Mark Holloway

MeetMe conferencing with Call Manager Express allows a CME phone to go off hook, press the MeetMe softkey, and dial the MeetMe bridge number to start a conference. Any other CME subscriber or off-net party may dial the MeetMe number to join the conference.  By default CME does not play tones when participants join or leave the bridge. Adding this capability requires the creation of a voice class for both the join tones and leave tones where tone frequencies are defined and then later referenced in the dspfarm profile.  In order to perform the following configuration you should already have hardware conferencing configured on CME.  Reference this post for assistance.

1. Configure a number to be a MeetMe Conference Bridge

ephone-dn  10  octo-line
number 4454 no-reg primary
conference meetme

2. Configure an ephone template that includes the MeetMe Softkey. This is so CME subscribers can easily start a conference.

ephone-template  1
softkeys remote-in-use  CBarge Newcall
softkeys seized  Redial Endcall Meetme Pickup Gpickup Cfwdall Callback
softkeys connected  Hold Endcall LiveRcd Park Trnsfer Confrn

ephone  1
privacy-button
mac-address 0022.90BA.2CC0
ephone-template 1
busy-trigger-per-button 4
button  1:1 2:2

3. Configure the appropriate voice class to define Join/Leave tones

voice class custom-cptone conf-leave
dualtone conference
frequency 300 900
cadence 300 150 300 100 300 50

voice class custom-cptone conf-join
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50

4. Assign each voice class to the dspfarm profile used for conferencing

dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
conference-join custom-cptone conf-join
conference-leave custom-cptone conf-leave
associate application SCCP

The Broadsoft Software Management Guide provides an example for configuring active/backup Ethernet interfaces (called mode 1) when configuring bonded Ethernet on a single Red Hat Enterprise server.  Only one Ethernet interface is transmitting and receiving traffic at a time while the other interface is in standby mode.  My preferred approach is 802.3ad Link Aggregation Control Protocol (LACP). LACP creates aggregation groups that share the same speed and duplex settings across Ethernet interfaces on a server. It utilizes all slaves in the active aggregate according to the 802.3ad specification.  802.3ad requires an Ethernet switch that is capable of supporting LACP.

## Red Hat Enterprise Linux ##

#vi /etc/modprobe.conf
alias eth0
alias eth1
alias bond0 bonding
options bonding miimon=50 mode=4   (note: miimon = frequency in milliseconds to monitor interfaces)

==AS1==
#vi /etc/sysconfig/network
NETWORKING=yes
NETWORKING_IPV6=no
HOSTNAME=as1
NETWORKDELAY=25

#cd /etc/sysconfig/network-scripts
#vi ifcfg-bond0
DEVICE=bond0
BOOTPROTO=static
ONBOOT=yes
USERCTL=no
BROADCAST=192.168.126.255
IPADDR=192.168.126.145
NETMASK=255.255.255.0
NETWORK=192.168.126.0
GATEWAY=192.168.126.1

#vi ifcfg-eth0
# Broadcom Corporation NetXtreme II BCM5708 Gigabit Ethernet
DEVICE=eth0
BOOTPROTO=none
ONBOOT=yes
MASTER=bond0
SLAVE=yes

#vi ifcfg-eth1
# Broadcom Corporation NetXtreme II BCM5708 Gigabit Ethernet
DEVICE=eth1
BOOTPROTO=none
ONBOOT=yes
MASTER=bond0
SLAVE=yes

## Cisco Switch ##

interface Port-channel30
description Port-Channel to as1
switchport
switchport access vlan 12
switchport mode access
spanning-tree portfast

interface GigabitEthernet3/24
description [as1] IBM x3650 m2 – eth0
switchport access vlan 12
switchport mode access
spanning-tree portfast
channel-group 130 mode active

interface GigabitEthernet6/24
description [as1] IBM x3650 m2 – eth1
switchport access vlan 12
switchport mode access
spanning-tree portfast
channel-group 130 mode active

The only reason I can foresee Broadsoft not providing this approach in the Software Management Guide is due to the requirement of an Ethernet switch capable of supporting LACP and the associated configuration.  Using active/standby “mode 1″ bonding does not require any additional configuration beyond RHEL, so while it is more convenient and simplistic, this configuration does not fully maximize the capability of the servers.  This can be especially useful for the Media Servers, Conference Servers, or the Application Servers where Broadworks log files or Radius CDR’s are being offloaded to a separate server.

