Step 1 Activate the network connectivity to CUE
from the CME router.
int ISM0/0
ip unnumbered GigabitEthernet0/0.34
service-module ip address 10.34.x.251
255.255.255.0
service-module ip default-gateway 10.34.x.1
no shut
!
ip route 10.34.x.251
255.255.255.255 ism0/0
Note: For 'ip unnumbered'
command to work, you need to make sure that 'proxy-arp' is enabled on the
original interface.
Step 2 Configure a dial-peer on the CME router
for CME phones to call voicemail number. CUE interface with CME is using SIP
protocol only whether you are using SCCP or SIP CME.
dial-peer voice 780
voip
description **** VOICEMAIL DIAL PEER ****
destination-pattern xxxyyy
session protocol sipv2
session target ipv4:10.34.x.251
codec g711ulaw
voice-class sip transport switch udp tcp
voice-class sip bind control source-interface
GigabitEthernet0/0.34
voice-class sip bind media source-interface
GigabitEthernet0/0.34
dtmf-relay sip-notify rtp-nte
ip qos dscp cs3 signaling
CUE supports G711ulaw codec
only. In case the dial-peer is using other codec, the CME will invoke a
transcoder else the call will fail.
Step 3 Configure Voicemail access on Voicemail
button.
voice register
global
voicemail 5222121
!
telephony-service
voicemail 5222121
Step 4 Configure Call forwarding on directory
numbers to voicemail
voice register
dn1
call-forward b2bua busy xxxyyy
call-forward b2bua noan xxxyyy timeout 30
call-forward b2bua unregistered xxxyyy
!
ephone-dn
11 call-forward busy xxxyyy call-forward noan xxxyyy time 10
call-forward unregistered xxxyyy
Step 5 Configure MWI. We will cover this is
separate section
Step 6 Configure voicemail application on CUE
Remote connectivity
(SSH/Telnet) to CUE module can't be established. You need to console to the CUE
module from the router. The credentials of the CUE module will be same as the
router (AAA or Local).
This feature is
introduced in CME 10.0 - 15.3(3)M. It provides support for new SIP phone device
without changing the IOS version .
Before
In the subsequent
CME releases, these new phones would be added into the list of supported phones
for CME with the necessary code changes. Currently all the new SIP phone
devices which are not yet supported on CME can register to CME as 3rd party SIP
phones and get generic SIP line features like Call Hold, Call Resume etc.
After
Fast track support
would provide a new configuration utility to provide the phone characteristics
of a new SIP phone device. As part of this feature, CME code would be
enhanced to retrieve the phone characteristics of a new SIP phone device and
allow the registration of these new SIP phone devices. With this new
configuration utility, existing SIP features on CME would be made
available to the new SIP phone devices.
Some new phones do
not add new features, but just change the display, change the button layout,
etc. Some new phones are created due to cost cutting to provide fewer features
or provide the same features with different hardware. These kinds of new phone support
which do not change software protocols, do not need CME new feature support are
the targets of this feature. To take advantage of the IOS rich parser
capability, the configuration will be done using CLI commands.
Forward Compatibility
New SIP phone model
is configured using the Fast-Track configuration approach . When CME is
upgraded to a later version which has the built-in support for this new SIP
phone model , the Fast-Track configuration for the SIP phone model gets removed
automatically
If the CME is
downgraded to version which does not have the built-in support then the
Fast-Track configuration should be applied again manually.
Restrictions:
Provisioning of new SIP
Phones is supported using IOS CLI commands only, No GUI and SNMP support.
Only XML format of
phone configuration file will be supported. Text format will not be
supported by this feature.
Cisco Legend phones
ATA,7905,7912 supports only TEXT, such new phone types are not supported.
Only built in supported
phones can be used as reference phone while configuring a new SIP phone
model
New phones having a
"new call flow" , "new message flow" or "new
configuration file format" that are not supported in CME will not be
supported by this feature as it needs code changes at CME.
