CDT to CDT call issue

Hi ,
We are getting issue in CDT to CDT calling. When i am making any call from 1 CDT to other CDT call got disconnect itself and showing noanswer message on CDT screen. We are using BCM version 7.0.7.0 SP07 .
CEM server logs
11:52:14.782 (08976/Sm2:NewCallSM ) TRC> [***] QUE:DEST_ALLOCATED : WaitInQueue = (186) {'CALL_ID': '65588BB453944E52BADDA71DD3196FB5'}
11:52:14.783 (08976/Sm2:NewCallSM ) ERR> [EXC] : NewCallSM.WaitInQueue : Exception occurred
11:52:14.783 (08976/Sm2:NewCallSM ) ERR> <type 'exceptions.TypeError'> : argument 1 must be dict, not str
11:52:14.785 (08976/Sm2:NewCallSM ) ERR> File: .\UniMain.py     (4617)  Func: WaitInQueue              <None>
11:52:14.785 (08976/Sm2:NewCallSM ) ERR> File: .\UniMain.py     (4955)  Func: HandleNextDest           <None>
11:52:14.785 (08976/Sm2:NewCallSM ) TRC> [CMD] CALL:CALL_DISCONNECT = (187) {'_CMD': 'CALL_DISCONNECT'}

Hello
It's great to see this community working and people helping out each other!
Chand, thanks for your contribution. However, brief comments here:
Anyone searching for help on similar issue - please review http://scn.sap.com/thread/3582307, especially Adele Davison's comments.
As to replacing files:
See Tomi Halmela's comment.
Most likely a typing error, but in the official release builds, there shouldn't be .py files (except for PDCconfig and customizer samples). In this context, UniMain.pyc is different.
If installation/upgrade goes ok, generally there should be no need for copying/replacing files manually. If some file is not installed correctly, usually there should be errors or warnings in the IA ui/log, and the recommended action is to re-apply the changes with IA.
Kind Regards,
-Lasse-

Similar Messages

  • Call issues on iPhone 4 after upgrading to IOS 5.1.1

    Ever since I upgraded to the IOS 5.1.1 my iPhone 4 has been having call issues. When I make or receive calls the other person can hear me for 5 seconds and then they can't hear me, but I can hear them. It is almost like the mute button is on, but I check it every time and it isn't on. I reset the factory settings, I reset the network settings and I checked the SIM card and nothing has fixed the problem. My phone has NEVER given me problems before and this is really frustrating. Anybody know what is causing this or how to fix it?? Please help!
    Thanks!

    Apple has not been forthcoming on the explanation. My Siri worked on my iphone4 (not 4s) immediately after upgrading to IOS5, but I take it they have redirected the servers that used to support Siri to only work with Iphone4s.
    It is one thing for apple to only bundle the improved Siri with IPhone 4S, or  for them to remove Siri from the Ap store.
    But it is another thing all together for apple to purchase a company, and then cease the service that many of us paid for.
    Attempts to contact apple at apple store and via email have been unhelpful.
    Apple Fan--

  • Incoming calls issue in Third Party SIP Phone

    Hi,
    Yesterday I configured my third party sip phone which is yealink in this case on cucm and successfully registered it with cucm, despite of registration i have some calling issue in this phone. I am able to make outbound calls from this phone to any other phone however issue is related to inbound calls.I tried calling its DN from anywhere but call disconnect after sometime. Also didnt get any proper sip session trace in RTMT. Kindly suggest some step to sortout this issue.
    Thanks

