User@IP-of-cucm showing up in call history

I have several different types of telepresence codecs running TC 7.2 registered to CUCM 10.5. I have a SIP trunk to a VCS control. If I make an outbound call from CUCM to the VCS domain, it shows up in the call history of the cucm endpoint as user@ip-of-cucm, rather than the display name configured in the endpoint I am calling (please note this is not a "Use Fully Qualified Domain Name in SIP Requests" issue. I have this option checked and the far end does see the caller ID as a SIP URI with FQDN). If I call from that same VCS registered endpoint into the CUCM registered endpoint, then the correct display name shows up in the call history of the CUCM endpoint. I've combed through a lot of SIP settings on the CUCM side but haven't found the right one yet. Does anyone know what settings I should be changing?
Thanks, Mike

Hi Mike,
We are facing the same issue for a call between DX70 and SX20. The SX 20 is running TC 7.1
If i upgrade the SX 20 to TC7.3, would the issue be resolved or will it still persist as the domain names of our CUCM and VCS are different ?? Given below is the call flow
DX70 --> SIP Route Pattern --> VCS SIP Trunk --> SX20

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CUCM 8.6.2 Cant view call history on certain Phones

Have 3 specific user's who cannot view any call history wha so ever, Cant find much in relation to this issue via the web, Thy are the only one's experiencing the issue on the system. I have reset the phone to no avail. also rebuilt the phone from scratch in CM. having no luck figuring it out.
any help or idea I can try

Hi Jason,
The phone stays in upgrading....
We need to fix this first.
Fix the issues related to DHCP & TFTP on the switch to which the phone is connected so that the phone can get the config from the DHCP server.
Sample DHCP config on switch:
ip dhcp pool dhcpconf
network A.S.D.W 255.255.255.0
default-router A.S.D.1
option 150 ip A.S.D.X ----->TFTP server of CUCM
lease infinite
ip address dhcp
Interface config:
interface FastEthernet0/1
switchport voice vlan 1
spanning-tree portfast
Check the routing issues between phone and CUCM TFTP server.
Upgrade the phone to the latest firmware and check if you have any issues.
Note: Do not disconnect the phone while the firmware is upgrading.
Regards,
Tirtha

When I log onto my macbook, my user account isn't showing up. All that shows up is a gray screen with only two boxes saying "Name" and "Password". How do I fix this?

There was one time where my wi-fi wasn't connecting so I tried to turn it back on. A window popped up but I had alrdy clicked enter before I read it, but it said something like "...will change settings, is that okay...?" And then wi-fi still would not connect. So I tried restarting my computer, but when it turned on, my user account did not show up and all there was on the screen was two boxes that asked for name and password. I was the only person using my macbook so I never set up a password for it. Has anyone had this problem? How do I fix this?

