Archive for the ‘Features’ Category

Customer reported a problem about their UCCX version 8.5.1.11001-35 for CAD Agents.

This is what they say about the problem:

We have a UCCX Agent with a 9951 Cisco Phone with 2 Lines.
First line is the private Line
Second Line is the agent Line.
Agent is logged into CAD with his agent extension.

Contact Center gets a customer call for the agent and the agent accepts the call on Agent Line.
Then the agent wants to ask a colleague or wants to transfer the call and presses the call transfer key (on CAD)
The transferred Call is answered by the colleague and they start talking.
The customer hears Music on hold.
So far everything is normal.

Now a new call comes in on the private Line and hits the CFNA to voicemail, or the caller hangs up.

The issue is that at that point with no interaction of the agent or his colleague the transferred Call from the agent to the colleague gets disconnected.

After the issue occurs the call from the private line is on the voicemail.
And the first call (Contact center) still hears music on hold.

This is the fact on all of our phones when an agent is logged in with CAD.

= = =

Later we found there was a bug CSCts44173 hitting this UCCX Version so we asked customer to upgrade to a fixed Version. Customer upgraded the UCCX to 8.5.1.11003-32 but the issue was there for external calls. Calling internally fixed the issue.

After going through the logs we found that the person who was testing had a “Remote destination” setup for his mobile number and this is why they were facing the issue.

So, when customer is facing this very issue make sure they are not using “remote destination” setup.

You may have come across this that though you are logged into your profile through Extension mobility but when you try to access your “Personal Directory” you will be prompted for a userid and pin. Unfortunately, this is by design and cannot be changed. There is a workaround to escape entering userid and pin which I found after getting several requests from customers. If you are among one of those who is annoyed with this then following is the procedure to get rid of it. You will need Admin rights on Call manager to make these changes.

  1. Create a new phone service for Personal directory named PAB
  2. Use this URL http://server-name-or-ipaddr:8080/ccmpd/login.do?name=#DEVICENAME#&service=​pab​
  3. Add the Parameters “name” “pin” and “userid” with no default value (Case sensitive)
  4. Now go to the phone and Subscribe to “PAB”
  5. In the name field enter Phone MAC address with SEP like SEPAABBCCDD9876
  6. In the pin field enter user pin
  7. In the user id enter user id of that user
  8. Click Subscribe and save
  9. Reset the phone

This will now add a  new service called “PAB” under services so you can go in Service and select “PAB” to go directly into Personal directory and the system will not ask you userid and pin.

You may also add this service to a button on the phone. Just add a Service URL type on your phone from “button template” and then you can access PD from that button.

You may also ask users to subscribe to this service by logging into CCMUSER and then adding this service from Device > Services.

The same procedure applies for FASTDIALS except the following changes:

  1. Name the new Service FastDials
  2. The URL is http://server-name-or-ipaddr:8080/ccmpd/login.do?name=#DEVICENAME#&service=​fd
  3. Add only “pin” and “userid” in Parameters
  4. Subscriber Phone or user to this service

Here are screenshots from my Communicator as how this service will appear:

With the missed call logging for shared lines feature, the administrator can configure Cisco Unified Communications Manager Administration, or the phone user can configure Cisco Unified CM User Options, so Cisco Unified Communications Manager logs missed calls in the call history to a specified shared line appearance on a phone.

This will only work if users who are sharing the number are logged in using Extension mobility.

You can configure it here under extension number by going into each phone:

If the box is checked, missed calls will be logged and if it is not checked then missed calls won’t be logged.

I had an interesting case where SIP calls over a SIP trunk were dropping after like 75 minutes. The duration was not confirmed as sometimes it use to drop even before 75 minutes.

I looked into few CCSIP debugs (debug ccsip messages) and found that the ‘BYE’ message was actually coming from our end (Call manager/Gateway).

