"click to call" posibillity in cisco phones ?

Hello
Is there any way to call numbers on cisco phones directy from clik on website ?
I am looking for sollution like in skype :
https://support.skype.com/en/faq/FA12264/what-is-the-skype-add-on
Users install add on on computer webbrowsers and numbers are automicaly highlited and then user can click this and call
I am looking this for hardware voip phones.
Is this possible on any cisco or non-cisco phones ?
Sorry for my bad english

Skype is working to provide a Click to Call update as close to a Firefox browser update release as possible. The current Skype click to call plugin is compatible with Firefox 26.
Follow the latest Skype Community News
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Similar Messages

Calls record on Cisco Phones

Hi,
I just want to know that if i will change my cisco ip phone with another cisco ip phone (same model or different model) is it possible to move all the missed, received and placed calls data onto the new phone?
These phones would be registered to Cisco CallManager 7.1.5.

Hi,
the phone maintains logs of all missed, recived and placed calls.
you cannot move them..
http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7962g_7961g_7961g-ge_7942g_7941g_7941g-ge/8_0/english/user/guide/62614241log.html
regds,
aman

Click to Call--highlighted phone numbers on contac...

In my gmail contact list, most of the the phone numbers are highlighted for use with Skype Click to Call. However, the phone numbers with country codes for France are not highlighted. In previous versions of click to call, they were highlighted along with all the other numbers. I have tried uninstalling and reinstalling Click to Call. No change. Suggestions?

Hi and welcome to the Skype Community,
1. Unfortunately "Click to call" is not available on Safari for iPad because the Safari Web Browser in iOS does not support the concept of plug-ins (like Safari on Mac does). That means Skype cannot hook into the browser to extend the functionality.
2. Just so I understand you correctly: You want phone numbers in Emails highlighted in the desktop application "Outlook 2013" as well, correct?
Follow the latest Skype Community News
↓ Did my reply answer your question? Accept it as a solution to help others, Thanks. ↓

Click to Call and Presence Options with Sharepoint

Hi everyone,
What options are there for providing click to call (CTC) and presence with Sharepoint? I have the following options (and thoughts) so far.
Deploy CUPC (Not going to happen, we're happily a Lync 2010 house but would provide CTC and unsure about presence for sharepoint)
Deploy CUCILync (Seems overkill, but would provide CTC and presence)
Deploy CUPS and use RCC via Lync (I keep on hearing that RCC was meant to be killed off in Lync 2010 and probably won't exist in future. Would provide CTC, unsure about presence)
Deploy click to call widget (Would be great to avoid another desktop application. Would only provide CTC)
Deploy Quad (Don't expect the Sharepoint guys buying in to that one...)
Use WebDialer API (Sounds like it would provide CTC)
Use existing JTAPI integration (This should be a given... Provides CTC and presence).
We currently have an existing Intranet, which provides click to call abilities/phone presence information using JTAPI. A team is completing a Sharepoint proof of concept and the question has come up how they can complete click-to-call and phone presence.
Thanks,
Mark

The first suggest to fix this error is to make sure that CIPC is running properly and that
a phone is selected to use with the Click to Call application.
This message is often presented unnecessarily when the ser invokes Click to Call but has not
entered their username and password into Click to Call Preferences then the user is
presented with the error "Click to Call cannot find Cisco IP Communicator. Please ensure
it is running or select another phone." rather than "Click to Call is not fully
configured."
If you have configured the user options correctly then you need to check the Click to Call log
files to try and diagnose the problem you're having. They are located at:
* Windows XP - C:\Documents and Settings\[Windows User Account Name]\Application
Data\Cisco\Click to Call\Log
* Windows Vista - C:\Users\[Windows User Account Name]\AppData\Roaming\Cisco\Click
to Call\Log
Please rate all useful posts