Since I'm not in front of your computer I can't tell what you see - I can only guess and give a hint. So once again, if the disk name on your Mac is "Macintosh HD", when you boot in Recovery Mode, and start up the Terminal in "Utilities" menu, and then type in the command:
    df -k
you should see a list of "volumes" - one of which is named "/Volumes/Macintosh HD" (but yours could be anything) and in that folder there is another folder called "Users" and in that folder is the names of the home directories of the user accounts (and the names that you can enter into the login screen) that you are looking for. So if you are not familiar with using the Terminal you should say so and I can give more details. But assuming you're OK with typing in commands to theTerminal this is what I would try:
    ls -l /Volumes/Macintosh HD/Users
and you should see something like the following (my disk name is "Yosemite" so I typed "ls -l /Volumes/Yosemite/Users")
ML9Book:~ xxxx$ ls -l /Volumes/Yosemite/Users
total 0
drwxr-xr-x+ 11 Guest      _guest   374 Oct 25 19:23 Guest
drwxr-xr-x+ 33 Jack       staff   1122 Oct 24 10:49 Jack
drwxrwxrwt   13 root       wheel    442 Dec 27 2013 Shared
drwxr-xr-x+ 38 admin      staff   1292 Oct 25 16:59 admin
drwxr-xr-x+ 15 x13        staff    510 Oct 23 22:41 x13
drwxr-xr-x+ 12 514        staff    408 Oct 23 12:24 x14
So now reboot your mac into normal mode - and you will see the login screen with the two empty boxes.
So the above output tells me that there are 3 users - Jack, x13 and x14 (there is always a Shared user and there might or might not be a Guest user since it depends on how your setup was setup) - and these 3 are the names that I could try and enter into the login screen (in the top or first box). You can try no password, but I don't think you can create accounts without passwords, so this will likely fail. In that case you need to reboot in recovery mode and run the Terminal command again (from the Utilities menu) and do a resetpassword command. Another window will pop up and from there you select the Volume (in my case I selected the Yosemite volume - in your case you would select the "Macintosh HD" if that is what your volume name is). Then select the user account you want to reset the password from the drop-down list of user accounts (it will show you the long and short names, and the short name is what was in the listing above from the "ls -l" command), then enter the new password in the indicated box, and then enter it again in the next box, to make sure you didn't make a typo, and enter a password hint in the last box (optional). Then click the "Save" button and you will see another dialog box pop up telling you that the password was changed but not the Keychain password (hopefully you remember this one) - click the OK button then reboot back in normal mode (click on the Apple logo in the top left corner of the screen and click on Restart).
Does this help?

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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Online number is not showing up in caller id

I live in the U.S. and my online number is not showing up when I call someone. Is there a way to fix this

unleashyourgeek wrote:
I am constantly having problems with clients not seeing my phone number when I call. Why is this happening? Your answer here was very short and not very specific as to how to resolve this. I have been fighting this for more than 2 months now but I keep paying for a number that people can't see when I call them.
I'm sorry I did not give an answer that met your level of specificity. The original poster did not post the number he/she was trying to call. This is also a worldwide forum mostly inhabited by Skype users along with some Skype employees who post from time to time. I'm not a Skype employee, just a user who makes use of many features of Skype to avoid paying more $$$ to the telephone companies.
When Skype users have Online Numbers, these numbers are not automatically used as Caller ID when calling telephones from a Skype account. This way, the Skype users hold the control over whether or not to have a working phone number show up as Caller ID. Most who have Online Numbers want them used as Caller ID, but the users - and not Skype - get that level of control. In my opinion, that's a good thing.
The surest way to see the current Caller ID settings for your Skype account and change them is to log into your account on the Skype web site (use the "account" link on the upper-right corner of the Skype web site). When you log in, you will see the current settings, and the link to make changes. If you have Online Numbers that can be used for Caller ID, you can select them. Online Numbers in the USA and the UK are examples of the numbers that can be used as Caller ID from Skype, and numbers from Mexico and Australia can't be used as Caller ID. You also have options to use different numbers as Caller ID for calls to telephones in those countries, and have a "default" number used for Caller ID anywhere else. In my case, my default number for Caller ID is my US Online Number, and my UK Online Number is used as Caller ID for calls to the UK.
Even with these settings properly configured, there are still occasions where your number (an Online Number, or a mobile number which has gone through the verification process with two SMS messages) will not appear on the Caller ID displays of the phones you call. This is generally outside the control of Skype. If you are using a number that is not a "local" number in the country that you are calling, it too may not show up on the Caller ID display. Instead of posting on the community forums, using the Support link at the top of this page may be the better way to go. You can go through the knowledge base trying to find a solution, and end up opening a support ticket with Skype. If you have a subscription, you may be able to go to a live chat with Skype support.
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
I am not a Skype employee. No soy un empleado de Skype.

