outbound call

We currently using UC520 and having strange problem. When we call through SIP for outbound call DTMF tone is not transmitting. and if I press multiple numbers with longer duration, It makes continuous tone on the other hand.
My current dialpeer uses DTMF relay RTP- NTE. I will attach sh run and debug ccsip message file. please help me this is being quite critical.

Seems like your SIP Trunk Service provider is not getting the RFC2833 packets or not interpreting them correctly.
I see on the log that you pressed 1, 2 & 3 and the UC520 sends all these digits out (each digit will have 7 packets - this post explains how to read this debug - https://www.myciscocommunity.com/message/14045#14045). So I know its not the UC520 - below is a snip for digit 1
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B86 timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1      Pkt:04 00 00 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B87 timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:04 00 00 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B88 timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:04 00 00 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B89 timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:04 01 90 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8A timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:84 03 20 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8B timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:84 03 20 >>
Sep 10 16:26:26.114:          s=DSP d=VoIP payload 0x65 ssrc 0x15FE9166 sequence 0x4B8C timestamp 0xBCB7190A
Sep 10 16:26:26.114:          Pt:101    Evt:1       Pkt:84 03 20 >>
In terms of next steps:
- Is there any router or NAT or firewall device between the UC520 WAN interface and the SIP Trunk service provider? If so maybe good to check if you can bypass this device or if this device blocks RFC2833 packets
- Ask the SIP Trunk SP what they see on their end for the RFC2833 packets - you can send them a wireshark capture off the UC500 WAN port if that is more convincing than a debug log. Who is the SP by the way?

FAX doesn't working through SIP

Hello.
Fax Machine is connected to SPA 122. Cisco CUCME is used. SPA 122 doesn't support SCCP (only SIP). Fax machine works as ordinary telephone device, but doesn't work as fax machine. What is a problem?
Thanks in advance.

Hi @brkgb ,
I understand that you are able to print, but you can't send a fax. I would like to help you.
Can you make a copy from the glass and ADF?
Print out a Fax test.
Touch the setup button, (wrench)
arrow to Service menu, press OK, then right arrow to fax service, OK, then run fax test.
You should have 4 passes if everything is communicating properly.
We need to check some settings on the printer for faxing.
Private receive:
On the printer touch the setup button, (wrench) right arrow to fax setup menu, OK, fax receive setup, OK, private receive, OK, turn it off.
Error correction:
Go to the setup button, (wrench) right arrow to fax setup, all faxes, error correction, OK, turn it off.
Speed:
Go to the setup button, (wrench) right arrow to fax setup, OK, all faxes, OK, fax speed, OK, change it to medium, OK.
Clear saved faxes:
Go to the setup button, (wrench) right arrow to fax functions menu, clear saved faxes, OK, OK to clear.
Try to send a fax again.
If you are still having issues then go through the steps to see if it will resolve this issue.
Cannot Send or Receive Faxes.
If you need further assistance, please let me know the following information.
Did the fax test pass?
What type of connection do you have? (Analogue, DSL, Cable Modem)
Are you using the 2 wire phone cable?
Do you have distinctive ring set up?
Is there any voice mail set up?
Do you have a answering machine?
(Where is it connected to?)
Where is the fax cable connected to? (modem, wall jack)
Is there a phone also connected the printer?
Have a great day!
Thank You.
Please click “Accept as Solution ” if you feel my post solved your issue, it will help others find the solution.
Click the “Kudos Thumbs Up" on the right to say “Thanks” for helping!
Gemini02
I work on behalf of HP

Telco Messages via PRI (VG) connected to CUCM 9.X via SIP Trunk??

Hello
Questions:
Should I hear tel-co message on an IP Phone if a call that is meant to be long distance is sent to the gateway without a one (1). Currently these calls simply ring until disconnect, but work properly if the user dials a 1, the user expects to get a message from the provider and I am wondering if the SIP trunk between GW and CUCM is not allowing it?
Scenario:
IP Phone --> CUCM (SIP Trunk) --> ISR 2901(PRI) --> PSTN
In CUCM we have a local pattern 9.[2-9]XX[2-9]XXXXXX
If the IP Phone dials a number that is actually a long distance number, which would require a 1, but they dial it as a 10 digit local number, the call gets to the gateway and IP Phone hears ringing but it, rings until it eventually disconnects. At this point I believe the gateway is sending a cause code back to CUCM and I would expect error message back from the telco informing the caller of the issue, such as "Please dial 1 before a long distance number". Is there a specific setting on the SIP trunk and or Gateway to achieve this?
See Q931 Debug
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9838F
Exclusive, Channel 15
Display i = 0xB1, 'London Hydro'
Calling Party Number
UCS5-GW-02# i = 0x2181, '5193334444'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '5191112222'
Plan:ISDN, Type:National
*Jul 23 15:33:28.363: ISDN Se0/0/1:23 Q931: RX <- STATUS pd = 8 callref = 0xC3FA
Cause i = 0x80AB28 - Access information discarded
Call State i = 0x01
*Jul 23 15:33:28.411: ISDN Se0/0/1:23 Q931: RX <- CALL_PROC pd = 8 callref = 0xC3FA
Channel ID i = 0xA9838F
Exclusive, Channel 15
*Jul 23 15:33:28.415: ISDN Se0/0/1:23 Q931: RX <- PROGRESS pd = 8 callref = 0xC3FA
Cause i = 0x80FF - Interworking error; unspecified
Progress Ind i = 0x8088 - In-band info or appropriate now available
Thanks
Richard

