Have phones 2001,2002,2003 in SRST mode in a router. Calls coming into the system from PSTn should hit 2001 first and then on 2003.
alias 1 6175222001 to 2001 preference 1 cfw 2003 timeou 4 huntstop
alias 2 6175222002 to 2001 preference 1 cfw 2003 timeout 4 huntstop
alias 3 6175222003 to 2001 preference 1 cfw 2003 timeout 4 huntstop
Dialplan pattern doesnt have any effect in the called number when alias is configured. So you have to configure full 10 digits in the alias command. Alias
The CCM server name that you give under System > Server should correspond to the actual name of the server used during install. You cannot change this later.
In addition to supporting message buttons for retrieving personal messages, Cisco SRST allows the automatic forwarding of calls to busy and unanswered numbers to voice-mail systems. Voice-mail systems with BRI or PRI access can log in to the calling phone's mailbox directly. For this to happen, some Cisco CallManager configuration is recommended. If your voice-mail system supports Redirected Dialed Number Identification Service (RDNIS), RDNIS must be included in the outgoing SETUP message to Cisco CallManager to declare the last redirected number and the originally dialed number to and from configured devices and applications.
Step 1 
From any page in Cisco CallManager, click Device and Gateway.
Step 2 From the Find and List Gateways page, click Find.
Step 3 From the Find and List Gateways page, choose a device name.
Step 4 From the Gateway Configuration page, check Redirecting Number IE Delivery – Outgoing.
call-manager-fallback
voicemail 92145425600 —- when somebody hits message button it calls main site's voice mail pilot number
call-forward busy 92145425600 — for calls that are busy, redirect to main site's voice mail pilot number
call-forward noan 92145425600 timeout 15 — for calls that are not answered, redirect to main site's voice mail pilot number.
call-manager-fallback
cor incoming <corlist> <list no> <list of numbers> ### applies corlist to outgoing calls from the IP phone to the router (SRST)
cor outgoing <corlist> <list no> <list of numbers> ### applies corlist to incoming calls to the IP phones from router (SRST)
List number can be from 1-20.
List of numbers should be separated by a dash and space to specify a range of phone extensions.
ex: cor incoming test 1 3001 – 3002
Applies cor list "test" and creates list 1 applied to numbers 3001 to 3002.
Default cor list
call-manager-fallback
cor incoming test default
This command applies cor-list Test to all incoming calls (w.r.t SRST router).
call-manager-fallback
no huntstop
alias 1 9725425979 to 5001
alias 2 9725425979 to 5002
alias 3 9725425979 to 5003
pickup 9725425979
This way when a call comes to 9275425979, it will randomly route calls to one of the three numbers 5001,5002,5003. Using pickup softkey any phone can pick up the call ringning on the phones
alias command also has a preference parameter. This is mostly used if you are entering specific matches rather than wild cards in alias command. For example, if you want to use an alias for IP Phone 4000 to send calls to 5000 if the phone is not registered, then make sure that you give a higher preference in alias as opposed to what is entered in the max-dn command
call-manager-fallback
max-dn 10 dual-line preference 1
alias 1 4000 to 5000 preference 5
Alias and Huntstop
The alias huntstop keyword is relevant only if you have also set the global no huntstop command under call-manager-fallback.
The alias huntstop keyword allows you to turn huntstop behavior back on for an individual alias, if huntstop is turned off globally by the no huntstop command. Setting the huntstop keyword on an individual alias stops hunting at the alias, making the alias the final member of the hunt sequence.
SRST call-forwarding for busy and no answer can be set system wide
call-manager-fallback
call-forward busy <number>
call-forward noan <number> timeout 10
SRST vs CCM Callforwarding
Call-forward-all settings under regular Callmanager mode, is not retained by the phones when they go into SRST mode.
Call forward number transformations
You can also transform the extension number for call forwarding by using wild cards.
call-forward busy 40..
If a user calls 5010 and the user is busy, then the call will be forwarded to 4010.
Call forwarding on individual lines
alias 1 1001 to 1001 preference 1 cfw 2001 timeout 20
In this example, you have created a second dial peer for 1001 to route calls to 1001, but that has preference 1 and call forwarding to 2001. Because the preference on the dial peer created by the alias command is now a lower numeric value than the preference that the dial peer first created, all calls come initially to the dial peer created by the alias command. In that way they are subject to the forward as set by the alias command, instead of any call forwarding that may have been set globally.
This feature can be used to create a basic hunt group setup using Callforwarding logic.
To create an IDle URL, CIPPS needs to be installed.
It requires Microsoft Outlook installed on the server with CDO (Colloboration Data Object)
May require a reboot.
and it never worked!
Refer 3.3.3 SR4a
a. Automated Alternate Routing Enable – False by default, to enable set it to True
b. Peer Mode Enable Code —? Entarapeeee – This enables the code to enable Peer mode Database. Default is 0000, Maximum length – 4 — Check this out!!!!
c. Sync between ADP and Phone config – Default is True. If set to false Cm will not update ADP when updating speed dials, ip phone services, directory numbers etc.
d. Max no of device level trace – default is 12, max is 256, min is 0
e. URL Help – help hyperlink in the gateway web pages. (/help”>http:///help). Once you add a gateway like 6608, you should see a help hyperlink which takes you to product specific help.
f. Dependency records is set to False (disabled) in 3.3.3 sr4. Its enabled in 3.3.3
g. Show Ring settings ——- This will enable whether user can change ring settings or not. By default its set to False. If set to true, user can change ring settings from user webpage (http://cmserver/ccmuser)
Ring settings mean, how to ring when phone is idle and when phone is in use.
ex: ring normally, ring once, flash only, do nothing.
h. CDR parameters (6 parameters) – study these when you study CDR. ***REMEMBER***
i. Default Network Locale and Default User Locale – If a phone or device pool is not configured with a locale, it chooses what ever locale that is configured here.
j. URL Authenticate, URL Directories, URL Idle, URL Idle Time (sec), URL Information, URL Messages
Find out what is URL Authenticate ********REMEMBER*********
URL Directories – for showing corporate directory on phone. Remember to change all Enterprise paramater server names to ip address.
