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Laura Douglas
Level 6
Level 6

These questions were asked by attendees during the "How to Migrate to Cisco Unified Communications from Legacy Phone Systems" Workshop.  Over 125 questions were received and the team is working on providing responses to all of them.

This document will be updated as the questions are answered. Subscribe to this post to get notified via email when updates are made.  You can subscribe to this entire "Migration Corner" space  by clicking on the "Receive email notifications" or "View feeds" links on the Migration Corner Overview page.

If you didn't make it to the webcast, but are interested in seeing the replay then you can watch the replay on the Cisco Knowledge Network web site.

Question 1: What is the primary factor that decides the migration "path" a customer ends up choosing?

Answer 1 : Typically this is a business decision as opposed to a technical one. For example a customer may need to perform extensive infrastructure upgrades i.e. adding Power over Ethernet, prior to rolling out IP Telephony and budget constraints may dictate that such an upgrade be carried out over numerous phases. Similarly, the cost of deploying IP Telephony may also dictate a phased approach as opposed to a parallel/flash-cut approach. 

Question 2: Which is the "right" method - phased or parallel?

Answer 2: There is no "right" or "wrong" answer as to which migration path is the correct one. Rather it is dependent upon the customers business objectives/priorities. One customer may be faced with a piece of equipment that has since gone "end-of-support" and needs to move onto another solution quickly thereby leaning more towards the parallel/flash-cut approach versus another customer who may need to move specific "communities of interest" ahead of others and would therefore choose the phased approach. Again, there is no "right" or "wrong" answer when it comes to which way a customer would choose to migrate.

Question 3: How can traffic studies be conducted on Cisco IP Telephony products, such as Call Manager, to to determine the traffic load in CCS or Erlangs on trunk groups such as PRIs?

Answer 3:  Cisco Unified Communications Manager (CUCM), along with other serviceability components, provides the CDR and Reporting (CAR) tool, which allows you to generate traffic reports.

Question 4:  How can we implement Cisco Contact Center without CUCM in place?

Answer 4:  There are multiple ways to implement a Cisco Contact Center without CUCM in place.

Cisco Unified Intelligent Contact Management (ICM) supports peripherals such as private branch exchange (PBX), ACD, IVR, or Web or e-mail server.

This integration is done though the Cisco Unified Intelligent Contact Management (ICM) Peripheral Gateways (PG).

A list of which ACD  releases are currently supported by Cisco Unified Intelligent Contact Management (ICM) Peripheral Gateways (PG) is available at:

http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/icm_enterprise/acd_supplements/icmacdmx.pdf

Cisco Unified Customer Voice Portal (CVP) can also be deployed to provide IVR and queuing functionalities and transfer calls without CUCM in place, using the standalone VXML or VRU only functional deployment models.

With Cisco Unified Contact Center Express, either CUCM or CUCME could be deployed.

Question 5:  Can you explain more why qsig is better and pls specify which protocols you are comparing with?

Answer 5:  QSIG is an established protocol that has had a lot of its interoperability issues resolved whereas H.323 and SIP still have some way to go in this area - please visit the Cisco Interoperability Portal which can be found at the following address http://www.cisco.com/go/interoperability for examples of various PBX's along with a selection of protocols for interoperating with Cisco Unified Communications Manager.

Question 6: You mentioned to start with small branches, then migrate the head office, however, most of our clients start with the core (head office), then move to the edge (branches), what is your input on this?

Answer 6:  It depends on the needs of the customer. For some, it may be best to convert the smaller offices first, get that right, and then move to the larger locations. For others, there may be a business need to do the larger offices first. 

Question 7: Is there a particular path which was better for establishing "hard ROI"? Was there a "hard ROI" in a phase-in approach?

Answer 7:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 8: How well does unified messaging work with Novell Groupwise?

Answer 8: Unity supports Novell Groupwise clients; however, this configuration places additional load on the Exchange message store server(s).