Yes, Early media needs to be turned on the SIP Profile that is associated to the SIP trunk. The setting is SIP Rel 1xx options. It needs to be set for Send PRACK for all 1XX messages. If you have CUCM 7.x , this setting is a service parameter.

Tiff2icns doesn't work via ssh w/o window manager

I need to run tiff2icns on a Mac mini (Mac OS X 10.4.6) via a remote ssh connection. Unfortunately this doesn't work if I don't have a window manager session open on the Mac at that time. Is this a bug in tiff2icns, or am I doing something wrong? I want to run tiff2icns without having a window manager open on the remote system.
Franz
Mac mini Mac OS X (10.4.6)

any ideas?

External DVD doesn't work via USB on macbook pro

My old internal DVD drive in macbook pro doesn't work. I want to use my old DVD burner from my old PC. I connect it to enclosure for disk.
This stuff i plug in to USB in my macbook. It can read CD/DVD but not burn CD and DVD. What can I do? Is there any drivers for USB to make it work?

The 60 watt old style T-connector for MacBooks is about $20 on eBay http://shop.ebay.com/i.html?_nkw=60W+POWER+ADAPTER+FOR+APPLE+MACBOOK&_sacat=0&_o dkw=60W+POWER+ADAPTER+FOR+APPLE+MAGSAFE+MACBOOK+A1184&_osacat=0&_trksid=p3286.c0 .m270.l1313
The 85 watt old style T-connector for MacBook Pros is about $25 on eBay http://shop.ebay.com/i.html?_nkw=85W+POWER+ADAPTER+FOR+APPLE+MACBOOK+pro&_sacat= 0&_odkw=60W+POWER+ADAPTER+FOR+APPLE+MACBOOK&_osacat=0&_trksid=p3286.c0.m270.l131 3

Custom Windows 8.1 .WIM doesn't work via WDS, works OK manually

Hello,
I have a very peculiar situation; I've created a custom Windows 8.1 Enterprise x64 image, but when I deploy it using Windows Deployment Services on Windows Server 2012 R2, using the Windows 8.1 Enterprise x86 boot.wim as the Windows Deployment
Services boot image, it deploys successfully, but malfunctions;
it doesn't activate [against our KMS server]
when a new user logs on, the process stalls; if you logoff then logon, Microsoft Office gives you a 1603 error.
File Explorer [I think] doesn't work (apologies for vagueness; I absolutely
wasn't expecting this and was somewhat floored by the symptoms)
HOWEVER, if I manually deploy the SAME .WIM image, copy in the
SAME unattend.xml, and manually replace %MACHINEDOMAIN% and %MACHINENAME% with the right values, it works fine.
HOWEVER, I've already verified that Windows Deployment Services is working perfectly well because I've added the Windows 8.1 Enterprise x64 install.wim as an install image, and used the unattend.xml to automate it. In other words, with the
install.wim from the .ISO, everything is working fine, but with a custom image, it malfunctions.
I don't know why it's malfunctioning; nothing looking like a root cause appears in any event logs. I have injected some drivers into my custom image, but these are for relatively obscure devices such as scanners and smart card readers, which
aren't even on the computer I'm testing this on. The test computer would benefit from
chipset, video and network drivers, but the inbox ones work OK. In any event, they work fine when I manually deploy.
Have more-or-less replicated this on test machines that are virtual, desktop and laptop, so I have no reason to believe the target device is a factor.
For now, Windows Deployment Services doesn't have a client unattend file.
This is a very peculiar problem. My existing, production environment has
Windows 7 Enterprise in a custom image
Windows Server 2008 R2
Windows Deployment Services
working perfectly, and was looking to maintain the same approach, but simply update it.
Would welcome your ideas.
Kind regards,
Anwar

Hi,
For this issue,I think it is related to the custom image.
I suggest we don't add any drivers to the custom image to test it.
Meanwhile,I suggest you upload following logs here for further research. SkyDrive or similar network sharing would be better to use to share your files.
C:\WINDOWS\PANTHER\setupact.log
C:\WINDOWS\PANTHER\setuperr.log
Regards,
Kelvin hsu
TechNet Community Support

