$ means ”end of digits” and is used for a more specific match. If you have two dial-peers with the same pattern and one with $ and the other without $, then call routing logic will always match the one with $, because that is more specific. Even if you give a lower preference to the dial-peer with $, still it will be matched because by default the dial-plan hunting algorithm is more specific match first and then preference.

dial-peer voice 10 pots

destination-pattern 1000

port 1/0/0

dial-peer voice 11 pots

destination-pattern 1000$

port 1/0/0

preference 5

In this case, always dial-peer 11 will be used to route calls to 1000, even though it has a lower preference. 

Unity Admin > Configuration > Settings

Replicate Cisco Unity Directory Objects

Unity Admin > Configuration > Settings

Schedule options: Use 24-hour time format for conversation and schedules

Default: 12-hour format

Unity Admin > Configuration > Settings

Address messages to other subscribers:  Enable spelled name search

Default: Enabled

  Click on link to enlarge the pic.

To find out which channels are in service, which are out of service, use command,

sh isdn service. Here is a sample output from a Cisco IOS router.

 BR1#sh isdn service
PRI Channel Statistics:

%Q.931 is backhauled to CCM MANAGER 0×0003 on DSL 0. Layer 3 output may not appl
y

ISDN Se0/0/1:23, Channel [1-24]
  Configured Isdn Interface (dsl) 0
   Channel State (0=Idle 1=Proposed 2=Busy 3=Reserved 4=Restart 5=Maint_Pend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3 3
   Service State (0=Inservice 1=Maint 2=Outofservice 8=MaintPend 9=OOSPend)
    Channel :  1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4
    State   :  0 0 0 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2 2
BR1#

1. Create a file by the name hideuser.ldif (can be any name) in Notepad

2. Lets take the example of hiding ac user for Attendant Console. Put the following contents in the file and save it.

dn: cn=ac, ou=users, o=cisco.com
changeType:modify
replace:Description
Description:CiscoPrivateUser

3. Go to command line and run the following command.

ldapmodify -h <server name> -p 8404 -D “cn=Directory
Manager,o=cisco.com” -w <DCDAdmin Password> -c -f hideuser.ldif

Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_tel_num_trans:
   Mail From=OFax@iptel.com, Phone Number In=FAX=1, Phone Number Dial=
Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_tel_num_trans:
   Incoming Dial-peer(Tag=202, Matched Digits=10, Destination Pattern=12742551004)
Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_match_digits:
   Outgoing Peer(Tag=201, Destination Pattern=1274255100., Strip Enable=TRUE)
Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_tel_num_trans:
   Incoming Dial-peer(Tag=202), Outgoing Dial-peer(Tag=201), Cover Page=FALSE
Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_new_rcpt:
   Envelope To=FAX=12742551004@pstnsw.gateway.com, Length=34,
   MMccb(Incoming Dial-peer=202, Outgoing Dial-peer=201,
   Telephone Number Dial=12742551004, Sub Address=, Cover Page=FALSE
Sep 17 16:49:09.887: //-1/xxxxxxxxxxxx/MSPI_OFF/mspi_offramp_new_rcpt:exit@1127
Sep 17 16:49:10.907: //-1/46E6CB8B80DA/MSPI_OFF/mspi_get_ani:
   in_str=”Outgoing Fax” <OFax@iptel.com>
Sep 17 16:49:10.907: //-1/46E6CB8B80DA/MSPI_OFF/mspi_get_ani:
   in_str=OFax@iptel.com
Sep 17 16:49:10.907: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_rfc822_header:
   Envelope From=OFax@iptel.com, Size=15
   From Personal Name=Outgoing Fax
   From=”Outgoing Fax” <OFax@iptel.com>
   To=<FAX=12742551004@pstnsw.gateway.com>
    Notification Type=0
Sep 17 16:49:10.907: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_rfc822_header:
   MMccb(Called Number=12742551004, Calling Number=, Incoming Dial-peer=202)
Sep 17 16:49:10.911: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.911: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.923: ISDN Se1/0/0:23 Q931: pak_private_number: Invalid type/plan 0×0 0×1 may be overriden; sw-type 13
Sep 17 16:49:10.923: ISDN Se1/0/0:23 Q931: Applying typeplan for sw-type 0xD is 0×2 0×1, Called num 2742551004
Sep 17 16:49:10.927: ISDN Se1/0/0:23 Q931: TX -> SETUP pd = 8  callref = 0×0093
        Bearer Capability i = 0×8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Calling Party Number i = 0×0081, N/A
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0xA1, ‘2742551004′
                Plan:ISDN, Type:National
Sep 17 16:49:10.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.931: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.939: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.939: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.943: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.943: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.947: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.947: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.951: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONNECTING)
Sep 17 16:49:10.951: ISDN Se1/0/0:23 Q931: RX <- CALL_PROC pd = 8  callref = 0×8093
        Channel ID i = 0xA98381
                Exclusive, Channel 1
Sep 17 16:49:10.955: ISDN Se1/0/0:23 Q931: RX <- ALERTING pd = 8  callref = 0×8093
        Progress Ind i = 0×8088 – In-band info or appropriate now available
Sep 17 16:49:10.959: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   MIME Type=12, Buffer Length=1536
Sep 17 16:49:10.959: //163/46E6CB8B80DA/MSPI_OFF/moff_save_buffer:
   Disabled Receiving!
Sep 17 16:49:21.123: ISDN Se1/0/0:23 Q931: RX <- CONNECT pd = 8  callref = 0×8093
        Display i = ‘hqph2′
Sep 17 16:49:21.127: %ISDN-6-CONNECT: Interface Serial1/0/0:0 is now connected to 2742551004 N/A
Sep 17 16:49:21.127: ISDN Se1/0/0:23 Q931: TX -> CONNECT_ACK pd = 8  callref = 0×0093
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI/mspi_bridge:
   MMccb(State=CONNECTING, Type=Offramp), Destination Call Id=166
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:28.931: //163/46E6CB8B80DA/MSPI/mspi_bridge:
   Offramp Rcpt Enabled
Sep 17 16:49:40.963: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:40.963: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:40.967: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:40.967: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:40.979: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:40.979: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:40.983: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:40.983: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:40.999: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:40.999: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.007: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.007: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.011: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.011: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.023: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.023: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.027: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.027: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.039: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.039: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.043: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.043: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.055: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.055: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.059: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.059: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.071: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.071: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.083: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.083: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.087: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.087: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.099: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.099: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.099: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.099: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.111: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=1536, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.111: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.111: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_put_buffer:
   Mime Type=12, Buffer Length=200, MMccb(Info Type=10, State=CONFERENCED)
Sep 17 16:49:41.111: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   Mime Type=12
Sep 17 16:49:41.111: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_send_buffer:
   BUFF_LAST
Sep 17 16:49:51.079: %CDP-4-DUPLEX_MISMATCH: duplex mismatch discovered on FastEthernet0/1 (not full duplex), with SCA041201H1 4/21 (full duplex).
Sep 17 16:50:17.491: //163/46E6CB8B80DA/MSPI/mspi_bridge_drop:
   MMccb(State=DESTROYING, Type=Offramp), Destination Call Id=166
Sep 17 16:50:17.503: %ISDN-6-DISCONNECT: Interface Serial1/0/0:0  disconnected from 2742551004 , call lasted 56 seconds
Sep 17 16:50:17.503: ISDN Se1/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0×0093
        Cause i = 0×8290 – Normal call clearing
Sep 17 16:50:17.523: ISDN Se1/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0×8093
Sep 17 16:50:17.527: ISDN Se1/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0×0093
Sep 17 16:50:17.531: //163/46E6CB8B80DA/MSPI/mspi_call_disconnect:
   Cause Value=normal call clearing (16),
   MMccb(State=DISCONNECTING, Type=Offramp)
Sep 17 16:50:17.531: //163/46E6CB8B80DA/MSPI/mspi_offramp_call_history:
   MMccb(Disconnect Cause=normal call clearing (16) 16)
Sep 17 16:50:17.531: //163/46E6CB8B80DA/MSPI_OFF/mspi_call_get_offramp:
Sep 17 16:50:17.531: //163/46E6CB8B80DA/MSPI_OFF/mspi_call_get_offramp:exit@2527
Sep 17 16:50:17.531: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_call_history:exit@2399
Sep 17 16:50:26.567: //163/46E6CB8B80DA/MSPI_OFF/mspi_offramp_dispose_context:
   MMccb(State=DISCONNECT_DONE)

OUtput of debug fax mspi all on T37 router. (First few lines are from debug isdn q931 to show the fax call come into the t37 router)

A successful fax inbound from the router to Unity will throw the following debugs.

