Call Manager cuts digits!!

Hi,
I work with CCM 4.1(3). In my number plan i use extensions with 6 digits and all have te same format, XX9XXX. When a user fails dial a number and dial XX9XXX CCM match with another route patter with this form 3XXX. Call manager way to match with a dial number is:
1st. search in own database extension (here my problem)
2nd. if don`t match with a extension in its own database search like rout pattern.
Thanks to all.
picledo

Oscar,
CallManager has no concept of it's own database extension vs. external extension when doing route matching. It relys on closest match routing with CSS partition preferencing. I would need to know the exact scenario to diagnose why the 3xxx pattern is being selected (DNIS info, CSS layout, Patterns partition, etc.). CCM comes with a good add-on utility, Dialed Number Analyzer. I would recommended using this to determine why the incorrect pattern is being selected.
Please rate any helpful posts
Thanks
Fred

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Cisco Call Manager / Translation Patterns

We have a Cisco Call Manager in a large metro area with 10 digit dialing.
The problem is some of the numbers we dial in the digit still require a 1 because they are technically long distance.
Currently I have to add multiple rules to either add a 1 or not depending on the exchange. If the user adds a 1 when they don't need it they will get a "you don't need to dial a 1 with this number". If they don't add a 1 when they need it they get a "you must dial a 1 with this number"
I'm sure there is an easier way to do this especially because I only need to dial 10 digits on my cell phone no matter what.
thanks,

There is no quick solution short of build specific route patterns based on exchange dialed when you have same area code requiring 10 or 11 digit dialing, this comes down to training and having users try both and having both 1[2-9]XX[2-9]XXXXXX and [2-9]XX[2-9]XXXXXX patterns defined.

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What you translate is dependent on what the telco is sending you and the value configured for Significant Digits in the gateway configuration. If the telco sends you 10 digits and the Significant Digits field is set to All, then you would translate the entire 800 number.
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How does Tandberg Gatekeeper call Call Manager SCCP extensions

Am I missing something here? I have CUCM 8.6 with h225 trunk built to Tandberg Gatekeeper (works fine). I have two Tandberg 150 video endpoints that are registered to the Tandberg Gatekeeper, extensions 8001 and 8002 (alias is what you call it I guess). The video enpoints can dial eachother by their extensions and work fine. The CUCM SCCP phones can dial extension 8001 and 8002 and ring the video phones to make voice only calls, which is fine. Had to build route patterns/lists/groups for this. How the heck do you dial a CUCM SCCP phone, like extension 1450 from the Tandberg video endpoint? Theres nothing in the Tandberg Gatekeeper to configure to route this call from a Tandberg Gateeper registerd phone to the Call Manager SCCP phone. This is just a simple lab setup.

Anthony,
Its interesting you say that because that is the total opposite of my experience. I'm currently working on a CME 9.5 to Exchange 2013 deployment where most users have DIDs.
When a call comes in to the extension to which the DID points, and the call rolls over to Exchange typically we receive a message stating that there is no mailbox associated with the extension because Exchange is reading the first diversion header information
(Wireshark shows two headers and the DID is always numbered 1) which has the DID number rather than the redirecting extension. I've had to either add the last 3 digits of the DID to the users mailbox or manually transform the SIP INVITE diversion header
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PLAR on Call Manager

I'm attempting to set up a direct PLAR to a line appearance on 79XX sets. I would like to have the FXO ring the line appearance, (I get this piece working)but I'm unable to access the line on the 7960 in this case and draw dialtone from the FXO. I would like to go directly to the FXO without the need for a dial pattern. Is there a way to achieve these results on Call Manager? I'm running 4.13(es94)version 4.1(0.11) Any help thoughts accepted and appreciated.

