c 60 with CUCM 7.1.5

Similar Questions

• Unity connection 8.6 Migration to 11 with CUCM 8.0

  Due to the right I'm going to intend a Migration of 8.6 connection unit to 11.0 of connection of the unit before I can improve CUCM. I am sure that the answer is Yes, but would there be problems running 11.0 of connection of the unit with CUCM 8.0.

  Nothing fancy on the face of connection of the unit, no unified messaging ect.

  New UC 11.0 should just registered the EPSC ports?

  Thank you

  Dan

  Looks like you need at least version 8.5 for the SCCP and SIP integration CUCM to be supported. Check the compatibility of connection of the unit guide below.

  http://www.Cisco.com/c/en/us/TD/docs/voice_ip_comm/connection/compatibil...

  Combinations for Cisco and unity connection Unified Communications Manager of the supported Version

  Table 22 Version combinations of supported for the connection of the unit and Cisco Unified Communications Manager

  Cisco Unity Connection	Cisco Unified Communications Manager
  8.5 and later	11.5 (1), 11.0 (1), 10.5 (2), 10.5 (1), 10.0 (1) 9.1 2, 9.1 (1), 9.0 (1), 8.6 (2), 8.6 (1), 8.5 (1)

  SIP: http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/connection/compatibil...

  -Terry

  Please note all useful posts

• SX 20 with CUCM 8.6.1

  Hello

  I need to record 20 peripheral SX with CUCM 8.6.1, but SX 20 is not available in the list of devices!

  How can I register it?

  Regds

  $

  Hello

  CDP file will be used for the endpoint SX20 push updates of software and the package devce for the CUCM, where it must be installed separately.

  as mentioned in the document provided by Vivek:

  Installation instructions

  Note: Cisco recommends that you apply this device for a package

  maintenance window.

  Note: Packages of devices are installed on the active partition and cannot be

  uninstalled;

  Cisco recommends that you take a backup DRS prior to installation.

  For more information on how to do a proper backup, see the Recovery Guide disaster for each version of Cisco Unified Communications Manager.

  Procedure

  Note: Apply this package of device to all of your Cisco Unified Communications

  Manager servers, starting with the publication server and the TFTP server.

  Note:

  When you apply a device Package to activate the new device support, a broad cluster restart is necessary for these devices to record successfully.

  A

  Cluster - wide restart is not required when you apply a Package of device for the update of the existing firmware/media. Cisco Unified Communications Manager 8.6, you perform all software installations and upgrades by using the software update menu options. The system can load and process only the software that Cisco has approved. You may not install or use third party or Windows - software applications that you used a previous version of Cisco Unified Communications Manager.

  1 from your web browser, go to the web page of Cisco Unified Communications Administration of the operating system.

  2. in the menu of the software upgrades, select Install/Upgrade.

  3. fill in the appropriate values in the section where the software and click Next.

  4. in the available software drop-down list box, select the device package file

  Click next

  5. check that the MD5 has the correct value. Click Next.

  6. in the warning box, verify that you have selected the right firmware and

  then click on install.

  7. check that you have received a message of success.

  Note:

  If you restart the cluster, skip step 8.

  8. stop and restart the TFTP server.

  9 reset the devices concerned to upgrade the devices to the new load.

  Hope that helps

  Concerning

  KR Ravi

• MGCP FXO does not register with CUCM on VG204XM

  I have a VG204XM running MGCP with CUCM 10.5. When the port is changed and reset him CUCM config push down to the VG204XM there is an error (it is identical to the Bug CSCtr78616 reported on a router 28xx in 2011)

  In the newspapers, I see that, once the router receives the config it tried to configure AALN/S0 / / 0 (voice-port 0/0)
  However, for some reason, it seems that the router is to convert the port to a 65535 instead of 0

  "cmapp_xml_get_interface_info: slot = 0 vic = 65535 port = 0".