Question 9:  How did you figure out the Licensing for the Sites that converted to Unity Voicemail before the PBX's? Did they have the VM licensing only or bought CUWL? What did they do after the upgrade with all the PIMG's it adds up :=)

Answer 9: Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 10:  What is the average bandwidth in Kb used when in a conversation on VoIP?

Answer 10:  It depends on the codec used for the call. If G.711 is used, bandwidth used will be around 80Kb. When G.729 is used, each call use around 30Kb per call.

Question 11:  Whats the difference between SIP/H.323 and Qsig trunking?

Answer 11:  QSIG is a signaling protocol originally used on ISDN trunks. Normally, referring to QSIG trunks implies TDM-based trunk technology. SIP and H.323 are signaling protocols based on IP technology. Please note that QSIG can be "tunneled" over H.323 trunks (H.323 Annex M) and SIP (ECMA-355).

Question 12:  What release of software does Cisco require for the Nortel Opt. 11c PBX to have?

Answer 12:  It depends on the type of integration you want.  For traditional ISDN PRI protocols (DMS-100, NI-2, 4ESS/5ESS), I think older Meridian 1 releases should support it. For QSIG, I think at least Nortel Rel 24+.  For SIP you'll need at least Rel. 4.0. Please visit the Cisco Interoperability Portal, URL is posted above, for specific details on PBX software releases.


Question 13:  Do you often find Cisco UC overkill for some small-medium businesses needs, how do you address this?

Answer 13:  Cisco addresses requirements specific to small-medium businesses. You can find more information on this at: http://www.cisco.com/iam/unified/ipt701/SMB/UC_Solution_for_Small_and_Medium_Businesses.htm

Question 14:  Based on your experiences, during deployment, what unforseen issues have created the largest problems, from both the customer's and the installer's perspective.

Answer 14:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 15: What is the best practice for emergency operations phones that need to remain in service  on the same number during cutover?

Answer 15:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 16:  What is the best method for migrating from a Nortel Symposium contact center to a Cisco Contact Center Manager?

Answer 16:  I believe you're referring to Cisco Intelligent Contact Management (ICM) software, which is a component of Cisco Contact Center offering. Customer's requirements and environments dictate what is the best migration method. Contacting your Cisco account manager will be helpful in determining the best migration strategy for your company. You can also refer to the Contact Center Solution Network Reference Design Guide for more information on Cisco Contact Center Enterprise: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/ipcc_enterprise/srnd/75/c7overvw.html

Question 17:  is there documentation showing how to integrate Unity with an NEC 2400 pbx?

Answer 17:  Check out the Cisco interoperability Portal at www.cisco.com/go/interoperability. It has numerous documents showing how to integrate a variety of vendors PBXs, including NEC, Avaya, Nortel, Siemens, etc to Cisco Unity.

Question 18:  if PBX doesn't support QSIG, what's the next best one?

Answer 18:  When using ISDN PRI trunks, I'd say using NI2, DMS-100, or 5ESS protocol, which are standard ISDN protocols just about all PBX's supporting ISDN should be able to support. With any of these protocols, bi-directional calling/called name delivery should be expected but please refer to the Cisco Interoperability Portal for more information.

Question 19: Can you use Cisco Contact Center Manager on a Nortel system?

Answer 19:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 20: Any issues using SIP trunking versus PRI trunks?

Answer 20:  Using SIP trunks may limit certain advanced features such as Path-Replacement, Reroute as a lot of the legacy PBX vendors are playing catch up with respects to SIP. Migrating legacy PBX to perform SIP functionality can result in such features not being available. CUCM supports both protocols.


Question 21:  What, if any, tools are recommended to assist?

Answer 21:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 22: Does CUCM observe Ecma or ISO standards for qsig

Answer 22:  CUCM supports both ECMA and ISO QSIG standards.


Question 23:  Are there any wireless handsets/equipment available? Cellular is not an option here.

Answer 23:  Cisco offers the 7921 wireless handsets.

Question 24:  What changes are required for the enterprise network, besides adding voice enabled routers to make sure that adding voip traffic doesn't result in data latency.

Answer 24:  Basically, making sure that Quality of Service (QoS) is implemented in order to ensure call quality.