The FXO trunk line functionality is availabe with Callmanger Express, but not with Callmanager. You can give the appearance of this functionality by doing the following.
1. Create a speeddial on the phone of 9.
2. Go to Callmanager service parameters and change the Speed Dial Await Further Digits parameter to True.
Now the user can begin dialing the number when they choose this "line". Of course the route patterns will still have to be in place, but it gives the appearance of seizing a line.
Hope this helps. If so, please rate the post.
Brandon

A question about call manager traces for Sip phones.

So today I create a sip based ip communicator and pressed the new call button and heard a dial tone. I started typing my telephone number. Half way through, I heard another secondary dial tone (which indicates mis-configured route pattern somewhere) .
However, When I look at the call manager logs, I do not actually see the digits that I was typing. With SCCP, I can see the keypad button press messages in the traces, but here, I cannot see the pressed buttons in my CUCM traces. Can anyone help with telling me how I can see button presses going to call manager . All I can see are the logs below which came up as soon as I got the dial tone and the final sip invite messages. I see nothing in-between.
|SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.xx.4.xx on port 56714 index 31809 with 973 bytes:
[6387070,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.x.x.66:56714;branch=z9hG4bK00005b1e
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=00ffb00bc50a00340000499f-00006ab4
Call-ID: [email protected]
Date: Sat, 14 Feb 2015 14:17:40 GMT
CSeq: 19 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:56714;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="18" state="partial" entity="sip:[email protected]">
<dialog id="12" call-id="[email protected]" local-tag="00ffb00bc50a003300006390-00002d4f"><state>trying</state></dialog>
</dialog-info>
SIPStationD(12991) - processCommonDialogNotifyInd: Did 12 Sending Notified SIPOffHook to new Cdfc