  Since the port above does not exist on the router the survey attempt a mistake.

  Mar 27 18:07:59.278: % SYS-5-CONFIG_I: configured from console by console
  Mar 27 18:07:59.278: cmapp_xml_get_interface_info:
  18:07:59.278 Mar 27: =.
  Mar 27 18:07:59.278: cmapp_xml_get_interface_info: slot = 0 vic = 65535 port = 0
  18:07:59.278 Mar 27: Interface_name: AALN/S0/0
  Mar 27 18:07:59.278: controller:
  18:07:59.278 Mar 27: addr: 65535/0/0
  Mar 27 18:07:59.278: interface:
  Mar 27 18:07:59.278: voice_port: voice-port 65535/0/0
  Mar 27 18:07:59.278: dial_peer: voice dial-peer 9990655350
  Mar 27 18:07:59.278: dial_peer_port: 65535/0/0
  Mar 27 18:07:59.278:
  Mar 27 18:07:59.278: cmapp_xml_configure_router: verification of the Deleted interfaces
  Mar 27 18:07:59.278: cmapp_xml_get_intf: looking for a vdb [65535/0/0]
  Mar 27 18:07:59.278: cmapp_xml_configure_router: analog ports are ready
  Mar 27 18:07:59.278: cmapp_xml_find_corr_intf_cfg:
  Mar 27 18:07:59.278: cmapp_xml_find_corr_intf_cfg: old_cfg is NULL
  Mar 27 18:07:59.282: cmapp_xml_intf_cfg_changed:
  Mar 27 18:07:59.282: old intf configuration is null
  Mar 27 18:07:59.282: cmapp_xml_cfg_analog_intf:
  Mar 27 18:07:59.282: cmapp_mgcp_get_card_type_for_analog_intf: looking for a vdb [65535/0/0]
  Mar 27 18:07:59.294: cmapp_xml_cfg_router_intf: intf configuration: [voice-port 65535/0/0]
  Mar 27 18:07:59.294: cmapp_xml_cfg_router_intf: configuration [echo - cancel enable]
  Mar 27 18:07:59.298: % SYS-5-CONFIG_I: configured from console by console
  Mar 27 18:07:59.298: cmapp_xml_cfg_router_intf: 395 line router configuration error
  Mar 27 18:07:59.298: cmapp_xml_send_msg_to_callmgr: CMBH_MSG_CFG_ERR of sending
  Mar 27 18:07:59.298: * info ERROR: the router configuration failed: voice-port 65535/0/0
  echo - cancel enable
  end

  I have a VG224 gateways running with MGCP CUCM 10,5 and no error

  Everyone knows about this problem on VG204XM?

  Hello

  Please check

  https://Tools.Cisco.com/bugsearch/bug/CSCui83022/?reffering_site=dumpcr

  Auto config MGCP fails for VG204XM
  CSCui83022
   

  Description

• Connection of the UNIT with CUCM integration

  Hello

  Can someone please advice me on the underside, I CUCM (PUB + 1 SUB) cluster and node stand-alone 1 CUC.

  need to know the best Cisco recommended configuration

  1.0 order of AXL servers (what should we put 0 - PUB or SUB)

  2.0 tftp and order serever CUCM under port, (PUB, order SUB) group settings

  Please advice

  Thank you

  Purna

  Axl Server always be PUB and SUB.
  Pub - 0
  Sub - 1

  TFTP server must first be Sub and PUB.

  Sub - 0
  Pub - 1

  This is the best way and the recommended method to configure / integration with CUCM with connecting the unit

  Sent by Cisco Support technique iPhone App

• SX20, EX90 with CUCM

  Hello

  the customer wants to buy SX20 and Ex90, and he needs information at my side, Questions:

  -Did he need a VCS? or CUCM is enogh for whom?

  -What is the version CUCM is supported for SX20, EX90?

  -How add these endpoints to CUCM?