CUCM 8.6 Dropped call transfers involving SIP phones

Hi All,
I am a developer who has been tasked with figuring out why call transfers are being dropped by Cisco CUCM when the original call comes from a SIP phone. This scenario works:
Cisco phone calls another Cisco phone, which transfers the original call to a SIP phone
These scenarios do not work:
SIP phone calls Cisco phone, which transfers the original call to another Cisco phone
SIP phone calls Cisco phone, which transfers the original call to another SIP phone
I have researched the Call Manager traces to the best of my ability, and I see some info in there that could potentially point to the source of the problem. I am just unable to understand what the trace means:
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/active_CcDisconnReq: ccDisconnReq.onBehalfOf=Media : ccDisconnReq.s.sv=2 : ccDisconnReq.c.cv=47 |1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/Stack/Info/0x0/sipConstructContainerContext #### Created container=0xb0b42f58|1,100,71,1.1^*^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendReasonHdr: appendReasonHdr - Invalid Disconnect Cause(cause=47), No Reason Header Appended|1,100,63,1.93259^10.10.10.85^*
10:23:08.672 |//SIP/SIPCdpc(1,74,2342)/ci=24377698/ccbId=175645/scbId=0/appendRPIDHdrForOriginalCalledParty: SIP device does not Support Orig Dialled Phone nego: 0|1,100,63,1.93259^10.10.10.85^*
I have been wondering whether this could be a codec issue, however the SIP phones we are using are configured with the following codecs:
G711U
G711A
G722
ILBC
GSM
and our SIP software is also set to accept the first codec offered by the remote side. It seems from the SIP client logs that G722 is being used as the codec to communicate with the Cisco phones, but perhaps I'm misinterpreting.
I have attached a CUCM trace of a call from a SIP phone (ext. 491) to a Cisco handset (ext. 170) where the Cisco handset attempts to transfer the call to another SIP phone (ext. 492). The trace snippet shown above is from this log.
I would really appreciate it if someone more experienced with VoIP/SIP/CUCM could take a look and offer any ideas on what the issue might be, and also how we might be able to address it. I can try to provide more info about our CUCM configuration if needed.
Thanks in advance!

