Cisco phone 3905 pickup group

Hi,
i have a CME 10.0 with cisco phone 3905.
The pickup group don't work correctly. When i press the softkey gpickup, i heard the reorder tone.
The phone have firmware 9.4.1
Can help me?
Thanks,
Simone

No workaround but in CCM 4.2 there is a new feature that will help you to see what is the call that you are going to pickup:
Call Pickup Notification
This new feature allows users to receive an audio and/or visual alert when a call rings on a phone in pickup groups in which they are a member. For multiple line phones, the alert is available for pickup groups associated with the primary line only.
You can configure the following notification parameters in the Call Pickup Group Configuration window:
Type of notification (audio, visual, both, or neither)
Content of the visual notification message (called party identification, calling party identification, both, or neither)
Number of seconds delay between the time the call comes into the original called party and the notification to the rest of the call pickup group members
You can configure the type of audio notification that is provided when a phone is idle or in use in the Directory Number Configuration window.

Similar Messages

Pickup group - Select which phone to pick up

Hello,
I've greated a Pickup group with several phones. Suppose several phones inside that pickup group are ringing, is there any way to specify which one you want to pick up?
Best regards,
Francisco Sedano.

No workaround but in CCM 4.2 there is a new feature that will help you to see what is the call that you are going to pickup:
Call Pickup Notification
This new feature allows users to receive an audio and/or visual alert when a call rings on a phone in pickup groups in which they are a member. For multiple line phones, the alert is available for pickup groups associated with the primary line only.
You can configure the following notification parameters in the Call Pickup Group Configuration window:
Type of notification (audio, visual, both, or neither)
Content of the visual notification message (called party identification, calling party identification, both, or neither)
Number of seconds delay between the time the call comes into the original called party and the notification to the rest of the call pickup group members
You can configure the type of audio notification that is provided when a phone is idle or in use in the Directory Number Configuration window.

Is there a way to answer a PickUp group if you are on the phone?

I have a user with a standard 7942 IP phone with a single DN on a CUCM 8.1 server. She can place a call on hold and make or receive a new call.
However, she's also part of a call Pickup group. She's stating that there isn't a softbutton option to pickup a call from this pickup group if she's on her current line.
Is there something that can be set to allow her to answer the pickup group if she's currently on the phone?
I know that's very basic...if there's something else I need to include configuration wise, please let me know. I just don't want to clutter up the question.

Thanks Jaime,
Does that mean that there isn't an alternative keypress or similar that will allow her to pick up something in the call pickup group short of hanging up on the existing call?

Forced Authorization Code For Cisco SIP Phone 3905

Hi Team,
Can i configure Cisco SIP Phone 3905 to use Forced Authorization Code ? I am using Call Manager 9.X.
Regards,
Praful Sartape

Hi Praful,
Yes you right for CME but for CUCM some sip phones supports FAC , as far as 3905 is concerned it supports FAC .
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/3905/8_6/english/admin_guide/IP05_BK_CDEEDD7F_00_admin-guide-3905_chapter_0101.html#IP05_RF_A5029279_00
I didnt find link for CUCM 9.x but that above link will help.
And also check this thread- https://supportforums.cisco.com/thread/2125452
Rate all the helpful post.
Thanks
Manish

FXS Ports & Pickup Groups

Is there a way to make an analog phone connected to an FXS port a part of a call pickup group that contains both analog phones & IP phones? I setup a lab and used MGCP to add the gateway and I was able to add the DN associated with the FXS port to a call pickup group. However, I am unable to figure out how to answer the call from the analog phone when another IP phone in the call pickup group is ringing.
Thanks in advance

Hi
You are going down the right track with this.
http://forum.cisco.com/eforum/servlet/NetProf?page=netprof&forum=Unified%20Communications%20and%20Video&topic=IP%20Telephony&CommCmd=MB%3Fcmd%3Dpass_through%26location%3Doutline%40%5E1%40%40.1dde5372/0#selected_message
See this other post I made (for a different purpose, but the principal is the same - it just opens up features available to IP phones for FXS ports by registering them using SCCP).
Regards
Aaron
Please rate helpful posts...

