Cisco Jabber 10.5 call pickup feature - just for added contacts?

Hi everyone,
we are checking out the new Cisco Jabber for Windows client. The call pickup feature is the best new feature so far.
The only issue we are running into: as we are using global pickup groups devided by office we now see every call coming in for everyone.
I would suggest that it would be much better if you could choose, for which group of contacts you would like to enable the call pickup.
Is there a roadmap for this in the future of Jabber call pickup feature? Would be a very appreciated Addon.
Thanks & Cheers!

Roadmap questions cannot be discussed here, either post this question on a partner forum, or reach your AM for this.

Similar Messages

MSVS-2010 users: Can you *PLEASE* verify whether the "View Call Hierarchy" feature works for you?

I recently decided to educate myself in the intricacies of plugin development and have been experimenting with the set of Samples provided. Supposedly, Visual Studio 2010 (the official IDE for this development) has a very useful feature named "View Call Hierarchy", as described here:
http://dhondiyals.wordpress.com/2011/07/20/call-hierarchy-a-handy-feature-in-visual-studio -2010/
My problem is that that feature never works for me in C/C++ projects. Regardless of what function name I right-click, I always get this error message:
       "A function with the name 'xyz' could not be located"
Note: For C# projects, the feature works perfectly.
I have seen on the net some suggestions, from "remove the SDF (Intellisense) files" to "Reinstall Visual Studio".
I have 5 Visual Studio Pro installations in different computers and the feature does not work ever, in any one of them. I have tried with the Adobe source code, I have tried to create a basic MFC project and the feature does not work. I have tried C and C++ and it does not work. Well, you get the idea.
Can you folks share your experience with this issue?
TIA

Karl Heinz Kremer wrote:
Based on the Acrobat XI SDK documentation, MSVS 2010 is NOT supported for
plug-in development:
http://livedocs.adobe.com/acrobat_sdk/11/Acrobat11_HTMLHelp/Plugins_Cr eatingSimplePlug.34.2.html
Karl Heinz Kremer
From your link:
"The project files for the sample applications are created and supported only in the listed compiler versions."
The above is wrong. I have dowloaded the following:
- Acrobat SDK 9.1: Comes with Visual Studio 2008 project files.
- Acrobat SDK X:    Comes with Visual Studio 2009 project files.
- Acrobat SDK XI:   Comes with Visual Studio 2010 project files.

Just for a contact

hi anji Reddy Wangala,
i have heard about u....... very short tempered. congrats for month topper.
regards
Reena

reena , thanks for ur kind info and plan for treat.
<b>naaaaaaaa , from Anji.I am also observing his postings.its good to have to peoples like anji who can deliver solutions w/o copy and paste</b>
Anji ,
lets plan for treat.
Regards
Prabhu

Cisco jabber for android no ask outgoing call

Is available:
1) cucm 8.6
2) The samsung galaxy S4 mini smartphone (android 4.2.2) with the client of cisco jabber for android v 9.1.4
On the smartphone, in the cisco jabber settings, it is specified - to ask for each outgoing call through what to ring.
If to open the address directory of phone and to click a call, the call is made through SIM a card.
How to set up, what the request through what to make an outgoing call would be output?

Take in mind the supported list of devices, in my experience it can be troublesome if you want to uses some of the brands not listed (I played with a Huawei and a LG without luck)
Check http://www.cisco.com/c/en/us/products/collateral/unified-communications/jabber-android/data_sheet_c78-649887.html (Device and OS Requirements)
Regards.
Rolando Valenzuela.

Cisco Mobile 8.0 vs Cisco Jabber (iOS)

Strange problem that a few users on my campus started noticing, that I also have been able to replicate. I can sit at my desk (approximately 15ft. from an access point) and make a call with Cisco Jabber. The call will be dropped within 2 minutes consistently. With the same server settings, etc. loaded into Cisco Mobile 8.0 I can make a call to the same number, from the same location and stay connected for 30 minutes or more (as long as I tested). 2 or 3 other iOS users have been able to replicate this problem. At least 2 of us are on iOS 5 and I believe the 3rd user is on IOS 4.3.x.
Ideas?
Thanks
~Tom

We're having this same issue. We're running CUCM 8.0.3. Cisco Mobile 8.x worked great - but the latest Jabber for iPhone (8.6.5.18781) is experiencing these issues. We are also running iOS 5.1.1 (the latest).
This is causing alot of frustration for our users as I'm sure it is for other people also.
Thanks,
Jon