PSTNSw#
Sep 17 14:55:15.972: ISDN Se1/0/0:23 Q931: RX <- SETUP pd = 8  callref = 0×0009
        Bearer Capability i = 0×8090A2
                Standard = CCITT
                Transfer Capability = Speech 
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98383
                Exclusive, Channel 3
        Display i = ‘hq vg248′
        Calling Party Number i = 0×0081, ‘2742551004′
                Plan:Unknown, Type:Unknown
        Called Party Number i = 0×80, ‘16285253800′
                Plan:Unknown, Type:Unknown
Sep 17 14:55:16.016: %ISDN-6-CONNECT: Interface Serial1/0/0:2 is now connected to 2742551004 N/A
Sep 17 14:55:16.016: ISDN Se1/0/0:23 Q931: TX -> CALL_PROC pd = 8  callref = 0×8009
        Channel ID i = 0xA98383
                Exclusive, Channel 3
Sep 17 14:55:16.016: ISDN Se1/0/0:23 Q931: TX -> CONNECT pd = 8  callref = 0×8009
Sep 17 14:55:16.032: ISDN Se1/0/0:23 Q931: RX <- CONNECT_ACK pd = 8  callref = 0×0009
Sep 17 14:55:29.132: //137/5CF666368032/MSPI_ON/mspi_call_setup_request:
   Outgoing Peer Tag=204
   Envelope From=FAX=2742551004@pstnsw.gateway.com
   Envelope To=ifax@iptel.com
   Mime Outer Type=2
Sep 17 14:55:30.132: //137/5CF666368032/MSPI_ON/mspi_check_connect:
   MMccb(Count=0)
Sep 17 14:55:30.132: //137/5CF666368032/MSPI_ON/mspi_check_connect:
   SMTP Connected To The Server !
Sep 17 14:55:30.132: //137/5CF666368032/MSPI/mspi_bridge:
   MMccb(State=CONNECTED, Type=Onramp), Destination Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=0),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=1),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=2),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=3),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=4),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=5),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=6),
   Source Call Id=138
Sep 17 14:55:42.480: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=7),
   Source Call Id=138
Sep 17 14:55:42.484: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=8),
   Source Call Id=138
Sep 17 14:55:42.484: //137/5CF666368032/MSPI_ON/mspi_xmit:
   MMccb(State=CONFERENCED, Type=Onramp, Buffer Count=9),
   Source Call Id=138
Sep 17 14:55:42.484: //137/5CF666368032/MSPI_ON/mspi_onramp_xmit:
   EOD Received
Sep 17 14:55:43.752: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=10)
Sep 17 14:55:43.752: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=9)
Sep 17 14:55:43.752: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=8)
Sep 17 14:55:43.752: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=7)
Sep 17 14:55:43.764: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=6)
Sep 17 14:55:43.764: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=5)
Sep 17 14:55:43.764: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=4)
Sep 17 14:55:43.764: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=3)
Sep 17 14:55:43.768: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=2)
Sep 17 14:55:43.932: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   MMccb(Call State=CONFERENCED, Buffer Count=1)
Sep 17 14:55:43.932: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   EOD Received; Status=0
Sep 17 14:55:43.932: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   Message Id=00012006095531752@PSTNSw.iptel.com
Sep 17 14:55:43.932: //137/5CF666368032/MSPI_ON/mspi_onramp_buff_finished_callback:
   Defer Destroying
Sep 17 14:55:46.932: //137/5CF666368032/MSPI_ON/mspi_onramp_defer_disconnect:
   MMccb(Call State=DESTROYING)
Sep 17 14:55:46.932: //137/5CF666368032/MSPI_ON/mspi_onramp_defer_disconnect:
   Send Disconnect Event
Sep 17 14:55:46.932: //137/5CF666368032/MSPI/mspi_bridge_drop:
   MMccb(State=DESTROYING, Type=Onramp), Destination Call Id=138
Sep 17 14:55:46.932: //137/5CF666368032/MSPI/mspi_call_disconnect:
   Cause Value=normal call clearing (16),
   MMccb(State=DISCONNECTING, Type=Onramp)
Sep 17 14:55:46.932: //137/5CF666368032/MSPI_ON/mspi_onramp_call_history:
   MMccb(Disconnect Cause=normal call clearing (16) 16)
Sep 17 14:55:46.932: //-1/xxxxxxxxxxxx/MSPI/mspi_free_ccb:
   Total Allocated MMccbs=1, Total Inserted MMccbs=0
Sep 17 14:55:46.940: %ISDN-6-DISCONNECT: Interface Serial1/0/0:2  disconnected from 2742551004
, call lasted 30 seconds
Sep 17 14:55:46.944: ISDN Se1/0/0:23 Q931: TX -> DISCONNECT pd = 8  callref = 0×8009
        Cause i = 0×8290 – Normal call clearing
Sep 17 14:55:46.960: ISDN Se1/0/0:23 Q931: RX <- RELEASE pd = 8  callref = 0×0009
Sep 17 14:55:46.964: ISDN Se1/0/0:23 Q931: TX -> RELEASE_COMP pd = 8  callref = 0×8009

Enable DNS resolution: 

ip domain-lookup

ip name-server <Unity server>

ip domain-name gateway.com (domain of gateway)

Enable T37: 

fax interface-type fax-mail  —- (reload the router)

Enable T37 Onramp: (Think of Onramp as coming ON to the network)
mta send server unity40.iptel.com port 25  —- (mail server – unity in this case)
mta send with-subject both —– (sends ani and dnis in subject)
mta send mail-from hostname pstnsw.gateway.com  ———– (Mail from hostname)
mta send mail-from username $s$
mta send return-receipt-to hostname pstnsw.gateway.com ———— (NDR from hostname)
mta send return-receipt-to username $s$
 !

application
 service onramp flash:app_faxmail_onramp.2.0.1.3.tcl

dial-peer voice 203 pots
 service onramp
 incoming called-number 16285253800
 direct-inward-dial
!
dial-peer voice 204 mmoip
 service fax_on_vfc_onramp_app out-bound
 destination-pattern 16285253800
 information-type fax
 session target mailto:ifax@iptel.com      ——– This should be the Inbound Fax mailbox that Unity monitors.

Enable T37 Offramp: (Think of Offramp as going OFF the network)

mta receive aliases pstnsw.gateway.com ———– (receiving alias name)
mta receive maximum-recipients 10 —————- (default is 0, set this to something greater than 0)
mta receive generate permanent-error
!
application
 service offramp flash:app_faxmail_offramp.2.0.1.1.tcl

 dial-peer voice 201 pots
 destination-pattern 1274255100.
 port 1/0/0:23
 forward-digits 10
!
dial-peer voice 202 mmoip
 description Offramp inbound from Unity
 service offramp
 information-type fax
 incoming called-number 1274255100.
 dsn delayed
 dsn success
 dsn failure

For MWI to work, the unity ports and MWI on/off numbers should have a CSS that included the phone lines.

For MWI to work, the unity ports and MWI on/off numbers should have a CSS that included the phone lines.

