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

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"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.
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UCCX Call Recording on 7912 phone

Hello All,
Does anybody know if you are able to do UCCX call recording on a 7912 phone? I do not see the option within the 7912 device configuration page to span the port to the PC even though the 7912's do have two ethernet ports on them and daisy chain the PC onto the network. The firmware for the 7912's is 8.0 (3). The latest and greatest firmware is 8.0(4) so i do not beleive it to be related to that - I am thinking unfortunately that they 7912's cannot perform call recording and will need to be replaced.
Thanks in advance for the feedback!!
Paul

Correct, the 7912 does not support Built in Bridge. You would need to use Desktop Recording or SPAN. Or switch to something like the 7911.
(See https://supportforums.cisco.com/thread/302779)
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Recording of cisco phones

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Ask the Expert : Call Recording with Cisco Unified Communication Manager (UCM)

Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn and ask questions about Cisco Unified CM call recording solution that provides the ability to record customer conversations for compliance purpose. This topic will cover an overview, configuration and troubleshooting of the call recording feature.
Monday, January 19th, 2015 to Friday, January 30th, 2015
Harmit Singh is a technical leader with the High Touch Technical Services (HTTS) and Technical Assistance Center (TAC) Unified Communications teams based in Bangalore. He has broad experience in Cisco Unified Communications infrastructure solutions. He has 10 years of experience working with large enterprise and service provider networks. He also holds CCIE certifications (#20012) in Voice and Collaboration as well as Red Hat and VMware certifications.
Mohammed Noorulla Khan is a customer support engineer in High-Touch Technical Services (HTTS) Unified Communications teams based in Bangalore. His areas of expertise include Cisco Unified Communications Manager, Gateways, and Jabber. He has over 6 years of industry experience working with large enterprise and service provider networks. He also holds CCIE certifications (#35741) in Voice and VMware certifications.
** Remember to use the rating system to let Mohammed and Harmit know if you've received an adequate response. **
Because of the volume expected during this event, the experts might not be able to answer every question. Remember that you can continue the conversation in the Collaboration, Voice and Video community, subcommunity, IP Telephony, shortly after the event. This event lasts through January 30th 2015. Visit this forum often to view responses to your questions and those of other Cisco Support Community members.

Hi Maheshwar,
Thank you for your query. Please find my response below:
1> Do we support recording with HCS environment and which 3rd party vendors are validated with HCS based call control 10.1.1?
Answer: Whether you use a standalone UCM cluster, UCCE or HCS, call recording would be supported across the board in the same manner.
Please refer to the following link:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cust_contact/contact_center/hcs-cc/10_0_1/Install_and_Config/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs/CHCS_BK_ICC270D0_00_installing-and-configuring-cisco-hcs_chapter_011.html#CHCS_RF_T1105284_00
Option
Notes
Recording
All Recording applications that are supported by Unified CCE are supported on HCS for CC. For details, see Recording section in Agent and Supervisor Capabilities.
With respect to which 3rd party vendors have been validated, marketplace.cisco.com is a good place to crosscheck this info. You will find a Cisco Compatible Logo against the partners listed there. The logo is used to signify that the PARTNER product has undergone technical interoperability testing with the Cisco product specified. The interoperability testing is conducted by a third party laboratory based on testing criteria set forth by Cisco. PARTNER is solely responsible for the support and warranty of its product. Placement of the PARTNER product or information pertaining thereto, on the Cisco Marketplace website does not constitute an offer to sell the PARTNER product in any way. For further information on the PARTNER products, please visit the PARTNER company website.
Please refer to the following link and use the search field under Collaboration Technology:
https://marketplace.cisco.com/catalog/search?utf8=%E2%9C%93&search[q]=&search[technology_category_ids]=23%2C24%2C197%2C1940%2C1941%2C1921%2C1576%2C1897%2C1983%2C2418%2C26%2C198%2C1904&search[order]=tier&per_page=20&_=1421663854257&ts=1421663855441
2> Which end points are supported for recording via HCS call control?
Answer: The following link should help clarify this:
http://solutionpartner.cisco.com/web/sip/wiki/-/wiki/Main/Unified+CM+Silent+Monitoring+Recording+Supported+Device+Matrix
Please let us know if you have any follow up questions. Hope this is helpful.
Regards,
Harmit Singh.

