Configuring Voip
Hello,
I was working with Microsoft products previously and now I have changed my mind to switch to cisco networks. My company has decided to go with VOIP SOLUTION as our branches are connected with MPLS network. The project given to me is this way. I have to connect 2 sites which includes my head office and branch office to use voip (both the branches are already connected using MPLS network). My head office is having a PBX which will be integrated with voip and my branch office is not having any kind of PBX system.
Note: It will be a pilot project so please suggest a cheap hardware only for 2 sites if its approved then we will be buy good router in our head office to connect all other branches.
1) What kind of cisco routers shall I buy for both Sites ?
2) what kind of modules shall be intalled in them to start ?
3) What kind of IOS ?
4) Do I need to do something in my PBX ? for the integration in my head office ?
5) Which books do you prefer as a beginner ?
I hope the above mentioned scenario is completely transparent by any chance if you feel that i havent provided you the complete information dont hesitate to ask. Your earliest response will be appreciated.
Thanks
SKAK
Hi,
I would do this with two 2801, CCME feature, and cisco IP phones of your choice. What interface does the PBX have ? If doesn't have ISDN BRI or PRI, it will be hard to do a nice integration. Please search "CME" on CCO and start reading about it.
Note: it is fundamental that you locate a reputable cisco reseller that can help you with the prices and availability of the material, if not with the actual configuration.
Similar Messages
-
To configure VOIP for this 2FXO card
We are having one Cisco 3640 Router with One Serial Port and 2FXS(2nos) Voice card. We have configured the 2FXO card to the best of our knowledge. These 2 ports are configured in EPABX. Now we are able to get the dialtone but we are unable to make calls to US . Kindly let me know the commands to configure VOIP for this 2FXO card.
Thanks in advance
Regards
Sapna shahWe are having one Cisco 3640 Router with One Serial Port and 2FXS(2nos) Voice card.We have configured voice card FXO but now due to some unknown reason we are not able to call the US office. Kindly help us .Here is the configuration:
Can u let us know what errors we have made in configuruing steps .
Here are HYD-RTR#sh voice port
Foreign Exchange Station 1/0/0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Direct line for IOC
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/0/1
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Connected to Alcatel PABX thru trunk line
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 1 s
Region Tone is set for US
Analog Info Follows:
Currently processing unknown
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/1/0
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Second Direct Line For IOC
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing Voice
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Station 1/1/1
Type of VoicePort is FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is Second Line Connected to Alcatel PABX thru trunk line
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing unknown
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Foreign Exchange Office 2/0/0
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is plar
Connection Number is 100
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is groundStart
Number Of Rings is set to 1
Supervisory Disconnect active
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Type is pulse
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Foreign Exchange Office 2/0/1
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancel Coverage is set to 8 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call-Disconnect Time Out is set to 60 s
Region Tone is set for US
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Voice card specific Info Follows:
Signal Type is groundStart
Number Of Rings is set to 1
Supervisory Disconnect active
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Please enlighten us why we are not able to make calls .IS there any fault in the configurations made by us !
Thanks in advance -
Cisco 1861- Configuring VoIP- Using PSTN
I need help configuring a 1861 ISR. This SOHO will have 4 analog line or PSTN connecting to the router and want to be able to connect 4 Cisco Ip Phones 796X to this router and make VoIP calls out the local ISP and out the PSTN. I was wondering what will be needed so that I can implement VoIP using the Cisco 796x Series phones with 4 pstn lines and a local ISP connection so that calls can be forwarded out.
I have attached a configuration of an existing 1861 we have implemented but registered to our CallManager via CallManager Fallback. However I want this design to be independent and totally separate from our network.There's a pretty good chance you can just switch this over to a CME configuration rather than SRST. For CME you would configure 'telephony-service' after taking 'call-manager-fallback' out, and registering the phones directly there.
You would then create a SIP/H323 trunk to CCM for connectivity to other phones and calls.