Configuring pickup groups

We have a site that has a PagePac unit connected to a FXS port. CM is
4.1.3sr2.
When the receptionist steps away / end of day, they forward the main
number to the overhead night bell.
The Night Bell is in a pickup group, as are all the phones, so that
after hours anyone can pickup incoming calls.
The problem is that because all phones are in the same pickup group, any
user can pickup anyone else's ringing phone, which they don't want. The
other issue is that once a user hears the overhead, they press Pickup
and the incoming call starts ringing at that phone, they get CallerID,
and then can't hit iDivert to send it back to the main number's VM box
if they don't want the call. Instead, they have to wait for RNA in
order for the call to end up in the main number VM.
I investigated the other pickup options:
Group Pickup might work, but means that it's now a two-step process to
pickup a call, but it will allow me to put the phones in one pickup
group and have the overhead in another pickup group. I would remove
"Pickup" from the softkey template for the phones.
I tried to configure OPickup, thinking that I could put the phones in a
pickup group where the pickup group number was in a a partition that wasn't in their calling search
space, and have the overhead in an associated pickup group that _was_ in
their CSS. Doesn't work; CallManager won't let you configure a DN with
a pickup group whose partition isn't in your CSS.
Any other ideas on how to implement this functionality?

Cisco AS recommends "longest idle hunt"mechanism among the members to be the most fair for the hot line staff. For overflow condition, Cisco AS suggest that we use vm dn. However, if you put the vm dn in the longest idle hunt group, you occasionally will have users rolling directly into the vm. To avoid this, create a hunt group with "first available hunt " and put the pilot of the "longest idle hunt" (in a nested fashion) first and then the vm DN (with always route option checked) second.

Hunt Pilot / Pickup Group

I have a scenario that is giving me a problem with pickup groups and I need some assistance. I have 3 extensions (1410, 1411 & 1412) that are part of a pickup group. That pickup group works fine. However, we have an after hours line group with a hunt pilot of 1420 that rings phones 1410 & 1600(broadcast). When a call goes to 1420 it rings 1410 & 1600 just fine but users at 1411 & 1412 cannot pick up the call ringing on 1410. Says call is not available. Is there a different way to set that up so it can pick up that line? I thought maybe adding 1420 to the pickup group but that does not appear to be an option. We are running CM 5.1.
Thanks

Hi Chris,
Hope all is well with you! You are on the right track with adding 1420 to the Pickup Group,but you are hitting this bug CSCsb12946 which prevents this feature from working (still applicable in 5.1);
Set enterprise parameter (System->Enterprise Parameters->Cisco Support Use 1) Cisco Support Use 1 to CSCsb42763 (case sensitive)
This operation added listbox (Route Plan->Route/Hunt->Hunt Pilot->hunt pilot XXXX -> Call Pickup Group) Call Pickup Group.
From this doc;
http://www.ciscotaccc.com/kaidara-advisor/voice/showcase?case=K11612191
1.- Enter
CSCsb42763 (in mixed case) in the Enterprise Parameter 'Cisco Support
Use 1' under the new group 'Cisco Support use Only' and click update. Only the value
CSCsb42763 in this parameter can enable the configuration. Change the
value to anything else to disable this functionality.
2. Go to Hunt Pilot Configuration page and select a call pickup group from the Call
Pickup Group dropdown list and click update.
CSCsb42763
Call Pickup fails on Hunt List
Issue:
Call pickup fails for hunt list when call is CFA from Route Point in 4.1(2)ES41. This is not the designed functionality for this issue and will break the Call Pickup logic.
Workaround:
Use multiple shared lines & CFNA between each shared line.
Related Bugs
Call Pickup fails when ringing line list member
Symptom: Selecting PickUp or GPickUp does not pickup the call from any of the members of a line group. Condition: CM 4.x Workaround: None. Feature not supported. Further Problem Description: Defect CSCsa66224 may appear to have allowed this feature to work in earlier versions.
Call Pickup from line group member fails after es installed
Symptom: A call is CFNA to a hunt pilot/line group. The call can no longer be picked up. Conditions: Calls CFNA to a hunt pilot. 4.1(2)ES36. Workaround: None
Hope this helps!
Rob

Cisco UCM Call Pickup

We recently migrated our old phone system to Cisco IP Telephony and just getting used to UCM. Don't know if this is the correct place to ask, but here is my question to ask. I have two users that are in a call pickup group. (Secretary and Director) The secretary wants to be able to pickup the Directors line if she's not available, but the Director does not want to receive any calls from the Secretary. The director has no means of answering the main line for any reason and kind of have a conflict with doing this thru call pickup. How can I accomplish this? Any suggestions are greatly appreciated. Thanks.