Question 25: Are there different license requirements depending on implementation?

Answer 25:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 26: In reference to Bally's answer about bandwidth, is that 90k per leg of the call?

Answer 26:  Regarding bandwidth consumption per voice call, whenever using G.711 codec, each call will use around 80-90Kb of bandwidth. More information on this topic can be found at: https://communities.cisco.com/docs/DOC-6303/edit?containerType=14&container=2787

Question 27:  My understanding of a full UC system also includes Exchange, will this topic be discussed sometime during the presentation?

Answer 27:  I don't believe we will get to this level of detail during the presentation....Depends. Cisco Unity voicemail can use Exchange as its repository for messages. Cisco Unity also supports IBM Lotus Notes. Cisco Unity Connection uses a propritary message store. 

Question 28:  What does Cisco recommend for monitoring performance and QoS?

Answer 28:  Cisco CUCM comes with serviceability tools such as the CDR and Reporting (CAR) tool. This tool allows you to monitor system traffic and QoS. To monitor system/components performance, Cisco's Real Time Monitoring Tool (RTMT) can be used.


Question 29:  Is there a way to avoid H.323 Gateway? I mean is it possible with out H.323? if yes then which router?

Answer 29:  A gateway is necessary in order to connect your VoiP platform to a TDM-based platform. You can use H.323, SIP, or MGCP as the gateway protocol. 


Question 30:  Are the wireless handsets IP/WiFi based?

Answer 30:  Cisco 7921 wireless handsets are 802.11a/b/g based.

Question 31:  In a pure 10 digit dialing scenario (all location DNs are 10 digit) is there a need for Trunk Access codes?  Pros or Cons going with no access codes after the migration?

Answer 31:  Dial plans are dependent upon the network configuration and if this is a private network then there is no limitation on what dial plan is used. However I am sure that at some point there needs to be a way out to the public switched telephone network, and this will require an access code, such as "9".


Question 32: CDR and Backup 7.x Why do i not see the option to uncheck CDR to not back it up

Answer 32:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 33:  Most of our currently deployed Call Manager offices have single points of failure, a single router or single switch.  Do you have a check list or single document that show what you need to provide 5-9s reliability for a small office?

Answer 33:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 34: Is attendant console being phased out?

Answer 34:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 35:  Does it make any sense to have SRST in a LAN?

Answer 35:  If the LAN is stable and has redundancy, SRST is not required. SRST is normally used over WAN links. 

Question 36: I'm having a disconnect, in a total IP centric dial-plan, why can't the callmanager determine the route pattern whether it is a PSTN call or internal?

Answer 36:  That is a way around and there is no issue with that, as long as the internal dial plan uses a standard DP with respects to local, long distance and international calls.

Question 37:  we currently have over 7000 users is it possible to transfer station info from one  system to the other

Answer 37:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 38:  For large analog installations like hospitals, is there a better solution than a stack of VG224s?

Answer 38:  There is a higher density analog gateway, the 48-port VG248.

Question 39: The previous screenshot/document , explaining all the phone features, is available somewhere into the cisco website ?

Answer 39: A detailed explanationn of ip phone features can be found at: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/phone_a_to_z/english/user/guide/az_user.html

Question 40:  When proposing solution to customer, how does Cisco "guesstimate" the cost for install of unified Communications, as well as complex integration of current PBX and special applications proposed.

Answer 40:  I think you need your Sales Team engineer to understand the network and the customers requirements. If it includes VOICE then obviously this is more costly.

Question 41:  What would be the direction to take so as to not have to replace the current VoIP phones when migrating to Cisco.  Would you recommend SIP phone registration for all sets ?

Answer 41: Yes, as long as CUCM can support the VoIP phones. Some third party SIP phones are supported by CUCM and there is a list so talk to your account manager to obtain the list of supported thirdy party phones.


Question 42:  Is the voice quality between G.711 and G.729 a noticable difference?

Answer 42:  There is a slight difference in voice quality. G.711 is better, that but it's not that much different, IMHO. Voice quality is subjective , however.

