Call Manager forwarding alerts
Hello. Does anyone know of any integrations between Cisco Call Manager and other network mgmt platforms like HPOpenView NNM or Ciscoworks? Can Call Manager send alerts to other Network Management platforms?
I know that there is that plugin that Cisco provides, but I am seeking the same thing. To know about call thresholds, etc.
Let me know if you found anything.
Similar Messages
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7960 call forwarding UI w/o call manager hindered by firmware?
I am in the process of moving from a MGCP hosted VOIP provider to another provider and SIP. What I'm irritated about is the Call Transfer methodology on the Cisco 79xx phones. What I have is a process that requires an end-user (while in a call) to press MORE, BlndXfr, phone number, and DIAL. It's also similar for Supervised Transfer. Of course I prefer, what I consider normal, is a user to press TRANFER, number and hang up or wait for a supervised transfer then hang up.
What I've been told is that since the provider isn't using the Cisco Call Manager the firmware on the phone isn't playing well and the there's no ability to change it. Seems that Polycom isn't affected as there's ability to have it "my way".
Is this true? While in a call with a non-Call Manager Cisco 79xx one has to hit the MORE softkey to get to the transfer options?What is the CallManager version you are using? What is the phone load using?
-
I have iPhone 6 model A1549, bought from and currently using with Verizon. You all must be aware that we can use the same phone for GSM carriers like at&t as well. Now we all know that some of the features related to conference calling are not available on CDMA carriers, like we cant have more than 2 callers, and we can't manage conference calls by making the conversation with one party private for some time and again switching back to conference mode, disconnecting a particular party instead of having to end the entire call etc. reference link- iPhone: Understanding phone features - Apple Support These features are available on GSM models and carriers, I just wanted to ask if I use a GSM SIM from a GSM carrier on this model, do I get these additional conference call management options or I am restricted to Verizon like experience while being on at&t (and while using the at&t SIM in this model)
Yes, I am dealing with this right now on at&t with a 6 plus. I've been down the same road as you. I got it to work with a windows phone yesterday but not my 6 plus when I switched back to the iPhone. My wifes 6 works fine. I believe it is an at&t issue and not an 8.3 issue. After dealing through several levels of support at at&t, they are telling me it has to do with their transition to hd voice. Some people are having this issue because they are updating their system and some peoples phone profile, on phones that have hd voice capabilities, are not getting added to their line and call forwarding will not work without it. I have a trouble ticket open to their engineering dept and they promise it will work for me on or before the 24th of april. Engineering has to build my phones profile so call forwarding will work with my iPhone.
This is frustrating because like you, I need to forward my calls because I have no signal in my office and need to forward to another phone so I can get my calls. If at&t would quit dragging their feet and enable wifi calling, this wouldn't be an issue for me. I would suggest you put more pressure on at&t and escalate this until it gets resolved. Or put your sim in a phone that does not have hd voice capability and see if it works then.
Good luck. I'm keeping my fingers crossed this will be fixed for me on or before the 24th. We'll see.
EDIT: I see you did put it in another phone and it worked. That backs up that the issue is at&t and their hd voice upgrades. They had no explanation as to why some work and some don't, but have them make sure a profile for your phone is on the line. If not, they'll need to build one for the phone so that it''l work. At least that's the story for now. Like I said, I'll know more in a day or two. -
Looking for recommendations and experiences with implementing 4.2.3. I have to decide between 4.1.3 and 4.2.3 with Call Manager and Unified Messaging. Also looking to eventually upgrade to 5.1. Any help is appreciated
a quick list of 4.2 features/enhancements includes:
user features/enhancements -
* call pickup notification
* one touch call pickup
* one touch group pickup
* other group pickup
* directed call park w/BLF
* login/logout of hunt groups
* CCM assistant on phone
* complete transfer on-hook
system features/enhancements -
* AAR support for calls on no bandwidth
* call forward on no-register
* device mobility
* h323 overlap sending/receiving
* h323 annex M.1 support for h323 GW & h225 trunks
* mlpp enhancements
* v.150 secure modem support
administrative features/enhancements -
* support for password aging, complex PWs, oneTime PWs with LDAP
* voice quality stats on cal-by-call basis
please see the following link for more CCM 4.2 info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042402c.html
a quick list of 5.0 features/enhancements includes:
sip trunk & endpoint support features/enhancements -
* native support for sip devices
* cti for ISP phones
* presense information for sip devices
* fault, config, accounting, performance, security enhancements for sip support
* sip trunk enhancements for external applications
* third party sip devices supporting RFC 3261
* sip line side RFCs
* sip trunk RFC support
licensing features/enhancements -
* each device corresponds to a device license unit
* DLUs must be purchased to cover the number of devices connected to CCM
* third party sip devices require DLUs
localization features/enhancements -
* many language support enhancements
please see the following link for more CCM 5.0 info:
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_data_sheet0900aecd8042403e.html -
Forwarding alerts to a 3rd party monitoring tool
I'd like to be able to forward alerts recieved by the Enterprise Controller to our Unicenter NSM server. I have no desire to use Ops Center's incident assignment capabilities.