Platform information:

CCM: System version: 7.1.3.32900-4

CUBE-10#sh ver

Cisco IOS Software, 3800 Software (C3825-ADVIPSERVICESK9-M), Version 12.4(24)T4, RELEASE SOFTWARE (fc2)

Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2010 by Cisco Systems, Inc.
Compiled Fri 03-Sep-10 09:15 by prod_rel_team

ROM: System Bootstrap, Version 12.4(13r)T, RELEASE SOFTWARE (fc1)

CUBE-10 uptime is 27 weeks, 3 days, 44 minutes

voice-card 0
!
!
voice call send-alert
voice rtp send-recv
!
voice service voip
allow-connections sip to sip
fax protocol cisco
sip
options-ping 180
!
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711alaw

!

voice class sip-profiles 1
request INVITE sip-header Allow-Header modify “, UPDATE” “”

!

dial-peer voice 10 voip
description *** Outbound to GXXX – SIP Provider ***
translation-profile outgoing OUTBOUND
huntstop
destination-pattern 9T
voice-class codec 1
voice-class sip early-offer forced
voice-class sip profiles 1
session protocol sipv2
session target ipv4:10.0.222.69
dtmf-relay rtp-nte
fax-relay sg3-to-g3
no vad

!
!

gateway
timer receive-rtp 1200

!
sip-ua
retry invite 1
retry response 2
retry bye 2
retry cancel 1
retry options 1
timers trying 200
!

!

!

Debug Output:

Jul 11 12:17:54.579 BST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

BYE sip:079XXXXXXX@10.0.222.69:5060 SIP/2.0

Via: SIP/2.0/UDP 10.0.222.65:5060;branch=z9hG4bK100B17D
From: “anonymous” <sip:anonymous@10.0.222.65>;tag=B891B2E8-282
To: <sip:079XXXXXXX@10.0.222.69>;tag=3519367399-656588
Date: Mon, 11 Jul 2011 11:15:27 GMT
Call-ID: D687F817-AADB11E0-A725957A-CA524E0B@10.0.222.65
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1310383074
CSeq: 128 BYE
Reason: Q.850;cause=16
Content-Length: 0

Jul 11 12:17:54.591 BST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.0.222.65:5060;branch=z9hG4bK100B17D
To: <sip:079XXXXXXX@10.0.222.69>;tag=3519367399-656588
From: “anonymous” <sip:anonymous@10.0.222.65>;tag=B891B2E8-282
Call-ID: D687F817-AADB11E0-A725957A-CA524E0B@10.0.222.65
CSeq: 128 BYE

Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH

Contact: <sip:079XXXXXXX@10.0.222.69:5060>

Content-Leng

= = = =

This is what I have done to fix the problem:

I added these commands under sip-profile:

voice class sip-profiles 1
request INVITE sip-header Allow-Header modify “UPDATE,” “”      < < <  This was already there
request REINVITE sip-header Allow-Header modify ” UPDATE,” “”
response 200 sip-header Allow-Header modify “UPDATE,” “”
response 180 sip-header Allow-Header modify “UPDATE,” “”

The next step was to increase the Service Parameter “SIP Session Expire Timer” which you can find under Service Parameters > Call manager:

That was it, it fixed the issue and customer confirmed having a call for more than 3 hours.

I had a request to do a quick write up on how to configure extension mobility properly.

These are the steps involved (considering Call manager Linux versions):

1 – Configure extension mobility service by going into Device > Device settings > Phone Services

The service URL is : http://call-manager-IP:8080/emapp/EMAppServlet?device=#DEVICENAME#

If you want to set it up as an Enterprise service then check that box “Enterprise Subscription” otherwise leave it. If you check that box, extension mobility service will appear globally for all phones and you won’t be able to find this service under phone Subscribe services in Step 2. If you don’t check that box then move to step 2

2 – Under phone, susbcribe this service as follows by dropping down the menu at top right corner and then going into “Subscribe/Unsubscribe Service”

3- Staying on the same phone page, scroll down and check the extension mobility check box otherwise no one will be able to login:

4 – Create a User device profile Device > Device Settings > Device profile. Any higher generation phone profile will work on lower one but that’s not true the other way around. So a Cisco 7960 device profile can be used to login oo a 7912 phone but a Cisco 7912 UDP won’t login on a Cisco 7960 phone.

5 – After creating UDP, the most important and sometimes missed step is to subscribe the Extension mobility service again like step 2 at UDP level

6 – Go into user from from User management > End user and add that device profile to it. Also select the primary extension at that user page.

Login and you should be good to go. Any issues, it would either be related to service not subscribed at UDP level or a UDP not associated with a user.

Web Link for EM:

There can be a situation where your Pub is down or inaccessible and users wants to login or logout. As EM service is dependent on Publisher they won’t be able to login/logout and will get “host not found”.

There is a workaround where users can login to their phones through a web link.