Sep 11 00:41:59.171: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1001@10.1.21.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
Remote-Party-ID: <sip:911@10.1.11.4>;party=calling;screen=no;privacy=off
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>
Date: Mon, 11 Sep 2006 00:41:59 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Supported: 100rel,timer,resource-priority
Min-SE:  1800
Cisco-Guid: 703035978-1080365531-2182204663-2878018674
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1157935319
Contact: <sip:911@10.1.11.4:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 238

v=0
o=CiscoSystemsSIP-GW-UserAgent 9779 8954 IN IP4 10.1.11.4
s=SIP Call
c=IN IP4 10.1.11.4
t=0 0
m=audio 19270 RTP/AVP 0 101
c=IN IP4 10.1.11.4
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Sep 11 00:41:59.179: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
CSeq: 101 INVITE
Server: Brekeke OnDO SIP Server (rev.172)
Content-Length: 0

Sep 11 00:41:59.215: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Wed, 13 Sep 2006 12:47:08 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Timestamp: 1157935319
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Allow-Events: telephone-event
Remote-Party-ID: <sip:1001@10.1.21.50>;party=called;screen=no;privacy=off
Contact: <sip:1001@10.1.21.49:5060>
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Length: 0
Sep 11 00:42:00.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKBAEC
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Wed, 13 Sep 2006 12:47:08 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Timestamp: 1157935319
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
Allow-Events: telephone-event
Remote-Party-ID: <sip:1001@10.1.21.50>;party=called;screen=yes;privacy=off
Contact: <sip:1001@10.1.21.49:5060>
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Type: application/sdp
Content-Length: 223

v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 10.1.21.50
s=SIP Call
c=IN IP4 10.1.21.50
t=0 0
m=audio 24678 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Sep 11 00:42:00.571: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1001@10.1.21.49:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.11.4:5060;branch=z9hG4bKC1569
From: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
To: <sip:1001@10.1.21.49>;tag=16777329
Date: Mon, 11 Sep 2006 00:41:59 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Route: <sip:10.1.21.49:5060;lr>
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

Sep 11 00:42:03.447: %MV64340_ETHERNET-5-LATECOLLISION: FastEthernet0/1, late collision error
Sep 11 00:42:06.183: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:911@10.1.11.4:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.21.49:5060;rport;branch=z9hG4bKc06f1e40a2ac9bac.3
Via: SIP/2.0/UDP  10.1.21.50:5060;branch=z9hG4bK6134c268
From: <sip:1001@10.1.21.49>;tag=16777329
To: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
Date: Wed, 13 Sep 2006 12:47:09 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
User-Agent: Cisco-CCM4.1
Max-Forwards: 69
CSeq: 101 BYE
Record-Route: <sip:10.1.21.49:5060;lr>
Content-Length: 0

Sep 11 00:42:06.191: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.21.49:5060;rport;branch=z9hG4bKc06f1e40a2ac9bac.3,SIP/2.0/UDP  10.1.21.50:5060;branch=z9hG4bK6134c268
From: <sip:1001@10.1.21.49>;tag=16777329
To: <sip:911@10.1.11.4>;tag=2A28BE60-16CA
Date: Mon, 11 Sep 2006 00:42:06 GMT
Call-ID: 2ACF6A6D-406511DB-8214CCF7-AB8B1472@10.1.11.4
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 101 BYE
Reason: Q.850;cause=16
Content-Length: 0

How to trigger via zone behaviour?

Local zones – always enable outvia on the destination zone

Remote zones – use invia or outvia on the remote zone

Via-zone gatekeepers differ from legacy gatekeepers in how LRQ and ARQ messages are used for call routing. Using via-zone gatekeepers will maintain normal clusters and functionality. Legacy gatekeepers examine incoming LRQs based on the called number, and more specifically the dialedDigits field in the destinationInfo portion of the LRQ. Via-zone gatekeepers look at the origination point of the LRQ before looking at the called number. If an LRQ comes from a gatekeeper listed in the via-zone gatekeeper’s remote zone configurations, the gatekeeper checks to see that the zone remote configuration contains an invia or outvia keyword. If the configuration contains these keywords, the gatekeeper uses the new via-zone behavior; if not, it uses legacy behavior.

For ARQ messages, the gatekeeper determines if an outvia keyword is configured on the destination zone. If the outvia keyword is configured, and the zone named with the outvia keyword is local to the gatekeeper, the call is directed to a Cisco Multiservice IP-to-IP Gateway in that zone by returning an ACF pointing to the Cisco Multiservice IP-to-IP Gateway. If the zone named with the outvia keyword is remote, the gatekeeper sends a location request to the outvia gatekeeper rather than the remote zone gatekeeper. The invia keyword is not used in processing the ARQ.

crypto isakmp policy 1
encr 3des
authentication pre-share
group 2
lifetime 10800
crypto isakmp key cisco address 10.1.1.1
crypto isakmp key cisco address 10.1.1.2

!
!
crypto ipsec transform-set CM esp-3des esp-sha-hmac
mode transport
!
crypto map CM 1 ipsec-isakmp
set peer 10.1.1.1
set transform-set CM
match address 101
crypto map CM 2 ipsec-isakmp
set peer 10.1.1.2
set transform-set CM
match address 102

!
access-list 101 permit ip host 10.2.2.2 host 10.1.1.1

access-list 102 permit ip host 10.2.2.2 host 10.1.1.2
interface Serial0/0.101 point-to-point
ip address 10.2.2.2 255.255.255.0
frame-relay interface-dlci 101
crypto map CM

To configure a virtual template for MLP LFI on a frame-relay subinterface, you have to remove the DLCI configured under that subinterface first before reapplying it back with the ppp keyword.

int s0/0

no frame-relay interface-dlci 30

 frame-relay interface-dlci 30 ppp virtual-template 1

Apply any LLQ, IP settings, ppp configs under the virtual template interface.

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

When you park a call, Callmanager will search the available Park numbers configured on the server that is processing the calls and will also look up the partitions that are available in the CSS of the phone and will pick the first available part number that is defined in a partition that is reachable by the phone.

When you go to another phone to retrieve the parked call, Callmanager will search for the parked call by using the CSS of the device.

This can create an issue if you create same park numbers in different partitions in different servers. It is recommended to give different park number ranges for different servers.

Make sure the call pickup group numbers does not overlap with the exiting dial plan

Call Pickup features include Pickup, GPickup, and OPickup:

•Pickup allows you to answer a call that is ringing on another phone within your “group” (a collection of extensions that your system administrator defines).

•GPickup allows you to answer a call ringing on a phone in another group.

•OPickup allows you to answer a call ringing on a phone in another group that is associated with your group

show sysdb /sw/info/filesys  — will show how much disk space is being used for voicemail. It will also show disk usage in percentage

To compact the database.

# offline

database compact

For H323

debug h245 asn1

For SIP

debug voip rtp session named-event

Sample output

H323

Aug 21 21:07:37.902: h245_decode_one_pdu: more_pdus = 0, bytesLeftToDecode = 7
Aug 21 21:07:37.902: H245 MSC INCOMING ENCODE BUFFER::= 6D810447200063
Aug 21 21:07:37.902:
Aug 21 21:07:37.902: H245 MSC INCOMING PDU ::=

value MultimediaSystemControlMessage ::= indication : userInput : signal :
{
signalType “9″
duration 100
}

SIP

Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 00 00  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:04 01 90  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>
Aug 21 21:07:37.902:          Pt:101    Evt:9       Pkt:84 03 20  <Snd>>>

* Remove ip address of serial subinterface interface

* Remove DLCI of subinterface.

* Its recommended to do this configuration on the remote site first.

int s0/0.9

no ip address

frame-relay interface-dlci 900 ppp virtual-template 1

int virtual-template 1

ip add 10.1.1.1 255.255.255.0

ppp multilink

ppp multilink interleave

ppp multilink fragment-delay 10 — milliseconds.

service-policy output VATS

** You will need to disable frame relay traffic shaping at the main interface level for this to work **

Configure LLQ for voice 

policy-map BR2-HQ
class RTP
priority percent 33
class Sig
bandwidth percent 2
class class-default
fair-queue
Configure Voice activated FRTS:

policy-map VATS  ———–  Line rate is 768kbps, CIR is 384kbps. Use 95 % of CIR for shape average and shape adaptive.
class class-default
shape average 729600 3648 0 
shape adaptive 364800
shape fr-voice-adapt deactivation 50
service-policy BR2-HQ
Configure FRTS class-map and apply VATS policy to the class-map

map-class frame-relay BR2-HQ
frame-relay fragment 480  ——— configures end-to-end FRF.12 fragmentation based on CIR of 384kbps
service-policy output VATS

interface Serial0/0
no ip address
encapsulation frame-relay
no dce-terminal-timing-enable
frame-relay fragmentation voice-adaptive deactivation 50
!
interface Serial0/0.201 point-to-point
ip address 10.201.2.2 255.255.255.0
frame-relay interface-dlci 201
class BR2-HQ

1. Check the dial plan to find out if the calls between the sites are long distance or international.

2. If the calls are long distance, then you need to create only 1 AAR group with the prefix of 91. If the calls are international, then you would need to create a AAR group for each site.