Configuration
configureterminal
voice register
pool-type pool-type
addons
max-addon count
descriptionstring
gsm-support
num-linesnumber
phoneload-support
reference-pooltype
pool-type
telnet-support
transport transport-type
xml-config xml-tag value
exit
Command or Action
Purpose
Step 1
enable
Router> enable
Enables the
privileged EXEC mode. Enter your password if prompted.
Step 2
configureterminal
Router# configure
terminal
Enters the global
configuration mode.
Step 3
voice register pool-type
Router(config)#
voice register pool-type 9900
Enters the voice
register pool configuration mode and creates a pool configuration for a Cisco
Unified SIP IP phone in Cisco Unified CME.
If the new phone
type is an existing phone that is supported on Cisco Unified CME release, you
get the following error message:
ERROR: 8945 is
built-in phonemodel, cannot be changed
Step 4
addonsmax-addons
Router(config-register-pooltype)#
addons 3
Defines the
maximum number of add-on modules supported in Cisco Unified SIP IP phones.
max-addons —The maximum allowed value
is 3. The configured add-on modules can be used while defining the pool
for the new SIP phone model using the existing type command as shown
below:
type
[addon 1 module-type [2 module-type]]
Step 5
descriptionstring
Router(config-register-pooltype)#
description TEST PHON
Defines the
description string for the new phone type.
Step 6
gsm-support
Router(config-register-pooltype)#
gsm-support
Defines phone
support for Global System for Mobile Communications (GSM) support.
Step 7
num-lines max-lines
Router(config-register-pooltype)#
num-lines 12
Defines the
maximum number of lines supported by the new phone.
max-lines —If this parameter is not
configured, the default value 1 is used.
Defines phone
support for firmware download from Cisco Unified CME. You can use the load
command in the voice register global mode to configure the corresponding
phone load for the new phone type if it supports phone load.
Step 9
telnet-support
Router(config-register-pooltype)#
telnet-support
Defines phone
support for Telnet access.
Step 10
transport {
udp | TCP }
Router(config-register-pooltype)#
transport TCp
Defines the
default transport type supported by the new phone.
If this parameter
is not configured, UDP is used as the default value. The session-transport
command configured at the voice register pool takes priority over this
configuration.
There
are two types of conferences supported by CME
Ad-hoc Conferences
In
ad-hoc conferences, the initiator will use Conf softkey
to dial parties and join them into conference (there
are other advanced methods to establish conference as well) . Ad-hoc conferences can be completed using
software CFB or hardware CFB.
Software CFB
- Software CFB uses router CPU - Software resources can
provide a max of 3 participants (SIP, SCCP, or PSTN) with G711 codec only.
For mixed codecs transcoder should be invoked. In case XCODER isn't
available, conference can't be completed. - The max number of software
ad-hoc simultaneous conferences depends on CME platform. - Each phone can be part of one conference at a time
For
software CFB nothing needs to be configured since its enabled by default and
can't be disabled. A
workaround to disable it is to enable hardware CFB without configuring DSP
resources.
To
change some default configuration for software CFB:
Number of Conference Sessions and Gain
As
mentioned earlier, the maximum number of conferences is platform dependent.
However, this can be limited manually using,
The
gain keyword offers the option to raise
the volume of the stream for all call legs (VoIP/Tele).
Note: Although its applied under
SCCP CME, its applicable for phone SIP/SCCP phones.
Modify Conference End Call behavior
By
default, once the initiator disconnects the call, all conference parties will
be disconnected (regardless if they are SCCP/SIP/Tele). This behavior can be
changed per phone.
For
SCCP initiators, here are the available options:
no keep-conference: This is
the default mode, where the conference is dropped once initiator hangs up
or press EndCall softkey. Also, initiator can use Conf softkey to drop the
last participant
keep-conference (No keywords
used): The initiator can use EndCall softkey to terminate the
conference or hang up to disconnect from conference while other two
parties are connected. Conf softkey can be used to break the conference but
stay connected to both parties while one is on-hold
drop-last: This changes the
action of Conf softkey which will disconnect the last party from
conference.
Endcall: This changes the
action of Endcall softkey which will disconnect the initiator from
conference while keeping other parties connected. The same will be the
case if initiator hangs up
local-only: When the
initiator disconnects, parties will remain connected (using transfer) only
if one of them is local to CME
For
SIP initiators, those options aren't available. Its either all parties
disconnected or leave them joined once the initiator disconnects.