Unable to Externally Register Phone Services on Cisco Jabber Client via Expressway E

Hi,
I have currently deployed Cisco Jabber along with Expressway C and E for external regeneration without VPN. I have successfully registered IM and Presence service externally on the internet and I am able to chat with other Jabber users.
I am not able to register phone services for Cisco Jabber client who are registering externally over the internet without VPN.
I have checked that all the users are able to use IM and Presence along with Phone services in the internal network and over the internet using VPN. I have configured the required DNS SRV records on both the internal and external DNS Servers.
I am attaching a screen shot of the Jabber Client that is registered over the internet along with this post for your reference.
Appreciate if you can share your thoughts on the same.
Please do let me know if you need any further informaiton.
Thank you.
Regards,
Joseph Chirayath.

I am testing with an android device, and I had to add a "digest user" on the BOT device in order for this to work. Phone services are now connected.

Need Cisco Jabber Client in personal profile

Cisco Jabber earlier it was there in appstore for BB8520 but after I done research I found that it was available now for Enterprise But problem here was I cannot enable workspace as currently our company or my client does not have BlackBerry Enterprise Services…
Is there any way to install Cisco Jabber on personal profile?

it was good to have 8520. Missing Jabber on z3.

Why did VZW remove the call blocking feature?

I believe the other carriers left it in and it is an excellent feature. Does VZW have any plans of including this in an update? This is a great phone but I am annoyed that VZW feels the need to restrict its freatures.

Sorry to hear that a feature is not available, dansx6. I unfortunately do not have any new details or info on whether the call blocking feature will be added back on at a later time. If something is announced regarding a new update for the G3, it should be on display at this link: http://bit.ly/n5snpq .
If you have any additional questions, we are here to help out.
NicandroN_VZW
Follow us on twitter @VZWSupport

Transfer VOIP Calls Between Cisco Desk Phone and Cisco Jabber For IPhone 9.5

Does anyone know how to transfer an active voip call from a Cisco IP Desk Phone to Cisco Jabber for IPhone? I can transfer a call from Cisco Jabber for IPhone to my Cisco IP Desk Phone no problem. I put the call on hold and then click "Resume" on my Cisco IP Desk Phone. However I cannot do the same but the other way around. If I put the call on hold on my Cisco IP Desk Phone, I see "no active call" on my Jabber client. The only information I could find slighlty relevant was using the Mobility Key/Remote Destination Profile feature however this defeats the object as this will forward to an external number, e.g. mobile and I just want to transfer the call within the VOIP environment between the two devices that are using the same directory number.
I am using Cisco Call Manager 9.1(2), Cisco Presence 9.1 and Cisco Jabber for IPhone 9.5.
Any help would be greatly appreciated.
Kind Regards,
Paul Parker.

Did you ever find an answer to this ?
I am seeing the same behavior and trying so see if I can put calls on hold and pick them up both ways also.
The only answer I seem to have found is to use park instead
That would/should work but I would just prefer to hold/unhold
Just not sure why we would not be able to hold/unhold on what is essentially a "shared" line
Does anyone have this working for them ?

Call/video not working between Cisco jabber for Windows and VCS control C40s

Hello,
I've been struggling with no luck how to make a call using Cisco Jabber for Windows 9.6.0 registered to CM 8.6.2 with intercluster ICT to another CM 8.6.2 where we have a VCS Control 7.0.2 via GK H225, and all C40s are registered as H.323.
The VCS has interworking between H323 and SIP, however not sure if there is any problem with that. Assuming it is ok, not sure either if I'm facing any interoperability issue because in my remote site I have C40 (H323 registered at VCS and SIP listening mode) and cisco jabber for windows which is SIP based.
If is not possible, would I be able to change my C40 from H323 to SIP at VCS, or have both H323/SIP registered at VCS? If so, will I need to change as well instead of GK I'll have to establish a SIP Trunk between the CM and VCS?
Another thing I do not believe either I would be able to have one VCS connected with two clusters, right?
I'm just trying to find a solution in case my current topology is not compatible, but feel free if you have any better idea to make it work.
Anyway here is what is happening:
When I make a call from my cisco jabber windows to C40 using alias number. The call is being redirected just fine to the C40 and it rings, however when someoene or the auto answer picks it up, the call dropped right away.
However, if I enabled the MTP in my CSF device, the call gets longer before dropping. I was even able to see my jabber " start video" turns green, before was grayed out all the time and the call dropped faster. I hear a fast busy tone.
I'm able to provide SDI traces, logs, diagnostic sip/h323 calls from VCS in order to know for sure if this is an incompatible issue or something I can workaround.
Let me know if someone of you are interested in read these logs or could point me on the right direction.
Thanks!

