IOS Conference Resources

voice-card 0     //enables motherboard DSP chips
dspfarm  //allows for DSPs for PRI card and VWIC interfaces
dsp services dspfarm   //allows transcoding and conferencing

sccp local l0   //binds SCCP process to interface

sccp ccm 10.1.1.1 identifier 1 priority 1 version 7.0+
sccp ccm 10.1.1.2 identifier 2 priority 2 version 7.0+

dspfarm profile 1 conference
associate application sccp
maximum conference-participants
maximum sessions
dspfarm conference


dspfarm profile 2 transcode
associate application sccp
codec g711ulaw
codec g729r8
maximum sessions


dspfarm profile 3 mtp
associate application sccp
codec pass-through
maximum sessions software 5000

sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register IOS-BR2-CFB1
associate profile 2 register IOS-BR2-XCODE1
associate profile 3 register IOS-BR2-MTP1

• Same process for CME except sccp ccm bind commands point to CME ip source address

Cisco Call Control Discovery & Service Advertisement Framework

Cisco SAF ( Service Advertisement Framework) - works on IOS routers and switches.
Cisco CCD (Call Control Discovery) - publishes/receives DNs, routes and how to get there and routing rules etc.  CCD runs on top of SAF

SAF-FP is the forwarder protocol and is used on SAF forwarders and advertises the info about services over IP.  SAF can be ran on any IP routing protocol, EIGRP, OSPF, BGP etc

• It uses EIGRP transport layer for service advertisement (does not rely on EIGRP for routing)
• Uses IP protocol 88, link local multicast, uses split horizon and DUAL to prevent service advertisement loops

SAF Clients can publish their services and when the SAF forwarder detects the SAF client has been removed/disconnected - it will stop sending out its services

SAF configuration for voice gateway router to be SAF Forwarder

router eigrp SAF

service-family ipv4 autonomous-system 8    //AS number needs to match throughout

sf-interface loopback 0   //if not defined, by default it will be across all interfaces

no split-horizon

topology base

service-routing xmcp listen       //Default XMCP listening port is 4788 (CM setting needs to point to 4788 too)

client username hqcucm password 0 cisco123

domain 8 default

client username sbcucm password 0 cisco123

domain 8 default

client unauthenticated     //this is for clients that do not authenticate/ not possible to authenticate -eg CME router//

domain 8 default

Show command:    show service-routing xmcp server 

Configuration to enable CCD over SAF

voice service saf

channel 1 vrouter SAF asystem 8

subscriber callcontrol wildcarded

show voice saf dnDb all    - shows the DN pattern that have been learned.  

Learned DNs can also be pulled by going to Call Manager

For cluster running on CM and will be a SAF Client, just CM needs to be configured as SAF client, no config needed for voice gateway

SAF configuration for SAF Client on CME. 

router eigrp SAF

service-family ipv4 autonomous-system 8

sf-interface loopback 0

no split-horizon

exit-sf-interface

topology base

exit-sf-topology

exit-service-family

CCD Configuration for CME 

voice service saf

profile trunk-route 1

session protocol sip interface vlan 31 transport tcp port 5060

profile dn-block 1

pattern 1 type extension 3xxx    //sets the patterns that should be published to the forwarder

profile callcontrol 1

dn-service

trunk-route 1

dn-block 1       //call control profile - combines the trunk route and dn block to send out CCD advertisement

channel 1 vrouter SAF asystem 8      //this allows learning/listening of patterns from the SAF forwarder

subscriber callcontrol wildcarded

publish callcontrol 1

show eigrp service-family ipv4 8 neighbors - shows neighbor relations that are running SAF

SAF Client Configuration on Call Manager

Go to Advanced Features>SAF>SAF Security Profile >

Enter descriptive name and username - set to username and password specified earlier on the IOS SAF config

Then go to Advanced Features>SAF>SAF Forwarder - set client label to match the "external-client" configuration in EIGRP and select the SAF Security Profile that was just created earlier and enter the IP address of the SAF Forwarder 

For the SAF Forwarder Port - enter in the port 4788 which is the default for XMCP

Expand the "show advanced settings" and select the Call Manager that should be participating in.  This will connect the CM to the SAF forwarder.  

