Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity to discuss catalyst switches and IP phones with Cisco expert Faraz Aladin. Faraz is a Senior Technical Marketing Engineer and has 12 years of networking experience. Feel free to post any questions relating to catalyst switches and IP phones.
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This is a somewhat related question.....
I have been running into LOTS of customers who standardize on hardcoding speed/duplex on most of their switches. Now anybody who has been troubleshooting AVVID stuff knows that if the phone and switch are not set to auto negotiation, this may lead to speed/duplex difference on each end of the link partner. Which in turn, leads to many misc. port errors on the switch and choppy conversations, echo, etc. on IP phones.
Finally my question.....will there ever be the option to set speed and duplex on IP phones? It is my understanding that the ability to hard code BOTH ends of a link needs to exist if you are going that route.
It's true that quite a few of the switch ports out there in production have their speed & duplex settings hard coded because auto-negotiation would not work properly with NIC cards all the time. The problem stemmed from the PHY's used in NIC cards and switch ports where the PHY's were not exactly compliant with the 10/100 auto-negotiation standard in their implementation at both ends.
Most of the newer PHY's used on switch ports as well as NIC cards do not have this issue and we have found that switches (or line cards) delpoyed in the last 1-2 years do indeed auto-negotiate properly.
Having said that Cisco is investigating the possibility of making the switch ports on the phones configurable for speed & duplex. Over the next few months we are going to look at the issues surrounding this topic and address it with an official statement that will provide an insight to customers that have ran into this problem.
Problem Description: I am trying to do some pre-design work on the new Catalyst 4224 switch. The design docs indicate that by default, the 8-port FXS blade takes 2 of the 6 DSP's, leaving four DSP's for the Digital T1 (MFT). With four DSP's allocated to the Digital T1, you can only use 16 DS0's off the T1. Is there any way to utilize all 24 channels of the T1 module and still have DSP's for the FXS blade? Can additional DSP's be added at the factory. My customer wants to have a full T1 (24 DS0's) and also have support for the 8-port FXS blade. Is there any way to do this?
You cannot add more DSP's in the field or at the factory.
Catalyst 4224 is positioned in the branch office where one would use the Multi-Flex Trunk Module to transmit (and receive) both voice & data on the same T1 from the carrier. This means that one would not have all 24 DS0's for voice. As such the implementation can still be valid if atleast 8 DS0's are used data leaving full DSP support for 16 DS0's and 8 FXS ports.
Also considering that fact that it has only 24 ports and the fact that normally there is an over-subscription between the number of users & DS0's provisioned, it is unlikely that there will be a one-for-one co-relation between the two and 16 DS0's will be sufficient for 24 users.
Having said that, if the implementation did need all 24 DS0's for any reason, then the only possibility will be to de-activate the FXS ports via SW that will become available very soon. This will release the DSP's that can be used for DS0's without support for any kind of FXS devices.
I'm in the very early planning stages for a small (200-phone) AVVID installation in an existing Catalyst 2900 switched network. I want to run voice and data on separate VLANs. Can I install the telephones and build up the VLAN infrastructure before I install any Call Managers, or is CM required to program VLAN settings on the phones? Will the switch continue to function if the CM goes down? I guess in general I'm concerned about whether I can run the Catalyst/VLAN/79xx infrastructure without CM.
You can set up the VLAN now and prepare for the AVVID installation, or set up the CM and move it into the VLAN after, The 2900 switches will work fine if the Cm goes down the data part of the network will be ok, you need the CM for the phones
Perhaps I wasn't clear when I said "switches". We're going to use the built-in switches in the phones to split the VLAN trunks from the 2900 switches into voice (internal to phones) and data (2nd switch port). I wanted to make sure they would work by themselves.
It will work. The switch inside the phone will work in terms of passing the data traffic thru it on the link from the phone to wiring closet switch as long as the phone has power and the phone has a valid VVID which would be configured on the wiring closet switch.
As long as the phone has power, the PC that is daisy chained behind it will work in the VLAN you want it to be. The phone will get its VLAN from the wiring closet switch (if VVID is configured on the switch) and will keep looking for a CM to register with (assuming it has an IP address) but it will not hamper your data networking. You can install the cM at a later time.
Hi, just have 2 questions :
1: Does Cisco plans to sell inline power "Access switches" that have more than 24 ports (referring to the 3524-PWR). Typically, i'm designing an IP Telephony deployement in a 10 stairs building, and all I can deploy at each stairs for the access is 3524 (and putting all the IP phones on the backbone Cat6Ks is not a good solution, and BTW i won't have enough slots!!!)
What's the best equipment right now for having many many FXS ports ?, (VG200 has few, routers can't hold too many... the best one I see is Cat6K with FXS cards)
Thx in advance
Hi, we are in the process of installing a voip solution for a customer. We are using the catalyst 4006 switch. It has 6 slots, 5 of which you can install 48 port inline power modules. There is also a cat4003 that will allow up to 2 48 port inline power modules.