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
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Call Records from Call Manager

I am trying to figure out how I can get call records from Call Manager. I can get CR form our PBX but cannot figure out the Call Manager... Any ideas?
Thanks,
Dennis

Hello
There are 2 ways to get call record.
Cisco has a built-in CDR Report Analysis. This application is accessable from the CallManager
Go to Application, Install Plugin, install the CDR Reporting and Analysis plugin in CallManager. Then go to Servicability and go to Tools then select CDR Analysis and Reporting
Default login is admin/admin
2nd method is via SQL Server where there is a databae call CDR . You can customize a ASP page to retrive the data you want from there
enjoy

Cisco Unified WFO - Call Recording and Quality Management with Extension Mobility agents

Hi All,
We're considering Cisco Unified WFO - Call Recording and Quality Management for a customer running UCCX 8.0, agents on multiple WAN sites, all agents using extension mobility.
The documentation I've been able to find describes three different recording methods:
Using Desktop Recording service (Endpoint) to record from an agent’s desktop.
Server Recording - Uses SPAN (not so good for remote sites)
Network Recording - Uses CUCM recording service / SIP trunk / phone's built in bridge.
Network recording or Desktop recording should be suitable for the customer but it seems that Extension mobility is not supported. Extension Mobility is not mentioned in the 8.5 installation guide, it is mentioned as ‘not supported’ in the 8.0 guide as follows:
'Server Recording and Network Recording have the following limitations:
• Extension mobility is not supported.'
Neither version's documentation specifically mention extension mobility in relation to the desktop recording method, though I realise this is a similar approach to the 'server recording' method.
So the question I have is: Is extension mobility supported in any way on version 8.0, or version 8.5 for recording? And if so which recording method(s) are supported?
Thanks,
Jonathan

Hi,
I had more luck asking questions over at the Calabrio forum - they make the software and Cisco re-brand a version of it - there is some good info on their portal (http://portal.calabrio.com), you have to register but it's fairly painless. The answer I got was:
"QM Desktop recording has always supported extention mobility as it determines the recorded user by the desktop user's login. Extention mobility was not supported for Server and Network recording until the Calabrio QM 8.6.2 release in April 2011 and will be added to Cisco QM starting with QM 8.5.2 in June 2011"
Regards,
Jonathan

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

Can we record calls while talking on phone ?

I understand that the new iPhone 3GS has feature to record classroom lectures, etc. Similarly, can we record calls while talking on phone ??

Right now , No , might be possible in the future with an App.

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

DOCUMENTATION - Cisco Unified Workforce Optimization - Call Recording

Hello,
I'm trying to sell the Cisco Unified Workforce Optimization - Call Recording option in a customer, but he is asking me to send him some documentation about capacities and functionalities. I'm not able to find this kind of information of the website. Can anybody help me? Thanks!

Hi,
Please check the following link. hope this helps!!!
http://www.cisco.com/en/US/products/ps8293/products_data_sheets_list.html
Thanks,
Dass
Please rate useful posts

Recording Multi-Line Phone Calls--TRx Phone Recorder

Anyone using their Mac (preferably a Mac Mini) to automatically record phone calls from two analog phone lines?
Thanks,
Robert

I wouldn't be able to give exact steps, but I would get an RJ-11 to USB adapter first. Then I'd use a VoIP program, like Skype (I don't know if Skype supports this, but maybe some other program does), to record the conversation. I think that's about all it takes, although finding a program might be a bit difficult.

Recording for Cisco IP Phones and Cisco C90 Codec

Hello
We are looking for a solution that is capable to record both Cisco IP Phones and Cisco Codec C90.
We are using CUCM 9.X for IP Phones and VCS 7.X for Cisco Codecs.
Is their any third party solution available for both the requirements or do i have to go with TCS and any other third party recording solution.
Thanks & Regards
Aniket Patil

My reply may be too late to be of any help to you, but for the benefit of others:
Be sure you understand the different types of PoE out there. The Linksys PoE switch only supports the newer IEEE 802.3af PoE standard.
The 7940, 7960, 7905 and other older Cisco phones only support Cisco pre-standard PoE and thus will not work with the 802.3af Linksys Switch.
To use this switch, you will need to make sure you are using the newer 7070, 7961, 7941 phones with support both pre-standard and 802.3af PoE.
All the best,
John

Cisco Unified WFO - Call Recording and Quality Management stops recording with conferenced translator

I'm having an issue whenever one of our employee's conferences in an external translator, as soon as they bridge the customer into the call with the translator QM stops recording. I can hear the intial conversation with the customer, and then the employee put the customer on hold and call the translator service. Only when the two are bridged together the call always stops recording.
We are performing all recording on the server side, and are not using Quality Manager Desktop agents.
Any help would be much appreciated. Thanks!
-Chris

So I found the following information listed below. I don't manage the Cisco Unified CM portion of our telco system. Can we limited the ourselves to a single Codec, and would this even resolve the issue. Does this cause other issues if we didn't limit the devices that are recording to a single codec?
Recording IssuesThis topic explains how to diagnose and resolve problems that occur with contact
recordings.
Calls for devices configured for Network Recording drop when you try to conference
or transfer a call.
Symptom. When you try to transfer or conference a call and one or more devices on
the call configured for network recording, the transfer or conference fails and
parties drop off of the call.
Cause. Cisco Unified CM does not support codec changes for devices that are
configured for call recording. The codec must remain the same throughout the life
of the call. For conference calls, the conference bridge must support the codec
used before the conference completes.
Solution. Update the Cisco Unified CM configuration to ensure that no codec
changes occur for network recorded devices.

Calls record on Cisco Phones

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