Leslie, so here is what I found from the traces....
To understand the difference we need to understand how cucm performs call transfers from a sccp signalling point and a sip signalling point
SCCP
When the transfer key is pressed
1. CUCM sends a CloseReceiveChannel and StopMediaTransmission to the IP phone involved in active media (referenced by the callids)
NB, here CUCM updates the call state on the phone to a call state of 8 which is "Hold"
2.CUCM tells the held party to listen MOH from MOH server
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5..CUCM sends a CloseReceiveChannel between the held phone and MOH server (to tear down the media)
6. Next CUCM sends a CloseReceiveChannel and StopMediaTransmission to the transfering party & transfered party to remove Xferring party from call
7. finally CUCM sends OpenReceiveChannel between the original called party and the transfered party..and call is done
For SIP signalling. when the first transfer key is pressed
1. CUCM sends invite (re-invite) with an inactive SDP (a=inactive) to indicate a break in media path
2. CUCM sends a Delayed offer to insert MOH or to resume Media stream
NB: CUCM expects a sendrecv offer with SDP to the DO. (NB:if cucm gets an inactive offer SDP in the 200 OK instead of providing a send-recv offer SDP, the media path remains in an inactive state and causes calls to dropcall will drop),CUCM sends an ACK with sendonly to the 200 OK
3.CUCM establishes newcall leg with the intended transfered destination..Once this call is connected
4.CUCM receives a new Transfer instruction from the transferring phone to connect the held party
5. Next CUCM sends a re-invite with an inactive SDP to indicate a break in media path to MOH (in attempt to complete transfer)
6.Next CUCM sends an inactive SDP to indicate a break in media path between transfering party & transfered party to remove Xferring party from call
7. Next CUCM sends a DO re-invite to connect the transfered party. The far end then sends 200 OK with the required SDP to connect the call
Now having explained all of these, we need to look at where the call is failing for SIP-----SCCP----SIP calls without MTP
lets look at succesful SCCP-----SCCP-----SIP without MTP
Point 4 above
++++++++Extension 170 presses the transfer button to connect the two calls (Callid=24378483)+++++++++++++
(0003395) SoftKeyEvent softKeyEvent=4(Trnsfer) lineInstance=1 callReference=24378483
Point 5 above
++++Next CUCM closed the media between extension 160 and MOH server callid=24378480(this is the only active call on this callid)+++
(0003396) CloseReceiveChannel conferenceID=24378480 passThruPartyID=16777845. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
Point 6 Above
+++++Next cucm closes the call between extension 170 and 490 callid=(24378483)++++++++
(0003395) CloseReceiveChannel conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
(0003395) StopMediaTransmission conferenceID=24378483 passThruPartyID=16777847. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a8b(10.10.10.139)
Point 6 above for the sip side (since the destination is SIP, to tear down media to SCCP phone, so as to connect the caller to the xfered party)
+++++++Next CUCM sends a re-invite with a=inactive SDP to the sip phone ++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885626,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23332dbee978
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
o=CiscoSystemsCCM-SIP 192115 2 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0
m=audio 24560 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=inactive-----------------------------------------------------Inactive
Still part of Point 6 for SIP signalling
++++++++++++Next sip phone responds with a 200 OK recevonly SDP +++++++++++++++++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885628,NET]
SIP/2.0 200 OK
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
a=ptime:20
a=recvonly-------------------------------------a=recvonly
Finally Point 7 above..
+++++++++++++=Next cucm sends a DO re-invite to extension 492-sip phone++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885630,NET]
INVITE sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
+++++++Next we get a 200 OK from sip phone with sdp=sendrecv+++++++++=
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 62220 index 1890 with 683 bytes:
[885634,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK233534ffec4a
Contact:
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
Call-ID: [email protected]
v=0
o=- 18077 11099 IN IP4 10.10.10.104
s=yasdjip
c=IN IP4 10.10.10.104
t=0 0
m=audio 16574 RTP/AVP 9 101
a=rtpmap:101 TELEPHONE-EVENT/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
+Now CUCM sends an OpenReceiveChannel and start media xmission to sccp phone (callid=24378480) with media parameters of sip phone++++++
(0003396) OpenReceiveChannel conferenceID=24378480 passThruPartyID=16777848 millisecondPacketSize=20 compressionType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierIn=? sourceIpAddr=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104). myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
(0003396) startMediaTransmission conferenceID=24378480 passThruPartyID=16777848 remoteIpAddress=IpAddr.type:0 ipAddr:0x0a0a0a68000000000000000000000000(10.10.10.104)
remotePortNumber=16574 milliSecondPacketSize=20 compressType=6(Media_Payload_G722_64k) RFC2833PayloadType=101 qualifierOut=?. myIP: IpAddr.type:0 ipv4Addr:0x0a0a0a89(10.10.10.137)
+++++++++++=Next Phone sends its ACK+++++++++++++++
(0003396) OpenReceiveChannelAck Status=0, IpAddr=IpAddr.type:0 ipAddr:0x0a0a0a89000000000000000000000000(10.10.10.137), Port=20352, PartyID=16777848
+++++++++++=Next CUCM sends ACK to 200 OK from SIP Phone+++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 62220 index 1890
[885635,NET]