Can't help you with the second question.
In-line power is only available on the 4006 and not the 4003 because only the 4006 chassis has the ability to accept the Power Entry Module (PEM) and also has traces on the back-plane which allow the DC power to be supplied to inline-power capable line cards. To enable in-line power on the 4006, you must have the Catalyst 4000 Auxiliary DC Power Shelf and at least two power supplies (WS-P4603-2PSU). The power shelf can accept up to three power supplies (WS-X4608) for N+1 redundancy. At least two are required for in-line power to work. Special cables (which are included with the power supplies) are used to attach each power supply to the PEM (WS-X4095-PEM). Finally, you must have an in-line power capable line card in the chassis. The WS-X4148-RJ45V is a 48 port in-line power capable 10/100 Ethernet switching module. It is similar to the daughter-card on the Catalyst 6000 module. The Catalyst 4006 switch operates identically to the Catalyst 6000 switch, in respect to in-line power detection and delivery.
Cisco is looking at a 48 port version of the 3524-PWR. This may suit your purpose but it is not in a timeframe which you can plan on. Anything more than 3-6 months is something which can't be commented on from a timeline perspective. Stay tuned.
2nd Q has to do with a high density analog gateway. This is coming in Q4 of this calendar year. It will have 48 ports and will be feature rich in terms of MWI, Conferencing, etc.
I have a customer who is trying to use FXS ports on a VG200 for traditional modem lines. It does not seem to be working. Is there a command that will make this work? Please advise.
I'm aware mine is not a very technically detailed question, but I thought you could help.
Say I want to give the possibility to people from city A to call to city B. I'd would like to implent this by using IP phones at city A and an IP-PSTN gateway at city B. Which pieces (router+VXO, call manager...) will I be needing if this can be done.
So, at Central site you will have CallManager plus IP Phones. At the remote site you won't have a CallManager which implies you will be using a Centralized Call Processing Model. In this case all you need at the remote site is a gateway (typically a Cisco router like 175X, 26XX, 36XX etc which can act as an H.323 gateway) that can support the appropriate number of FXS ports for regular analog phones and FXO ports for trunks to the PSTN. You could use digital trunks like PRI, T1 as well.
Good question. Is SIP used at all in Cisco's present Call Manager solution? Are there any indications that it may be adopted in the near future?
I'm connecting 3524XL-PWR and IP Phones without
PCs behind IP Phones.(and CCMs reside somewhere.)
When I configure the port as an access port
of the VVLAN, not as trunk,
it looks like they work fine.
switchport access vlan 6
Why the switch inside the IP Phone works fine
when the opposite 3524 port is not a trunk port?
I understand 3524 doesn't speak DTP.
Is the 3524 simply ignoring tagged frames,
and is the inside switch simply sending
tagged frames thinking that 3524 port is trunk?
In addition, the IP Phones seem to know the
VVID while I'm not configuring "voice vlan"
as above. How do they know that?
What is the recommended swtich config when
there's no plan to connect PCs to IP Phones?
Let me add few more comments.
The reason I don't make the port trunk is
because my customer wants port security.
I understand in order to enable port security
it must be an access port.
I am trying to build a small IP Phone/CM network - 50 phones, MCS-7825 and Unity Unified Messaging. The client has existing 3548's and an existing 3662 (empty) which I need to leverage in the design. I would like to use Power Patch Panels to drive the phones and put an NM-HDV-T1 as well as NM-2VICs with for FXS and FXO termination. My question is this: Without a Cat 4006 or 6500, where do I get DSP resources for Conference Calls? Do I even need them or can I use the CM as a Conference Bridge? Can I use a second HDV module as a DSP farm? How about the Enhanced HDV?
Thanks in advance for your help.
You can use the Voice Media Streaming Application that ships with the CallManager to do G.711 Conferencing. Since this is all in one single location (assumption), codec type should not be an issue i.e. everybody is G.711
Your other choice is (as you stated) you will be able to use the DSP's on the Network Module (available now or very shortly - pls confirm with appropriate people) to achieve conferencing.
Fina;;y if you do need a standalone DSP box (lets say to be put at a remote site) then one will be available Q4CY01.
Can you realistically attach 24 IP Phones to a 3524xl-pwr switch? I seem to see intermittent problems with one or two ports "dying" when I go to that 24th phone for any length of time... wierd thing is, when I plug a non-powered device into the port (one that wasn't working with a phone attached), it does work... Does anyone routinely load up their 3524s with 24 phones? Could I just have half dozen or so weak power supplies?
When connecting an IP phone to a network that has DHCP, what happens when you require a static IP address. Is there a way of manually programming your phone so that it has an ip address other than through DHCP?
How does this work through the internet? Can you build a web interface for that phone?
You can provide a ststic address to the phone (you will need to give it a static TFTP server address as well in this case) but you have to do it form the phone front panel interface. Cannot do it over the web or internet.