This is a brief explanation of how to retrieve the Music on Hold files from Cisco Unified Communications Manager 7 and load the files into a router’s flash for remote-site streaming directly from the router.  Alternatively, if you have Call Manage Express you can use music-on-hold.au file. The only difference between them is the actual music is different.

In order to complete this you must have a Secure FTP (SFTP) server available on your network and reachable by the UCM Publisher or Subscriber.  If you don’t already have an SFTP server on your network you can use CentOS, FreeBSD, RHEL, Mac OS X, or any Unix-based platform where SFTP is natively part of the OS.

Perform the following steps.

1. SSH to the UCM Publisher or Subscriber node.
2. Enter the following command file list activelog mohprep/* and note the list of files displayed.
3. In this case we want to SFTP the file SampleAudioSource.g729.wav to our SFTP server
4. Enter the following command file get activelog mohprep/SampleAudioSource.g729.wav
5. The following information will be displayed. Enter the appropriate SFTP information when prompted.

___

Please wait while the system is gathering files info …done.
Sub-directories were not traversed.
Number of files affected: 1
Total size in Bytes: 2702728
Total size in Kbytes: 2639.3828
Would you like to proceed [y/n]? y

SFTP server IP: 177.1.10.2
SFTP server port [22]:
User ID: mark
Password: *********

.
Transfer completed.

___

At this point the file SampleAudioSource.g729.wav now resides on the SFTP server.  Copy this file to a TFTP server and issue the command copy tftp flash on the router the file should reside on.  The same server I used for SFTP is also a TFTP server.  Any Unix-like OS is capable of supporting SFTP and TFTP.

router# copy tftp flash
Address or name of remote host []? 177.1.10.2
Source filename []? SampleAudioSource.g729.wav
Destination filename [SampleAudioSource.g729.wav]?
Accessing tftp://177.1.10.2/SampleAudioSource.g729.wav…
Loading SampleAudioSource.g729.wav from 177.1.10.2 (via Serial0/3/0.1): !!
[OK - 332600 bytes]

332600 bytes copied in 51.848 secs (6415 bytes/sec)

The following undocumented command is useful for debuging H323 messages.  This command cannot be found in the help-context of IOS as noted below.

router# debug gatekeeper main 10

router# debug gatekeeper ?
gup Gatekeeper GUP messages
load Gatekeeper load-balancing messages
servers Gatekeeper Servers

The great thing about this command is the simplicity in the debug output.

Click the screenshot for a sample output.

CSIM is an undocumented IOS command.  When configuring a router to act as a voice gateway it is possible to test outbound calls by originating calls from the router.  If the router is configured correctly then the far end number will ring and may be answered. Use the csim start dialstring hidden command to initiate simulated calls to whichever real-world E.164 number is desired. This allows you to determine whether you can properly go offhook from the router to the PSTN, send digits, and complete a call to the destination phone. You can modify the POTS dial-peer appropriately to account for long-distance access codes and other prefixed digits as necessary. In the example below, the POTS dial-peer can match on any string of digits starting with “9”, and all digits that follow the “9” are played out voice-port X/Y/Z.

dial-peer voice 1 pots
destination-pattern 9T
port 1/0:0

r1# csim start 918005551212

csim: called number = 18005551212, loop count = 1 ping count = 0

csim err csimDisconnected recvd DISC cid(28)

csim: loop = 1, failed = 1   csim: call attempted = 1, setup failed = 1, tone failed = 0

Despite the call working in this case CSIM will always display failed=1.  There is no real explanation for this.  CSIM can also be used with PRI, CAS, and if calls are not completing it is recommended to begin analyzing the call flow using debug commands. For example debug isdn q931 will show the Tx and Rx messages (including ISDN Cause Codes) between the local router and the far end TDM equipment.

Here are other examples of POTS dial-peers which reflect different ways to pass digits to the TDM network.

Any of these now sends the entire string “12345678” to the PBX:

!
dial-peer voice X pots
 destination-pattern 1234....
 port 1/0:0
 forward-digits all
!

or:

!
dial-peer voice X pots
 destination-pattern 1234....
 port 1/0:0
 no digit-strip
!

or:

!
dial-peer voice X pots
 destination-pattern 1234....
 port 1/0:0
 prefix 1234
!