Timesheet Approval Configuration - Fixed Approval Routing

Hello,
Currently our timesheet configuration is through chain approvals. Team member submits to their manager, manager does approval and sends to me (timesheet manager). I have Fixed Approval Routing unchecked, as I would like for the Manager to be able to select
me as manager to approve. However, after unchecking Fixed Approval Routing, it doesn't seem to pop up for managers to select me as the timesheet manager. They just get the dialogue box to click OK when approving. Is there a configuration I'm missing? I tried
this for every manager by delegating and it seems it just goes directly to their timesheet manager without having a message to send to. Previously, it would work fine, but after a few configuration tweaks it stopped popping up to select timesheet manager. 
Johnny Eltawn, Project Server Admin

For Project Server 2013, Try following:
Add yourself in the "Timesheet Managers" list, which can be set under "Server Settings" -> "Timesheet
Managers"
Update Team member and make yourself as "Timesheet
Manager"
thru "Manage User" or thru "Resource Centre."
Update following setting on "Server
Settings" page, in the "Time and Task Management"
section, click "Timesheet Settings and Defaults"
Select the "Fixed Approval Routing" check box.
Under "Task Status Manager Approval", choose "Enabled" if you want managers to be able to approve individual
timesheet lines. You can also choose to "require line approval before timesheet approval."
Hope this will help you.
Raushan Kumar |
LinkedIn

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       Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
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       Match Rule=DP_MATCH_DEST; Called Number=90862157774
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       Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
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       Match Rule=DP_MATCH_DEST; Called Number=90862157774T
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         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
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       Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
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       Match Rule=DP_MATCH_DEST; Called Number=90862157774
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       Result=Success(0) after DP_MATCH_DEST
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       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
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         3: Dial-peer Tag=1
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    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:11 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397231
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:12 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397232
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 09:27:14 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327397234
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 244
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
    s=SIP Call
    c=IN IP4 192.168.22.1
    t=0 0
    m=audio 18258 RTP/AVP 8 101
    c=IN IP4 192.168.22.1
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    I made some changes in the router configuration.
    I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
    The debugging is changed now. I can send and receive a respond from SIP server. But  It shows an error: SIP/2.0 404 Not Found
    Then it moves to ISDN line, and use this line to make a call.
    102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
    102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774T
    102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
    103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
    Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
    From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>
    Date: Tue, 24 Jan 2012 14:45:47 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327416347
    Contact: <sip:[email protected]:5060>
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 19412 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Received:
    SIP/2.0 404 Not Found
    From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
    Call-ID: [email protected]
    CSeq: 101 INVITE
    Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
    Content-Length: 0
    103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    ACK sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
    From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
    To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
    Date: Tue, 24 Jan 2012 14:45:47 GMT
    Call-ID: [email protected]
    Max-Forwards: 70
    CSeq: 101 ACK
    Allow-Events: telephone-event
    Content-Length: 0
    103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
    103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
    103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=211
    103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=20018
    103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=0862157774
    103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
       No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
    103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
       Result=NO_MATCH(-1)
    103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
    2801(config-dial-peer)#
    Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
    But it didn’t affect anything.
    Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
    Really stuck and don't know where to look at.
    Any help will be highly appreciated.
    Thanks.

    Hi Dan.
    Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
    I use Cisco ASDM for ASA to make changes.
    There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44)  for a few ports.
    Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
    For NAT:
    I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
    (AS TRANSLATED) UDP 5060
    Because there is already translation for the Server.
    Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
    116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
       Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
       Peer Info Type=DIALPEER_INFO_SPEECH
    116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
       Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
    116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
    116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Match Rule=DP_MATCH_DEST; Called Number=90862157774
    116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
       Result=Success(0) after DP_MATCH_DEST
    116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
       Result=SUCCESS(0)
       List of Matched Outgoing Dial-peer(s):
         1: Dial-peer Tag=9000
         2: Dial-peer Tag=2
         3: Dial-peer Tag=1
    116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
    116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:25 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505305
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:26 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505306
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:27 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505307
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
    Sent:
    INVITE sip:[email protected]:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
    Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
    From: "Sam " ;tag=D4410748-1C9D
    To:
    Date: Wed, 25 Jan 2012 15:28:57 GMT
    Call-ID: [email protected]
    Supported: 100rel,timer,resource-priority,replaces
    Min-SE:  1800
    Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
    User-Agent: Cisco-SIPGateway/IOS-12.x
    Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
    CSeq: 101 INVITE
    Max-Forwards: 70
    Timestamp: 1327505337
    Contact:
    Expires: 180
    Allow-Events: telephone-event
    Content-Type: application/sdp
    Content-Disposition: session;handling=required
    Content-Length: 247
    v=0
    o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
    s=SIP Call
    c=IN IP4 10.1.1.101
    t=0 0
    m=audio 18782 RTP/AVP 8 101
    c=IN IP4 10.1.1.101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=ptime:20
    I'll add Incoming dial-peer now.
    Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
    Appretiate your help.
    Thanks a mill.

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    Requsitioner creates a Shopping Cart where Approver A is the approver.
    3.
    Requsitioner enters approval preview and Approver A is displayed as approving manager
    Questions is:
    Is there any way of displaying that Approver B is setup as an active substitute and that
    Approver C is a passive substitute?
    There is a function module BBP_PDH_WFL_APPROVAL_SIMULATE where approvers are saved to a table et_adhoc_agents
    Perhaps this table could be "looped" through and for each approver, add a line if there exist any substitutes.
    I heard about a function module that retrieves substitutes when provided a userid, is this any FM like this?
    Possible other solutions?
    I have a feeling that this might be solved by using standard SAP functionality, perhaps someone knows anything about this?
    Thanks for all future help!
    Kind regards,
    Teodor

    Hi,
    There is one Function Module which is used for retrieveing the substitutes available for a specific user.
    this is not SRM Function module , but is used in SRM.
    FM is
    RH_SUBSTITUTIONS_GET
    ---> with import parameters
    USER_TYPE                       US           
    USER_NAME                          
    SEARCH_DATE                    
    ACT_PLVAR                                    
    READ_ONLY_ACTIVE                X            
    USER_TYPE IS ALWAYS   US
    USER_NAME IS INPUT FOR THE USER
    SEARCH_DATE IS THE DATE ( EACH SUBSTITUTE WILL HAVE START DATE AND END DATE).
    READ_ONLY_ACTIVE = X RETRIEVES ONLY ACTIVE APPROVERS AND
    READ_ONLY_ACTIVE = EMPTY RETRIEVES BOTH ACTIVE AND PASSIVE APPROVERS ON GIVEN DATE.
    >
    THIS FM YOU HAVE TO USE WHEN THE APPROVAL PREVIEW IS SHOWN IN SRM.

  • Need Help for configuring Floating static route in My ASA.

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    Cryptochecksum:eea6e7b6efe5d1a180439658c3912942
    : end
    i think half of the configuration stil there in the ASA.
    Diagram.
    Thanks
    Roopesh

    You have missed the last command in your configuration, Please check it again
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    route ISP2   0.0.0.0 0.0.0.0 3.3.3.3
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    frequency 10
    sla monitor schedule 123 life forever start-time now
    track 1 rtr 123 reachability
    You can do NAT in same way, here the logical name of the interface will be different.
    Share the result
    Please rate any helpful posts.

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