    Dear Manish,
    Call normally dicsonnected after 30-40 sec with termination code 102 in session trace. PFB SDI  trace with 5030 is Thirdparty sip phone and 5033 is c7945. Looking forward for your suggestion.
    CallingPartyNumber=5033
    |DialingPartition=
    |DialingPattern=5030
    |FullyQualifiedCalledPartyNumber=5030
    |DialingPatternRegularExpression=(5030)
    |DialingWhere=
    |PatternType=Enterprise
    |PotentialMatches=NoPotentialMatchesExist
    |DialingSdlProcessId=(0,0,0)
    |PretransformDigitString=5030
    |PretransformTagsList=SUBSCRIBER
    |PretransformPositionalMatchList=5030
    |CollectedDigits=5030
    |UnconsumedDigits=
    |TagsList=SUBSCRIBER
    |PositionalMatchList=5030
    |VoiceMailbox=
    |VoiceMailCallingSearchSpace=PT-LHR-LOCAL:PT-Local:Unityvmpt:PT-F6-Local:PT-ISL-LOCAL:PT-KHI-LOCAL:PT_Operator_LHR:PT_Operator_KHI:PT_Operator_ISL
    |VoiceMailPilotNumber=7103
    |RouteBlockFlag=RouteThisPattern
    |RouteBlockCause=0
    |AlertingName=Syed Ahmer
    |UnicodeDisplayName=Syed Ahmer
    |DisplayNameLocale=1
    |OverlapSendingFlagEnabled=0
    12:17:38.028 |//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to 172.16.200.21:[5062]:
    [23928282,NET]
    INVITE sip:[email protected]:5062 SIP/2.0
    Via: SIP/2.0/UDP 10.100.200.11:5060;branch=z9hG4bK1ca0cc6e317649
    From: "Syed Ahmer" ;tag=8787406~039e2a80-8561-4586-8954-d01ed2aa12c8-246211918
    To:
    Date: Thu, 30 Jan 2014 07:17:38 GMT
    Call-ID: [email protected]
    Supported: timer,resource-priority,replaces
    Min-SE:  1800
    User-Agent: Cisco-CUCM8.5
    Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
    CSeq: 101 INVITE
    Expires: 180
    Allow-Events: presence
    Send-Info: conference, x-cisco-conference
    Alert-Info:
    Contact:
    Remote-Party-ID: "Syed Ahmer" ;party=calling;screen=yes;privacy=off
    Max-Forwards: 70
    Content-Length: 0
    |14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::fireSignal() varId = 0|14,100,50,1.14103336^10.163.14.4^SEP00230432C828
    12:17:38.028 |EnvProcessUdpHandler::fireSignal - SEND: index = 0, handler = 0xaf299320|*^*^*
    12:17:38.028 |EnvProcessUdpPort::fireSignal - SEND, destination = 172.16.200.21:5062|*^*^*
    12:17:38.028 |EnvProcessUdpPort - EnvProcessUdpHandler::send(buff, 850, 172.16.200.21:5062)|*^*^*

  • 2.2.1 update and call issues

    Hi all,
    I was very happy with my iPhone and the 2.2 software update, things were running nicely and I hadn't had any troubles, which is why I decided to go ahead and install the 2.2.1 update when it was released - and I have had nothing but trouble...
    I live in an area where Vodafone coverage is not great, there are however certain areas where I can go to get almost full reception; this was the first problem that I noticed - I no longer get full reception here, it's now two or sometimes three bars.
    The second problem I have is when I go to make a phone call, when the call is establishing it now takes sometimes up to 40 seconds of silence to get to the ringtone! With 2.2 it almost seemed instantaneous, and now there is this massive delay which is infinitely infuriating!
    I think this second issue may be linked with the third, which is pretty much the opposite. When I end a call there is another long delay, though not normally as long as the setup delay - 15-20 seconds. Pressing the home button takes me back to the home screen with the green bar at the top saying the call is in progress, pressing the green bar takes me back to the call and allows me to press the end call button again, to no avail. All this whilst the call timer is still counting.
    The last issue is with the calls itself... Now when I make a call, about 50% of the time it will drop out in the first 30 seconds of the call, I will not get a call failed message (which I had seen a couple of times in the past); it will just sit there like the person on the other end of the phone has gone silent and the call timer will still be incrementing. This coupled with the call end and redial delays make for an almost 1 minute round trip from the call dropping out to me calling back - this is extremely infuriating!
    Now, I thought I could just roll back to the 2.2 update when everything was peachy... How wrong I was; it appears that the baseband has been altered in such a way with the 2.2.1 update that the 2.2 update cannot be restored to the phone, I've even gone to the lengths of jailbreaking the phone only to realise that the baseband cannot be rolled back, even if the firmware is.
    So, this puts me in a situation where I have an infuriating phone to use because I cannot roll back to the last known good update for it - it was reasons like these call issues that I moved away from Windows Mobile and Symbion devices. My iPhone was great until this update - I just want it back to good working order!
    I've tried restoring the phone, rebooting the phone, restoring to a 2.2 release (gives a #1013 error when trying to update it). Is there any way I can get this back - or is there a 2.2.2 release planned that will solve any of these issues?
    Cheers,
    --ryan

    Have you done a reset (hold the sleep/wake & home button until the Apple logo appears)? A reset will often correct many issues you may have with the phone.
    If a reset or restore does not correct the issues, you could try a new SIM from ATT or make an appointment at the Genius Bar to have the phone looked at.
    This is a user to user forum. No one here knows when or what the next update will contain.