  -regarding bandwidth, which is good the bandwitdth to make audio and video calls is very clear and quality? up to 6 Mbps?

  -What is the version for SX20 EX90 must be suppoeted with CUCM or VCS?

  Please, your answers are very important to me

  Thank you

  Hi Lkayyali,

  Please find the BW resolutions depending on requirements.

  profiles of optimal definition for system supporting 1080 p

  w288p30 w448p30 w576p30 720 p 1080 p 30 p 30

  256kbit/s normal 512kbit/s 768kbit/s 1152kbit/s 2560kbit.s

  average of 128 Kbps 384kbit/s 512kbit/s 1152 kbit/s 1920kbit/s

  128 Kbps high 256kbit/s 512kbit/s 768kbit/s 1472kbit/s

  profiles of optimal definition for system supporting 720 p 60

  w144p60 w288p60 w448p60 w576p60 720 p 60

  Normal 128kbit/s 512kbit/s 1152kbit/s 1452kbit/s 2240kbit.s

  average of 128 Kbps 384kbit/s 768kbit/s 1152 kbit/s 1920kbit/s

  128 Kbps high 256kbit/s 512kbit/s 768kbit/s 1152kbit/s

  Please feel free to rate this answer in case it is useful

• 3241 GW ISDN with CUCM 10.5 or VCS

  Hi all

  What is the best practice to deploy ISDN GW 3241.

  It connects with CUCM or VCS control?

  How to connect it with CUCM or VCS?

  Kind regards

  There are some basic steps in the Quick Start Guide and there is more information on the different ways to configure it in this thread: https://supportforums.cisco.com/discussion/11580051/isdn-gw-3241-admin-config-guide

  Wayne
  --
  Remember the frequency responses and mark your question as answered as appropriate.

• Cisco C40 unsubscribe automatically with CUCM 8.6.2

  Hello

  I have Cisco C40 connected to CUCM 8.6.2.

  First, he is registered with CUCM and I was able to call on another phone C40.

  Then, C40 automatically BSW of CUCM and isn't recording with CUCM now.

  I added C40 via device-> phone-> Add new-> then selected Teleprecense C40

  C40 SW version is: 5.1.5.297625

  I joined C40 and CUCM screenshot. It shows not recorded, which proves C40 registerd at least once.

  I could make and receive calls with the C40 for a few minutes. Subsequently recorded automatically and now

  error message is status: failed

  Suggest pls.

  Rgds

  Rajesh

  Hi Renon,

  Please below changes made in the SIP settings:

  change SIP profile 1 Type at Cisco, since it is native enrolled and default transport for TCP

  * c xConfiguration SIP URI 1 profile: "[email protected] / * / "

  "

  * c xConfiguration SIP profile 1 DisplayName:

  * c xConfiguration SIP DefaultTransport 1 profile: TCP

  * c xConfiguration SIP profile 1 TlsVerify: Off

  * c xConfiguration SIP profile Type 1: Cisco

  * c xConfiguration SIP profile 1 outgoing: Off

  * c xConfiguration SIP profile 1 Proxy Discovery 1: manual

  * c xConfiguration SIP Proxy 1 address of profile 1: "172.31.131.10."

  * c xConfiguration SIP profile 1 Proxy Discovery 2: manual

  * c xConfiguration SIP Proxy 2 address of profile 1:

  * c xConfiguration SIP profile 1 Proxy Discovery 3: manual

  * c xConfiguration SIP Proxy 3 address profile 1:

  * c xConfiguration SIP profile 1 Proxy Discovery 4: manual

  * c xConfiguration SIP Proxy 4 address of profile 1:

  * c xConfiguration SIP profile 1 1 LoginName authentication:

  * c xConfiguration SIP profile 1 1 password:

• Telepresence EX60 with CUCM 8.5 - How to integrate?

  Good morning people,

  I'm building a lab with 8.5 CUCM and presence Server 8.5

  I'll install the Cisco TelePresence EX60, and I wonder how I do integration with CUCM.