3. Check the dial plan to see if the calls to the other site are using the local gateway only. If yes, then you can reuse the route pattern. Also you can reuse one of the existing CSS that has access to that route pattern as the AAR CSS.  If no, then create a route pattern to specifically match the AAR calls and add it to the long distance or international partition for that site.

voice translation-rule 1
rule 1 /\(^[2-9]………$\)/ /91\1/ — for national calls
rule 2 /\(.*\)/ /9011\1/ — for international calls

!
!
voice translation-profile ANI
translate calling 1

When using IPIPGW, you need to define a zone prefix to route calls. invia zones does not use the default technology.

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.

A. Unicast G711 everywhere

1. Create region for MoH. G711 with all other regions.

2. Do not enable multicast in audio source, server, MRG

3. Enable only G711 in service parameter (Default)

B. G711 multicast everywhere

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 5.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in infrastructure (igmp snooping in switches, pim sparse-dense in routers)

C. G711 multicast everywhere, remote site uses MoH from local SRST flash

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 1.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in HQ and Remote LAN infrastructure (igmp snooping in switches, pim sparse-dense in routers), Disable multicast in WAN infrastructure.

5. Enable SRST MOH feature from flash.

6. Enable loopback interface in SRST router for PSTN users to hear MoH.

D. G711 multicast at HQ, G711 unicast at Remote

1. Create region for MoH. G711 with all other regions.

2. Enable multicast in audio source, server, MRG. Set hop count to 1.

3. Enable only G711 in service parameter (Default)
4. Enable multicast in HQ and Remote LAN infrastructure (igmp snooping in switches, pim sparse-dense in routers), Disable multicast in WAN infrastructure.

5. Create two MRG, one with multicast enabled for HQ and one with no mulitcast for BR1.

E. G711 unicast at HQ site, G729 unicast at remote

1. Enable Service parameter for G711 and G729

2. Use the HQ device pool for MOH server (assuming HQ device pool uses g729 for remote)

3. Do not enable multicast

F. G711 multicast at HQ, g729 multicast at remote

1. Enable Service parameter for G711 and G729

2. Use the HQ device pool for MOH server (assuming HQ device pool uses g729 for remote)

3. Enable multicast in audio source, server, MRG

4. Enable multicast in infrastructure

G. g729 unicast at remote using transcode

1. Enable only G711 in service parameter

2. Put MoH server in HQ device pool (assuming HQ uses G729 for remote)

3. Add a transcoder to MRGL of MoH server

I. HQ uses G711 multicast, remote uses G729 unicast, all PSTN uses hear tone on hold

1. Use HQ device pool for MoH server

2. Enable multicast on Audio source and server.

3. Create 2 MRG, one with multicast, one without multicast

4. Enable g711 and g729 in the service parameter

5. Create a third MRG and MRGL with no MOH and apply that directly to gateways

If the remote device does not support the codec of the MOH stream, then MOH server can trigger a transcoder. You would need to configure a MRGL with transcoder resource in the MoH servers device pool. In this case, CCM does not use the trancoder of the remote device. It will only use the transcoder of MOH server.

TRAP uses the Default Outdial Restriction table.

a. setup meetme conference, conf bridge as necessary. Let meetme number be 1900

b. setup a route point for calls from external and internal callers as the dial-in bridge number. Let this be  1800. Cfwdall this CTI route point to voice mail.

c. Setup a Call handler meetme in Unity. Under transfer settings, tell it to ring subscriber at extn 1900. Use Supervise transfer and Check Ask Caller’s name option.

d. Create a call routing rule under forwarded calls in Unity, and set teh Forwarding station to be 1800 (CTIRP). Attempt a transfer to callhandler meetme that you just created in step c.

Unity will ask the caller his name and transfer him to the conference bridge. Conference bridge members who already have joined the conference will hear, “Call from <callername>”

Under caller input page, lock a key (0 through 9, * #) to a particular action if you want Unity to perform that action as soon as you press that key. The “Allow callers to dial an extension durin greeting” has no effect if the keys are locked. To allow callers to dial an extension during the greeting, dont lock the key to a particular action

 If you have any of your keys set to Ignore key, the greeting that is played is usually System greeting “I do not recognize that as a valid selection”. This is the Error greeting. Expose the Error greeting using Advanced Settings tool from Tools Depot and you can re-record the Error greeting to a message of your choice.

Automatic Work setting is at the CSQ level (Or at Agent Based Routing settings level) (Default Disabled).

Automatic Available setting is available for each agent. (Default Enabled)

Default settings make the agent go to Ready status immediately.

Automatic Work Settings override Automatic available setting set at agent level, when Automatic work is enabled.  No matter what the Automatic available setting is, if Automatic Work is set to enabled, agent goes into Work state after terminating the call. The length of the time he/she will be in Work state is determined by the Wrapup time period. (Upto 7200 seconds)

If Automatic Work is disabled, and Automatic available enabled under that agent, the agent is pushed to Ready status after the call

If Automatic Work is disabled, and Automatic available disabled under the agent,  the agent is pushed to Not-ready status after the call.

a. Use GetUser step to create a User object from a userid or extension. Store the result in a User variable called studentagent.

b. Use SelectResourcestep and change Routing target type to Resource. Specify Resource Target as studentagent. Set the timeout to a value < Call Forward No answer timeout set for the agent’s phone in Callmanager. One ring is 4 seconds.

c. With agent based routing, you route the call to a specific agent. Say caller calls into IPCC and selects option 1, the call is routed to agent1. Caller selects option 2, call is routed to agent 2 and so on and so forth.

** No matter how many agents are available (Ready status), call will be routed only to the particular agent specified in the script.

** With agent based routing, if the agent doesnt pickup the call, the agent is set to not-ready state and even if he logs back into the queue, call is not routed back to the  specified agent. So there is no point in putting a queueloop inside the Failed step.

** Agent based routing has separate Wrapup time and Automatic Work settings compared to  the CSQ. No matter what you set at the CSQ level, it doesnt affect the agents behaviour after he terminates the call. To force the agent to work state when doing Agent based routing, go under RmCm->Agent Based Routing settings -> set the Automatic Work to enabled and Wrapup time to X number of seconds.

**

Set the Service Parameter, H225 Device Connect Timer to either 0 or 1.

Default is 0 and media cut through is enabled. To disable set it to 2.

Change the service parameter “Accept Unknown TCP Connection” to True

Default is False

If you have two gatekeeper trunks, one in a G711 region and the other in a G729 region and you setup the Route list with G711 as the first priority and G729 as the second priority and the gatekeeper does not have enough bandwidth to process a G711 call, then it will fallback and use the G729 trunk to process the call.

You do not need to enable BRQ to make this happen.

Rule of thumb.

a. First analyze and group the time-period for which you want the partition to be active. This could be business hours (M-F, 8-5 for say).

b. If the time slots cannot be grouped into one time period, define multiple time periods.

c. Group all timeperiods into one time schedule.

d. apply time schedule to partitions and partitions to CSS.

e. calling devices should use this CSS (Gateways/ phones etc)

f. Note that when you want differential call treatment based on ToD, you need to create a tie on the dialed number. To create the tie, you have to define the same number (say 1005) twice  one in each partition P1 and P2. P1 may be a TOD enabled partition, while P2 may be a normal partition. P1 will be listed above P2 in the calling device CSS.

The following tip may be utilized when you are asked to forward a call from CM / CME to Voicemail and have the caller leave a message. Then the message should be delivered to multiple user inboxes and also  light up mwi on the appropriate phones.

a. Setup CM/CME to forward calls to VM. (say  a route point with number  1005)

b. Create a PDL in Unity with members as the individual subscribers who needs to receive the message.

c. Create a Callhandler CH with number 1005. Go to Messages – > Message recipient and select this PDL you created in step 2.

For a gateway to register with a gatekeeper, there should be at least one voip dial-peer defined in the gateway.

If it is a CME router, then when you define ephones, that will create dial-peer and hence that will work fine.

We setup a IPIPGW with 3 gatekeepers and 2 callmanagers.

IPPhone1—Callmanager 1—–GK1———–(IPIPGW/VIAZONEGK)———–GK2———Callmanager2—IPPhone2

GK1 configs:

int lo 0

ip address 172.12.100.1 255.255.255.0 

gatekeeper
 zone local HQ-RTR ipexpert.com 172.12.100.1
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 20*
 gw-type-prefix 1#* default-technology
 no shutdown
!