Best Practice to Integrate CER with RedSky E911 Anywhere via SIP Trunk

We are trying to integrate CER 9 with RedSky for V911 using a SIP trunk and need assistance with best practice and configuration. There is very little documentation regarding "best practice" for routing these calls to RedSky. This trunk will be handling the majority of our geographically dispersed company's 911 calls.
My question is: should we use an IPsec tunnel for this? The only reference I found was this: http://www.cisco.com/c/en/us/solutions/collateral/enterprise-networks/virtual-office/deployment_guide_c07-636876.htmlm which recommends an IPsec tunnel for the SIP trunk to Intrado. I would think there are issues with an unsecure SIP trunk for 911 calls. Looking for advice or specifics on how to configure this. Does the SIP trunk require a CUBE or is a CUBE only required for the IPsec tunnel?
Any insight is appreciated.
Thank you.

you can use Session Trace in RTMT to check who is disconnecting the call and why.

Not receiving the 486 message from CUCM to Genesys via SIP trunk.

I have setup where Genesys is used along with CUCM 9.1
Below is the snapshot how it will look for call flow.
PRI----V.G----CUCM---SIP trunk (created in CUCM)-----Geneys server.
Query here is for outbound call from SIP softphone to PSTN, where if the PSTN user cancel the call.
the SIP phone is still assuming the call is continuing and after 40 sec its getting disconnected.
after looking in to the sip traces... it looks like that SIP trunk from cucm is not sending the user busy message 486....
(checked in V.G and its giving user busy)...but in the CUCM its not getting sent to the genesys...
After some time in genesys server itself send the 480 Temporarily Not Available message...
I assume I should get the 486 message from CUCM to genesys when the PSTN party disconnect the call without answering.
Please assist.

From logs what i can see is after one min call legs stops transmitting and receiving packets.
You certainly need to check this Genesys support for this behavior , as far as I know there is no problem either with the CUCM or with VG.
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:15 tx:3765/602400 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:16 tx:3387/568856 rx:3838/614080
Tele 0/3/0:15 (1514) [0/3/0.31] tx:76760/76760/0ms g711ulaw noise:-68 acom:3 i/0:-64/-62 dBm
1477 : 1515 22820290ms.1 +0 pid:0 Originate connecting
dur 00:01:26 tx:4330/692800 rx:3387/541760
IP 10.129.0.45:32596 SRTP: off rtt:0ms pl:67720/140ms lost:0/1/0 delay:55/55/65ms g711ulaw TextRelay: off
1477 : 1514 22820290ms.2 +0 pid:0 Originate active
dur 00:01:30 tx:3387/568856 rx:4515/722400
Tele 0/3/0:15 (1514) [0/3/0.31] tx:90290/90290/0ms g711ulaw noise:-68 acom:3 i/0:-67/-61 dBm
Rate all the helpful post.
Thanks
Manish

WD Hard Drive doesn't work via Fire Wire to Mac OSX Lion or Mountain Lion.

I have a 1 terrabyte WD Hard Drive that works perfectly fine connected to Snow Leopard but isn't even recognized by any Mac running Mountain lion!!!
How do I fix this?????

I agree with alkhater. From reading other posts about the issue (Googled "Western Digital hard drive not appearing in Disk Utility") it sounds like you need to update your driver. If you need help locating the correct driver, please let us know the exact hard drive that you have.

Script doesn't work via Task Scheduler

I've got the script below that works fine in an elevated PS window but when run as a sceduled task produces a last run result of 0x80070001 which I think is strangely something to do with Windows backup after some googling!
$LimitDay = 21
$limit = (Get-Date).AddDays(-$LimitDay)
# Path to Recurse
get-childitem -Path \\mainserver\backups\myfolder -Force |where {$_.psiscontainer -and $_.lastwritetime -le $limit}|foreach { remove-item $_.fullname -recurse -force -confirm:$false }
The task is set to run with the highest privileges and usin a domain user service account, I have tried to deleteand recreate the job as well.
I'm trying to remove old backup folders based on their modified date and the contents, but not the contents when the modified date is valid e.g. server backups so config files date's never change but the folder to be deleted should still be recognised.

Hi,
in the Scheduler try to call the script with the whole path like this:
powershell.exe -noexit &'C:\DATA\Scripts\myscript.ps1'
Regards
I'd use -File instead (and drop -noexit).
Action: Start a program
Program/script: C:\Windows\System32\WindowsPowerShell\v1.0\powershell.exe
Add arguments: -File C:\Path\To\Script.ps1
Don't retire TechNet! -
(Don't give up yet - 13,085+ strong and growing)

HT4437 Airplay doesn't work via my iphone anymore

I've had my airport express for years and have been able to airplay via my iphone 4s. All of a sudden the icon as vanished and i can't play through the phone...
I have restarted phone and soft reset the airport but no luck. any advise?
Thanks, Jay

Read here:
http://support.apple.com/kb/ts4215

CUCMBE dtmf doesn't work via SIP Trunk to PSTN

Similar Messages

Maybe you are looking for