ACK sip:[email protected]:62220;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK23366067b8c0
From: ;tag=192115~d8e94532-127d-4dca-bba0-64b1675da032-24378484
To: ;tag=5B0E9816C2CA6D70F3166FB972EDE4C2
Date: Tue, 19 Feb 2013 21:44:45 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192115 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.137
b=TIAS:64000
b=AS:64
t=0 0
m=audio 20352 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Now at this point all is well...and the call is connected....
Now here is where the call is failing on the SIP-SCCP-SIP call without MTP
From Point 2 above, CUCM sends a DO to insert MOH, and then gets response, then sends an ACK to 200 Ok back to SIP Phone..
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881160,NET]
ACK sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22035ecc1fcb
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:38:50 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Max-Forwards: 70
CSeq: 102 ACK
o=CiscoSystemsCCM-SIP 190666 3 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------IP address of MOH server
t=0 0
m=audio 4000 RTP/AVP 0--------------------------------MOH port 4000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendonly---------------------------------------------------------sendonly
+++++NOW Point 6 above (SIP) CUCM sends a=inactive to break media path to MOH server to connect caller and xfered party++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.104 on port 53361 index 1810
[881161,NET]
INVITE sip:[email protected]:53361;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Date: Tue, 19 Feb 2013 17:39:04 GMT
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 164
v=0
o=CiscoSystemsCCM-SIP 190666 4 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 0.0.0.0--------------------------------------------------------------------Media IP is 0.0.0.0
t=0 0
m=audio 4000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=inactive---------------------------------------------------------------------media inactive
At this point, we should get a response back from the sip phone...
and here is what we got..
++Trying which is expected++++
//SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 331 bytes:
[881162,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK22045bb7f918
From: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 103 INVITE
To: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Content-Length: 0
++++++++Then we get a BYE+++++++++++++++
/SIP/SIPTcp/wait_SdlReadRsp: Incoming SIP TCP message from 10.10.10.104 on port 53361 index 1810 with 576 bytes:
[881163,NET]
BYE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.104:53361;branch=z9hG4bKa2vdQvR7J9OiMyjU;rport
Contact:
Max-Forwards: 70
From: "492" ;tag=97C34E1FB9A11F83DD8D8F5BA4C87C57
Allow: OPTIONS, INVITE, ACK, REFER, CANCEL, BYE, NOTIFY
Supported: replaces, path
User-Agent: Acrobits Softphone Business/2.4.8
To: ;tag=190666~d8e94532-127d-4dca-bba0-64b1675da032-24378214
Call-ID: 1CCA5149B966DC89AE0F752B8EF86480BC7102DB
CSeq: 3 BYE
Content-Length: 0
So this is the root cause of the problem. Your SIP phone does not know how to respond to multiple media break between it and the MOH server.
The difference between this and the succesful SCCP-SIP--SCCP, is that the held party was a sccp phone, hence the sip phone only has to process one a=inactive SDP message, where as in the SIP-SCCP-SIP, the help party was sip, so the sip phone has to process two a=inactive SDP messages
Now what is the difference when MTP is involved! A Big difference. MTP stays in the media path. So there is never a break in media and no inactive SDP attribute is sent. The flow looks like below:
for the initial call...The SIP phone sends its media to MTP and likewise the SCCP phone
SIP------Media------MTP------------Media-------SCCP Phone
When the new destination is dialled and transfer is commited,
SIP-------------media----MTP--------media---------MTP
The final invoite sent to connect 492 and 491 has MTP as the IP address to connect Media to.
++++++++Ivite to 492 ++++++++++++++
INVITE sip:[email protected]:61303;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231a3b24b862
From: ;tag=192048~d8e94532-127d-4dca-bba0-64b1675da032-24378472
To: ;tag=78FF5BF6C019A55EA020B69BB6A767E2
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 214
v=0
o=CiscoSystemsCCM-SIP 192048 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195---------------------------------------------------------------the MTP ip address
t=0 0
m=audio 25038 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
+++++++Invite to 491 +++++++++++++++++
//SIP/SIPTcp/wait_SdlSPISignal: Outgoing SIP TCP message to 10.10.10.94 on port 50376 index 1887
[885429,NET]
INVITE sip:[email protected]:50376;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.10.10.195:5060;branch=z9hG4bK231b78d1b56
From: ;tag=192046~d8e94532-127d-4dca-bba0-64b1675da032-24378467
To: "491" ;tag=F13CE94DE942C47680356A647DC7F916
Date: Tue, 19 Feb 2013 21:24:59 GMT
Call-ID: AE7045FFB2D8D9C28D54651473A14A5D41B5B93C
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM8.6
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Remote-Party-ID: "Leslie2" ;party=calling;screen=no;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsCCM-SIP 192046 1 IN IP4 10.10.10.195
s=SIP Call
c=IN IP4 10.10.10.195----------------------------------------MTP
b=TIAS:64000
b=AS:64
t=0 0
m=audio 25030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Wao! That was a long one isnt it...It was fun too.
So now you can look at your sip phones and see if they can accept two inactive sdp messages within the same call. That way you can remove MTP. otherwise you will have MTP involved in every single call involving a sip phone, even if they do not involve transfers
Please rate all useful posts
"opportunity is a haughty goddess who waste no time with those who are unprepared"