The easiest way to give the secretary the ability to answer the director's calls is to add the director's DN to the secretary's phone. This will allow incoming calls to the director's phone to ring at the secretary's phone simultaneously. Secretary can answer the director's calls and see if the director is on the phone or not.
I hope I understood your question correctly.
"The secretary wants to be able to pickup the Directors line if she's not available, but the Director does not want to receive any calls from the Secretary."
HTH,
If this helps, please rate.

Adding an extension to a Call Pickup Group

I am new to call manager 7.0 and just wanted to know if adding an extension to a Call Pickup group was as easy as just going to the directory number configuration page of the extension and selecting a call pickup group within the "Call Forward and Call Pickup Settings?" Would there be anything else I would need to do?

nope, that's it
Assigning a Call Pickup Group to Directory Numbers
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fscpickg.html#wp1056555
HTH
java
if this helps, please rate

7962 Can't use pickup group

Hey guys, maybe someone can help me out here. I have a pickup group and about 15 phones that are using it. One of the phones, a 7962, cannot pickup the calls. The CSS does have access to the pickup group partition. I've reset the phone several times and even rebuilt the phone completely. Stare and compare reveals no difference in the configs.
The user reports when they press the pickup softkey they get a key not available message. I have confirmed this is the behavior experienced when the line is not a member of a pickup group at all.
CUCM trace reveals nothing in the logs from that phone except for keepalives during a test call. It's like the phone never requests the pickup from CUCM. Another phone that is able to use pickup and answer the call has all kinds of messages from pickupmanager in the SDL.
I opened a TAC case but they are taking their time getting back to me. Any ideas?
Thanks!

Bug below references 7960 phones:
CSCsb74215
CallPickup is Broken !!!
1. I have Three 7960 Phones ---> Phone A ( 1400) in CallPick Group -->1384020005 , Phone B ( 1399 ) ---> in Callpick Group ---> 1384020005 .
Now Phone C ( 1403) Calls Phone A ( 1400 ) , Now don't answer the call on Phone A , Pick the Call from Phone B as both are in same Pickup Group , mesage displayed " No Call Available for Pickup ".
Known Affected Releases: (1)
5.0(1.51.35)
If you don't believe bug is related:
What is the call manager version?
phone firmware version?
Is the line on the 7962 a shared line? If it is a shared line, is part of the same pickup group?
Hope this helps,
Please rate helpful answers,
Thanks.

Call manager 5 in cisco phone

Hello All,
Maximum 3 call managers can be added in CM group and one SRST reference. So totally 4 CM servers but why we have CM server 5 in all cisco phones under network configuration.
-Murali

perhaps the plan at one point was to support more, and nobody ever bothered to adjust the page. In either case 3 +1 as you state is the max.
HTH,
Chris

Help needed for pickup group configuration

I have been handed the task of configuring a pickup group and the setting on a MDs phone and Secretary and need some help please.
There is a Pickup group that a user is already a member of. She is now also secretary to a General Manager and needs to be able to pick his calls up. I have added the GM as and extra line on the secretary's phone but now when the GM's phone rings, all the people in the pickup group can't tell whether it's the secretary's phone ringing or the GM's. Obviously the people in the pickup group don't want to be picking up the GM's calls so is there another way around this or is there a way that on the secretary's phone, her ringtone and the GM's ringtone can be different?
I have tried adding Audio and Visual Alert to the pickup group but no one in the pickup group wants to know who is calling who so I think they just need to be able to differentiate between the secretary and the GM by the sound of the ringtoneâ¦.
I hope that makes sense!!
Many Thanks!
Adam

the second line you put on the secretary's phone is that a shared line?
I would set it up as being just that, and make sure the ring setting is flash only. That shared line (i.e. the GM's extension) should not be part of a pickup group.
try that and see how get on