  Need to install the additional server? You simply add a SIP device in CUCM?

  Can someone help me?

  Thank you

  Bruno

  Hello.

  You must have CUCM 8.6 in order to utlize the full potential of the CUCM / integration EX60. You save the unit directly to the CUCM.

  I recommend you take a look at this guide, after that you did with your laboratory update.

  http://www.Cisco.com/en/us/docs/Telepresence/endpoint/codec-c-series/TC5/administration_guide/cts_endpoint_administration_ucm86_quick_start_guide_tc50.PDF

  I have attached a screenshot showing how it should look like the description of the device.

  Thank you

  Marius

• 320 TS Cisco with CUCM integration

  Dear experts,

  Is there a way we can integrate the TS with CUCM 320 without any conductor.

  Means of SIP trunk or as conference bridge.

  BR

  Hi Muhammad

  Telepresence on 310/320 Media Server and Cisco TelePresence Server on a Virtual Machine do not support managed locally; mode on these platforms, the telepresence server must be managed by a system as a conductor of Cisco TelePresence

  Conductor for the unique conference bridge is free.

  BR Oleksandr

• can't make an external call with CuCM and H323 gateway

  Hi experts, I have a problem with CuCM and h323 Gateway. I use a 2811 router to make a gateway H323, CuCM server and Switch layer 2 build a VoIP LAB, this my network:

  

  and my router configuration:

  Current configuration: 2335 bytes

  !

  version 12.4

  horodateurs service debug datetime msec

  Log service timestamps datetime msec

  no password encryption service

  !

  router host name

  !

  boot-start-marker

  boot-end-marker

  !

  forest-meter operation of syslog messages

  !

  No aaa new-model

  !

  !

  !

  dot11 syslog

  IP source-route

  !

  !

  IP cef

  DHCP excluded-address 192.168.1.1 IP 192.168.1.10

  !

  IP VOICE dhcp pool

  network 192.168.1.0 255.255.255.0

  option 150 ip 192.168.1.2

  default router 192.168.1.1

  !

  !

  No ipv6 cef

  !

  Authenticated MultiLink bundle-name Panel

  !

  !

  voip phone service

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  Fax protocol cisco

  !

  !

  !

  vocal h323 class 1

  H225 timeout tcp establish 3

  !

  !

  !

  translation of the voice-rule 1

  rule 1 / ^ 9.

  !

  !

  voice translation-profile PSTN

  translate 1 called

  !

  !

  voice-card 0

  !

  !

  application

  Service flash aa1: its - cisco.2.0.2.0.tcl

  paramspace English language en

  paramspace English index 1

  Param aa-driver 0839959204

  paramspace location English flash:

  prefix English paramspace

  !

  !

  !

  Archives

  The config log

  hidekeys

  !

  !

  interface FastEthernet0/0

  IP 192.168.1.1 255.255.255.0

  automatic duplex

  automatic speed

  H323-gateway voip interface

  port of link voip H323-gateway 192.168.1.1

  !

  interface FastEthernet0/1

  no ip address

  Shutdown

  automatic duplex

  automatic speed

  !

  IP forward-Protocol ND

  no ip address of the http server

  no ip http secure server

  !

  !

  !

  control plan

  !

  !

  !

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  !

  voice-port 1/0/0

  !

  voice-port 0/0/2

  !

  voice-port 0/0/3

  !

  !

  !

  1010 Dial-peer voice pots

  Service aa1

  incoming called number 08395959204

  direct line to inside

  port 0/0/0

  !

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  !

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  !

  !

  !

  !

  Line con 0

  line to 0

  line vty 0 4

  Cisco password

  opening of session

  !

  Scheduler allocate 20000 1000

  end

  I have configured the gateway h323 and attach files to router parttern on follow CUCM

  What is the problem and how to solve?

  Thanks for all the help.