Skype Click to call interferes with ClipBoard

I have found the strange behavior that as soon as Click to call has recognized a phone number (be it in IE or FF) on a webpage, it disables functionality of the clipboard: when trying to copy/paste within an MS Office program, only the text is copied and all the formatting info or the original formula in Excel is not copied anymore.
For that reason I had to disable and remove Skype CtC (I have version 5.10.0.9560). Problem is solved now, but nice feature of Skype is gone.

Me too! I thought it was a Google Chrome problem. I even rebuilt my laptop over it and spent a whole day on the phone with Geeksquad! I just uninstalled Click-to-call and now the problem is gone.
This is a really big deal, people use copy/paste all the time and expect the formatting to work. I'm telling everybody to uninstall click-to-call and/or don't install it in the first place. I hope somebody gets serious and fixes it. And, Skype will have to acknowledge the problem and tell me it is fixed before I will give click-to-call another shot.
Not trying to be a jerk - that is just the truth of it.

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

Contacts Capable of "click to call" in phone browser?

Hello,
I hope someone can answer this question for me. When GW Webaccss contacts are opened in a browser on a phone, android or iphone, is the phone number associated with an individual contact "click to call"? Meaning does it appear as a link that accesses the cell phone's native dialer when tapped?
Thank you in advance for any insight or answer you can give me on this.
David

samsa1mi <[email protected]> wrote:
> Thank you. That answers my question. I only have a few users that
> require this capability (although I'm sure many would like it), so I was
> hoping for a quick fix by going from GW 8 to GW 2012.
>
> The Data Synchronizer is now in the near future for me.
>
> Thank you again.
>
Yeah. I just tried the GWMail app (from the App Store) and the numbers
aren't clickable there either.
Danita - http://www.caledonia.net/blog

Can't click to call phone number in Calendar subject

can't click to call phone number was input in iPhone 5S calendar subject, but work in notes field. any idea?

I also can click to call numbers in email, and the numbers pasted into iCal's notes field are recognized (gray box when clicked) but they don't call.
On your suggestion I created a number in the notes field - it called. But until recently numbers pasted into the field as part of a document worked, too.
I'd rebooted my phone, but I tried the reset, too. Unfortunately it didn't help. Strangeness...
Thanks for the suggestion and troubleshooting ideas, anyway!

Cdr doesn't bill Cisco phone 6901 Call Manager release 9

Hello Engineers,
Actually I opened a TAC for a problem that I got in my customer's environment.
I have CUCM release 9 and Cisco phone 6901 - firmware: SCCP6901.9-3-2.loads
When I tried to see the calls from CAR I didn't get output from the CAR/CDR.
I did an upload to the firmware SCCP6901.9-3-1-SR1-3.loads.
After that all calls were ok.
I hope this post can help anothers engineers

Hi Karina,
Thanks for taking time out of your busy schedule to help others
here @ CSC Much appreciated!
Cheers!
Rob
"Seek it out and ye shall find "
- OneRepublic

Click to Call Widget 8.0 Outlook Phone Numbers

Hi,
I have a customer who is using the Click to Call 8.0 Widget.
They have contacts with the following numbers specified:
Business
Business 2
Mobile
Home
The Business, Mobile and Home numbers show up when I click the "Click to Call" toolbar button but the Business 2 number does not.
Is this a known limitation of the Click to Call widget? - if yes are there any workarounds?
On a related topic the C2C plugin seems to cause Outlook to crash on a few PC's. I am using the latest version available from the Cisco download site (8.0.2540.0).
This was released in July 2010. Does anyone have any suggestions for troubleshooting Outlook crashes? - there seem to be some Engineering Special versions mentioned on here - would it be worth asking TAC about these?