  • DECT phone outgoing call issue

    Has anyone had issues with DECT phone outgoing calls -  My new BT Studio 3500 can receive calls ok, but dialling out returns "this number is not recognised" - using my old analogue phone and it suffers no such issues.  Is there something up with my phone line - I have ADSL connected, this works fine with microfilter and analogue- even when I disconnect the ADSL line the DECT phone still has an outgoing call issue.  I even replaced the DECT phone with a another model previous, but the issue still remains. 
    Im not contracted to BT but go via TalkTalk, and just want to find out if I am setting the DECT phone up incorrectly!

    So rather than try and troubleshoot your problem, or take the phone in to have it checked by Apple for a hardware fault, you're just going to wait for something to magically change... OK, then.

  • IPhone 4 voice call issues

    My iPhone 4 sometimes goes into Facetime in the middle of a voice call. I don't have a screen protector that might block the proximity sensor. I have done a full reboot and am on the latest iOS. Any ideas what is wrong?

    I've been having a very similar issue that just started last night. I keep dropping calls and when I try to call out nothing happens at all. I cannot browse Safari through 3G either. It keeps coming and going. I was able to finally place a call out and then tried making another call later and it wouldn't work again. I tried the restore as well and nothing works!
    I do not know if I should contact AT&T or Apple for the issue. I do have the extended warranty that I purchased with the phone but about a week after I got the iPhone 4 I had to have it replaced for issues I am not even sure that they would replace it again. I do not have a home phone I rely solely on my cell so I cannot wait to have it fixed or replaced but I am not going to keep a phone that doesn't work especially after paying so much for it!

  • Voice Call issues

    Hello,
    We have Exchange 2010 and Blackberry 5 in our environment. This query is for a single user(User A)only who is having exchange active sync and Blackberry.
    When users call him via Blackberry address book, the call lands to user B instead of User A.
    If the user is called directly from voip, landline the issue does not occur.
    The issue is observed only if any user calls this user A to his blackberry. However, the issue is not observed on I phone. The phone has been wiped alreday.
    Regards
    Ajit

    Can you share the SIP dialog between CUCM and the MCU? INVITE, 100, 180/183, 200 OK, ACK?
    CUCM can do the packet capture directly from the CLI as long as you run it from whatever node the SIP trunk is being sourced from (primary of that trunk's CMG).

  • Iphone 4s phone call issues

    I just bought an iphone4s, white, 32gb. Sometimes when I make a phone call the person that I am ringing cannot hear me but I can hear them. I have to turn the phone off and then on again, then call them before they can hear me.  I dont have the phone in a cover as yet. Is anyone else having the same problem and is there a solution?

    brconflict1 wrote:
    Since the latest updates to iOS, I'm hearding more issues reported to my company where iPhone users are not able to be routed correctly in our IVR. From the PBX, I'm seeing such as the following example: Callers dialing an extension 312, for example are able to dial 31, but the 2 is not responding on the iPhone. When the caller does eventually get the 2 to go through, the PBX states 31 is an invalid extension, and then registeres the 2 as a new entry, which is our Sales Queue option. I have confirmed this on my iPhone 4S.
    Dialing in the main keypas dialer on the iPhone is fine. However, the additional, smaller keypad for IVR nagivation use is not.
    IF the keypad is fine on the iphone, but not smaller keyboard whatever that is, what makes it an iphone/Apple or software issue.

  • IPhone 4 Phone Call Issues

    About a week ago I noticed my iphone 4 (updated IOS) was having issues with phone calls.  I would make a call to someone and mid-call they could no longer hear me.  In other cases, I made a call to someone and they never could hear me at all.  Other cases, someone calls me and they cannot hear me.  Finally ther are cases where I call someone and mid-call there is a bunch of static and they cannot hear me.
    I spoke with Apple and they told me that I needed to restore the phone and install it as a new iPhone.  This should deal with the issue.  I did such steps and then synced the phone and tried to make a call.  The first three calls nobody could hear me.  The next call they could hear me for 2 minutes and then static and nothing.
    I have had a new SIM card installed and have reset the network as well.  In reading the Apple Support Community postings, it looks like the only long term solution is to acquire a new phone.  Is there any other solution anyone knows of to help remedy this situation.  I have had the iPhone 4 for 2 years without an issue and in the last week, this crops up.  It has not been in water, it has a case on it, I am not covering up the mics, the speaker phone is off. 
    Next to getting a new phone, any ideas?