GK2 Configs:

gatekeeper
 zone local LT-RTR ipexpert.com 192.168.10.46
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 40*
 gw-type-prefix 1#* default-technology
 no shutdown

IPIPGW config:

 voice service voip
 allow-connections h323 to h323

dial-peer voice 1010 voip
 destination-pattern 20..
 session target ras
 incoming called-number 40..
 dtmf-relay h245-alphanumeric
 codec transparent
!
dial-peer voice 1020 voip
 destination-pattern 40..
 session target ras
 incoming called-number 20..
 dtmf-relay h245-alphanumeric
 codec transparent
!

int fa0/0

ip route-cache same-interface

h323-gateway voip interface
h323-gateway voip id VIAZONE ipaddr 10.12.200.2 1719
h323-gateway voip h323-id PSTNSw
h323-gateway voip tech-prefix 1#

VIAZONE GK Configs:

gatekeeper
 zone local VIAZONE ipexpert.com 10.12.200.2
 zone remote HQ-RTR ipexpert.com 172.12.100.1 1719 invia VIAZONE outvia VIAZONE
 zone remote LT-RTR ipexpert.com 192.168.10.46 1719 invia VIAZONE outvia VIAZON
E
 zone prefix LT-RTR 20*
 zone prefix HQ-RTR 40*
 no shutdown

If you define a zone prefix for a local zone , then it will prevent gateways from registering using E.164 ID to that zone.

zone local zone10 aaa.com 10.1.1.1
zone prefix zone10 5000

With this configuration, none of the gateways can register to zone10 using E.164 ID’s.

The IPIPGW supports configuration of an RTP loopback dial peer for use in verifying and troubleshooting H.323 networks. When a call encounters an RTP loopback dial peer, the gateway automatically signals call connect and loops all voice data back to the source. In contrast to normal calls through the VoIP-to-VoIP gateway, RTP loopback calls consist of only one call leg.

dial-peer voice 30 voip
destination-pattern 2222
session target loopback:rtp
incoming called-number 2222
codec g711ulaw

Telephony call-legs: 0
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 1

13FB : 67 97945490ms.1 +20 pid:30 Answer 5004 active
dur 00:00:12 tx:631/100960 rx:631/100960
IP 10.200.200.250:17012 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

Usually you call the B-ACD application by matching the incoming dial-peer and calling the application using the service command. You can also set it so that you can call from IP Phones and FXO ports directly to AA application

dial-peer voice 10 voip
destination-pattern 1000 (this is the pilot number defined in AA)
codec g711ulaw
session-target ipv4:10.1.1.1 (use a loopback address of the router so that the dial-peer will always stay up)

dial-peer voice 11 voip
incoming called-number 1000
codec g711ulaw
service aa (this dial-peer will invoke the aa application)

If you monitor calls you can actually see three legs for the calls. See below

Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 2
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 3

13DD : 403 633897060ms.1 +1490 pid:20011 Answer 5004 active
dur 00:00:11 tx:0/0 rx:543/86880
Tele 50/0/7 (403) [50/0/7.0] tx:9320/9320/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm

13DD : 404 633898520ms.1 +30 pid:204 Originate 5060 active
dur 00:00:11 tx:543/86880 rx:316/50560
IP 10.200.200.250:17668 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

13DD : 405 633898530ms.1 +10 pid:205 Answer 5004 active
dur 00:00:11 tx:316/50560 rx:543/86880
IP 10.200.200.250:19166 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

When you install two IPCC servers the following considerations hold true:

 a. The first server you activate (CRS Engine and/or Datastore services) will become the active CRS box.

 b. The second server you activate (CRS Engine) will become the standby box for the active CRS engine.

 c. The second server you activate (Datastore engine) will become the standby box for the active Datastore engine. (Agent, Historical etc).

 d. When you create a jtapi user in the active box (first server in your cluster), the user name that is created in Callmanager is “jtapi_1″ the one signifying that its the jtapi user for the first CRS server. When you install the second CRS server, it will create another jtapi user named “jtapi_2″.

 e. The second server that you install will use the administrator username / password of the first box during the initial server setup. You should not use Administrator / ciscocisco password.

 f. If the active box has a jtapi group with 20 CTI ports ranging from 3201 to 3220, when you initialize the standby box, it will automatically create 20 more CTI ports from 3221 to 3240. 

 g. JTAPI groups that are created in IPCC coressponds to CTI ports created in Callmanager. The ports in Callmanager that belong to the active CRS box will have a description (if you leave it at defaults) as JTAPI Group #0 – 1. The ports that belong to the standby CRS box will have a description JTAPI Group #0 – 2

You can use transcoding resources available on a separate chassis with CME. You can go upto 5 sdspfarm sessions. CME will share all available resources.

CME configs:

ip source-address 10.1.11.4 port 2000 ———– 10.1.11.4 is the CME router.

sdspfarm units 2
sdspfarm transcode sessions 2
sdspfarm tag 1 mtp0013c3443150 —- this transcoder is on the same box as CME. Lets call this trancoder 1. This is on NM-HDV2
sdspfarm tag 2 mtp0009b7da6540 —- this transcoder is on a different box. Lets call this transcoder 2. This is on NM-HDV

Transcoder 1 configs:

voice-card 1
dspfarm
dsp service dspfarm

sccp local FastEthernet0/0
sccp ccm 10.1.11.4 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register mtp0013c3443150
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
codec g729br8  ————— g729r8 and g729br8 are not there by default
maximum sessions 1
associate application SCCP

Transcoder 2 configs:

voice-card 1
dsp service dspfarm

sccp local FastEthernet0/0
sccp
sccp ccm 10.10.10.1 priority 1
!
dspfarm transcoder maximum sessions 4
dspfarm
!

Two calls were placed to force the transcoding and CME did utilize each transcoder (The number of transcoding sessions per transcoder was set to 1)

sh sccp connection output on Transcoder 1: (call1)

PSTNSw#sh sccp conn
sess_id conn_id stype mode codec ripaddr rport sport

1 1 xcode sendrecv g729 10.1.11.4 2000 17040
1 2 xcode sendrecv g711u 10.1.11.4 2000 16416

Total number of active session(s) 1, and connection(s) 2
sh sccp connection output on Transcoder 2: (call2)

br2#sh sccp conn
sess_id conn_id stype mode codec ripaddr rport sport

3 3 xcode sendrecv g729 10.1.11.4 2000 18014
3 4 xcode sendrecv g711u 10.1.11.4 2000 19080

Total number of active session(s) 1, and connection(s) 2

Lets look at an example where you want to block international calls outside of business hours 8 am to 5 pm.

Remember the time you should configure is in 24 hours format. So 5 pm is 17:00.

after-hours block pattern 1 9011
after-hours day Sun 17:01 08:59
after-hours day Mon 17:00 08:59
after-hours day Tue 17:00 08:59
after-hours day Wed 17:00 08:59
after-hours day Thu 17:00 08:59
after-hours day Fri 17:00 17:00
after-hours day Sat 17:01 17:00

If you specify the time as 08:59, it includes the 60 seconds from 08:59:00 to 08:59:59. So calls will be blocked until 08:59:59. If you specify the time as 17:00, calls will be blocked till until 17:00:59.

Scenario 1

Two phones need to have a shared line. When a call is answered on the first phone and a second call comes in to the shared line, you need to have the call ring at second phone with no call waiting on the first.

ephone-dn 1
number 3001
huntstop channel
no huntstop

ephone-dn 2
number 3001
huntstop channel
preference 1
ephone 1
button 1o1,2

ephone 2
button 1o1,2

Scenario 2

Two phones need to have a shared line. When a call is answered on the first phone and a second call comes in to the shared line, you need to have the call ring at second phone with call waiting on the first.

ephone-dn 1
number 3001
huntstop channel
no huntstop

ephone-dn 2
number 3001
huntstop channel
preference 1
ephone 1
button 1c1,2

ephone 2
button 1c1,2

When you add CTI ports via IPCCX admin, it uses AXL to add into CCM database.

The service parameter “Maximum AXL Writes Allowed per Minute” determines how soon it can add to the database. By default it is 20. Give a higher value to speed up this process.