Hi James,
With regard to the Outlook Crash issue, you may be hitting the bug CSCtl57395 which is superceded by
CSCtj90662.
.This bug is superceded Though it is in "release pending" state, I could see a couple of customers being able to solve this issue with the latest ES published by TAC.
I would suggest you to open up a case with Cisco TAC and let them verify if you are hitting this bug.
Please rate helpful posts.
Regards,
Saurabh Agnihotri

Click to Call with Cisco WIM 4.4?

Hi eGain Experts,
Could anybody confirm that Click to Call with Cisco WIM 4.4 will work with UCCE? I understand Web Callback will work, however not too sure about Click to call. I know eGain CallTrack or ClickToCall can achieve this. Is anybody able to confirm if this feature is configurable with the Cisco WIM licensing arrangements? if so could somebody provide sample configuration on how to achieve would be very useful.
Thank you very much.
Regards,
Yavuz

Short answer no. All evidence points to UCCX only.
Sent from Cisco Technical Support iPad App

Strange problem with Extension Mobility and Click to Call

Can anyone explain how is it possible ? Any ideas, guys?
CUCM 7.1.3
PC1 with IP comm. and user1 is logged in to Extension Mobility + Click to Call. User1 can make a call using Click to Call.
PC2 with IP comm. and user2 is logged in to Extension Mobility + Click to Call. User2 can't make a call using Click to Call.
The following error appears on the PC2 screen:
"The call failed. Please ensure you are logged into your Extension Mobility device. If the problem persists contact your phone administrator"
Here is the log from PC2:
2010-02-03 12:49:46,781 [16] INFO - 1 devices returned from ParseDevices
2010-02-03 12:49:46,781 [16] DEBUG - 0) MY IPC - Cisco IP Communicator - SEP0022680B43E9
2010-02-03 12:49:48,703 [1] DEBUG - entering FindCallRecord - 26468949
2010-02-03 12:49:48,703 [1] INFO - matched tag with call record - 26468949
2010-02-03 12:49:48,703 [1] INFO - action - new call: ct:Click to Call;rt:20100203-12494870;pn:26468949;pt:;cn:desk phone ct:;desk phone rt:;desk phone pn:;desk phone pt:;soft Phone cn:soft Phone ct:;soft Phone rt:;soft Phone pn:;soft Phone pt:;soft Phone cn:
2010-02-03 12:49:48,734 [1] DEBUG - ClickToCallDialer server and port10.100.3.1:8443
2010-02-03 12:49:48,734 [1] INFO - make call through WD - 26468949
2010-02-03 12:49:48,734 [1] INFO - MakeCall: user(a.koltalo) to(26468949) with profile(a.koltalo;Extension Mobility Phone;;True)
2010-02-03 12:49:51,859 [1] ERROR - make call failure through WD - CALL_FAILURE_ERROR
2010-02-03 12:49:51,859 [1] DEBUG - entering WriteRecord - 26468949
2010-02-03 12:49:51,859 [1] INFO - record already exists, go through records to remove matched record - C:\Documents and Settings\Jevgenij\Application Data\Cisco\Click to Call\Data\Outbound\26468949.xml
2010-02-03 12:49:51,859 [1] DEBUG - entering ReadRecord - C:\Documents and Settings\Jevgenij\Application Data\Cisco\Click to Call\Data\Outbound\26468949.xml
2010-02-03 12:49:51,875 [1] DEBUG - push call record into stack
2010-02-03 12:49:51,875 [1] DEBUG - write record into file
2010-02-03 12:49:51,875 [1] INFO - outbound call record changed, fire event to notify
2010-02-03 12:52:08,484 [17] DEBUG - ClickToCallDialer server and port10.100.3.1:8443
2010-02-03 12:52:08,593 [17] DEBUG - entering QueryDevices - 10.100.3.1 - a.koltalo
2010-02-03 12:52:08,656 [17] INFO - return success from GetDevices -
User2 moves from PC2 to PC1 - run IP comm. do loggin to Extension Mobility and run Click to Call with his credentials. User2 can make a call using Click to Call
User1 moves from PC1 to PC2 - run IP comm. do loggin to Extension Mobility and run Click to Call with his credentials. User1 can't make a call using Click to Call
PC2 and PC1 - have the same configuration and software installed, both PCs are on the same LAN subnet. There are no any firewalls between PCs and CUCM server.