Call Control Discovery Set up on Call Manager 

Set up SIP trunk: Type SIP Trunk/ SIP/Call Control Discovery 

Make sure for inbound CSS - it is able to reach internal DNs

Set SIP trunk security profile and SIP profile to standard

Set up Hosted DN Group Info > Call Routing> Call Control Discovery> Hosted DN Group

Then create hosted pattern

Call Routing>CCD>Hosted Pattern - add new enter in pattern you wish to publish e.g. 1xxx and map to Hosted Dn Group

Then set up Advertising Service

Call Routing>Call Control Discovery> Advertising Service  and select the CCD trunk created earlier and also select the the Hosted DN Group as well - set feature to activated

Then set up CCD partition 

Call routing>CCD>Partition - Create a named CCD partition 

Then set up requesting service

Call routing>Call control discovery - Requesting service Info 

Enter name and route partition that you created earlier - the route partition is where the learned DNs will be placed into

Extension Mobility Cross Cluster

Extension Mobility Cross Cluster (EMCC) allows for your device profile/ extension from your home cluster to login to a phone in a different visiting cluster.

EMCC allows user in the visiting cluster to make calls using the gateway of the visiting cluster if the route list of the matched route pattern in home cluster matches the local resource group.

Call Routing of the user in visiting cluster is made by the home cluster.  e.g. call will be sent out via the home cluster even if EMCC user is logged into visiting cluster

EMCC builds the concatenated CSS based from

• adjunct CSS ( in roaming device pool)
• line CSS of EM device profile
• EM CSS on the device profile of the phone

EMCC Prep

• Activate Services
• Extension Mobility and Bulk Provisioning Service 
• Create Extension Mobility Cross Cluster IP Phone Server
• Service URL:http://10.10.13.11:8080/emapp/EMAppServlet?device=#DEVICENAME#&EMCC=#EMCC#
• Subscribe phone to EMCC service or enable as an enterprise parameter
• Check enable extension mobility on the device 
• Create Device Profile for EMCC user
• Associate end user to the device profile and check enable EMCC
• Get all certificates exported from CM pub in cluster A and B
• Export certificates to SFTP server via OS Administration >Security> Bulk Certificate Management > set SFTP address and credentials
• Consolidate all certificates
• Import All certificates to clusters
• Create dummy database entries for EMCC templates.  Dummy entries limit the number of concurrent EMCC logins
• BAT>EMCC>EMCC Template
• In template, enter in the device pool and template name and save
• BAT>EMCC>Insert/Update EMCC - set to update EMCC devices and select EMCC template that was just created and run immediately.  
• Go to Insert/Update EMCC again and set to Insert EMCC devices and set # of dummy EMCC devices and set to run immediately
• Create Geolocation Filter
• System>Geolocation Filter and checkmark geolocation criterias e.g. country, state, City or Township
• Configure EMCC parameters
• System Enterprise parameters
• Set Cluster ID
• set TFTP server for EMCC login device, EMCC Geolocation Filter, default server for remove cluster update 
• Set up EMCC SIP Trunk
• Device>Trunk>Add Trunk
• set to SIP trunk type, SIP device protocol and EM Cross Cluster trunk service type 
• Enter in trunk name, no need for destination address.  Set SIP trunk security profile and SIP profile
• Set EMCC Remote Clusters
• Advanced Features>EMCC>EMCC Remote Cluster>Add new
• set cluster ID of remote cluster to its cluster ID and save
• enable EMCC service, save and update remote cluster now

VGW- All about Translations

Translation Rules

• Voice translation rule is to be created first
• Second step to create a translation profile and associate the voice translation rule to the profile
• Apply the translation profile to a dial peer or voice port

In global config, enter voice translation-rule x where x is the tag identifier

You can have 15 rules under a voice translation rule prior to IOS 15.1.4M.  After 15.1.4M, you can have up to 100 rules under a voice translation rule.


The translation rule takes the number slice and substitutes it into the translated number slice grouping e.g.rule 1 /1234/ /4321/ translates 1234 into 4321.  There are different voice translation rule operators, also known as regular expression.