Cisco Phone 7945

I have cisco phone 7945 and I have upgraded it to SIP but the phone keep registering and never seem to register with my asterisk PBX.
Kindly view my SEP(mac).cnf.xml file for any error.
<?xml version="1.0" encoding="UTF-8"?>

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>admin</sshUserId>
<sshPassword>password</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Central Standard/Daylight Time</timeZone>
<ntps>
<ntp>
<name>172.100.101.229</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5062</securedSipPort>
</ports>
<processNodeName>172.100.101.229</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<backupProxy>172.100.101.229</backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>true</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>240</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>g729</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<phoneLabel>LIGHTNING</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<sipLines>
<line button="1">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>172.100.101.229</proxy>
<port>5060</port>
<name>101</name>
<displayName>101</displayName>
<autoAnswer>
<autoAnswerEnabled>10</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<authName>101</authName>
<authPassword>line secret</authPassword>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*99</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<contact>101</contact>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2">
<featureID>20</featureID>
<featureLabel>Menu</featureLabel>
<serviceURI>http://example.domain.ext/services/menu.xml</serviceURI>
</line>
</sipLines>
<voipControlPort>5060</voipControlPort>
<startMediaPort>16348</startMediaPort>
<stopMediaPort>20134</stopMediaPort>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile></softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<loadInformation>SIP45.8-5-4S</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>00:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<userLocale>
<name></name>
<uid></uid>
<langCode>en_US</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale></networkLocale>
<networkLocaleInfo>
<name></name>
<uid></uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL>http://example.domain.ext/services/authenticate.php</authenticationURL>
<directoryURL>http://example.domain.ext/services/directory.php</directoryURL>
<servicesURL>http://example.domain.ext/services/menu.xml</servicesURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>4</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>

Hi Godfrey,
Could you please collect packet capture from the back of the phone when it tries to register using the link:
https://supportforums.cisco.com/document/44741/collecting-packet-capture-cisco-ip-phone
If the IP phone port is not spanned, then you have to collect the sniffers from the switchport where the phone is connected to. You have to span the switch port and collect the sniffer captures
~Amit

Cisco phone proxy will support on cucm 8.6 or not

Hi as per document i can see that Cisco phone proxy . Without using vpn connect . Customer want to keep configured Cucm on there Jabber or cisco mobile on the iphone and need to get connected when ever they have an internet access .
And also by keeping a small end router at branches havin only one cisco phone needs to connect to the corprate office using proxy.
Is that possible and also is it possible on 8.6 cucm version . Here am just attaching a document info which says version supported
Supported Cisco UCM and IP Phones for the Phone Proxy
Cisco Unified Communications Manager
The following release of the Cisco Unified Communications Manager are supported with the phone proxy:
•Cisco Unified CallManager Version 4.x
•Cisco Unified CallManager Version 5.0
•Cisco Unified CallManager Version 5.1
•Cisco Unified Communications Manager 6.1
•Cisco Unified Communications Manager 7.0

I understand the VPN solution is a good one, but we have already 100 proxy phones deployed and from what I read I'd need to first set up a VPN phone on the cluster before deploying it in the field. This would be difficult to do with users all over the country. Or am I missing something?
I have a question about our ASA though if anyone can answer it. We have to move from an ASA 5510 to a 5520 due to the 100 UC proxy-phone license limitation (even though we have 70 phones it appears that a UC license is taken up for each TFTP server configured on the phone - primary and secondary). When I configured the 5520 I can register new phones no problem. None of the existing phones will register until the CTL file is manually deleted on each phone. Is there a way to seamlessly migrate to the 5520? Such as using the same certs, CTL file etc.. on the 5520? When I configured it it generated it's own self-signed key and CTL file. I did import the certificates from CUCM...