Sounds like a possible permissions issue on the workstation to me. Have you tried configuring one of your test users as the local admin on the workstation?

SPA 303 and some kind of Click to Call?

I recently purchased the a single SPA 303 configured to an anveo.com acount - no thrills just one line easy setup for my home office.
I was trying to find some kind of solution so I could click on a phone number on the web and it would automatically dial out from my SPA 303.
Hopefully there's just some program I can install on my computer that can communicate with the SPA 303 without the need for a server.
I saw some stuff regarding CUCME but that seems like an overkill for what I'm trying to do. Also saw some stuff like Call Ctrl which has vanished from the internet (company went bust?).
some users on dslreports forums recommended a script which performs something akin to anveo web call, which places the call to the destination and then to my SPA phone - but frankly that's not what I want.
Thanks in advance!

sorry the site's Ui was really laggy that day
I found this cool chrome extension called Cisco Dialer.
It works wonderfully with a simple setup (just put in the phone's IP address) hover over a gmail contact's number and a little phone pops up on the right hand side - click it and it automatically dials it through your cisco phone. I just wish this extension was chrome wide, not just for gmail contacts.

E71x & Call Connect for Cisco

After getting the runaround from both Nokia and AT&T regarding the functionality of Call Connect for Cisco on this model of device (Nokia bounced me to AT&T, saying the E71x was AT&T's custom model, AT&T bounced me back to Nokia, saying that Nokia was the manufacturer), I finally got someone that told me Call Connect would function.
On that advice, we ordered (4) E71x's from AT&T.
They arrived today, and I'm trying to install the .SIS Call Connect package.
PC Suite is installed, the phone is talking to my PC.
I attempt to install CC, and I get:
"Application not compatible with phone. Continue anyway?"
I go ahead and continue, it checks certificate validity, gives me details on "License Manager".
I click continue, checks certificate validity again, and starts installing (progress bar time).
I wait.
The Nokia CC Cisco EULA comes up, I click "OK" to agree to it, and receive the warning:
"Nokia CC Cisco
Phone model is not supported! Press Cancel to abort installation."
Obviously, cancel aborts. If I continue, the progress bar returns, grows, and eventually errors out with an odd tone:
"Unable to install. No access."
Then dumps me back to the menu.
Has anyone played with this yet?

Hrmm...
I see that it doesn't show the Nokia VoIP 2.3 - is that required for Call Connect? I thought Nokia VoIP was a seperate functionality.
When speaking with both AT&T and Nokia prior to our purchase of these phones, I was assured that they had the VoIP capabilities.
That list appears to be inaccurate in some aspects for the E71x - it doesn't list the following for the E71x:
Nokia AV 2.5mm
Nokia microUSB Cable CA-101
USB Mass Storage
The E71x has all of these.
I'm having another problem too, locating any 802.1X configuration, which is also advertised on this model...
Message Edited by kythri on 12-Jun-2009 04:34 PM

"click to call" posibillity in cisco phones ?

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