Question 43:  So how does Cisco provide guidance and budgetary cost to customer? % of equipment? or other models?

Answer 43:  Normal situation would be that we Cisco are in a competitive situation with other vendors. The key advantage we have when working with customers that support legacy equipment is all the interop testing that we have done and verified as working. Our portal helps sway vendors just based upon the application notes that we publish.


Question 44:  how long will cisco use proprietary POE modules vs the 802.3af based POE modules. Will all cisco phones work with both standards?

Answer 44:  Cisco has been offering 802.3af-based modules for several years now. All newer Cisco IP phones support 802.3af

Question 45: i hear something about CCME as SRST IOS images....whats that about?

Answer 45:  Both Cisco Unified SRST and Cisco Unified Communications Manager Express in SRST fallback mode provide survivability at remote sites when the WAN link between the central site and the remote site fails. CCME and SRST do not provide the same level of features, however. For more information on this, please access the SRST Q&A found at: http://www.cisco.com/en/US/prod/collateral/voicesw/ps6788/vcallcon/ps2169/prod_qas0900aecd8028d113.html

Question 46: what is cisco solution for wireless ip phone deployment within a site.

Answer 46:  Cisco offers the 7921 wirelss handsets.

Question 47:  Can you create failover for your remote sites over VPN tunnels if the dedicated circuit fails?

Answer 47:  if you route calls over a vpn connection you have to make sure the link has qos configured or your audio qualtiy will be terrible. 

Question 48: besides the number of Ports what would be the advantage of using the VG224 over an ATA186

Answer 48:  The VG224 analog gateway is IOS-based; offers higher density, better administration and fax/modem relay support.

Question 49:  Do you recommend deploying the 7.x version of CUCM or shall we play it safe and stay with 6.x for now?
Answer 49:  Cisco CUCM 7.X is the latest available version. This version has been around for considerable time, and is considered stable. 

Question 50:  In supporting non-Cisco phones, do we still need to worry about the DLUs or has that been replaced by better licensing?

Answer 50:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 51: We are a Nortel shop with approx 60 sites.  We have started our migration to Cisco with 3 sites on Cisco.  Does Cisco Unity offer "express messaging" (direct to users mailbox) from a external number or calling from a Nortel phone from a remote site?

Answer 51:  Yes, Unity enables you to send a message directly to the recipient as an option, without the need to login as a subscriber.

Question 52:  Bob Close, re: your western bank migration. Did you migrate from Siemens phone mail to Unity? MWI worked OK?

Answer 52:  Yes it did - the integration was done using analog pots lines with DTMF as the sgnalling protocol. It provided basic "features", including MWI, but it did work.


Question 53:  have many migrations been done on IPv6?

Answer 53:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 54:  What version CUCM will G.722 be supported?

Answer 54:  G.722 is part of the phone features, so it depends on the model of phone but I believe 7.X will support G.722.  More clarification: G.722 was supported by CUCM long time ago, at least from 4.x. Not sure which phone model supports G.722 but if it does, the phone will report that capability to CUCM. CUCM has an option to turn off G.722, however.

Question 55:  What is the typical customer adoption of softphone endpoints as opposed to hardware VoIP phones?

Answer 55:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 56:  on one of your slides, you show that skinny is used btw the endpoints and the router with SRST, is it possible that SIP is used between the endpoint and the router

Answer 56:  Cisco SIP phones can work in "fallback mode" when losing connectivity to CUCM; however, I believe that using SIP phones in SRST mode is not supported.

The local SIP gateway that becomes the SIP registrar acts as a backup SIP proxy or redirector and accepts SIP Register messages from SIP phones. It becomes a location database of local SIP IP phones that are set up for dual registration. Dual registration allows SIP IP phones to simultaneously register with both their primary and their fallback registrar devices. That is, when a SIP IP phone registers with a Cisco Unified SIP SRST gateway, it simultaneously registers with the main proxy and SIP redirect server for coverage in case of a WAN failure.