    There is nothing to sort out.
    Basic troubleshooting from the User's Guide is reset, restart, restore (first from backup then as new).  Try each of these in order until the issue is resolved.
    If the issue still occurs, take the device to Apple for evaluation as there is obviously a hardware issue.

  • Calling issue with Cisco 7937 conference station

    Hi Friends,
    I am facing issue wiht Cisco 7937 conference station, our customer have various branch offices accross the world. All branches are connected over MPLS through service provider( SIP service provider) . there is a centralized CUCM and remote office have SIP Voice gateways .
    When making calls from once remote site to another using Cisco 6921 phones calls working fine
    When making calls from once remote site to another using Cisco 7937 conference station to make call  any phone at remote office, calls are getting disconneted, remote phone rings when calls,  but its gets fast busy tone when other party picks up the phone and  not able to talk.
    I suspect the issue with Codec but we have configured transcoders  in VG and registered with CUCM
    Please help me if any one experience such issue earlier.
    Regards
    Siva

    hi Basant,
    1. Actually tow phones A and B are registerd with centralized CUCM, A and B are located in two different locations, RTP traffic between And B pass through service provider. 
    Call Flow --> Phone A ---->CUCMRouterpattern--> SIP trunk ----> Voice gateway--->Service provider cloud---> Respective Voice Gateway---> CUCM -- Phone B
    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
    sip
    rel1xx supported "track"
    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
    !Application: CUE Running on ISM
    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
    duplex auto
    speed auto
    interface ISM0/1
    description Internal switch interface connected to Internal Service Module
    no ip address
    shutdown
    interface Vlan1
    no ip address
    ip forward-protocol nd
    no ip http server
    no ip http secure-server
    ip route 0.0.0.0 0.0.0.0 10.198.2.1
    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
    ip route 192.54.47.0 255.255.255.0 79.121.254.81
    ip access-list extended SIP-InBound
    permit ip host 77.37.25.46 any
    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
    voice-port 0/1/3
    compand-type a-law
    timeouts initial 60
    timeouts interdigit 60
    timeouts ringing infinity
    caller-id enable
    no ccm-manager fax protocol cisco
    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#

  • Site to Site calling issue - Cisco 2911 Dial Peer Configuration

    My customer dials from remote site to main site to their main site number, the call by-passes their auto attendant and goes directly to any random available party. 
    At first fingers were pointing to the their PBX, however we noticed one of their sites that wasn't managed by our company did not have the issue.   We cut that site over to our service and the issue started right up.  I believe it is possibly due to the way the dial peers are configured and how the calls route into the PBX.  Unfortunately I do not understand much about them and curious to know if anyone has any history on a issue similiar to this or any input whatsoever?
    Cisco equipment/Dialpeer config below ........
    co IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.2(4)M4, RELEASE SOFTWARE (fc2) - Cisco CISCO2911/K9
    dial-peer voice 100 voip
     description --- VoIP Dial-Peer ---
     translation-profile outgoing 7digit
     huntstop
     preference 1
     service session
     destination-pattern .T
     progress_ind setup enable 3
     session protocol sipv2
     session target sip-server
     incoming called-number .T
     voice-class codec 99  
     dtmf-relay rtp-nte
     fax-relay ecm disable
     fax rate 14400
     fax nsf 000000
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 150 voip
     permission none
     description 900 block
     huntstop
     destination-pattern 1900T
     session protocol sipv2
     session target sip-server
     voice-class codec 99  
     dtmf-relay rtp-nte
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 151 voip
     permission none
     description 900 block
     huntstop
     destination-pattern 900T
     session protocol sipv2
     session target sip-server
     voice-class codec 99  
     dtmf-relay rtp-nte
     ip qos dscp af41 signaling
     no vad
    dial-peer voice 101 pots
     description --- INCOMING Calls from PBX ---
     incoming called-number .T
     direct-inward-dial
    dial-peer voice 1001 pots
     description --- Calls to the PBX ---
     preference 3
     destination-pattern .T
     port 0/0/1:23
     forward-digits 4
    Here is some ISDN debug information
    BAD CALL
    Protocol Profile = Networking Extensions
    0xA11C0201420201008014484152545F20484F54454C535F434C4159544F4E
    Component = Invoke component
    Invoke Id = 66
    Operation = CallingName
    Name Presentation Allowed Extended
    Name = XXXXXXXXXXX
    Display i = ''XXXXXXXXXXX''
    Calling Party Number i = 0x2180, ''XXXXXXXXXX''
    Plan:ISDN, Type:National
    Called Party Number i = 0x80, ''6551''
    Plan:Unknown, Type:Unknown
    Aug 19 16:10:47.242 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB15
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: RX <- CONNECT pd = 8 callref = 0xAB15
    Channel ID i = 0xA98381
    Exclusive, Channel 1
    Aug 19 16:11:02.634 GMT: ISDN Se0/0/1:23 Q931: TX -> CONNECT_ACK pd = 8 callref = 0x2B15
    GOOD CALL
    Protocol Profile = Networking Extensions
    0xA116020144020100800E475245454E204D4F554E5441494E
    Component = Invoke component
    Invoke Id = 68
    Operation = CallingName
    Name Presentation Allowed Extended
    Name = XXXXXXXXXXXXXXXXXX
    Display i = ''XXXXXXXXXXX''
    Calling Party Number i = 0x2180, ''XXXXXXXXXX''
    Plan:ISDN, Type:National
    Called Party Number i = 0x80, 'XXXX''
    Plan:Unknown, Type:Unknown
    Aug 19 16:15:07.999 GMT: ISDN Se0/0/1:23 Q931: RX <- ALERTING pd = 8 callref = 0xAB17
    Channel ID i = 0xA98381
    Exclusive, Channel 1