Make sure all users in CD have telephone field defined. Otherwise people wont be able to dial through a search in CD.
URL IDLe is used to display an image like logo of the company. Use IP Phone productivity services to perform this task. ********REMEMBER TO DO TASK**********
URL Idle timer = 0 (disables IDL url feature)
URL Information – press I page will invoke this web page.
URL Services – When services button is pressed, this url is invoked.
IP Phone Proxy address – To access all the above URLS a proxy can be specified.
k. Enable All user search – This if set to true, will allow blind search from Corporate directory in phone. Default is true.
l. User search limit. This is the number of users that are returned for a blind search from CD. Default is 64. Maximum is 500. (minimum is 1)
a. Configure AAR groups, prefixes as required
b. Enable AAR service parameter (set it to true)
c. Apply AAR groups to lines, gateways, voice mail ports etc
d. Apply locations to phones, gateways, voice mail ports
e. Apply AAR CSS to phones and gateways
f. Make sure that you have appropriately set the external phone number mask. In AAR group prefix settings use only 91 or 9 depending upon whether its a Local or LD.
This assumes that Locations, Route pattern and a CSS for use by AAR, have already been defined
Create an AAR group name
a. Prefix digits with in “AAR Group” — This defines prefix used for calls with in the same AAR group. For ex. two sites in the same metroplex, when calls route via AAR, the above prefix is used.
b. Prefix between AAR groups — This defines prefix used when calls go from one AAR group to another.
*** Note *** Remember to add a 9 as trunk access code.
Remember to add a 1 for long distance
Add the group under line configuration of a phone.
Add AAR CSS under phone configuration. This CSS defines access restrictions to route patterns that will be used to dial out via PSTN.
Create an AAR group name
a. Prefix digits with in “AAR Group” — This defines prefix used for calls with in the same AAR group. For ex. two sites in the same metroplex, when calls route via AAR, the above prefix is used.
b. Prefix between AAR groups — This defines prefix used when calls go from one AAR group to another.
*** Note *** Remember to add a 9 as trunk access code.
Remember to add a 1 for long distance
Add the group under line configuration of a phone.
Add AAR CSS under phone configuration. This CSS defines access restrictions to route patterns that will be used to dial out via PSTN. You don’t need to create a new CSS for AAR. You can reuse the least restrictive CSS that is already available.
User locale – language display on the phone.
Network locale – Tones and cadence according to country.
Install network locale file to support tones and cadence from different countries.
—-
When studying MGCP configuration, check out MGCP tone download feature.
“ccm-manager download-tones” command.
Refer : System -> Device pool
If a CSS for Auto reg. is specified, and a phone auto registers with the callmanager, this CSS will be applied to the phone’s CSS. Line’s CSS is not affected.
SRST reference has 3 fields (In CM3.3)
IP address, SRST reference name and TCP port which should match with what is configured on the router.
In CM 4.1 two more field were added:
a. SRST Secure – check box. If checked will use SRST in secure mode (encryption)
b. SRST certificate provider port – Default is 2445.
Region — > if gsm codec is used in regions, phones will default to g729.
if u use g723 as codec, phones (7960, 7940) wont work, you will get a reorder tone.
System – > Region
Supported Codecs – > g711, g723, g729, GSM, Wideband.
Calculated with 30 ms sampling rate for voice packets.
g711 – 80 kbps
g723 – 24 kbps – Used with 30VIP and 12 SP+
g729 – 24 kbps
wideband – 272 kbps ( Press i twice on phone, Rx type and Tx type show up as Link16K)
gsm – 29 kbps
—————-Video codecs—————–
a. g722 – 80kbps codec
b. g728 – 16kbps LBR codec
The video call bandwidth if set to None, will not allow allocation of bandwidth for video between this region and other region.
The video bandwidth if specified (say 384 kbps) is the maximum bandwidth allowed for 1 video call (mix of audio and video bandwidth) between this region and specified region. The audio stream in the video uses g722 (80 kbps) or g728 (16kbps). The video stream may use the rest of the bandwidth.
——————–Region versus Location settings—————————-
Region setting defines per call bandwidth setting between two regions.
Location setting define total bandwidth available between two locations.
Location bandwidth >= Region bandwidth between two sites.
System – > Device Defaults
Loads, Device pools and Phone Button Templates (configurable in this page) are used only when there is no equivalent parameter specified under each device’s (phone) configuration page.
These settings will apply only for auto registered devices. When you manually add a phone, Device pool and Phone button template are required items (*).
Refer : System – > Cisco Callmanager Page
Ethernet Phone Port – Default is 2000. We changed to 4000. Phones started using TCP port 4000.
Digital Port – Default is 2001. This is used by DT-24+ cards
Analog Port – Default is 2002 . This is used by 6624-FXS card for Catalyst 6509.
MGCP Listen port – Default is 2427. This can be changed to any port.Make sure to configure the same port in the MGCP gateway ( router config).
MGCP keep-alive port – Default is 2428.
All these ports can be configured from the range 1024 – 49151.
Dependency records show cross relationships between various services / components with in Callmanager. Its enabled by default in CM 3.3.3. Its not enabled by default in 4.x
CCIE Voice notes 