To login using URL:

http://<CUCM-SUB IP Address>/emapp/EMAppServlet?device=MACADDR&userid=USERID&seq=XXXXX

XXXXX = Pin

Example:

http://192.168.70.11/emapp/EMAppServlet?device=SEP001E4AF0F9E4&userid=ajon&seq=12345

To logout the user:

http://&lt; CUCM-SUB IP Address>/emapp/EMAppServlet?device=SEP001646D97913&doLogout=true

I came across this issue with one of  our customer last week where a call was coming in to receptionist at Site ‘A’ but she was not able to transfer the call to someone at Site ‘B’. They had Centralized deployment and all phones on branch sites were registered to Central site. The call was dropping as soon as she would press ‘hold’ to transfer the call. This is what I was getting on Site ‘A’ gateway:

2621-VG01#
.Feb  7 09:25:14.805: ISDN Se1/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0002
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98382
Exclusive, Channel 2
Called Party Number i = 0x81, ‘700481’
Plan:ISDN, Type:Unknown
.Feb  7 09:25:14.837: ISDN Se1/0:15 Q931: TX -> CALL_PROC pd = 8  callref = 0x8002
Channel ID i = 0xA98382
Exclusive, Channel 2
.Feb  7 09:25:14.961: ISDN Se1/0:15 Q931: TX -> ALERTING pd = 8  callref = 0x8002
Progress Ind i = 0x8188 – In-band info or appropriate now available
.Feb  7 09:25:22.877: ISDN Se1/0:15 Q931: TX -> CONNECT pd = 8  callref = 0x8002
.Feb  7 09:25:22.945: ISDN Se1/0:15 Q931: RX <- CONNECT_ACK pd = 8  callref = 0x0002

.Feb  7 09:25:41.934: ISDN Se1/0:15 Q931: TX -> DISCONNECT pd = 8  callref = 0x8002
Cause i = 0x80FF – Interworking error; unspecified
.Feb  7 09:25:42.118: ISDN Se1/0:15 Q931: RX <- RELEASE pd = 8  callref = 0x0002
.Feb  7 09:25:42.122: ISDN Se1/0:15 Q931: TX -> RELEASE_COMP pd = 8  callref = 0x8002

Now this error had no explanation whatsoever and it was quite hard to troubleshoot.

I knew that for a transfer function to work properly across WAN (G729-G729) you need a MTP. The MRGL had a group with MTP as well as transcoder. I noticed MTP was added as part of one group where announciator, CFB etc were added with it as well. Transcoder was added seperately. I think I read somewhere that Call manager sometimes take Transcoder as MTP if MTP is not added in a seperate group. As transcoder can’t do G729-G729, it will fail to provide supplemantary services to the gateway and eventually the call will drop. As a good design approach, always place MTP in a seperate group and at the top of MRGL.

I removed MTP from that group, placed it in a seperate group and prioritized it at MRGL.

Made test calls and they all worked fine.

The users which are part of a global directory can setup their own personal directory, speed dials etc by access CCMuser webpage.

The link for any Call manager would be:

https://x.x.x.x/ccmuser

x.x.x.x = CUCM Pub

The username and password will be same as mentioned under User Management > End user.

You can restrict users to access certain things from Enterprise Parameters.

Background Image on Gen 2 and Gen 3 Cisco IP phones can be changed by using the following procedure.

I will be showing you how to do for a Cisco 7965 IP phone but the same procedure can be used for other IP phones (like 7941. 7961, 7945 etc). The only difference between different family of phones will be the Image sizes.

1) Resize the Images as per 7965 Phone.
2) Full size image— 320 pixels (width) X 212 pixels (height) and the thumbnail image— 80 pixels (width) X 53 pixels (height).
3) Save the full size file as ‘large.png’ and the thumbnail as ‘small.png’ – (you can give any name as long as it match exactly in List.xml)
4) Create a notepad file called “List.xml” – do change the extension to .xml.
5) Add the following lines in List.xml

<CiscoIPPhoneImageList>
<ImageItem Image=”TFTP:Desktops/320x212x16/small.png”
URL=”TFTP:Desktops/320x212x16/large.png”/>
</CiscoIPPhoneImageList>

6) Save the file and double click it to open it, the file should open in a web browser
7) Load the Image files and List.xml on CUCUM TFTP


8 )Repeat the same process for the two Image files
9) Upload all the files on both Publisher and Subscriber
10 Restart TFTP service

Go to the phones > settings > user settings > Background Image and select the image from there.

The information about customization can be accessed from Cisco Support link.