Question 57:  When in SRST mode what features/functionality is lost as compared when everything is working normally

Answer 57:  Whenever IP phones operate in SRST mode, only basic phone functionality is provided. IP phones will be able to place/receive calls, perform transfers, call forward and three-party conference calls.

Question 58:  I have heard that many networks are starting to use SRST as a standalone, without it acting as a failover, but as the main system. Can you comment on this?

Answer 58:  This would not be SRST, but UCME because Unified Call Manager Express can act as a stand alone CCM, wheras SRST is meant for failover scenario.


Question 59:  How do we integrate Fax machines that are on dedicated analog lines using ATA 186 adaptors?  Do you recommend a fax server?

Answer 59:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 60:  Are DSP's like cheese on pizza...you can never have too much?

Answer 60:  That is a general rule, but let's not get greedy...

Question 61:  Who maintains inside wiring currently maintened by a CPE vendor?

Answer 61:  Normally this is the service provider that is currently supporting the current PBX equipment.

Question 62: My customer is going to route the 1-800 number through the legacy system (PRI to Nortel PBX) to the the Meetingplace, and the meeting place is not connected to the Nortel PBX. What are the solutions?

Answer 62:  Information on how to connect/integrate Meetingplace to your PBX can be found at: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/4x/42mtplc.html#

Question 63:  If my carrier is using SIP trunking that they push from there PSTN to my system, and I am using QSIG  between my PBX and CM - will there be issues in call quality, etc?

Answer 63:  No issues, assuming QoS etc. is configured correctly. 


Question 64:  It's my understanding that codec G.729 is not supported for Fax, are there any plans to resolve this?

Answer 64:  in VoIP networks, when fax/modem tones are detected, they are either passed through to the remote fax/modem automatically via G.711, or are relayed using fax/modem relay mechanisms. 


Question 65: You mentioned legacy pbx requirements, can you elaborate is there any integration with legcay v/m, pbx, ivr, acd systems

Answer 65:  If you are asking if Cisco integrates with 3rd party legacy systems, the answer is yes.  Check out www.cisco.com/go/interoperability.  Shows our integration for a variety of these scenarios. 


Question 66:  If running G.722 or G.711 enterprise wide is there any reason for hardware media resources?

Answer 66: You would not need any transcoders if the codec was consistent. If bandwidth is not an issue i.e. LAN deployment, I would go with G.711.

Question 67:  Is it possible to provide line-side ISDN BRI functionality?

Answer 67:  Line side BRI would be an ST interface. I believe we do not support this.

Question 68:  LDAP direcotory integration was not recommended in v4.  Is it now a good option?  Can we share btween clusters that are in single tree, multiple domain?

Answer 68:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 69:  When configuring QoS on the infrastructure is "Auto QoS" good enough as a starting point?

Answer 69:  Auto QoS is a good starting point whenever a small-to-medium businees needs to deploy VoIP quickle and lacks the time/resources to deploy QoS Services. For additional information on AutoQoS, refer to a White Paper found at: http://www.cisco.com/en/US/technologies/tk543/tk879/technologies_white_paper0900aecd800a8561.html

Question 70:  I realize Unity offers sending messages directly to a users voice mail by preceding it with a * when calling internally, but how is this done when calling externally?

Answer 70:  It depends on how the Unity is configured.  But generally, once the external caller get the general greeting, you can enter the extension number and it will take you directly to the user's VM.

Question 71:  I have a number of questions regarding LDAP implementation.  Cisco Documention is at times too overwhelming.  What's a good source for questions and requests for enhancements?

Answer 71:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 72:  Is there a maximum number of remote site end users that should be supported using SRST versus their own subscriber? Or, what is recommended as maximum number of end users per SRST?

Answer 72:  This information can be found in the SRST 7.1 support website; specifically at: http://cco/en/US/docs/voice_ip_comm/cusrst/requirements/guide/srs71spc.html 

Question 73:  One of the slides shows the sizing for several ISR routers, what about the 3200 series that now supports Call Manager Express, what is the possible sizing on those?
Answer 73:  The 3250 can support up to 20 IP phones and a total of up to 100 DN's. The 3270 can support up to 48 IP phones and a total of 240 DN's.