    I done the configration via CCA  and the running conf i can see two voip dial peer. this is the site where all trunk line roured. Customer from other site2 needs to call outside by taking line from site1.
    dial-peer voice 2100 voip
    corlist incoming call-internal
    description **CCA*INTERSITE inbound call to SITE 1
    translation-profile incoming multisiteInbound
    incoming called-number 82...
    voice-class h323 1
    dtmf-relay h245-alphanumeric
    fax protocol cisco
    no vad
    dial-peer voice 2101 voip
    corlist incoming call-internal
    description **CCA*INTERSITE outbound calls to SITE2
    translation-profile outgoing multisiteOutbound
    destination-pattern 81...
    session target ipv4:192.168.50.1
    voice-class h323 1
    dtmf-relay h245-alphanumeric
    fax protocol cisco
    no vad
    no dial-peer outbound status-check pots

  • Lync 2013 Mobility Client - Video Call Issue (Everything Else works).

    Hey Technet’ies,
    I have been struggling for a couple of weeks now to resolve one of the final issues with a Migration/Upgrade Project on Lync Server 2013.
    I have nearly every Lync Client Scenario working well except the Mobility Client (outside of the domain) which cannot successfully complete a Video
    Call to a Windows Desktop Client (inside the domain) – the toast pops up, but when we click accept - it just times out after 15 seconds
    K
    The strange thing is every other Video Call scenario works;
    Mobility Client (Outside Domain)
    àWindows Desktop Client (Inside Domain)
    = Unsuccessful
    Mobility Client (Inside Domain)
    à Windows Desktop Client (Inside Domain) = Successful
    Mobility Client (Inside Domain)
    à Windows Desktop Client (Outside Domain) = Successful
    Mobility Client (Outside Domain)
    à Windows Desktop Client (Outside Domain) = Successful
    Mobility Client (Inside Domain)
    à Mac Desktop Client (Inside Domain) = Successful
    Mobility Client (Inside Domain)
    à Mac Desktop Client (Outside Domain) = Successful
    Mobility Client (Outside Domain)
    à Mac Desktop Client (Outside Domain) = Successful
    Mobility Client (Outside Domain)
    à Mac Desktop Client (Inside Domain) = Successful
    Mobility Client (All Scenarios)
    à Mobility Client (All Scenarios) = Successful
    Our setup is pretty standard;
    All OS servers are Win Server 2012 R2
    Lync 2013 Standard FE Server
    Lync 2013 Edge Server
    Reverse Proxy IIS AAR (Also Win Server 20012 R2)
    Lync 2013 Desktop Clients, Lync 2011 Mac Clients, IOS and Android Mobility Clients – all latest versions.
    External NAT (via Cisco Routers ACL’s)
    Internal Windows Firewall Currently off on all Servers
    All Certs / DNS Configured as per Technet Recommendation
    I have reviewed / analysed a number of log/trace files – looking for some/any information on the problem;
    Mobility Client (cmlogX.log)
    Windows Desktop Client (*.etl) / Microsoft Message Analyzer
    Lync 2013 Server (CSController)
    With all my looking I could not make out an error except a timeout (15 seconds) on the Mobility Client log (which I currently can’t find).
    Clearly we are missing something and I have gone up and down our implementation a number of times, the only semi-unknown is our old 2010 Pool / Central Management Server (which is currently pretty much redundant but the majority of users still on).
    I would really appreciate any feedback anyone can provide or any insight anyone may have on this challenge.
    Thanks in advance, Alex.
    Alex