• ^ - matches the expression at start of the line e..g 
• $ - matches expression at end of the line
• / - marks start and end of the matching and replacing set
• \ - Escape - removes meaning of the next character
• -  - hyphen is used to indicate a range e.g. 1-9  1 and all the digits through 9.  
• [1456] - only matches one of the single characters in the brackets mentioned in the list e..g 1 or 4 or 5 or 6
• [^1456] - cannot match any of the single digits in the list .e.g cannot match 1 or 4 or 5 or 6
• .   - matches any single character.  
• * - repeats the previous regular expression ZERO or more times e.g.   /^818\(.*\)/     /515\1/  
• If 8183334880 is dialed, it will take the dialed number and replace 3334880 instead of the 1 group number slice because "." matches any number and the * will match "." an infinite number of times so it could even be dialed number 81811111111111 and it would be translated to 51511111111111.  Basically allows for anything, including null e.g. if no match, it will still translate 
• + - repeats the previous expression ONCE or more times e.g. basically same as * operator with exception of null. 
• ? - Repeats the previous regular express zero or one occurrence of the previous expression ( Use control + v key and then press ? key to enter this in) e.g. /^818\(.?\)/ /515\1/ when 8183334880 is dialed, it will translate it to 51533
• ( ) - groups digits into sets 
• & - Brings all matched digits into the replacement string

Translation Profiles

**Place holder**

CUCM - CallBack

     The Cisco Call Back feature allows you to receive call back notification on your IP phone when a called party line becomes available. 

1. IP phone user A calls IP phone user B in the same cluster. 
2. If IP phone B is busy or there is no answer, IP phone user A activates the Cisco Call Back features through the CallBack softkey. 
1. The CallBack softkey can be activated while the call is ringing (RING OUT state). Pressing the CallBack softkey while the call is ringing cancels the call. It does NOT work if the call gets sent to voicemail.
2. The CallBack softkey can be  activated after the call has ended (ON HOOK state).
3. The CallBack softkey can be activated during a consultation transfer (CONNECTED TRANSFER state).
3. IP phone B becomes available. Phone B either went ON HOOK after their call ended. Or they got back to their desk and picked up their phone (OFF HOOK state).
4. IP phone A receives an audible alert and visual notification that the DN is available. Since Cisco CallManager remembers the dialed number, IP phone user A can then press the Dial softkey to reach IP phone user B.

CallBack Configuration Requirements:

1. Enable Cisco Extended Functions (CEF) on all servers via Serviceability
2. Add the "CallBack" softkey to the ON HOOK, RING OUT, and CONNECTED TRANSFER states.
3. Apply the CallBack softkey templates to the phones.
4. (Optional) Review and modify the CallBack Service Parameters as needed
5. Test CallBack!

Callback FAQ
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_qanda_item09186a0080191057.shtml#intro









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VGW - MGCP the Easy Way

     Media Gateway Control Protocol (MGCP) is a is a plain-text (sent over UDP on port 2427) protocol used by call-control devices (like Communications Manager), known as media gateway controllers (MGCs) or call agents (CAs), to take control of a specific port on a gateway. Cisco Communications Manager knows and controls the state of each individual port on the gateway.

     If you want the fastest and automated way of configuration an MGCP gateway, you can utilize the "ccm-manager config server" CLI command. Having this automated MGCP configuration does not allow for partial/fractional T1 PRI's, but what you can do is enable the automation, modify the T1 manually, and disable the automation after all of the commands have been populated. Here's the steps in setting up an MGCP gateway:

1. On the gateway, set the "ip domain name" and "hostname"(eg. BR1.company.com)
1. This provides the full domain name which MGCP utilizes to register the gateway
2. Configure the gateway on Cisco Communications Manager
1.  CM Administration > Device > Gateway
2. Add New
3. Select specific Gateway Type (eg. Cisco 3945)
4. Select MGCP as the protocol
5. Set the full domain name of the router and CUCM Group
6. Set the appropriate modules 
7. Configure each of the endpoints/circuits
3. From the Gateway's CLI configure MGCP
1. mgcp bind control source-interface <voice interface or loopback>
2. mgcp bind media source-interface <voice interface or loopback>
3. ccm-manager config server <CUCM IP Address Primary> <CUCM IP Address Secondary>
4. ccm-manager config
5. ccm-manager mgcp

 MGCP Messages:

Command	Message Name	Sent By	Description
AUEP	AuditEndpoint	CallManager	Determines the status of a given endpoint.
AUCX	AuditConnection	CallManager	Retrieves all the parameters associated with a connection.
CRCX	CreateConnection	CallManager	Creates a connection between two endpoints.
DLCX	DeleteConnection	Both	From CallManager: Terminates a current connection.
			From Gateway: Indicates that a connection can no longer be sustained.
MDCX	ModifyConnection	CallManager	Changes the parameters associated with an established connection.
RQNT	NotificationRequest	CallManager	Instructs the gateway to watch for special events such as hooks or DTMF tones. It is also used to instruct the gateway to provide a signal to the endpoint (for example, dial tone and busy tone).
NTFY	Notify	Gateway	Informs the Cisco CallManager when requested events occur.
RSIP	RestartInProgress	Gateway	Informs the Cisco CallManager that an endpoint or group of endpoints are taken out or placed back into service.