Its very important to know that keep-conference feature requires call transfer to
enabled. Once the initiator disconnects, call transfer will be initiated to
join the remaining parties together. This isn't a problem in case all phones
are part of same CME system since transfer is allowed. However, if one of the
parties is remote (e.g. SIP trunk), in this case call transfer (either blind or
consult) should be enabled to keep the other parties connected.
Note: During call conference,
the display on initiator will be showing last called number. Also, on parties,
initiator will be displayed as calling number. On initiator screen, Conference keyword will be displayed in
the place of system message. In case the initiator disconnects, the display
will be updated to show the on-call parties as calling numbers (assuming
keep-conference is enabled).
Note:
MOH isn't suppressed from the conference call incase any of the parties places
the call on-hold
Hardware CFB
- Once hardware CFB is enabled,
software CFB will be disabled. - Allow more than 3
participants per conference - Does not support the
local-consult transfer method (transfer-system local-consult command) - In case of multi-codec legs,
CFB can do transcoding as well as conferencing (it won't invoke separate
XCODER) - Hardware CFB isn't supported
for SIP phones. SIP phones will still use software CFB with max of 3
participants in case SIP phone is initiator. However, SIP phones can be parties of hardware conference (SCCP
phone will be initiator). - Each phone can be part of one conference at a time - For CFB to work, DNs should
be configured to act as bridges with conference ad-hoc/meetme feature enabled. Those DNs
can't have other features such as call forward and can't be assigned to
ephones. - Once conference is triggered,
each party will be transferred to one of those conference DNs acting as
its bridge. Therefore, the number of
conference DNs configured should be enough to accommodate the max number
of participants configured combined with max number of conference
sessions. - Those DNs can't be
single lines (conference ad-hoc/meetme command won't be taken and will
throw error).
They can be either dual or octo lines. E.g. for one conference with 5
parties, we should have 3 dual-line conference DNs or one octo-line
conference DN.
ephone-dn20octo-line
number 3020
description **** HW CFB ***
conference ad-hoc
SiteC#show call leg active summary
GLElog A/O FAX T CodectypePeer AddressIP R:
G17B4L 372NORGT33g711ulawTELEP3002
G17B9L 374NORGT24g729r8VOIPP3001135.9.75.10:18688
G17B4L 377NORGT21g711ulawTELEP3020
G17B9L 378NORGT21g729r8TELEP3020
G0L 379NORGT21g711ulawVOIPP142.6.66.254:2000
G0L 37DNORGT21g729r8VOIPP142.6.66.254:2000
G17C6L 37ENANST17g711ulawTELEP32143003
G17C6L 37FNORGT17g711ulawTELEP3020
G0L 380NORGT17g711ulawVOIPP142.6.66.254:2000
- Since the parties will be
transferred to conference DNs, then CME call transfer should be enabled
(especially for external parties to join). - Once the conference is
established, the display on all parties will show To
Conference. - If any party disconnect from
conference, the audio stream of other remaining parties won't break and
will remain connected to CFB. If the number of remaining parties become 2,
the CFB will be release and the stream will break to reconnect directly
between parties as point-to-point call. Also, the display will be updated
with new DNs. - The keep-conference feature
under ephones/voice pools is ignored. If theinitiator disconnects from conference,
other parties will remain connected through CFB as long as the number of
participants is more than 2. - Also, max-conferences option
under telephony-service is ignored. The max number of conferences depends
on dspfarm configuration. - Similar to software CFB, MOH
isn't suppressed from conference - Conference initiator should
have at least one dual-line or two single-line DNs to be able to start
conference. - Initiator with octo-line DN
will select an idle channel from that DN to establish new call. In case
free line isn't available, the conference can't be completed. CME won't
select idle channels from other DNs within the same phone similar to single/dual lines behavior. - CME admin can assign admin
role to any of the ephones within a conference. The conference
administrator can:
1.
Dial in to any conference directly through the conference number
2.
