Jun 14

Cisco Meeting Server Displaying Reconnecting for the Web Admin

Deploying a cluster of CMS servers, I had one host that didn’t want to play the game. After enabling the Web Admin service for port 445 I could not reach the Web Admin portal. The CMS Server just sat at “Reconnecting”.

I changed the Web Admin port back to 445. No Luck
I tried both Chrome, IE and Firewfox browsers. No Luck
I used both IP Address and CN Name to access the Web Admin. No Luck.

Version was 2.1.5 (the version Cisco ships with CMS as of June 2018).

I proceeded to download the latest CMS Server version which is 2.3.4 (31st May Release Date). I upgraded the CMS Server, change the Web Admin port back to 445 and I can now access the Web Admin portal.

Not too sure why I only had this issue on this one CMS Server.. I had deployed 4x CMS Server exact same version same hardware.

Webadmin status command output.
webadmin status - cms

Mar 15

Packet Capture on Cisco IOS-XE

Quick configuration snippet for capturing packets on an IOS-XE based Device. You can find the full Cisco article on this link.

  1. Define the location where the capture will occur:monitor capture CAP interface GigabitEthernet0/0/1 both
  1. Associate a filter. The filter may be specified inline, or an ACL or class-map can be referenced:monitor capture CAP match ipv4 protocol tcp any any
  2. Start the capture:monitor capture CAP start
  3. The capture is now active. Allow it to collect the necessary data.
  4. Stop the capture:monitor capture CAP stop
  1. Examine the capture in a summary view:show monitor capture CAP buffer brief
  1. Examine the capture in a detailed view:show monitor capture CAP buffer detailed
  1. In addition, export the capture in PCAP format for further analysis:monitor capture CAP export ftp://192.168.0.1/CAP.pcap
  1. Once the necessary data has been collected, remove the capture:no monitor capture CAP
Feb 25

WFO QM Not Syncing with CCX

A quick little blurb regarding WFO QM and CCX DB Connection.. I’ve setup QM a number of times now.. But I know the below is a little gotcha for newbies.. and speaking from experience this can be very frustrating!

Problem is when attempting to connect the QM Server to the CCX DB. No error is displayed in the Post Install setup.. However, you do not see any CCX info such as Agents when configuring the via the Admin Portal. When attempting to run a manual sync via the Admin Portal, it errors out.

The Sync Log show the below error:

ERROR SYNC2000 Failed to connect to side A of an ACD connection. Will try side B

Resolution is to replace the hyphens with underscores in the server name. This piece of information can be found in the install guide for QM.. Obviously my eyes perused over this key piece of information!

Feb 20

ATA190 Faxing to PSTN via ISDN and CUCM SIP Trunk

I’ve been deploying a lot of CUBE environments of late with faxing working from the T.38 protocol. I did want to jot something down regarding faxing to the PSTN via ISDN using an ATA190 registered to CUCM (doesn’t really matter what version) and connected the Voice gateway via a SIP Trunk.

I’m based in Australia and the fax settings I’ve found to be most successful as follows.

ATA190 Device Configuration page in CUCM

  •  Ring Voltage = 70V
  • Ring Frequency = 25Hz
  • Fax Mode = NSE Fax Pass-through g711alaw

 Voice Gateway (Cisco IOS)

Under the Voice Service Voip Menu.

  • modem passthrough nse codec g711alaw

CUCM SIP Trunk configured as per normal, no special configuration. I have aLaw configured, I found  uLaw (which is also acceptable in Australia) was working for outbound faxing but not for inbound faxing, however I have had some case where the opposite is true.

Just a matter of making logical/strategic changes to the faxing configuration to ensure both directions are operating correctly.

Feb 06

Cisco Jabber Last Logged in Report

To find out the last logged in times for Jabbers in CUCM required the below shell command. I found the command on the Cisco Support Forum.. I have added the reference link to the bottom of this blog. I though I would extend the the process of getting the info out of CUCM and into a spreadsheet to make some sense of the data.

SSH to the CUCM Publisher Server and execute the below SQL command.

run sql select e.userid, cd.timelastaccessed from enduser as e, credentialdynamic as cd, credential as cr where e.pkid=cr.fkenduser and e.tkuserprofile=1 and e.primarynodeid is not null and cr.tkcredential=3 and cr.pkid=cd.fkcredential order by cd.timelastaccessed

Jabber last Logged In

Snippet of the results.

jabber-last-logged-in-1

I had logging enabled in Secure Shell as the results will span past the shell buffer. Open Excel and import the log file generated. I use ‘Delimitated’ and separated via ‘space’.

After the log file has been imported into excel, I use the Unix to excel time formula to make sense of the date.

Formula is =CELL/(60*60*24)+”1/1/1970″

*Note: ensure the format of the cell is ‘Date’.

Save the spreadsheet, and you now have a full list of all users and their last logged in date for Jabber.

References:

Cisco Support Forum: https://supportforums.cisco.com/t5/unified-communications/jabber-report/td-p/2957556

Feb 02

CSR 12.0 Extension Mobility Sign-in Options

Extension Mobility in Cisco Collaboration Services 12.0 has expanded with a couple of useful options. When configuring Extension Mobility sign-in service, you can now elect to setup a further two types of sign-in options.

These are Primary Extension and PIN known as Login Type ‘DN’. The second is using the Self Service User ID and PIN, known as the login type ‘SP’. This is on top of the current User ID and PIN, know as the login type ‘UID’.