Here is a more detailed explanation of how SIP calls notify cucm when they go off hook to make a call. The digit dialled here is 4080
+++++ Analysis of SIP Phone making a call +++++++++
The user picks up the phone and the IP Phone sends a NOTIFY to CUCM to indicate the start of a new dialog. This dialog begings by an offhook event
00869539.002 |14:58:13.837 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 976 bytes:
[46240,NET]
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:13 GMT
CSeq: 11 NOTIFY
Event: dialog
Subscription-State: active
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 350
Content-Type: application/dialog-info+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8" ?>
<dialog-info xmlns:call="urn:x-cisco:parmams:xml:ns:dialog-info:dialog:callinfo-dialog" version="10" state="partial" entity="sip:[email protected]">
<dialog id="6" call-id="[email protected]" local-tag="544e42f26d0b001d00007cc9-000044a3"><state>trying</state></dialog>
</dialog-info>
++++ CUCM SIP stack processes the new connection for the phone+++++++
00869540.001 |14:58:13.837 |AppInfo |//SIP/Stack/Info/0x0/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 2 (SIP_NETWORK_MSG), for event 1 (SIPSPI_EV_NEW_MESSAGE)
00869540.002 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipTransportProcessNWNewConnMsg: context=(nil)
00869540.003 |14:58:13.837 |AppInfo |//SIP/Stack/Transport/0x0/sipConnectionManagerProcessNewConnMsg: gConnTab=0xe81c0d70, addr=10.50.16.1, port=52910, connid=2748, transport=TCP
++++ Next CUCM allocates a call id for this call +++++
00869546.002 |14:58:13.838 |AppInfo |LineControl(66) - Get call instance=1 for CI=24419584
+++Next CUCM sends a 200 OK to the NOTIFY request for the new dialog ++++
00869555.007 |14:58:13.839 |AppInfo |//SIP/Stack/Transport/0x0xe7df4d48/sipTransportPostSendMessage: Posting send for msg=0xefbe9910, addr=10.50.16.1, port=52910, connId=2748 for
00869555.008 |14:58:13.839 |AppInfo |//SIP/Stack/Info/0x0/act_dialog_pending_resp_event: Changing from State: SUBSCRIBE_STATE_DIALOG_PENDING to state SUBSCRIBE_STATE_ACTIVE
00869556.000 |14:58:13.839 |SdlSig   |SIPSPISignal                           |wait                           |SIPTcp(1,100,71,1)               |SIPHandler(1,100,79,1)           |1,100,14,31314.75^10.50.16.1^SEP00909E9D106C |*TraceFlagOverrode
00869556.001 |14:58:13.839 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46241,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00002531
From: <sip:[email protected]>;tag=544e42f26d0b001e000056e7-0000311c
To: <sip:[email protected]>;tag=1822746380
Date: Mon, 16 Feb 2015 12:58:13 GMT
Call-ID: [email protected]
CSeq: 11 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
++++ The IP Phone sends its connection ID to CUCM, its ip address and its port number+++++++++
00869541.001 |14:58:13.838 |AppInfo |SIPStationInit: connID=2748, SEP00909E9D106C, 10.50.16.1:52910, Routed signal by connection index to (1,100,73,66)
++++ Next CUCM informs us that the NOTIFY message is for an offhook event ++++++
00869542.003 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd: Notified Dialogs - Did 6 State trying
00869542.004 |14:58:13.838 |AppInfo |SIPStationD(66) - processCommonDialogNotifyInd:   Did 6 Sending Notified SIPOffHook to new Cdfc
00869542.010 |14:58:13.838 |AppInfo |SIPStationD(66) - processSIPOffHook Primary Call Not-Found
00869543.000 |14:58:13.838 |SdlSig   |SIPOffHookInd
+++ The next thing is the USER dials a digit on the phone ++++++
This is where it gets a little complicated. So lets examine this. The first digit that is dialled generates an INVITE to CUCM like this:
In this example the user dialled "4" first so we see an "INVITE sip:4@host-IP"
00869559.002 |14:58:14.064 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 1445 bytes:
[46242,NET]
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Call-ID: [email protected]
Max-Forwards: 70
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 INVITE
User-Agent: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=tcp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "Emre ESEN" <sip:[email protected]>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Content-Length: 373
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 21020 0 IN IP4 10.50.16.1
s=SIP Call
t=0 0
m=audio 20250 RTP/AVP 0 8 18 9 116 124 101
c=IN IP4 10.50.16.1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
+++++ NEXT CUCM sends a trying for the INVITE it received +++++++++++
00869562.001 |14:58:14.065 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46243,NET]
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000015ec
From: "Emre ESEN" <sip:[email protected]>;tag=544e42f26d0b001d00007cc9-000044a3
To: <sip:[email protected];user=phone>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 101 INVITE
Allow-Events: presence
Content-Length: 0
++++NOW CUCM evaluates the DTMF supported by the phone to determine how to inform the phones to send the remaining dtmf digits++++
From the INVITE cucm concludes that KPML and rtp-nte is supported
00869566.009 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: KPML Supported.
00869566.010 |14:58:14.066 |AppInfo |setEndpointsDtmfCaps: Detected inband DTMF support
Next CUCM generates kpml event pkg which is going to be used to receive the remaining digits from the phone
00869590.001 |14:58:14.067 |AppInfo |SIPEventPkg::SIPEventPkg 0xe4a1d1e0 scbId[16725], event name[kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3], id[]
+++ Next CUCM sends a SUBSCRIBE to the IP phone for kpml event +++++
00869594.001 |14:58:14.068 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46244,NET]