    Hi Andrew,
    Thanks for the quick contact!!
    Sorry - typo in my post, I have now corrected it;
    Mobility
    Client (Outside Domain) àWindows
    Desktop Client (Inside Domain) = Unsuccessful
    So I completely missed that blog post of yours (out of the 100's I have read) and you are the first person to clearly say (in simple terms) to point the lyncweb.contso.com to the public IP address on my internal DNS, I have just made that change and see
    if it will correct the issue - will wait for replication. 
    Also I am running IIS / AAR as Reverse Proxy (not ISA) and had the Internal DNS Host A of lyncweb.contso.com and lyncdiscover.contso.com point back to my Reverse Proxy Internal NIC (i.e. 10.1.1.32) as the External NIC is isolated on in the Perimeter Network,
    is this correct?
    So it now goes like this
    *Inside Domain*
    lyncdiscover.contson.com --> 10.1.1.32 (Reverse Proxy Internal Nic) --> 10.1.1.26 (Front End Server)
    lyncweb.contson.com --> 201.183.0.1 (Reverse Proxy External Nic - Public IP) --> 10.1.2.16 (Reverse Proxy External Nic) --> 10.1.1.26 (Front End Server)
    Does this make any sense, I know ISA and IIS/AAR are pretty much interchangeable.
    Thanks again Andrew and look forward to any feedback,
    Alex.
    Alex

  • RFC call  - issue

    Hi all,
        I have a issue in makeing RFC call to a function module . The scenario is , i have a custom built transactions which in turn call few standard transactions in one server(A) and i want to open these transactions from another server(B) using RFC calls and hence i made a remote enabled function module in server(A) and i called this function module from server(B) using destination. But the transaction that i call inturn call another standard transaction in  its code which in turn call another standard transaction using the statement "LEAVE TO TRANSACTION" . Because of this the RFC connectivity is getting disconnected.
       Hence i made a asynchronous call by using the statement
       call function "xxxx" destination 'd' performing new task 't1' .
       This worked fine & i was able to execute the transaction. But this asynchronous call has destroyed the 2EE server process on the ITS . But with other normal RFc calls everything is running fine .
       What should i do now coz this asynchronous call is killing the performance and server. Can anyone suggest me anyother approach.
    Thanks & regards,
    Rajbarath

    Hi,
    In SM59 Check if Radio button for Trusted System is YES (Selected). Then it should not ask even after that if it asks then provide the password in Logon details in Logon and Security Tab available in SM59 for your Destination System.
    Thanks,
    Prashanth
    Edited by: Prashanth KR on Feb 11, 2009 12:13 PM

  • ABAP Webservice Call issue

    Hi
    I am trying to use webservice from ABAP ECC to external server (Webspehere ).
    I have configured RFC connection G Type HTTP connection to external server in SM59.
    Connection sucessfull.we use basic authentication scheme to access external server.
    I am able to call webservice method and getting the response in my program. but while calling second time it doesnot able to get the response. when I see the request message to external server second time, looks like some cookie adding to the request and this is causing no response back to my program.
    I have raised protocal to HTTP1.1 and also selected "No Cookies"  I got few tests sucessfullbut after some time same cookie issue.   Looks like somewhere request was going with cookies...but not sure where....
    anybody have this kind of issue before while calling webservice...? or how to make this cookie to not attach to request ?
    or is this cookie is because of authentication set to token system instead of basic authentication at external server ?
    Please suggest
    Thanks
    Praveen