I paid for a number, it doesnt show up on caller i...

i paid for a number, it doesnt show up on caller id and i cannot recieve calls on it?
22 beans for 3 months plus im paying for calls still? so what did i pay for?

bfgcb wrote:
We received a call from someone who didnt speak English, was obviously reading from a script, and the originating number didnt show up at all on caller ID nor was it listed in recent calls on the phone (which means theyve figured out how to hide that data). THe caller said something like he was calling from the Microsoft Windows Support Center (warning flag 2,because Microsoft doesnt care enough about end users to actually call them). I could only understand fragments of what he was saying. Malware, errors, concerned, etc. I asked him to repeat his name and provide his MS employee ID number, and told him I was going to call a friend who worked at Microsoft to confirm his employment. SURPRISE! He hung up. I really would like to identify the originating number so I can report it. Without a doubt it's a scammer.
It's a scammer using a system (probably international via the Internet) that spoofs (hides) all caller ID information, which is why there was no Caller ID information or entry in the call log. It's illegal so reporting them even if you could woukd have no effect. If I don't get valid caller ID information from a number I recognize I just ignore the call or quickly answer and immediately hang up.

Multiple OS users into the CUCM and CUC?

Does anyone know if it possible to have multiple OS users into the CUCM and CUC?

Hi ,
you can use the command set account username in Call Manager where username is what you want to configure.
regds,
aman

We can not clear all recents(30 recents call). When Tango show "No recents call". We touch other buttons and come back to Recents button again. It still show 30 recents call.

We can not clear all recents(30 recents call). When Tango show "No recents call". We touch other buttons and come back to Recents button again. It still show 30 recents call.

No one here is going to do anything about it. Send feedback to Apple.
http://www.apple.com/feedback/ipad.html
Basic troubleshooting steps.
17" 2.2GHz i7 Quad-Core MacBook Pro  8G RAM  750G HD + OCZ Vertex 3 SSD Boot HD 

I deleted a contact out of my iPhone 4S but when I go to send a message the contact appears although that individual is no longer in my address book. I deleted all text and call history pertaining to the deleted contact but still shows up.

I deleted a contact out of my iPhone 4S but when I go to send a message the contact appears although that individual is no longer in my address book. I deleted all text and call history pertaining to the deleted contact but still shows up. Does anyone have the same issue and how do I fix it?

dam122577 wrote:
but wont restoring as new delete all my info on my phone
Yes. That is why I added...
it will disappear over time due to non-use.

HT5622 My FaceTime shows one user ID, my iTunes shows a different one. How can I get everything on one?

My FaceTime shows one user ID, my iTunes shows a different one. How can I get everything on one?

You want to keep the iTunes ID since that is the one that you make purchases with. That should be your primary Apple ID email address.
Go to. Settings>FaceTime>Tap your ID and sign out. Then sign in with the iTunes/Apple ID email address.

My Iphone 5 updated to IOS 6.2 and now my contact names do not show on incomming calls or text messages. All i can see is the number of the person calling. how can i fix this?? please help

I updated my iphone software to IOS 6.2 yesterday and im not sure if it is a coincidence but now, my contact names do not show on incomming calls or messages. Also they do not show on past calls or recent history.
All my contacts in my phonebook still exist and are correct, the problem only occurs when receiving calls or messages. please help i need this fixing
Regards,
Frosty

Try to restore your iphone via iTunes or update to ios6.1.2.
If itunes gives an error with restoring (DFU/Recovery or just normal)
try this:
open your computer and search on windows; C://windows/system32/drivers/etc/hosts
open up the hosts file in note pad you will see IP adresses and more add another line and put in this:
#74.208.10.249 gs.apple.com
this is a by-pass to the cydia/saurik restore server, it has nothing to do with jailbreaking over voiding apple's warranty
let me know if succeeded!

My caller id is not working on iphone 5 - it beeps but doesn't show who is calling or let me click over

I have recently had my iphone 5 not show who is calling on call waiting. The phone beeps but doesn't show the number / per son that is calling, won't let me click over and won't show who it was in my call history. Help very annoying.

1st go into your settings then tap phone to see if call waiting is turned on or off. If call waiting isOFF turn it ON. You can use also Apple article HT 4515. I would contact my wireless carrier reguarding the caller id not showing up and also with help reguarding your call waiting as well
Hope this helps

CUCM 8.6.1.2000 - Call History not logging, and no ring tone access.