SUBSCRIBE sip:[email protected]:52910 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 101 SUBSCRIBE
Date: Mon, 16 Feb 2015 12:58:14 GMT
User-Agent: Cisco-CUCM10.5
Event: kpml; [email protected]; from-tag=544e42f26d0b001d00007cc9-000044a3
Expires: 7200
Contact: <sip:[email protected]:5060;transport=tcp>
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 424
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern criticaldigittimer="1000" extradigittimer="500" interdigittimer="15000" persist="persist">
    <regex tag="Backspace OK">[x#*+]|bs</regex>
</pattern>
</kpml-request>
+++ Next we get a 200 OK to the SUBSCRIBE from the ip phone ++++
00869595.002 |14:58:14.118 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 459 bytes:
[46245,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce719b37856
From: <sip:[email protected]>;tag=480227084
To: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 101 SUBSCRIBE
Server: Cisco-SIPIPCommunicator/9.1.1
Contact: <sip:[email protected]:52910;transport=TCP>
Expires: 7200
Content-Length: 0
+++ NEXT the IP phones sends the remaining digit dialled on the phone to CUCM +++
00869603.002 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 573 bytes:
[46247,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0
00869608.001 |14:58:14.183 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46248,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK000045c8
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1000 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++Next the IP phone sends the next digit. Here its important to note that the NOTIFY doesnt contain the next digit,
the NOTIFY is still the same as the first digit but the next digit is carried in the xml document attached to the NOTIFY.
At this point I will insert a paragraph from the RFC 4730 for SIP KPML
+++++++++++++
The event package uses SUBSCRIBE
   messages and allows for XML documents that define and describe filter
   specifications for capturing key presses (DTMF Tones) entered at a
   presentation-free User Interface SIP User Agent (UA). The event
   package uses NOTIFY messages and allows for XML documents to report
   the captured key presses (DTMF tones), consistent with the filter
   specifications, to an Application Server +++++++++++++++++++++++++++
00869609.002 |14:58:14.209 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46249,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
00869622.001 |14:58:14.210 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46250,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00003c9d
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1001 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Again we get the next digit ++++
00869624.002 |14:58:14.262 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46251,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1002 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="8" tag="Backspace OK"/>
00869637.001 |14:58:14.263 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.50.16.1 on port 52910 index 2748
[46252,NET]
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK0000310f
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
To: <sip:[email protected]>;tag=480227084
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
CSeq: 1002 NOTIFY
Server: Cisco-CUCM10.5
Content-Length: 0
+++ Finally we get the last digit ++++
00869638.002 |14:58:14.390 |AppInfo |SIPTcp - wait_SdlReadRsp: Incoming SIP TCP message from 10.50.16.1 on port 52910 index 2748 with 877 bytes:
[46253,NET]
NOTIFY sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.50.16.1:52910;branch=z9hG4bK00006c1c
To: <sip:[email protected]>;tag=480227084
From: <sip:[email protected]>;tag=544e42f26d0b001f0000092c-0000070a
Call-ID: [email protected]
Date: Mon, 16 Feb 2015 12:58:14 GMT
CSeq: 1003 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:[email protected]:52910;transport=TCP>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 209
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required
<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
Once digit collection is completed CUCM proceeds to finalise its digit analysis process.
Note that digit analysis is carried out for each digit that is recieved. I have only included the final DA here
00869648.003 |14:58:14.391 |AppInfo |Digit Analysis: star_DaReq: Matching SIP URL, Numeric User, user=4080
00869648.004 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes data: daRes.ssType=[0] Intercept DAMR.sstype=[0], TPcount=[0], DAMR.NotifyCount=[0], DaRes.NotifyCount=[0]
00869648.005 |14:58:14.391 |AppInfo |Digit Analysis: getDaRes - Remote Destination [4080] isURI[0]
00869648.012 |14:58:14.391 |AppInfo |Digit analysis: match(pi="2", fqcn="9106", cn="9106",plv="5", pss="", TodFilteredPss="", dd="4080",dac="0")
00869648.013 |14:58:14.391 |AppInfo |Digit analysis: analysis results
00869648.014 |14:58:14.391 |AppInfo ||PretransformCallingPartyNumber=9106
|CallingPartyNumber=9106
|DialingPartition=
|DialingPattern=4XXX
|FullyQualifiedCalledPartyNumber=4080
|DialingPatternRegularExpression=(4[0-9][0-9][0-9])
|DialingWhere=
+++++Once this is done CUCM then proceeds to send the call out to to the intended destination as configured in the RL ++++
00869701.001 |14:58:14.435 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.250.0.13 on port 5060 index 2754
[46256,NET]
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/TCP 10.28.132.111:5060;branch=z9hG4bKce931ee3d74
From: "Emre ESEN" <sip:[email protected]>;tag=16726~813ee89e-33db-4d58-9f6a-61542cc840ee-24419585
To: <sip:[email protected]>
Date: Mon, 16 Feb 2015 12:58:14 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces

FAX Machine like using Call Manager Express

Hi to all,
Let me share to your my problem, before we have this analog phone which connect to the our printer to have this fax machine to send fax letters, but since the phone line we used to send fax has been cut, so my boss wants me to set up a fax service using the extension number that we have in our CME (Call Manager Express) VOIP. can any one help me how to configure and set up one extension number of the VOIP using the Call manager Express Gui v7.1 or using the CLI. since i only have a basic knowledge in networking.
This topic first appeared in the Spiceworks Community

Much of Cisco's early SIP work involved proof of concept-proving that SIP worked in an IP telephony environment, with traditional features such as caller ID, call hold, call transfer, and three-way calling. Capabilities such as announcing a call transfer to the recipient before connecting the call (which involves setting up a media stream prior to connecting the call) had to be tested to make sure they would work through RTP streams and SIP-to-public switched telephone network (PSTN) gateways. Similarly, TDM out-of-band signaling (which passes information such as caller ID data) had to be passed via the SIP network.
Most of the early adopters of SIP technology were young service provider companies looking to differentiate themselves by providing IP-based services. They include Vonage, the IP-based telecommunications services provider that used Cisco SIP gateways and SIP analog telephone adapters (ATAs) to launch its service in 2001, and B2, a Swedish IP-based telephony service provider. But even companies such as Microsoft saw the promise of SIP as far back as 2001, when it launched the SIP-enabled Windows Messenger Update for its MSN .Net initiative. The Windows Messenger Update enabled MSN users to make SIP-enabled voice over IP (VoIP) calls from the MSN Messenger client, with Cisco providing the SIP infrastructure to Microsoft service provider partners.
Try URL:
http://www.cisco.com/en/US/netsol/ns340/ns394/ns165/networking_solutions_white_paper0900aecd80131325.shtml

What is the recommended Delay/Latency between Call manager and The SRST setup

Hi ,
Need to understand the readability between the CUCM and SRST.
What is the recommended Delay/Latency, Bandwidh between Call manager and The SRST gateway setup.
Regards,
Velu S

Hi Manish,
I've been struggling to get this information and what I could understand from the SRND is that this 80ms is just related to the Intra-Cluster communications (Between Servers UCS), there is no relation with SRST Gateways:
"The maximum one-way delay between any two Unified CM servers should not exceed 40 ms, or 80 ms round-trip time."
From that I assume that the RTD between the SRST and the Cluster should be based on the affirmation below:
"If a voice service is hosted across a WAN where the one-way latency is 200 ms, for example, users might experience issues such as delay-to-dialtone or increased media cut-through delays. For other services such as presence, there might be no problem with a 200 ms latency."

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

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

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

Call Manager 8.0 to 9.1 upgrade

We are currently running Call Manager 8, UCCE 8, and CVP 8. ICM/CTI 8.
We would like to upgrade Call Manager 8 to 9.1 first before upgrading UCCE, CVP, etc., it could be months before these are upgraded.
Does anyone know or foresee any issues if UCCE, CVP, etc., are not upgraded right away after CM is upgraded?
Any comment is appreciated. Thanks.

Look at the UCCE compatibility matrix as you need the exact versions to find out whether they will work together, of ir you'll need a single window on which to upgrade all.
HTH
java
if this helps, please rate
www.cisco.com/go/pdihelpdesk

Just downloaded Sales Call Manager App...I can't get it to work on either my Iphone 4S or My Ipad 2?

Downloaded the App Sales Call Manager. I can't get it to work on either my Iphone 4s or my Ipad2? Can anybody advise how to fix or even get back my money that i paid for it?

Contact the app developer for support. A link to their website will be in the apps description in the AppStore.