Question 74:  Do we have to have QoS to implement this?

Answer 74:  Yes. Having a QoS-enabled network is preferred when deplying VoIP.

Question 75:  Is cat6 cable the recommended cable for VOIP networks? Is there any issues with the 5e cables?

Answer 75:  Cat6 has better transmission performance; however Cat5E is perfectly fine for VoIP applications.

Question 76: What is best practice when connetting to the PSTN via SIP trunk, should incoming calls be pointed to an ISR Router(Border Element) or CUCM?

Answer 76:  We only support this type of scenario with CUBE (Cisco Unified Border Element), which runs on an ISR. 


Question 77:  If the customer has 8-wire GigE how is PoE handled?  All wires are used for the data.

Answer 77:  Power over Ethernet has been implemented in many variations before IEEE standardized 802.3af. 802.3af specifies the ability to supply an endpoint with 48V DC at up 350mA or 16.8W. The endpoint must be capable of receiving power on either the data pairs [Mode A] (often called phantom power) or the unused pairs [Mode B] in 100Base-TX. PoE can be used with any ethernet configuration, including 10Base-T, 100Base-TX and 1000Base-T. Power is only supplied when a valid PoE endpoint is detected by using a low voltage probe to look for the PoE signature on the endpoint. 

Question 78:  Bob: Because the QoS will give the voice packets higher priority?

Answer 78:  Yes. Voice Over IP really requires a minimum of two queues - with one being a high-priority queue dedicated for voice - everything else goes in the other "best-effort" queue.

Question 79:  I have a question about redundancy?? In a TDM network (specifically NEC PBX) redundancy is in the CPUs (dual processors/power/time division ...etc) What or how is the  redundancy set-up or designedured in a Cisco VOIP Network??

Answer 79:  CUCM provides redundancy via failover servers: ip endpoints/gateways are configured to home to secondary/tertiary servers in case of primary CUCM server failure.


Question 80:  Implementing PoE cards could be a substantial cost in itself. What % of customers opt for power supplies for each VoIP phone?

Answer 80:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 81:  Trivia question- What does ISDN stand for?

Answer 81:  Integrated Services Digital Network.

Question 82:  Can you talk about wireless handsets? do they really work? what problems do they have?

Answer 82:  We did a two part series "Collaboration Strategies in a Web 2.0 World: Better Collaboration on the Go Parts I & II" that may answer your questions.  http://www.ciscoknowledgenetwork.com/uc/archives.php 

Question 83: Can someone address the differences between using Cat 5E vs. Cat 6 for IP Telephony cabling?  What are we giving up by staying with Cat 5E for our future VOIP environment?

Answer 83:  The difference between cat5e and cat6 is in transmission performance. Cat5E is perfectly fine for VoIP applications.


Question 84:  Linux isn't susceptible to viruses?

Answer 84:  Yes it is, but due to the way we deliver this solution i.e. an appliance model, plus we include the Cisco Security Agent as well as implementing good security practices, we see very little if any issues.


Question 85:  Why does slide state "No NAT across Internet"

Answer 85:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 86:  Why would you limit calls on a per-site basis?

Answer 86:  Save bandwidth resources.  Being a packet-based solution each additional call would be allowed access to the IP circuit which would have a negative impact to the others. We stop this from happening by sizing the IP WAN correctly for the expected voice traffic then us Call Admission.  Control (CAC) to either block subsequent calls or divert them over the PSTN.

Question 87:  Is there a good QoS presentation available for Enterprise?

Answer 87:  www.cisco.com/go/designzone contains a large amount of information which also covers QoS.


Question 88:  What is the typical customer adoption of softphone endpoints as opposed to hardware VoIP phones?

Answer 88:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 89:  In reference to the 30,000 lines, Does that mean 15,000 phones with dual lines?

Answer 89:  Not necessarily. You will need to contact your Cisco Partner/Account Team in order to size Cisco Unified Communications Manager as a number of factors influence sizing. 

Question 90:  Will SRST automatically route internal calls to PSTN when the WAN is down ?