CTI ports take only 1 ms to reinitialize (ie, to get back to operation after a call is disconnected), but CMT groups will take upto 200 ms to reinitialize. IPCCX will not answer calls if there are no CMT ports available. So make sure you provision 10% more CMT ports than CTI ports.

VTA uses CAST protocol. Uses 4224 TCP port. This protocol helps in discovering remote VTA capable endpoints, communicates with IP phone as well as with Callmanager. To ensure proper QOS is given to CAST, on the phone port, create a service policy to mark TCP 4224 with DSCP CS3.

VTA uses two types of codecs.

a. H263 – min of 128kbps to a max of 1.5 Mbps per call

b. Wideband – 7.0 mbps per call

Step 1 Browse to C:\dcdsrvr\config, and open the file, UMDirectoryConfiguration.ini, in Notepad.

Caution You must open the file in Notepad. Opening the file in another text editor application may corrupt the file.

Step 2 In the UMDirectoryConfiguration.ini file, locate the key, UserDirAccess.

Step 3 Change the UserDirectAccess value to true.

Step 4 In Notepad, choose File > Save to save the file.

Step 5 Close the UMDirectoryConfiguration.ini file.

Step 6 Restart the IIS Admin Service and its dependent services by choosing Start > Programs > Administrative Tools> Services.

Step 7 Right-click IIS Admin Service and then choose Restart.

Step 8 A dialog box prompts you to restart dependent services. These services may differ depending on your configuration. Click Yes.

Step 9 Restart the dependent services.

One of the current limitations is that the JTAPI (CTI-quick buffer encoding [QBE]) signaling packets are unmarked (TOS = 0) when the Cisco Unity Express AIM transmits them. In order to correct this, use an access control list (ACL) on the router that has the Cisco Unity Express AIM installed to mark and prioritize the traffic.

The JTAPI signaling from the Cisco CallManager is correctly marked with a differentiated services code point (DSCP) value of CS3.

The JTAPI signaling protocol uses TCP port 2748. Dedicate 20 kbps per Cisco Unity Express site for this traffic.

All Real-Time Protocol (RTP) audio traffic from the Cisco Unity Express AIM or the IP phone is correctly marked with a DSCP value of 0xEF.

This example shows a sample configuration for this on the router where a.b.c.d is the IP address of the Cisco Unity Express AIM:

access-list 101 permit tcp host a.b.c.d any eq 2748
!
class-map match-all cti-qbe
 match access-group 101
!
policy-map cti-qbe
 class cti-qbe
 set dscp af31
 bandwidth 20
!
interface Serial0/1
 service-policy output cti-qbe

Since Cisco Unity Express uses JTAPI to interface with the Cisco CallManager, no MWI on and off numbers need be configured.

Callforward All

If the Call Forward all CSS is set to none, it will concatenate the line and device CSS. This applies to settings done via the phone as well as via the CCM user page.

Callforward Busy/No-answer 

If the Callforward CSS is set to none, it does not concatenate the line and device CSS.

Based on the ISDN number plan and type of the incoming call, the following values will be used to prepend the Caller ID.

National number prefix

International number prefix

Subscriber number prefix

Unknown number prefix

To enable Overlap receiving for PRI interfaces that you the PRI Backhaul mechanism

If an Attendant DN is configured on FXO port and this parameter is set to True, then Callmanager will limit outbound calls from that DN to only use the FXO port that is configured with that Attendant DN. You would need to create a route pattern and include this FXO port in the route list. Even though there are other gateways that could be listed above in the route list, it would still use this gateway only.

Also, none of the other phones will be able to use this gateway even though this gateway appears in the route list.

This parameter determines if you can resume a call from hold on a MGCP FXS shared line.

Default value is true

This is a global setting for Caller ID for all gateways. You can also set the Caller ID individually at the gateway level

In CCM 4.1, Callmanager concatenates the CSS of the device and lines and places the CSS of the lines in front of the CSS of the device.

As per SRND, For CTI ports and CTI Route points, it does the other way. ie, CSS of the device comes first followed by the CSS of the lines. This used to be the behaviour in CCM  3.1. This was not the case during testing. Even the CTI ports and CTI route points were using line CSS First

When you have a CMC and FAC enabled on a route pattern, you have to enter FAC first, before entering CMC code.

CMC codes do show up on CDR records (unlike FAC)

When you define FAC code names and auth codes, only FAC code names appear on CDR. (For added security). The auth codes themselves never show up.

When doing digit manipulation, first decide whether you want to do it at route pattern level or route group level. Route group level is recommended for following reasons.

 a. The end user will see the manipulated digits if you do digit manipulation at the route-pattern level. If you do it at route group level, the end user will not see that on his phone.

b. IF you do it at route pattern level, CDR will log the digits including the manipulated ones. So its recommended to do it at the route group level.

c. Any digit manipulation you do at route group level completely overrides the setting at route pattern level.

MGCP PRI backhaul does not support call preservation when transitioned from Callmanager to SRST and vice versa.

H323 does not support Facility IE. H323 supports only Display IE

MGCP supports both Display and Facility IE

EAG is implemented as a system script called checkaltgreet.aef. To activate EAG is a script, insert a Call Subflow checkaltgreet.aef into the script.

EAG is activated by the existence or absence of the greeting. It is stored as AltGreeting.wav. If this file exists on the system, the EAG is automatically activated and the special greeting is played whenever the script directs it to be played.

The number to call to record the greeting (to turn on the EAG feature) or to delete it is the Cisco UE pilot number associated with greeting management system (GMS).

Give param queue-manager-debugs 1 and from privilege mode, “debug voip application script”.

 Site2#
*Jul 17 00:44:53.311: //89//TCL :/tcl_PutsObjCmd:
proc init_perCallvars
*Jul 17 00:44:53.311:
*Jul 17 00:44:53.323: //89//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome Prompt and options menu ++
*Jul 17 00:44:58.383: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ No option selected +++
*Jul 17 00:44:58.455: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD Service Running <<<
*Jul 17 00:45:06.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ New incoming call to queue 3210 +++
*Jul 17 00:45:06.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NEW} {1153097106}} <<<
*Jul 17 00:45:06.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ 3210 Queue Length = 1 +++
*Jul 17 00:45:06.719: //89//TCL :/tcl_PutsObjCmd: TCL AA: >>> Destination Huntgroup = 3210 <<<
*Jul 17 00:45:06.747: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097106}} <<<
*Jul 17 00:45:06.767: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_001 –
*Jul 17 00:45:06.767: //89//TCL :/tcl_PutsObjCmd: TCL AA: — ERROR: Unable to play prompt; File not found –
*Jul 17 00:45:06.771: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH +++
*Jul 17 00:45:16.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097116}} <<<
*Jul 17 00:45:16.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:26.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097126}} <<<
*Jul 17 00:45:26.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:30.815: //89//TCL :/tcl_PutsObjCmd:
 proc act_CallSetupDone
*Jul 17 00:45:30.819:
*Jul 17 00:45:30.819: //89//TCL :/tcl_PutsObjCmd:
 IP IVR Leg Setup Status = ls_002
*Jul 17 00:45:30.819:
*Jul 17 00:45:30.823: //89//TCL :/tcl_PutsObjCmd: TCL AA: >> Agent did not answer; Received Call NOAN <<
*Jul 17 00:45:30.835: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NOAN} {1153097130}} <<<
*Jul 17 00:45:36.695: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Now playing second greeting prompt +++
*Jul 17 00:45:36.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NOAN} {1153097136}} <<<
*Jul 17 00:45:36.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:45.831: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY} {1153097145}} <<<
*Jul 17 00:45:45.879: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097145}} <<<
*Jul 17 00:45:46.163: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_000 –
*Jul 17 00:45:46.167: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH Livefeed +++
*Jul 17 00:45:46.219: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_001 –
*Jul 17 00:45:46.219: //89//TCL :/tcl_PutsObjCmd: TCL AA: — ERROR: Unable to play prompt; File not found –
*Jul 17 00:45:46.223: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH +++
*Jul 17 00:45:46.711: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097146}} <<<
*Jul 17 00:45:46.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:56.711: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097156}} <<<
*Jul 17 00:45:56.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:57.415: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> Call Abandoned Time = 64 Seconds<<<
*Jul 17 00:45:57.427: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD ++ Service Counter = 6 ++
*Jul 17 00:45:57.431: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Queue 3210 Statistics = {“Total calls presented to the queue: ” 5}
        {“Calls answered by agents: ” 1}
        {“Number of calls in the queue: ” 1}
        {“Average time to answer (secs): ” 20}
        {“Longest time to answer (secs): ” 20}
        {“Number of abandoned calls: ” 4}
        {“Average time before abandon (secs): ” 133}
        {“Calls forwarded to voice mail: ” 0}
        {“Calls answered by voice mail: ” 0} <<<
*Jul 17 00:45:57.435: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ 3210 Queue Length = 0 +++
*Jul 17 00:45:57.439: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> 3210 Queue Length = 0 <<<
*Jul 17 00:45:57.439: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> Queue 3210 is Empty <<<
*Jul 17 00:45:58.455: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD Service Running <<<

By default if conference initiator hangs up the call, the conference is terminated.