    Adding details.......
    Experiencing cookie issue while calling webservice to Tivoli system hosted on IBM websphere.
    Steps we have done as follows:
    1. Created a ABAP Proxy class from Tivoli WSDL.
    2. Configured T-Code LPCONFIG to point to above proxy class and logical port.
    3. Configured  SM59 RFC connection to point to Tivoli server using basic authentication mechanism (User ID/Password).
    4. Called webservice method from ABAP program.
    After above steps, we are able to call the webservice method successfully but second time we are experiencing Cookie being adding to the request and eventually no response back to our program.
                When we investigated this cookie issue with IBM Websphere people, they says thay have enabled SSO Config instead of basic authentication and that's why a cookie is being added to the request and fails. when they disabled SSO, we are able to call webservic method sucessfully sevaral times. but now this cannot be no longer disabled as other applications are using SSO enable option. So, we are thinking, is there something that can be done in SAP ECC itself as we see the option of using SSO in SM59 instead of User Id/Password but not sure How?
    If anybody has undergone this scenario/SSO config from ECC SM59 successfully, Please reply back.
    Thanks in advance
    Praveen

  • WebCenter Spaces UCM configuration - GET_ENVIRONMENT service call issue

    After installing and configuring UCM, then WebCenter Spaces, the startup of WebCenter should create a content folder in UCM. However the connectivity between WebCenter and UCM causes an error in the log files:
    On the UCM-side:
    <Nov 9, 2011 2:35:11 PM HKT> <Error> <oracle.ucm.idcibr> <UCM-CS-000001> <general exception
    intradoc.common.ServiceException: !csUnableToLoadEnvironment2!csUnableToExecMethod,loadEnvironmentVars
         at intradoc.server.ServiceRequestImplementor.buildServiceException(ServiceRequestImplementor.java:2115)
    Caused By: java.lang.NullPointerException
         at intradoc.common.TableUtils.getIndexList(TableUtils.java:34)
    On the WebCenter-side:
    <Nov 9, 2011 2:35:11 PM HKT> <Warning> <oracle.webcenter.content.integration.spi.ucm.UCMBridge> <WCS-55028> <Error calling UCM server associated with repository UCMConnection. The service GET_ENVIRONMENT was called with user weblogic at time 11/9/11 2:35 PM, and returned statuscode -32.
    oracle.stellent.ridc.protocol.ServiceException: Unable to load environment. Unable to execute service method 'loadEnvironmentVars'. The error was caused by an internally generated issue. The error has been logged.
         at oracle.stellent.ridc.protocol.ServiceResponse.getResponseAsBinder(ServiceResponse.java:135)
    I did some 'sniffing' of the socket requests between the two and here they are:
    WebCenter sends:
    REMOTE_USER=weblogic
    USER-AGENT=Java; Stellent CIS 11g; Oracle WebCenter 11g
    CONTENT_TYPE=text/html
    HEADER_ENCODING=UTF-8
    REQUEST_METHOD=POST
    CONTENT_LENGTH=204
    HTTP_HOST=CIS
    $$$$
    NoHttpHeaders=0&IsJava=1&IdcService=GET_ENVIRONMENT
    <?hda jcharset=UTF-8?>
    @Properties LocalData
    ClientEncoding=UTF-8
    IdcService=GET_ENVIRONMENT
    UserTimeZone=UTC
    UserDateFormat=iso8601
    ThreadID=15
    @end
    UCM replies with:
    Content-type: text/plain; charset=utf-8
    Content-Length: 531
    <?hda version="11gR1-11.1.1.5.0-idcprod1-110413T184243" jcharset=UTF8 encoding=utf-8?>
    @Properties LocalData
    StatusCode=-32
    blFieldTypes=StatusMessage message
    IdcService=GET_ENVIRONMENT
    ClientEncoding=UTF-8
    UserTimeZone=UTC
    UserDateFormat=iso8601
    NoHttpHeaders=0
    ThreadID=15
    StatusMessage=Unable to load environment. Unable to execute service method 'loadEnvironmentVars'. The error was caused by an internally generated issue. The error has been logged.
    dUser=weblogic
    blDateFormat=yyyy-MM-dd HH:mm:ssZ!tUTC!mAM,PM
    IsJava=1
    @end
    Thanks in advance for any insight into this issue.
    Edited by: Snowy on Nov 9, 2011 12:45 AM

    It turns out, the "Intradoc ServerPort" was not set in UCM. In Enterprise Manager: Content Server: Oracle Universal Content Management: Configuration, it was blank. As soon as I set this to "4444" and restarted it fixed this error.
    I'd like to thank all who contributed.

Maybe you are looking for