Ok,
I have looked at both logs. I have to mentinon though that you didnt
provide the log that shows the h323 setup between cucm and the VCS. This
is most likely because the call originated from a different cucm than
the ones you provided the logs from.
The call would have orginated from the first cucm in the cucm group of
this trunk: Name=RL_TRUNK_VIDEO
The cucm ip will be : 10.252.53.10.
This is the VCS log that confirms where the h323 request originated
from:
pr 10 22:50:29 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:29,187"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="54000"
Received RAS PDU:
Having said that here is my analysis of the logs that you sent..
Jabber sent an INVITE to CUCM and advertised all the codecs (audio and
video it can support)..
Observer that Jabber says it doesnt support G729 anexB
21:55:16.576 |//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message
from 10.223.20.73 on port 54677 index 90661 with 2220 bytes:
[862370,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.223.20.73:54677;branch=z9hG4bK000029d3
From: "4122107" <sip:[email protected]>;tag=00059a3c78000011000070b0
-00000e65
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Fri, 11 Apr 2014 01:55:16 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CSF/9.4.1
m=audio 19252 RTP/AVP 0 8 18 105 104 101
c=IN IP4 10.223.20.73
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 28878 RTP/AVP 97
c=IN IP4 10.223.20.73
++++Now lets observer the capabilites exchange during h245 negotiation
between cucm and VCS++++
Here CUCM advertises its caps to VCS (afterreceiving caps from VCS)
Note that G729A, G729AB, G729 is all advertised..
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,017"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : terminalCapabilitySet
capabilityTableEntryNumber 2,
       capability receiveAudioCapability :
g729wAnnexB : 6
       capabilityTableEntryNumber 3,
   capability receiveAudioCapability : g729AnnexAwAnnexB : 6
       capabilityTableEntryNumber 4,
       capability
receiveAudioCapability : g729 : 6
capabilityTableEntryNumber 5,
       capability receiveAudioCapability :
g729AnnexA : 6
++++++
After doing MSD (master slave determination, we move to the OLC phas e..
Here we see that the far end..c40 wants to use G729AB for media++++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,783"
Module="network.h323" Level="DEBUG": Src-ip="10.224.114.11" Src-
port="11163"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
   forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
     dataType audioData :
g729AnnexAwAnnexB : 20,
     multiplexParameters
h2250LogicalChannelParameters :
+++Next VCS sends G729AB as the codec to use to CUCM+++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,784"
Module="network.h323" Level="DEBUG": Dst-ip="10.252.53.10" Dst-
port="45660"
Sending H.245 PDU:
value MultimediaSystemControlMessage
::= request : openLogicalChannel :
   forwardLogicalChannelNumber 1,
forwardLogicalChannelParameters
     dataType audioData :
g729AnnexAwAnnexB : 20,
     multiplexParameters
h2250LogicalChannelParameters :
++++The next thing we get is an OLC reject from CUCM and this is where
th call drops++
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="45660"
Received H.245 PDU:
value MultimediaSystemControlMessage
::= response : openLogicalChannelReject :
forwardLogicalChannelNumber 1,
   cause dataTypeNotSupported : NULL
Apr 10 22:50:31 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:31,790"
Module="network.h323" Level="INFO": Dst-ip="10.224.114.11" Dst-
port="11163"
Detail="Sending H.245 OpenLogicalChannelRejResponse
+++We then receive a call release from cucm with cause code of 47:
resource unavailable++++
Apr 10 22:50:32 TWELDVCS01 tvcs: UTCTime="2014-04-11 01:50:32,365"
Module="network.h323" Level="DEBUG": Src-ip="10.252.53.10" Src-
port="50913"
Received H.225 PDU:
Q931
   Message Type: Release
Complete
   Call reference flag: Message sent from originating side
Call reference value: 0x7b
   Info Element : Cause
     Location: Usr
   Cause Value: Resource unavailable
   Info Element : User User
   Length = 22
Suggestions:
Change the region setting between the ICT trunk to VCS and Jabber to use
G711 and test again.

Cisco Jabber for Windows in Extend and Connect mode and making outbound calls

Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.