Answer 90:  Yes, that is one of the functionalities of SRST.

Question 91:  What is the typical customer adoption of softphone endpoints as opposed to hardware VoIP phones?

Answer 91:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 92: What is the recommended number of phones per subscriber?

Answer 92: Depends on server type. For the high end MCS-7845-H2/I2 server 7,500 phones is the maximum. You must use 1:1 redundancy when more than 7,500 IP phones are registered on the two primary subscribers because there cannot be more than 7,500 backup registrations on a single backup subscriber. With that in mind you could register 2,500 per server so that at any one a failure would not exceed the max a server could handle.

Question 93: Once WAN connectivity has been restored, will it automatically re-establish from the PSTN?

Answer 93:  Once WAN connectivity is re-established, IP phones operating in fallback mode (registered with the SRST router) will automatically revert back to CUCM. 

Question 94:  What is the acronym IDS and who developed and supports it?

Answer 94:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 95: Is it possible to put a TFTP server at a remote location and have the phones at that site source their information from that server?

Answer 95:  Yes, it is possible but we do not necessarily recommend this.

Question 96: I normally dial 205 to get a user in my home office. The WAN goes down so my phones fails over to the SRST router. Can it now automatically send my call to 205 out the PSTN and dial 123-456-7890 ?

Answer 96:  SRST supports the alias command, which provides a mechanism for rerouting calls to telephone numbers that are unavailable during fallback. Up to 50 sets of rerouting alias rules can be created.

Question 97: What is the optimum load on a subscriber?  80% of 7500?

Answer 97:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 98: Can CUCM consist of only one server playing the roles of pub/sub? and if we have only 2 servers can the pub be the backup of sub?

Answer 98:  Yes you can have a PUB and SUB running on one server, and in the other case you can have a separate PUB and SUB and if SUB fails PUB will backup.  Also another note if you use a PUB/SUB on the same node it will not offer any redundancy.

Question 99: please review losing publisher with Call Manager version 4 vs CallManager version 7. thanks

Answer 99:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 100: Why we need to CSS, one under the phone and one under the DN?

Answer 100:  Having to enter CSS settings on both at the device and at the DN level is not required. Line/Device CSS'es work together; however, the line CSS takes precedence over the device CSS. When both line and device CSS are configured on phones, the two CSS'es are concatenated and CUCM places the line CSS in front of the device CSS.

Question 101: Will there be any discussion on the distinction between the "traditional CSS" approach and the "Line/Device" CSS approach?

Answer 101:  Line/Device CSS'es work together; however, the line CSS takes precedence over the device CSS. When both line and device CSS are configured on phones, the two CSS'es are concatenated and CUCM places the line CSS in front of the device CSS. If route pattern/DN appears in two partitions, one contained in the line CSS and the other contained in the device CSS, CUCM selects the route pattern/DN that is listed first in the concatenated list of partitions, which in this case would be the route pattern associated with the line CSS.

Question 102: What is the fail-back process from SRST to central CM after the WAN comes back up?

Answer 102:  Upon WAN link re-establishing, idle IP phones automatically rehome to central Call Manager when CCM responds to keep alives.  Active IP phones stay homed to branch router till call ends, and then rehome to central Call Manager.

Question 103: What about support for Novell E-directory ?

Answer 103:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 104: Is there anything that needs to be done with the schema to integrate with Microsoft Active Directory?

Answer 104:  Cisco Unified CM 7.x relies on two separate components to satisfy the user provisioning and user authentication requirements independently. No changes to AD schema is necessary. 

Question 105:  Does LDAP Integration require an AD schema update?

Answer 105:  No. We simply copy the LDAP DB to CUCM and then add our extensions where necessary - the corporate AD remains untouched.

Question 106:  Can you sync the CUCM vs the Global catalog?

Answer 106: Yes. When you enable LDAP authentication with Microsoft Active Directory, Cisco recommends that you configure Unified CM to query a Microsoft Active Directory Global Catalog server for faster response times.

Question 107: Are you aware of anyone authenticating against Novell edirectory/NDS using LDAP?