  Your DP pointing CUCM is wrong, or your ÉRA OPX, depending on what you need.

  voice-port 0/0/0

  cover of the echo - cancel 32

  connection ERA opx 3100

  Description 08395959204

  activation of the caller ID

  Dial-peer voice 10 voip

  destination-model 1...

  Setup progress_ind allow 3

  progress_ind enable progress 8

  h323 voice-class 1

  session target ipv4:192.168.1.2

  DTMF-relay h245 alphanumeric

  Codec g711ulaw

  No vad

  Your DP is waiting for 1... and you send 3100, they must match.

  You might need to change the significantt numbers and entering CSS, but only you can know

  For outgoing, you are throwing 9 predot, but again, expect it in your RFP, again, change as you need.

  voice pots Dial-peer 2020

  PSTN description

  translation-profile outgoing PSTN

  destination-model 9 t

  Setup progress_ind allow 3

  progress_ind enable progress 8

  port 0/0/0

  Forward-digits all the

  You cannot transfer the numbers all send you also 9 for each call

  Change your configuration you want / need.

  HTH

  Java

  If it helps, please note

  www.Cisco.com/go/pdihelpdesk

• VCS Express with CUCM 10

  Hello

  There is a lot of confusion on Express C/E & VCS C/E, if anyone can confirm my following queries:

  With VCS VCS & Express control every endpoint to register with VCS control (no participation required CUCM or VCS control can't integrate with CUCM)?

  With Highway Core & edge that every endpoint is record in CUCM & CUCM will record Expressway Core?

  For calls B2B & movi Jabber, that we must have Expressway Center/periphery, VCS control/Express IS NOT support jabber movi & called B2B?

  Software Express C/E & VCS C/E are the same, but the licenses are different?

  Our clients have obtained

  1 x control VCS licensed

  1 x Express VCS under license

  1 x ISDN gateway,

  1 x TP driver

  1 x TP management suit.

  Kind regards

  Francisco

  With VCS VCS & Express control every endpoint to register with VCS control (no participation required CUCM or VCS control can't integrate with CUCM)?

  Yes, no CUCM is required, endpoints can also enroll in the VCS-E or it could register SIP to VCS - C proxy endpoints. A parallel installation of VCS CUCM with a trunk should also work with todays versions.

  The VCS will be only support standards basis of endpoints SIP and H323, its main purpose

  a for video settings. You won't be able to use it to register, provision or use of Cisco IP phones with it.

  With Highway Core & edge that every endpoint is record in CUCM & CUCM will record Expressway Core?

  See the C & E highway as a single unit to the external communication. The registrations are transferred by proxy

  CUCM outside.

  But that also means, you can't register 3 party at the highway and even end points

  Cisco endpoints must support the installation of the fast track C & E.

  The highway also runs things like XMPP for Jabber.

  For calls B2B & movi Jabber, that we must have Expressway Center/periphery, VCS control/Express IS NOT support jabber movi & called B2B?

  The C & E Highway can also manipulate calls business2business or another sip trunks / h323, but this infrastructure requires the media license.

  B2B calls also works very well with the installation of VCS.

  If you are talking about how to provide Jabber accounts and video Jabber:

  There are video Jabber (which is implemented by TMS and VCS) and Jabber (i.e. led CUCM) there.

  If you are talking about outside calls that he can just be regarded as regular b2b external calls.

  then they be it work with VCS or motorways on the site of the end.

  Software Express C/E & VCS C/E are the same, but the licenses are different?

  It is correct. Licensing Expressway is more attractive for users CUCM, but you already have

  have SPRDD in place that should work in all cases well.

  With CUCM, you get licenses Expressway in any case. So according to the type of use, you.

  B2B, internal calls... you can add a configuration of fast track to Jabber and the stuff of office home and

  use the VCs to B2B and part 3 records.