Hi guys,
I've set up Cisco Jabber for Windows to use Extend and Connect to control a remote PBX endpoint. I've configured the required CTI-RD device, remote destinations, associated the users to the line and added the devices to end-user controlled device. The extend and connect part is working flawlessly without any issues. I'm able to receive inbound calls on the remote PBX endpoint and control the call (hold, resume, transfer etc.) using the Jabber call window that pops up.
However, I'm unable to make any outbound calls via the Jabber client when in extend and Connect mode. Reading the Extend and Connect guide, I need to configure Dial Via Office (DVO) Reverse. So when the user initiates a Dial-Via-Office reverse call, CUCM calls and connect to the Extend and Connect device (CTI-RD). CUCM then calls and connects to the number the user dialled and finally connects the two call legs.
After attempting to configure DVO-R for Jabber for Windows in Extend and Connect mode following the CUCM feature services guide, i'm unable to get any outbound calls working. From RTMT, i am receiving the following Termination Cause Code: (27) Destination out of order. What i also notice is that there is no calling number for that trace either. I would've thought that the calling party would've been the Enterprise Feature Access (EFA) number.
Has anyone got this working or can provide some guidance?
Thanks.

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.

Hi Team, I wuold like to know if you have any app to make Firefox OS working with cisco Call Manager 10.5. Something like Cisco Jabber for Android or iOS.

I'm interesting on buying a Firefox Smart Phone, but
I would like to know if are any app to install on Firefox OS smart phone in order to work with cisco call manager 10.5.
Something like Cisco Jabber for Android o iOS.
Thanks,

Hi Itech,
If Cisco Jabber has a webapp, or mobile version of their website available, you should technically be able to access it through Firefox OS.
You may also search Firefox Marketplace for an alternative solution:
* [https://marketplace.firefox.com/]
- Ralph

Cisco Jabber for Windows unable to make calls in softphone mode

Hi,
I am facing an issue when I make a call from jabber in softphone mode to another jabber client my call fails and using wireshark traces I see SIP2.0 503 Service Unavailable.
I have followed the guide and was able to deploy it once before successfully. This is a lab environment and we don't have any CSS or partitions.
If someone can please guide me :)
Regards
Hassan