Use the ConfList soft key to list conference parties
3.
Remove any party from any conference
Meet-Me
- In meet-me conferences, the
creator presses MeetMe soft key before dialing the conference number. Other
meet-me conference parties only dial the conference number to join the
conference. - In case software CFB is used,
Meet-Me conferences can't be completed. - Max number of Meet-Me
conference parties is 32 for G711 codec and 16 for G729 codecs
Conference Softkeys
The
RmLstC, ConfList, Join, and Select
soft keys are not supported for software-based conferencing. Only Conf softkey is supported with software CFB.
- ConfList: Lists all parties in a
conference. For meet-me conferences, this soft key is available for the
creator only unlike Ad-hoc. Press Update to update the list of parties in the conference - Join: Joins an established call
to an Ad-hoc conference. You must first press Select to choose each connected
call that you want to join in a conference, then press Join to join the selected calls
to the conference. - RmLstC: Removes the last party
added to the conference. This soft key works for the creator only. - Select: Selects a call or
conference to join to a conference and selects a call to remove from a
conference. The creator only can remove other parties by pressing the ConfList soft key, then use the Select and Remove soft keys to remove the
appropriate parties.
Configuration Template
voice-card 0
dsp services
dspfarm
!
voice class custom-cptone CONFERENCE-LEAVE
dualtone
conference
frequency 1800
3210
cadence 1000
1000 2000 1000 3000 1000
voice class custom-cptone CONFERENCE-JOIN
dualtone
conference
frequency 1210
2800
cadence 3000
1000 2000 1000 1000 1000
!
sccp local
sccp ccm identifier version 7.0
sccp
!
sccp ccm group
description
HARDWARE CONFERENCE BRIDGE
bind interface
associate ccm
priority
associate
profile register
switchback
method graceful
!
dspfarm profile conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum
conference-participants
maximum
sessions
conference-join
custom-cptone CONFERENCE-JOIN
conference-leave custom-cptone
CONFERENCE-LEAVE
associate
application SCCP
!
telephony-service
conference
hardware
!!!!... Define the
digits that can be used during conference to mute/unmute. You can mute and
unmute your phone using the phone's mute button also. You must unmute the phone
in the same way that you muted it, either with the keypad or the mute button.
sdspfarm
conference mute-on mute-off
sdspfarm units
sdspfarm tag
max-ephones 10
max-dn 20
ip
source-address port 2000
transfer-system
full-consult
!
ephone-dn
dn-tag [dual-line | octo-line]
number number
[secondary number] [no-reg [both | primary]]
conference
{ad-hoc | meetme}
no huntstop
[channel]
!
ephone-template
conference
add-mode [creator]!!!... Enable adding parties to conference.
By default both creator and participants can add parties to conference.
conference
drop-mode [creator | local]!!!... Default is conference never drops.
You can tune it to drop conference when creator drops or when the last local
participants drops.
conference
admin!!!... Configures the ephone as the
conference administrator
CME support
transcoding between G711 and G729 for the following features:
3-way Ad-Hoc software
conferencing
Call-Forward or Call-Transfer
(one leg of a hairpin call is using G711 while the other is using G729).
MoH (G711 MoH stream is
transcoded to G729 phone)
CUE (G729 calls forwarded to
CUE which is using G711 only should be transcoded)
Restrictions
Meet-me conferencing not
supported
Multiple-party ad-hoc
conferencing not supported
By default codec
configuration on SCCP phones will be used for internal calls within CME. Also,
it will always be used in best effort mode trying to achieve minimum bandwidth
and DSPs utilization. For external calls crossing trunks, codec configuration won't
be used at all. External codec will be used by SCCP phones. This has been
changed by introducing dspfarm-assist
keyword which will force codec configuration to be used in internal and
external negotiations.
If the codec g729r8 dspfarm-assist command is
configured for a SCCP phone and a DSP resource is not available when needed for
transcoding, a phone registered to the local Cisco Unified CME router will use
G.711 instead of G.729r8. This is not true for nonSCCP
call legs; if DSP resources are not available for the transcoding required for
a conference, for example, the conference is not created.