Cisco Phone Control and Presence 8.6.1.1185 with IBM Lotus Notes 8.5.2 (Integrated Sametime Client 8.0.2) - No presence status visible

Hi community,
I am trying to integrate Cisco Unified Presence 8.6.1.10000-34 with IBM Lotus Notes 8.5.2 with the integrated Sametime Client version 8.0.2 via the Cisco Plugins 8.6.1.1185.
Phone control is working fine, whereas the presence status is not shown (= no handset symbol next to the Sametime user). When I look in the preferences of the plugin, I can see that the plugin has connected successfully to the CUCM (8.6.2.20000-2),whereas the connection to the CUPS has not been established.
The user id as well as the password are all the same on all systems. Here is a description of what I have configured via the ciscocfg.exe tool:
Feature Control:
- Enable Phone Status -> checked
- Enable Dial Using Cisco IP Communicator -> unchecked (not required)
- Enable Control Desk Phone -> checked
- Default Mode -> Control Desk Phone
Control Desk Phone Settings:
- Voicemail Pilot Number -> left blank (no voicemail)
- Cisco Unified Communications Manager
     - Servers -> IP address of CUCM
     - Read Only -> unchecked
     - Use as Default CUCM -> checked
     - Synchronize Credentials -> checked
          - Use Sametime Credentials -> checked
Use Secure Connection: -> not required
LDAP Phone Attributes: -> not required
Phone Status Settings:
- Cisco Unified Presence Servers -> IP address of CUPS
- Read Only -> unchecked
- Synchronize Credentials -> checked
     - Use Sametime Credentials -> checked
- Sametime User ID Mapping
     - Use Business Card Attribute -> MailAddress
     - Remove Domain -> checked
- Display Off-Hook Status Only -> unchecked
At the moment I don't see an error in the configuration, but maybe I am wrong. Could anyone please tell me what the error could be?
Thanks a lot in advance!
Kind regards,
Igor

Hi all,
here are some additions to my above post:
Servers and clients used:
1x CUCM 8.6.2.20000-2
1x CUPS 8.6.1.10000-34
1x IBM Lotus Domino Messaging Express Server 8.5.2
1x Sametime Entry Server 8.5.2 (on top of the Domino server)
2x IBM Lotus Notes 8.5.2 with integrated Sametime 8.0.2
2x Cisco Phone Control and Presence with Lotus Sametime (PCAP) 8.6.1.1185
2x Cisco Unified Personal Communicator 8.5.5.19839
Setup:
- CUCM, CUPS and CUPC are working fine, i.e. Desk Phone control via CUPC, as well as availability and presence status are working without issues
- IBM Lotus Domino server is the LDAP Directory, the Sametime Entry Server is installed on top of the Domino server and uses the Domino Directory
- User ID and password on CUCM/CUPS match the ShortName field and password in Domino
- The PCAP plug-in has been manually deployed to both Notes clients with the following configuration:
     - Enable Phone Status -> active
     - Desk Phone Control -> active
     - no credential synchronization for CUCM and CUPS, i.e. every user must fill the user details himself
     - Sametime User ID Mapping is implemented via the LDAP Attribute uid (which is equal to the user id in CUCM)
     - LDAP configuration filled in with details of the Domino server
Phone Control is working fine, also the connection to the LDAP server (Domino) is fine. However, when I type in the credentials for the CUPS server login, I can see (in Troubleshooting pane) that the user (pparker) is connected to the CUPS server for a short period of time and then gets disconnected. After that no connection is possible to the CUPS server, i.e. status is always disconnected.
I have collected the Tomcat (EPASSoap00010.log and security00010.log) logs via RTMT and compared them to the logs from the PCAP plugin. The relevant time period is from 15:14 to 15:17. In the Tomcat logs I can see that the authentication is successful (see attached files), however in the log of PCAP plugin I can see the following messages:
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.1 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG Credential is rejected. Nothing to retry ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.answerChallenge() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
2012/02/03 15:14:35.281 WARNUNG #### Connection rejected presence server ::class.method=com.cisco.sametime.phonestatus.cup.CUPPresenceWatcher.onPresenceServerConnectionRejected() ::thread=CT_CALLBACK.2 ::loggername=com.cisco.sametime.phonestatus.cup
I don't understand why the connection is rejected although the Sametime Internal ID and CUPS User ID match. Does anyone know what the issue could be?
All posts are very much appreciated!
Thanks a lot in advance!
Kind regards,
Igor

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