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CUCM - Service Parameters

     Service Parameters are global settings on multiple CUCM services (CallManager, IP Voice Media, Extension Mobility, etc.) that may apply to all servers (clusterwide parameters) or to only specific nodes (server parameters).

     Keep in mind that for Service Parameters, there is an "Advanced" button at the top of the page which presents additional Service Parameter options!


Service Parameters

Parameter Name	Suggested Value	Parameter Value	Description
CDR Enabled Flag	True	False	Enables Call Detail Records
CDR Log Calls with Zero Duration Flag	False	False	Enables CDR for calls that never connected or that lasted for < 1 sec
Station KeepAlive Interval	30	30	Interval between KeepAlives from IP phones to CUCM
Station and Backup Server KeepAlive Interval	60	60	KeepAlive interval between phone and secondary CUCM
T301 Timer (Alerting Timeout)	180000	180000	How long CUCM waits to receive the Alerting message
T302 Timer (Interdigit Timeout)	10000	15000	Timer after each digit is pressed before pushing the call through
Strip + on Outbound Calls	False	False	CUCM strip + on outbound for MGCP GWs and SIP Trunks. H.323 ALWAYS strips the + sign.
Built-in Bridge Enable	On	Off	Enables builtin bridge to save on conferencing resources. G.711 supported only.
Device Mobility Mode	On	Off	Allows roaming phones to dynamically associate with the local device pool
Display Device Mobility Location During Phone Registration	True	True	Displays the Device Mobility location when booting up
Off-hook to First Digit Timer	15000	15000	Time CUCM waits for the first digit to be dialed
Privacy Setting	True	True	Enables (True) or disables (False) the Privacy feature
Allow Peer to Preserve H.323 Calls	True	False	Allows H.323 GW to preserve an active call during a loss of connection to CUCM
SIP Min-SE Value	1800	1800	Minimum Session Expiration timer for SIP
SIP Session Expires Timer	1800	1800	Interval between session refresh requests
Call Park Display Timer	15	10	Time that Call Park number is displayed
Call Park Reversion Timer	60	60	Call reverts back to original phone if not answered
Party Entrance Tone	False	True	Determines if a tone will play when a party joins or exits a call with more than two parties (barge, cbarge, conference, join, meetme)
Block Offnet to Offnet Transfer	False	False	Blocks the ability to transfer an offnet call to another offnet number
Suppress MOH to Conference Bridge	True	True	Don’t play MOH on conference bridges
Drop Ad Hoc Conference	Never	Never	Ad Hoc Conference bridges stay up as long as two users are on the conference
Maximum Ad Hoc Conference	16	4	Maximum Ad Hoc Conference Participants
Maximum MeetMe Conference Unicast	16	4	Maximum MeetMe Conference Participants
Advanced Ad Hoc Conference Enabled	True	False	Conference Chaining and control Enabled
Forward Maximum Hop Count	8	12	Maximum times a single call can be forwarded or diverted. For Q.SIG calls, the maximum value is 15.
Forward No Answer Timer	16	12	The Global FNA timer setting
Max Forward Hops to DN	8	12	Max number of forwards allowed for a DN. Prevents call routing loops within CUCM
G.722 Codec Enabled	Enabled	Enabled	Enabled G.722 advertisement
Automated Alternate Routing Enable	False	False	Enables CAC Automated Alternate Routing
Single Button Barge/CBarge Policy	CBarge	Off	Sets the Single Button Barge feature to use barge, CBarge or disable
Allow Barging When Ringing	False	False	Allows users to barge before the phone is answered
Stop Routing on Out of Bandwidth Flag	False	False	Call routing behavior for calls through trunks when not enough bandwidth is available. Used for subsequent route lists and AAR/Locations.

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