To keep the conference parties connected when the conference initator leaves, configure under the  ephone “keep-conference“. When the initiator hangsup the remaining parties stay connected. If he uses the end-call soft key, the conference is terminated.

To keep the conference parties connected when the conference initator leaves by hanging up or pressing the end-call softkey, use “keep- conference endcall” option under ephone

For both internal and external calls

ephone-dn

caller-id block

* This does not apply for E1R2, because caller id blocking is a flag in the setup messages and E1R2 does not support that. This does not apply for FXO, because caller id is controllerd by Telco on FXO. Use translation patterns to change caller ID to null

Only external calls

dial-peer voice 10 pots

clid strip – to block only the number

clid strip name – to block only the name

To send only caller name

telephony-service

caller-id name-only

To block caller id on a call by call basis

telephony-service

caller-id block code *67

1. Disable ip domain lookup using command “no ip domain-lookup”

2. Disable exec timeout using command “exec-timeout 0 0″ under line vty mode.

1. BRQ – True

2. AAR – True

3. Call Diagnostics enabled – True

ANI is not available with E1R2 dtmf signaling. For all other E1R2 options, ANI is available and you need to select this option to have ANI send out.

Make sure you always specify MRGLs for Gateways. They dont default to the device pool to get the MRGL. (unlike phones, where if you set the MRGL to none, they default to the device pool)

When multiple Cisco Extended Functions are active within a Cisco CallManager cluster, the redundancy manager uses an algorithm to determine which Cisco Extended Functions service is active and which is the backup CEF. The Redundancy Manager uses the lowest IP address of the server that is running the CEF service as the active servicne. The remaining CEF services serve as backup services.

Create a Public distro list and assign it a number.

You can choose individual phones as members of the PDL.

Or you can create a Group, make this group a member of the PDL and make individual phones as members of the group. You can also add a GDM as a member.

MWI doesn’t seem to work for the line appearance for GDM extensions.

The following script allows to dial an extension at any time during the prompt, along with giving an option to press 1 or 2 to do different tasks (like dial by name or reach an operator etc).

If you need to integrate CME with a CUE on a different chassis, you would need the following.

1. Create a h323 dial-peer in CME gateway and point it to the gateway that hosts CUE.

2. Create a SIP dial-peer in the gateway that hosts CUE to point to CUE ip address.

You cannot create a SIP dial-peer on the CME gateway and point it directly to the CUE IP Address.

How to mark traffic coming in and out of CUE ?

class-map match-all SipControl
 match protocol sip
!

policy-map SipMark
 class SipControl
  set ip dscp cs3
 class class-default

 interface Service-Engine0/0
 ip unnumbered FastEthernet0/0
 ip route-cache flow  ——————————This command was used solely for the purpose of Netflow. Nothing to do w/marking
 service-module ip address 10.1.11.5 255.255.255.0
 service-module ip default-gateway 10.1.11.4
 service-policy input SipMark ——————– This policy map marks packets from CUE to CME (thats why its applied inbound). Thanks to Ted for correcting this. I had applied this service policy outbound in my earlier post

 dial-peer voice 20000 voip
 destination-pattern 3600
 session protocol sipv2
 session target ipv4:10.1.11.5
 dtmf-relay sip-notify
 codec g711ulaw
 ip qos dscp cs3 signaling  — This marks SIP control packets going from CME to CUE.
 no vad

Netflow output:

Protocol         Total    Flows   Packets Bytes  Packets Active(Sec) Idle(Sec)
——–         Flows     /Sec     /Flow  /Pkt     /Sec     /Flow     /Flow
TCP-WWW            447      0.0        25   335      0.0       0.4       1.8
UDP-NTP            218      0.0         1    76      0.0       0.3      15.4
UDP-other           34      0.0       260   201      0.0       9.1      15.5
ICMP                25      0.0        16   217      0.0      21.8      15.5
Total:             724      0.0        28   273      0.0       1.6       7.0

SrcIf         SrcIPaddress    DstIf         DstIPaddress    Pr SrcP DstP  Pkts
Se0/0         10.1.11.5       Local         10.1.11.4       01 0000 0303     6
Se0/0         10.1.11.5       Local         10.1.11.4       01 0000 0303    27
Se0/0         10.1.11.5       Local         10.1.11.4       11 8003 42BC   901
Se0/0         10.1.11.5       Local         10.1.11.4       11 13C4 13C4     3

10.1.11.5 is CUE, 10.1.11.4 is CME. Port 13C4 in hex translates to 5060. THis is the port on which SIP protocol listens for control packets.

Output of show policy-map interface

PSTNSw#sh policy-map int
 Service-Engine0/0

  Service-policy input: SipMark

    Class-map: SipControl (match-all)
      7 packets, 2911 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      Match: protocol sip
      QoS Set
        dscp cs3
          Packets marked 7

    Class-map: class-default (match-any)
      125 packets, 51142 bytes
      5 minute offered rate 0 bps, drop rate 0 bps
      Match: any
PSTNSw#
PSTNSw#

1. Out of bandwidth text — this text is displayed when AAR is not enabled and there is not enough location bandwidth

2. AAR Congestion rerouting text — this text is displayed when AAR is enabled and rerouting occurs

Decides how many seconds before answering the call automatically. Should be less than T 301 as well as Call forward no answer ring duration.

Built-in bridge enabled:

True: is needed to enable the bridge in the phone for barge in feature. Built in bridge supports only G711.
False: Default

Privacy:

Will decide if the call state information  is displayed on shared lines and also if barge is allowed via shared lines.

True: Default. No call state information is displayed across shared lines and barge is not allowed
False:

Always use Prime line

False: For an incoming call on any line, when you go off hook the call will be answered. Is the default.
True: Will always use the first line when you go off hook. Inorder to answer an incoming call on any other line, you need to hit the ringing line button.

Always use Prime line for voicemail

False: When you hit the messages button, will select the line which has a message waiting.

True: When you hit the messages button, will always use the first line

Call Classification classifies the call type. Valid values are On-net and Off-net. Default is off-net. This can be set at the gateway level as well. By default the gateway is set to Use System default.

This parameter works with Block off-net to off-net transfer. Default is false. Hence by default the system will allow trunk to trunk transfer.

1. T301 – timers between alerting and connect. This is the interval that Callmanager will keep ringing the remote phone and if no answer will drop the call after this interval. This should be more than the Call forward no answer and auto answer duration.

Default is 180 seconds

2. T302 – Interdigit timeout

Default is 15 seconds

Station KeepAlive Interval

Keepalive between phones and Primary Call processing server

Defauit: 30 sec

Station and Backup Server KeepAlive Interval
Keepalive between phones and Secondary Call processing server
Default: 60 sec

Tone on Hold -  3 options

0 – disabled for MGCP, play once for other devices

200000 – disabled for all device

any non zero value greater than 5 – plays tone on hold for all devices every that many seconds

Default: 10

To disable auto registration for any of the port set Sid 0 or Sid1 to 0 or empty. This will disable the port and not attempt a registration with CM. If you want to manually add the port in CM, put Sid0 or Sid1 as the mac-address of the appropriate port.

WHen calculating cir and mincir for FRTS, use 95 % of the CIR value. If CIR = 256k, it should be used as 256000. Do not multiply 256 with 1024.

Frame-relay fragment is typically not needed for link speeds above 768 kbps.

If CIR is 256k and link speed is 480kbps, frame-relay fragment should be based off of PVC CIR ie 256 K.

1. AdminUtility

C:\Program Files\Cisco\Bin

Used to change service account passwords.