Upgrade Call Manager from 8.6.2 to 10.5

Hello All,
We are planning to upgrade a Cisco Call Manager Publisher node from 8.6.2 to 10.5. We want to install a new publisher on a new environment, but we are struggling with some questions.
If we install the Pub node 10.5 can we migrate the configuration from the 8.6.2 to 10.5? Do we need a special tool for this?
If above doesn't work, we can still migrate the existing call manager towards the new servers and start the upgrade, the only problem we face there is that we have to change the publisher's IP because we want to use a new ip addressing scheme.
Kr,
Yannick Vranckx

Your best bet is to use the new tool: cisco prime collaboration deployment. This fits perfectly into what you want to do here and can easily help you with all aspect of the migration including the ip address change. You can learn how to use the tool here..
https://www.youtube.com/watch?v=JzG4kz1_hL4

What are the essentials needed before upgrading from call manager 9.1.1 to call manager 9.1.2?

I recently tried to upgrade my call manager in a lab environment from 9.1.1 to 9.1.2 but failed. The error stated that connection had been lost after 2 hours into it. Connect using CLI. Any help would be greatly appreciated.
Steve

You mean via GUI??
What does the upgrade status via CLI says??

Call Manager Migration 7.1.5 to 9.1 / Trunking : ICT vs. SIP

Hello All,
Currently studying for CCNA Voice and have been asked by my current employer to upgrade CUCM 7.1.5 to 9.1.1. There has been a time frame put on the deployment however I want to dig pretty deep into the deployment to learn as much as possible. I am going to start by configuring a trunk between the 2 call manager clusters. from there I will add users to the new call manager and try to convert slowly to the new call manager instead of one big cutover. I will place a route pattern over the trunk so that cucm 9.1 will route through cucm 7.1 while I slowly convert users to the new system. I will then deploy directory numbers in a Staging partition that is not associated with any CSS. Once I have all users and phones configured on the new 9.1 call manager I want to use the bulk admin tool to change the staging partition to the working internal_PT production partition. The users will be imported from LDAP. My questions are, does this sound like a feasible plan? Is there any other difference between a SIP trunk and ICT besides the SIP and H.323 protocol? ( I prefer SIP from what I have read) and will having the same users on both boxes interfere with call processing, if these users and phones are not active yet?

Just for the record. I found the solution to my problem. Checking more logs I read this:
The installation has encountered a unrecoverable internal error. For further assistance report the following information to your support provider.
"/usr/local/cm/script/cm-dbl-ontape_backup-install RU PostInstall 9.1.2.11900-12 7.1.5.30000-1 /usr/local/cm/ /common/component/database /common/log/install/capture.txt " terminated. Exceeded max time (240)
The system will now halt.
So I accessed the Dissaster Recovery Section on CUCM and deleted the tape backup device that was configured there. After deleting it the upgrade went well.

Can't remove registered ephone in call-manager-fallback

This ephone and dn keeps registering so long as call-manager-fallback is not shutdown.
RTR001#show ephone registered
ephone-1[0] Mac:0FD4.9DA0.D415 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED in SCCP ver 6/5 max_streams=1
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:7
IP:10.32.21.183 * 26602 SCCP Gateway (AN) keepalive 4 max_line 2 available_line 1 dual-line
port 2/0/21
button 1: cw:1 ccw:(0)
dn 2 number 4851 CM Fallback CH1 IDLE
Preferred Codec: g711ulaw
Lpcor Type: none
The MAC 0FD4.9DA0.D415 identifies port 21 on a Cisco VG224. After shutting down that voice-port, the ephone doesn't register when call-manager-fallback is enabled.

Well, that is one idea that I've already had, Linc, but I'm reluctant to use the "nuclear option" for obvious reasons. I'm actually wondering now if the Secure Cert / OD problem is affecting Profile Manager. See this thread: https://discussions.apple.com/message/23686348#23686348

Call Manager cuts digits!!

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