URLs for these options are below.

Login Type DN

URL: http://:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=DN

Login Type SP

URL: http://:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=SP

Login Type UID

URL: http://:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#&loginType=UID

Nov 14

MoH Silent – SIP CUBE to ITSP without MTP – OPTION 2

More MoH talk with ITSP and CUBE’s. I mentioned in earlier posts that playing MoH without an MTP can be achieved by creating sip-profiles to manipulate some of the SDP attributes. I recently have another MoH issues where the MoH stream was simply dead air or silent. Of course enabling the MTP on the SIP Trunk in CUCM resolved the issue.. however we want to avoid forcing an MTP for all calls.

I resolved this by removing the cmd “pass-thru content sdp” under the Voice Service Voip -> SIP config menu. In this case the sip-profile route was not working for me.. The above cmd negates the Gateway in the negotiation process, hence passing through codec and mtp negotiations. The potential problem here is the mismatch between CUCM and ITSP, we want the gateway to participate and effectively inter-work between CUCM and the ITSP.

If you have other options or methods that work to combat silence in MoH using an ITSP, please post.

Apr 30

Cisco Jabber cannot Call out to PSTN

Migrated to SIP Carrier and experience an issue where Deskphones could call out to the PSTN, however the Cisco Jabber softphones could not. The annunciator message was played from the carrier network. The remote device (being the device across the PSTN Network) would ring once.

Tracing the SIP messages, I could see the carrier was sending back a SIP reason of “Q.850;cause=41″. Looked up the code which is “Temporary Network Failure – Try again”. So not too much help there..

I analysed the SDP being sent to the carrier from the Cisco Jabber softphone vs the Deskphone and found the video and content sharing attributes were being passed out to the Telco. This Telco connection is just an audio PSTN service, so it would not support video.

I created a new Device Pool for the SIP Trunk to the CUBE, along with a new Region and set the Video to “None”. This effectively disables Video, hence any endpoint including Cisco Jabber establishing call, will not send the Video/Content media attributes. Telco is now happy and calls proceed through the PSTN.

FYI, have pasted below the differences in the SDP.

SDP – Cisco Jabber

v=0
o=CiscoSystemsSIP-GW-UserAgent 9138 8218 IN IP4 192.168.241.20
s=SIP Call
c=IN IP4 192.168.241.20
t=0 0
m=audio 17206 RTP/AVP 8 0 18 101
c=IN IP4 192.168.241.20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 51372 RTP/AVP 31
c=IN IP4 192.168.241.20
m=application 17458 RTP/AVP 125
c=IN IP4 192.168.241.20

SDP – Standard IP Phone (No Camera)

v=0
o=CiscoSystemsSIP-GW-UserAgent 7438 4222 IN IP4 192.168.241.20
s=SIP Call
c=IN IP4 192.168.241.20
t=0 0
m=audio 18034 RTP/AVP 8 0 18 101
c=IN IP4 192.168.241.20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

a=ptime:20

Apr 19

MoH – SIP CUBE to ITSP without MTP

No Music to PSTN using SIP Cube toward ITSP? Using an MTP would resolve this issue, however if using an MTP was not an option, then use the below SIP Profile statement to allow Music on hold to stream toward the PSTN.

Create (or append if you are already using sip-profiles) a sip-profile voice class and add statements.

voice class sip-profiles 1
request REINVITE sdp-header Audio-Attribute modify “inactive” “sendrecv”
request ACK sdp-header Audio-Attribute modify “sendonly” “sendrecv”
response 200 sdp-header Audio-Attribute modify “sendonly” “sendrecv”

Apply the sip-profiles globally or on the Dial-Peer.

** GLOBAL**

Voice service voip
sip
sip-profiles 1

** DIAL-PEER **

Voice-class sip profiles 1

Apr 10

Group Voicemail Alternative – Cisco Jabber

Departmental or Group voicemail is always requested when deploying a phone system. The traditional method of distribution lists worked great, however since the introduction of Jabber, there seem to be caveats in configuring these lists. Jabber will display the voice message no worries, this is great.. Except you can not accept or decline a voice message. So you will still need to log into the Voicemail Server via the phone and follow the prompts to accept messages.

Another method is to use Alternate Extensions (if the user doesn’t already have a mailbox). This is limited to a certain number of users who can access the mailbox via alternate extensions. But I’ve found in most cases.. The number of users wanting access to department or group mailbox is fewer than the limitation.

Alternate Extensions also allows Jabber to visually display the voicemail, allow to tag as unread, read, delete message etc.. And yes voice message are indeed synchronised across the Jabber clients that have access to the mailbox. Users can also forward voice messages from Jabber 11.8.

Outline on how to set up this alternate method is below.

The Jabber Service Profile must include the Voicemail Server configuration along with the Credentials field set to “not set”. ie this means the user must enter the credentials for the mailbox.

On the Unity Connection Server ensure the below is configured.

- Mailbox Password is set (This is not the PIN)
- Alternate Extension of the Jabber Users
- MWI Extension of the Jabber Users.

Assign the Jabber Service Profile to the End User in CUCM. Once the user logs into the Jabber client, select the Voice Messages Tab. The user will have to enter the credentials of the group mailbox.

On top of this, we can also configure single mailbox for the group. (if the destination email address is a Exchange/O365 mailbox).