• Range of Ports RTP endpoints TC changed after registering with CUCM

  Hello community,

  We have SX80 endpoints running TC7.3.2 and registered at CUCM 11.

  By default the TC ports rank 2326-2487 UDP RTP endpoints. After registering with CUCM, the range of ports have been changed to 16384 to 32766. In my view, WHAT CUCM change the codec settings.

  rtp_ports.PNG

  

  Based on the http://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/codec-c-series/tc7/administration-guide/administering-endpoints-running-tc72-on-ucm1051.pdf guide I couldn't see any setting to do with RTP ports.

  This is a problem for us, because on the firewall rules, we have configured and fixed ports for points endpoints use the default go 2326-2487 UDP. Because these endpoints and video infrastructure (servers of telepresence, TCS, VCS) are on different networks separated by firewalls. At present, when recording and join the conference bridge there are no media because ports have been changed. When I change the default port range, there is no problem with the media. I can record and call to the conference bridge hosted by a TP server successfully. And if I hit the button refresh on the web GUI of the codec, 16384 to 32766 changes the settings.

  Y at - it a setting that I could check on the side of CUCM?

  Thanks in advance.

  «By default the endpoints ports rank 2326-2487 UDP RTP TC.» After registering with CUCM, the range of ports have been changed to 16384 to 32766. "I believe THAT CUCM change the codec settings".

  Yes, it's true. Once your video endpoint is configured and registered on the CUCM, the CUCM will be more than just the default settings of your endpoint. It depends on the SIP profile has been used and applied for your endpoint. You need to check that the configuration of the SIP profile of the CUCM applied for your endpoint. In your case if it's SX80, check the SIP for SX80 profile and change the start and stop media ports. It comes to the RTP port range.

  In your case:

  Port of bootable media: 2326

  Stop Media Port: 2487

  rtp_cucm.PNG

  

  Kind regards

  Acevirgil

• Voicemail configuration base with CUCM and connections of the unit

  Hello again,

  I'm still very new to CUCM.  I am configuring a CUCMB 6.01 test deployment.  He has connections in the integrated unit.  I have 2 DNs assigned to users and Cisco IP phones.  I use MGCP (Cisco 3745 with 2FXO as gateway).  Now, I'm trying to extensions of voicemail installation for the users.  I read a lot of documentation on CUCM and unity, but I know that Unity Express is not what I need, and that you have not found information on the basics of implementing voicemail for users of CUCM.

  Users are DN 1001 and 1002 respectively.  They are part of a list of hunting (broadcast) so that the arrival of incoming calls, both phones will ring simultaneously.  Not sure if this has any bearing or not.  I would like to be able to put in place so that each DN rings on the IP phone say 6 times, then go to a Unity voicemail box configuration for that user.  I have already imported users 2 in the unit.  I ran the voice messaging port configuration wizard.  Everything looks as if it runs correctly, but I'm stuck really share say CUCM to route the call to the mailbox of the unit.  Probably the simplest part.  Ideas or help on this would be much appreciated.  Thank you in advance.

  Impetuous salvation,

  CUCM works this way

  So the fact that 1001 and 1002 receive calls via a group has a significant impact on this config.

  You must keep in mind that a call that is routed through the fighter pilot will always ignore the CFWD settings on phones of hunting member of the group. The normal way to control it then is through disconnection of the Group Hunt members phones combined with the front fighter pilot level settings.

  The following section describes how the parameters to call it forward on the phones of individual members Hunt are ignored when presented a call via the hunting feature. Here's a clip;

  Hunting and call forwarding

  The concept of hunting differs from call forwarding. Hunting allows Cisco Unified Communications Manager to extend a call to one or more lists of numbers, where each such list can specify an order of hunting which is selected from a fixed set of algorithms. When a call extends over a part of hunting of these lists and the party does not answer or is busy, hunting resumes from the next part of hunting. (The next party of hunting varies according to the current hunting algorithm). Hunting has so ignore settings call forward No Answer (CFNA), call forward busy (CFB) or call before all (CFA) for the party to attempt (group member line phones).