Hello,
A little background:
Recently, I setup a UCS C-210 M2 with CUCM 8.6.1 and restored our Pub and Sub to it. Also migrated to from physical Unity to Virtual CUC, same with Presence.
I've also setup a 2nd CUCM Sub, as well as new CUC and CUP subs on a second UCS C210. I also upgraded CUP to 8.6.4. before adding the Sub CUP node.
I'm thinking my problems began when I added the 2nd Subscriber CUCM node on our 2nd C210. Here is a description of what's going on.
- I cannot change ring tones on an IP phone. I get "Ring List Unavailable" when I try to change a ring tone. I noticed that my phone had changed over to the CTU ring tone around the time I brought up the VM environment. The CTU ring tone was a custom one I had loaded at one time.
- I cannot access any call history on my IP phone. My Jabber client logs call history just fine.
Seems to be a TFTP issue, but I restarted the Cisco TFTP service yesterday, and it did not help. I also reset the Device Pool, which didn't seem to do anything.
Also, I'm still a bit unclear on which services I should be running on my new Subscriber node.
Any help is very much appreciated. Thanks again, guys!!
-Jonathan

I had this issue and resolved it. The problem was that we had changed DHCP servers over to a different platform and the new DHCP server didn't have Option 66 defined to point the phones to a TFTP server. Once we re-specified the TFTP server IP's for option 66 in DHCP and reboot the phones the issue was resolved.

User Name and Password for JCO RFC call to BAPI

Hi all,
What I think I know:
--We do NOT have Single Sign On configured so don't tell me to use SSO please - I agree, but...
--We have a requirement to do a goods receipt which prints labels for the handling units
.....The printer to which the labels are directed depends on the user who is running the transaction
What I think this means
--We will need to specify a user name and password in the RFC call so the label will go to the correct printer
--I cannot use the IllumnLoginPassword (or whatever its name is) for the password
--I need to prompt the user for their password a second time after they login to our MII app
The problem
--I will need to store the password somewhere for the duration of the session
......In session variable that has been encryted
.............I didn't see an encryption action block so I could create my own
......In the database using database column encryption
.............A little bit of a pain, but not too bad
Any corrections, alternatives, ideas .... ???
Thanks,
--Amy Smith
--Haworth

Thanks for the attention guys. A little clarification.
1. I have been assuming that I cannot use the IllumnLoginPassword for the JCO SAP password in the action block. If this is NOT true, then it solves my whole problem.
2. It would not work to prompt a shop floor person for their password every time they do an operation completion. Well, at least
if I don't want to not get lynched!
3. I am planning on prompting people every time they log on for their ECC password and retaining it somewhere secure while they are logged on (and longer if they skip the logoff step.)
4. I have been focusing on how/where to retain the password, but also need a way to encrypt it during transmission. Jeremy said the applet/BLS would at least encode it for me. That is good.
--Amy Smith
--Haworth
Edited by: Amy Smith on Feb 18, 2010 1:30 PM

No Contact Names or Pics Showing for Incoming Calls?

I just got my phone yesterday. I have clean sync with my MacBook Pro. I thought that contact names should show for all incoming calls if that person and number are in my contacts. I also thought a photo would show if there was a photo in the contact as well.
???

I had a problem somewhat similar when I first bought my IPhone as well. My iPhone would show the numbers in international format with the 011 in front of all the numbers which made the contacts not show up when calls were incoming. I had to contact AT&T and spent 4 hours on the phone until someone actually figured out what was wrong and fixed it in less than 3 mins. It wasn't even the iPhone it had to do with the phone I previously had through Cingular/AT&T. The person said it was some sort of provisioning or something like that with some incorrect settings that screwed with the incoming calls only. This problem also would make the contact picture not show up as well for anyone in your list. This might help or maybe it won't. Good Luck figuring it all out. I still love my iPhone and haven't had too many problems with it

User@IP-of-cucm showing up in call history

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