Hi Hassan,
I found four calls in the trace file with the same calling/called numbers. I analyzed one of them.
Calling party - 42001
Called party - 42002
00267676.002 |17:15:14.335 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 122.129.75.131 on port 23600 index 689 with 2610 bytes:
[7525,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 192.168.1.60:54441;branch=z9hG4bK00005d94
From: "42001" <sip:[email protected]>;tag=3c77e6734e8e00180000789e-000014f2
To: <sip:[email protected]>
Call-ID: [email protected]
Max-Forwards: 70
Date: Tue, 21 Apr 2015 11:29:56 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CSF
Contact: <sip:[email protected]:54441;transport=tcp>;video;bfcp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "42001" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Recv-Info: conference
Recv-Info: x-cisco-conference
Content-Length: 1469
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 5552 0 IN IP4 192.168.1.60
s=SIP Call
b=AS:4000
t=0 0
m=audio 21996 RTP/AVP 9 104 105 0 8 18 101
c=IN IP4 192.168.1.60
a=rtpmap:9 G722/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 25898 RTP/AVP 126 97
c=IN IP4 192.168.1.60
b=TIAS:4000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42E01F;packetization-mode=1;level-asymmetry-allowed=1;max-fs=3601;max-rcmd-nalu-size=32000
a=imageattr:126 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]
a=content:main
a=label:11
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42E01F;packetization-mode=0;level-asymmetry-allowed=1;max-fs=3601
a=imageattr:97 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]
a=sendrecv
m=video 32652 RTP/AVP 126 97
c=IN IP4 192.168.1.60
b=TIAS:4000000
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42E01F;packetization-mode=1;level-asymmetry-allowed=1;max-fs=4099;max-rcmd-nalu-size=32000
a=content:slides
a=label:12
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42E01F;packetization-mode=0;level-asymmetry-allowed=1;max-fs=4099
a=sendrecv
m=application 5070 UDP/BFCP *
c=IN IP4 192.168.1.60
a=floorctrl:c-s
a=confid:4
a=floorid:2 mstrm:12
a=userid:4
a=setup:actpass
a=connection:new
a=sendrecv
00267702.007 |17:15:14.342 |AppInfo |Digit analysis: match(pi="2", fqcn="42001", cn="42001",plv="5", pss="", TodFilteredPss="", dd="42002",dac="0")
00267702.008 |17:15:14.342 |AppInfo |Digit analysis: analysis results
00267702.009 |17:15:14.342 |AppInfo ||PretransformCallingPartyNumber=42001
|CallingPartyNumber=42001
|DialingPartition=
|DialingPattern=42002
00267709.001 |17:15:14.343 |AppInfo |LBMIF: CI: 30582939 ASSOC    30582940
00267709.002 |17:15:14.343 |AppInfo |LBMIF: CI: 30582940 ASSOC'   30582939
00267715.000 |17:15:14.343 |SdlSig   |CcSetupReq                             |restart0                       |LineControl(1,100,174,27)        |Cdcc(1,100,219,42)               |1,100,14,10382.36^122.129.75.131^CSFHASSANRAZI |[R:N-H:0,N:2,L:0,V:0,Z:0,D:0] CI=30582940 CI.branch=0 sBPL.plid=65 sBPL.l=1 sBPL.pl=5 sBPL.msd=0 FDataType=0opId=0ssType=0 SsKey=0invokeId=0resultExp=Fbpda=F pi.piid=30 pi.l=0 pi2.piid=30 pi2.l=0 pi3.piid=30 pi3.l=0 FQCGPN=ti=1nd=42001pi=0si1 preXCgpn=tn=0npi=0ti=1nd=42001pi=0si1 cgPart= cgPat=42001 cgpn=tn=0npi=0ti=1nd=42001pi=1si1 cgpnVM= unXCgpn=tn=0npi=0ti=1nd=42001pi=1si1 cName=locale: 1 Name: UnicodeName: pi: 1 DD=tn=0npi=1ti=1nd=42002User=42002Host=192.168.201.101Port=5060PassWord=Madder=Transport=4mDisplayName=RawUrl=sip:[email protected];user=phoneOrigPort=0pi=0si1 origDD=tn=0npi=1ti=1nd=42002User=42002Host=192.168.201.101Port=5060PassWord=Madder=Transport=4mDisplayName=RawUrl=sip:[email protected];user=phoneOrigPort=0pi=0si1 preXCdpn=tn=0npi=0ti=1nd=42002pi=0si0 preXTagsList=SUBSCRIBER preXPosMatchList=42002 cdPart= cdPat=42002 cdpn=tn=0npi=0ti=1nd=42002pi=1si1 cdpnVMbox= localPatternUsage=2 connectedPatternUsage=2 itrPart= itrPat= LRPart= LRPat=42002 LR=tn=0npi=0ti=1nd=42002pi=0si1 LRVM= LRName=locale: 1 Name: UnicodeName: pi: 0 FQOCpdn=ti=1nd=42002pi=0si1 fFQLRNum=ti=1nd=42002pi=0si1 oPart= oPat=42002 oCpdn=tn=0npi=0ti=1nd=42002pi=0si1 oCdpnVM= oRFR=0 oName=locale: 1 Name: UnicodeName: pi: 0 ts=SUBSCRIBER posMatches=42002 withTags= withValues= rdn.l=0IpAddrMode=0 ipAddrType=0 ipv4=122.129.75.131:23600 region=Default capCount=7 ctiActive=F ctiFarEndDev=1 ctiCCMId=1 cgPtyDev=CSFHASSANRAZI callInst=0 confCallInst=0 OLF=1Supp DTMF=3DTMF Cfg=1DTMF Payload=101isOffNetDev=F bc.l=4 bc.itr=8 bc.itc=8 bc.trm=6 bc.tm=16 maxForwards=69 cgpnMaskedByRedirect=F callingDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 featCallType=0 callingUserId= UnicodeName: muteEnabled=0 associatedCallCI=0 featurePriority=1 nonTargetPolicy=0 unconsumedDigits= suppressMOH=F numPlanPkid =6807b55c-7ab5-4f8d-8fba-db383ecfaaa4 networkDomain= bitMask=0 SetupReason=0 