CCMService, CCMServiceRW, SQLSvc, CCMCDR,CCMUserf

2. CCMPwdchanger

C:\dcdsrvr\bin

Used to change the password for admin user accounts. Directory Manager,  CCMAdministrator, CCMSysUser, IPMASysUser

1. While copying and pasting the DHCP configs, forgot to change the default gateway for second scope
2. Did not set ntp master in router
3. forgot to set ntp in subscriber, had to go back and do it again
4. forgot to set framing and linecoding in controller
5. did not shut the voice port before removing ds0-group
6. forgot call pickup group and night service
7. ip phones with night service, not clear??? Is it for ephones or ephone-dns
8. when integrating unity with CME or CCM, always verify servers before saving the configs, because when you save unity services will restart
9. forgot to reset voicemail hunt list after adding,
10. forgot to change the framing at PSTN switch
11. at CME, the call was matching the wrong dial-peer.
12. forgot to set portfast for CCM and unity ports
13. forgot to add DNS entry in unity
14. forgot to add ds0 country code
15. forgot to add 91 to the after hours block pattern in CME
16. forgot to enter secondary dialtone
17. configured night service to 8:00 instead of 7:59
18. use bulk import wizard to add subscribers
19. infrasturcture – 18 minutes 3 sites, setting vlans, assigning to ports, trunking etc, dhcp pools.
20. messed up ip addressing in branch 2.
21. dont set mwi on-off first before setting number command in ephone-dn
22. created so many errors in bulk subscriber import
23. alias,lastname,firstname,dtmf
24. created live record first…but no way to send message to a PDL.
25. So created a call handler with extn 1000 and then set the greeting to blank, after message, toold it to take  a mesage and message receipient was set to all subscribers PDL. After message, send caller to hangup.
26. mwi is not working for callmanager subscribers. restarted unity to fix this.
27. route-group route list name incompatibility. deleted route groups and added it back fixed it.
28. use no ip domain-lookup on all routers.
29. for e1 r2, the first dial peer was alwasy gtetting matched (which had a destination pattern configured). And ANi was set to the destination pattern number in the first dial -peer. To avoid this situation, alwasy configure first dial peer with incoming-called number . so that this problem can be avoided.
30. used full international digits to dial branch 2.instead use 4 digit dialing. and prefix the 011331322 to the dialed digits (3xxx).
31. after hours block pattern i used 1900, 1976. instead should use 91900,91976 (because 9 is dialed).
32. night service – specify night-service bell on the source ephone-dn
33. specify night-service bell on the destination ephones.

1. How to configure ATA for voice VLAN?
2. Set speed and duplex for all ports
3. Do we need to configure spanning tree port fast or not for ip phone ports on NM-16ESW? Yes, configure spanning tree for NM-16
4. To force IP Phones to renew DHCP, disable and enable inline power
5. Overlooked the fact the BR2 is CME and configured dhcp option 150 to CCM
6. Always configure publisher as the auto registration server. Did not work for Subscriber
7. MSFC retains the access list applied to interface vlan even after the vlan interface was removed
8. Did not consider voice mail ports for max ephone and dn when doing the CME setup
9. Forgot to set the Callforward settings on phone, during initial setup
10. did not notice crs admin user name, created the usual user rather than what was in the question
11. no need to install crs editor, it is already installed
12. ICD extensions and shared lines??
13. configured the wrong agent user
14. forgot about ip phone agent telecaster user
15. CRS answers calls, calls agent, but does not patch call – Issues with jtapi client, had to update it using the jtapi client update tool
16. Idea – for easy editing of phone line settings create a dummy phone and add all line to that phone. You need to select only one phone and toggle between lines
17. Forgot to add EM Service before creating EM Profile, had to come back and do it.
18. When copying the EM service link from univercd, copy the example link since it has the entire link in one line
19. Configured AC user with password as cisco
20. Did not associate the attendant phone to ac user
21. Forgot to check call park retrieval for ac user
22. Change DNS to IP in update plugins
23. attendant console, calls to pilot number should be transferred to voicemail on no-answer. I configured the last member as voicemail pilot. This is wrong. You need to configure the call forward no answer on the hunt group members to voicemail and create a subscriber in unity with the extension of hunt pilot. Set the TCD Service parameter, Reset original called number on redirect to False
24. clearing a vlan doesnt remove the actual config from the port in 6500
25. set ata port also to trunking (vlan, auxiliary vlan).
26. didnt put spanning-tree portfast on all switchports (nm16esw and 3550sw)
27. lease 0 8 — means 8 hours, lease 8 means 8 days.
28. dont use auto registration – time waste
29. wasted 1.30 hours registering phones
30. configured wrong helper address at hq
31. unity message recording – set this in subscriber template (under each subscriber also) and under default subscriber COS.
32. used label command instead of description in cme… should have used description command.
33. modified default icd script by accident– dont do this…save as a new file and modify.
34. applied wrong file in the application..
35. put mgcp fax rate 7200 — but this is for t38 only.
36. didnt use telephony-service setup command to setup CME. Should use this to reduce time.
37. check against trivial passwords for security – this should be unchcekd so that you can set password to 54321.
38. call restriction table – ???? or 4 digits as maximum transfer digits.
39. Subscribers -> Account policy.
40. attendant console – created a 3rd member, for voicemail transfer. No need for this. when somebody doesnt answer the call it will roll over to voicemail and hit the pilot’s mailbox.
41. did uncheck fax relay ECM override on 6608. TO disable ECM this should be checked
42. didnt set fax relay speed settings to 7200 in vg248. this should be set to 7200bps. (global setting).
43. if far end gateway is 6608 set nsf to preserve in vg248

1. Create a MRG for each resource at each site. HQ conf, HQ MOH, HQ Xcode, same for BR1. Do not add any unused resources to any group. The devices with MRGL assigned can access only the resources in the groups and does not have access to any unused resources. Total 6 MRG’s required.

2. Create a MRGL for each type of device, one for phones one for gateways. Total 2 MRGL’s

3. if one site requires Mulitcast and another requires unicast. Then uncheck “Allow multicast” under the appropriate MRG. The audio source should be multicast enabled.

4. It seems gateways cannot access MRGL from the Device pool. So you have to apply MRGL at the gateway level. All other devices (phones etc) are able to access MRGL from the Device pool if you have not set a MRGL under the device.

mgcp
mgcp call-agent 10.1.1.1
mgcp ip qos dscp cs3 signaling
mgcp dtmf-relay voip codec all mode out-of-band

ccm-manager fallback-mgcp
ccm-manager redundant-host 10.100.1.50
ccm-manager mgcp
ccm-manager music-on-hold (if you need to stream moh locally from flash)

application
global
service alternate Default
dial-peer voice 10 pots
service mgcpapp
incoming called-number .
port 1/0:23
direct-inward-dial

1. voice class codec, g711 as first preference and g729 second

2. voice class h323 with timeout 3

3. dial-peer to match incoming call leg

4. two dial-peers to send calls to CCM. enable no vad on these dial-peers and also dtmf-relay
5. 4 outgoing dial-peers for 911, local, long and international

6. enable qos for signaling

voice class codec 1
codec preference 1 g711alaw (under most circumstances g711ulaw is enough, unless the site is in Europse or a country that uses alaw)
codec preference 2 g711ulaw
codec preference 3 g729r8
!
!
!
voice class h323 1
h225 timeout tcp establish 3

dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
port 1/0:1

dial-peer voice 2 voip
destination-pattern 3…
voice-class codec 1
voice-class h323 1
session target ipv4:10.1.21.39
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 3 voip
preference 1
destination-pattern 3…
voice-class codec 1
voice-class h323 1
session target ipv4:10.1.21.38
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad

To have the ATA prepend Tech prefix for outgoing calls, do the following

1. AutMethod = 0×00000004

2. PWD0 = 1#

If PSTN does not send ANI, then the gateway will use the destination pattern of the incoming dial peer as the ANI. The destination pattern needs to be a specific match like 911. It will not be used if there are any wild cards, like 9..

This is mostly seen as an issue in T1/E1 CAS.

Set the Cisco Telephony Call Dispatcher Service Parameter
Reset Original Called Party on Redirect to FALSE

To disable trivial password check go to

Subscriber > Account Policy >  Check against trivial passwords for extra security