Has anyone managed to forward alerts from Ops Center to a 3rd party monitoring tool ?Since posting my question, I have found this possible solution...
http://www.halcyoninc.com/products/NeuronIntegration/index.php
Halcyon appear to be an Oracle partner who have developed a product which integrates OC with popular 3rd party tools.
I see also that a user by the name of 'halcyon' sometimes posts in this forum, so I'm guessing that's them also.
Obviously this is a commercial solution and won't come for free, but I have struggled to figure out how this can be achieved independently. As you say, there are lots of questions and documentation is sparse. I'd prefer not to make a career out of it. ;)
The integration of OC with our Unicenter NSM tool with be my highest priority for when I get back from Xmas vacation in mid January. I will post up any useful information I discover.
Edited by: graham1137 on Dec 30, 2012 2:40 PM -
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 CdfcHere 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 -
Call Manager and AD authentication
We change our AD administrator passwords periodically. We recently had callmgr 4.1 installed on our servers. We can't login to the Administration pages of call manager since we changed the AD admin password. How do I fix this?
When you install the Active Directory plugin for CallManager, you must supply a user account with certain read privilegs to AD. This user account is used to bind to AD and check for valid credential sets when you login to CallManager via MLA, CCMUser etc etc.
So, if you change the password for the account which CallManager uses to bind to AD w/, it breaks.
You're not completely out of luck though. When you install MLA, you configure an account with full admin access that you can use at any point that does not authenticate against AD. Most of the time this account is ccmadministrator and the local administrator password of the box, but could have been altered during install. Refer to the following doc for more detailed info: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_callmg/4_1/sys_ad/4_1_3/ccmcfg/b07mla.htm.
Moving forward, I'd recommend creating a service account (denied interactive login to any boxes on your network) for the sole purpose of CallManager binding to LDAP. Make the password complex, mark it never to expire, user can't change, and you shouldn't have to change the password in the future.
Hope this helps! -
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 -
CUCME 8.6 Call not forwarding Voicemail
Hi frieds,
In our office we are using CUCME 8.6 on Cisco 2951 and unity express 8.5 in ISM module. As per our configuration whenever user is busy or not answering , the call will forward to voicemail. Totally we have 24 PSTN line. So we have an additional gateway 2901. The Issue I’m facing is that, when a PSTN incoming call coming through the second gateway(2901), if the extension is busy or not answering the call is disconnecting instead of forwarding to voicemail.
My 2951 configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface GigabitEthernet0/1
bind media source-interface GigabitEthernet0/1
registrar server.
Dial peer we are using for voice mail:
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad.
2901 Configurations
voice service voip
ip address trusted list
ipv4 172.16.19.80
ipv4 172.16.19.81
ipv4 172.16.19.82
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
dial-peer voice 99 voip
destination-pattern 1099
session protocol sipv2
session target ipv4:172.16.19.81
dtmf-relay sip-notify
codec g711ulaw
no vad
============================
Debug CCSIP Calls
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0xAF40FD8
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 5000
Called Number : 1099
Source IP Address (Sig ): 172.16.19.80
Destn SIP Req Addr:Port :
Destn SIP Resp Addr:Port :
Destination Name :
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 172.16.19.80
Source IP Port (Media): 25364
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Dec 15 15:23:23.448: //37497/8FD56BBAA9EA/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 47
Disconnect Cause (SIP) : 200
For your reference I here attach a network diagram
What the command which I missed?Check License status on your CUE, I had same issue.. Finally figured out its about license.. sh license status
Sent from Cisco Technical Support iPhone App -
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 VranckxYour 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.
SteveYou 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
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My wife and I have separate iPhoto databases but to make our lives easier, I want to point my iPhoto to her database. Before I do that, I need to export a bunch of photos from my database and import them into her library. However, these photos are
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PDF´s won´t open in Safari... Wazzup?
Hi! I seem to be unable to open any PDF´s in Safari- when ever I encounter a pdf, I have to copy the link, and paste it in Opera, in order to open it... Am I totally blond and beyond, or what?? I have the same problem on both my Imacs and my MacBook
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