  Cisco Unified Communications Manager offers the possibility to redirect a call when hunting fails (that is, when the hunt ends without a part of hunting to the respondent, due either to exhaust the list of numbers of hunting is to expire). If used, this final redirection includes an automatic call forwarding action. Therefore, the fighter pilot Configuration window includes call forwarding configuration concepts which are similar to those found on the directory number Configuration window.

  Example of a call hunt

  Although hunting is different from the transfer, Hunt often develops as a call that gets forwarded to a certain number of fighter-pilot. The protective cover of appeal extends from hunting to allow the final drive after two exhausts of hunting or expires.

  A typical call calling hunting can include the following phases:

  The appeal of hunting through groupings of posts set up according to the algorithms implemented for each group. Hunting is successful (if it meets a hunting party), exhausts (if all parts of hunting are tempted, but no response), or expires (if the deadline set by the Maximum hunting timer runs out before that all parties are tempted, and none of the parties who have tried to answer).

  For the purpose of this example, we assume that hunting does not succeed.

  If a form any of the final drive is configured, the call forward for a next destination; otherwise, the call is released.

  Maximum hunting timer

  The Maximum Hunt Timer field on fighter pilot Configuration window allows the administrator to enter a value (in seconds) to limit the hunting through a list of hunting season. After gaps in the time specified, if hunting was not successful, the call is transferred to a voicemail system, a specific compound number, or some personal treatment (if configured) or the call is released.

  The good doc.

  http://www.Cisco.com/en/us/docs/voice_ip_comm/CUCM/Admin/6_0_1/ccmsys/a03rp.html#wp1078922

  If the call goes unanswered or all Member phones are disconnected, then an attacker needs to be applied at the level of the "fighter pilot".

  Hunt the settings forward

  Before hunting without an answer - when did not answered the call which is distributed through the list of hunting at a specific time, this field specifies how to transfer the call.

  Destination this parameter contains the number of the directory to which calls are forwarded.

  Before busy hunt - when the call which is distributed through the list of hunting is busy at a precise moment, this field specifies how to transfer the call.

  Destination this parameter contains the number of the directory to which calls are forwarded.

  Maximum Timer Hunt - enter a value (in seconds) that specifies the maximum duration for hunting. (Used in conjunction with Forward busy hunting)

  In your case, it is best to configure the DN of a fighter pilot in the extension of alternatives under the 1001 and 1002

  mailbox.

  I hope this helps!

  Rob

• Integration of TANDBERG 1700 MXP configuration with CUCM (h.323/SIP)

  I tried to configure a CUCM with TANDBERG 1700 MXP 8.5 and I could not, if someone has already done and help me with this number SIP or H323.

  Thanks in advance

  Hello

  Mxp units can be saved on cucm as endpoints (base) of third parties.

  Please find the steps to enter mxp.

  The sip on mxp as uri and sip proxy address settings.

  1. sip mode: on

  2 SIP Uri: [email protected] / * / of cucm for example: [email protected]/ * /.

  3 server discovery: manual

  4. server address: ip address of cucm.

  5 server Type: cisco

  6. transport: whether auto snack but who do not work, change to udp.

  7 TLs: off

  8: authentication sip username: user you created on cucm eg 9011 or amlesh

  9 password: use the same password configured when creating user 9011 or amlesh

  CUCM config:

  1. Select the basic profile of third party.

  2. Add the mac address of endpoint mxp.

  3. Select the appropriate sip, phone etc models profile.

  4. under user digest and owner user id select the same user, for example: 9011 or amlesh

  5. save the configuration and assign DN even that we used in the uri on mxp sip settings.

  6. also associate device mxp the user using the association on cucm devices pane.

  Please let me know if that work for you. I tried this in my lab and works.

  Thank you

  Amlesh