routeClass=1 sideACmDeviceType=4 protected=1 ControlProcessType=0 tokens=0 isPresent=F transitCount=0 geolocInfo={geolocPkid=, filterPkid=, geolocVal=, devType=4} locPkid=29c5c1c4-8871-4d1e-8394-0b9181e8c54d locName=Hub_None deductBW=F fateShareId=StandAloneCluster:30582939 videoTrafficClass=Desktop oFromAnalogDvc=F bridgeParticipantID= callingUsr= remoteClusterID= isEMCCDevice=F lHPMemCEPN= cHPMemCEPN= uri=ti=1User=Host=Port=0PassWord=Madder=Transport=4mDisplayName=RawUrl=<sip:[email protected]:54441;transport=tcp>OrigPort=0pi=0si1 param=;video;bfcp M=Unknown ;rc=26624 Hdrs= CanSupportSIPTandN=true TransId=0 AllowBitMask=0x7bf UserAgentOrServer=Cisco-CSF OrigDDName=locale: 1 Name: UnicodeName: pi: 0 mCallerId= mCallerName=LatentCaps=null icidVal= icidGenAddr= oioi= tioi= ptParams= receivedPAID= routeHdr= routeCepn= requestURI= PCVFlag=F originallyHadISUP=F isIMSFinalRoute=F IMSMode=0 SideABibEnabled= 0 isCgpnNonPreemptable=F isCdpnNonPreemptable=F origDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 lastRedirectingDP=1b1b9eb6-7803-11d3-bdf0-00108302ead1 originalLRG= lastRedirectingLRG= nwLoc=0 rstr= FarEndDeviceName=CSFHASSANRAZI hdrMOH=0 CAL={v=ffffffff, m=ffffffff, tDev=F, res=F, devType=0}
00267715.001 |17:15:14.343 |AppInfo |LineControl(27) - 0 calls, 0 CiReq, busyTrigger=2, maxCall=6
00267715.002 |17:15:14.343 |Created |                                       |                               |LineCdpc(1,100,175,74)           |LineControl(1,100,174,27)        |                                         |NumOfCurrentInstances: 2
00267758.001 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 55 (SIPSPI_EV_SEND_FAILURE_MSG)
00267758.002 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/ccsip_spi_process_event: Send Error for event(0xe5139420)
00267758.003 |17:15:19.364 |AppInfo |//SIP/Stack/Error/0x0/act_idle_send_msg_failure: Send Error to 192.168.1.172:54901 for transport TCP
00267758.004 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/ccsip_set_cc_cause_for_spi_err: Categorized cause:38, category:186
00267758.005 |17:15:19.364 |AppInfo |//SIP/Stack/Info/0x0xe5136ad8/sipSPIInitiateDisconnect: Initiate call disconnect(38) for outgoing call
00267758.006 |17:15:19.364 |AppInfo |//SIP/SIPHandler/ccbId=4212/scbId=0/ccsip_api_call_disconnected: ccb->cc_disc_cause (38); ccb->sip_disc_cause(503)
00267794.001 |17:15:19.367 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 122.129.75.131 on port 23600 index 689
[7527,NET]
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP 192.168.1.60:54441;branch=z9hG4bK00005d94;received=122.129.75.131
From: "42001" <sip:[email protected]>;tag=3c77e6734e8e00180000789e-000014f2
To: <sip:[email protected]>;tag=4211~132b515c-f941-ca80-8023-9423185ada2d-30582939
Date: Tue, 21 Apr 2015 12:15:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Server: Cisco-CUCM10.5
Reason: Q.850; cause=41
Content-Length: 0
Seems like a network issue. TCP connection can't be established with the called jabber client, which as per the trace is 192.168.1.172.
Check for firewalls blocking the connection.
You may take a pcap on the called jabber client side or probably from the call manager, and check for TCP issues.
You mentioned the jabber client's IP as 192.168.1.60 but I see incoming invite from 122.129.75.131. What this IP?
HTH,
Atul

Unable to select 'Use Phone for audio calls' on Cisco Jabber Client 9.6 for Windows

Hi,
I have recently deployed Cisco Presence Server and integrated with Call Manager 9.1.2. I have successfully deplyed 6700 users on IM & Presence. Some of the users requested for Cisco Jabber with phone control.
I have added CFS client on the Call Manager and associated it with the same extension numbers from their desk phones. I am currently able to make audio and video calls for these specific users. I am currently using Cisco Jabber Client 9.6 for windows. I have users both daisy chained to their desk phones and who are not. Can you please confirm if it does make any difference.
Problem Faced -
I am not able to use to option use phone for audio calls. The phone comes down with a cross sign on it. At the same time Cisco Jabber by default uses the client and it works as expected.
Can you please let me know if any of you guys have faced a similar issue.
Please let me know if you need any information regading the configuration used.
Looking forward for your valuable comments.
Thank you.
Regards,
Joseph Chirayath.

Hi Will,
I am attaching the screen shot for the END USER on CUCM 9.X that has been configured on Cisco Jabber.
Please do let me know if you need any further information.
Thank you.

Cisco Jabber 10.5 call pickup feature - just for added contacts?

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