Answer 107:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 108:  So there is no integration on for a Novell based network ?

Answer 108:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 109:  I know that you can sync with "OU's" in LDAP directories from 6.X and up, I seem to remember there was a limit to how many OU's could be synced with.  Is this still the case?

Answer 109:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 110:  Is there support for Novell's e-directory

Answer 110:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 111:  We have a few remote sites with quite a few phones (80 - 300) and things like firmware upgrades take quite a while due to how tftp works. We also only have a single T1 between our sites. It was also related to my previous question. We are looking at the best way to support our larger sites... (2 of them)

Answer 111:  Not sure what the actual question is if you are referring to the size of the IP phone firmware files and distributing them to remote WAN connected phones then you should investigate the Peer Firmware Sharing feature. This feature essentially allows one IP phone to download the firmware and then distribute it locally to the other IP phones.

Question 112:  Is the interop for basic telephony only or is there interop for ACD call event/agent event data/features as well?

Answer 112:  Our group mainly does interoperability testing between PBX and CUCM for signaling and feature transparency.  Basic calls are covered along with supplementary services and advanced network features offered by type of integration.We do not conduct testing to detemrine ACD features interoperability.

Question 113:  It seems that the VG224 is end of sale?

Answer 113: No, the VG224 is not end-of-sale.


Question 114:  does the unified communications have anything built into it that would be equivalent to Avaya vectoring or VDN's

Answer 114:  Cisco IP Contact Center products provide functionality similar to Vectors/VDN to route/process calls. 

Question 115:  can you compare  VG224, FXS port and ATA for Fax support?

Answer 115:  There is a document found at: http://www.cisco.com/en/US/tech/tk652/tk777/technologies_tech_note09186a0080159cf3.shtml#topic1 containing a matrix which illustrates differences in fax relay/passthrough support among gateways. 


Question 116: A slide was presented that details integration between Cisco and Nortel for H323.  I am looking for documentation for this as we are unable to get them talking,  Nortel CS1000M Rel4.5 to CM7.0.02.  Do you have a knowledge expert or documentation for this?

Answer 116:  We've primarily concentrated on QSIG and SIP for our more recent application notes. However, we were able to make basic H323 calls using the config contained in this link: http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/pbx/interop/notes/445274no.pdf.  Nortel uses non-standard data for services such as calling name - basically tunnelling non-standard data through h323.

Question 117:  Can Marquis contact me regarding NEC interoperability?

Answer 117:  Documents illustrating interoperability of Cisco UC platforms and the NEC 2400 can be found at: www.cisco.com/go/interoperability

Question 118:  can UCSPT be used for compliance checking?  Example:  phones not needing the PC port enabled, can a survey generate a report of current phones that do not show an "active" link on the pc port?

Answer 118:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 

Question 119:  From a post-cut standpoint, is there a way to script adding new users on an ongoing basis without using LDAP?

Answer 119:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 120:  what is the link to  UCSPT

Answer 120: Subscribe to ucspt@cisco.com to receive release updates and UCSPT news.
Send an email to UCSPT support mailer and include the detailed problem description.
ucspt-support@cisco.com.

Question 121: When building a TDM trunk between a Nortel Meridian 1 PBX and a Cisco gateway using a 3825 WAN router would you recommend 24 channel T1 or a PRI using VWIC and PVDM2?

Answer 121:  You can use either/or; however, we have tested ISDN PRI connectivity as opposed to T1 CAS, and can provide documentation, found in the interoperability portal at www.cisco.com/go/interoperability

Question 122: Can we hear more about the SOHO office using a Cisco handset ?

Answer 122:  Answer TBD.  We are currently working on getting you a response.  We received over 125 questions during the webcast, and are working as quickly as possible to respond. 


Question 123:  My Nortel uses CallPilot, can we keep CallPilot?

Answer 123:  CallPilot can be used as a centralized voicemail platform supporting both Nortel and Cisco users; it does require that it continue to be hosted by the Meridian 1 along with an ISDN PRI QSIG trunk providing connectivity between Nortel and Cisco. 

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