If you would like PSTN connectivity to basic analog lines, you can get something like a VIC2-4FXO, a PVDM2-8, and route calls like that. For FXO lines they will be generally redirected to a single number via a 'connection plar' statement that leads to a receptionist's DN or an auto attendant. -
Help for basic VOIP function on 2821
Hi,
I'm new to VOIP. Sorry for this simple question.
We have two Cisco 2821 routers, two 7960G IP phones.
Each 2821 router has PVDM2-8 DSP module, HWIC-4ESW 4 port swtiching card, IOS 12.4(22)T3.
No any Call Manager software is loaded.
Can anyone point me to a sample configuration/how-to so I can make phone call from Router A to Router B?
Something as below:
Phone A <----> Router A <---- internal IP network ----> Router B <------> Phone B
Please note we don't have Call Manager software and we don't need fancy phone call features.
Thank you very much!
-AndrewYou need two analog phones. The VOIP phones you have 7960G will not work. They are dumm terminals. They need something to tell them what to do. A call agent (CUCM, CCM,) or any other call agent that can control cisco VOIP phones eg asterics.
When you do get 2 analog phones configure voip and pots dial peers
example
Configure this on both routers with the IP pointing to each other
dial-peer voice 10 voip
destination-pattern 5678 --No the analog will send to the other sides fxs port to ring the analog phone
session target ipv4:10.10.10.1 ------ ip address of the other router.
dial-peer voice 11 pots
destination-pattern 234 --- the number that will ring the phone plugged here.
port 1/0 -------fxs port where your analog phone is plugged.
Dial away.
You might find ths url helpful.
http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a00800e00d0.shtml -
I have 1MB wireless link from my ISP. I am using Cisco VOIP phones in my company. Over the months my VOIP quality is not good(on/off connectivity),though my internet is somehow working fine. When I talked to my ISP , they told me to use Cisco switch / router instead of Nortal. Moreover ISP also saying that the switch / router can be configured to full duplex/ half duplex because ISP equipment can be switched to half/duplex.
-->My questions.
1-what cisco router/switch should I use with voip phones ?
2-What duplex setting should be on the router/switch so voip quality be remains optimum.
3-Is 1MB link can be used for voip?
4-Is it possible to configure voip phones to full/half duplex ?
Please help. Thanks.Hi,
2. If you're currently running your phones over a wireless connection, this can cause problems. Wireless doesn't have guaranteed bandwidth and it's half duplex.
If you were to switch this out with a router and switch, you could make every link full duplex.
Full duplex is what you should run everywhere. The only modern technology you should be running half-duplex on is wireless, because that's just the way it works.
1. The router depends on your company size and what you want to use it for.
3. 1 MB link may be alright, but again, it depends on how many active phone calls, what your data usage is, and what codec you're running.
4. Phones are auto-negotiating, and they will use either. -
hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
DevangDial-peer matching information:
http://www.cisco.com/warp/public/788/voip/in_dial_peer_match.html
Analog DID for Cisco 2600 and Cisco 3600 Series Routers:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122t/122t2/dt_did.htm -
I configured Voip on Nokia 700 belle fp2 with the guide from this page. I prefer 3G connection so I set packet data connection over 3G and enabled AWCDMA setting in VoIP service settings. Like in the guide. I can make calls. Sound quality is excellent on booth sides. But I'm not reachable over VOIP. My phone doesn't make any signs when I try to call it. I quadruple check to be sure that every profile entry has been made correctly. I reconfigure settings for wifi connection and trying to establish connection over wifi bat the problem resist. My voip provider doesn't make any restrictions. Is there any explanation or better solution for this problem.
Solved!
Go to Solution.I find this document from year 2002. I capture the registering data and incoming call data with wireshark. I find out that my voip provider has problem with Call-ID header. While registering they answer with right Call-ID header. When I get incoming call the Call-ID header is not the same. There is no malfunction. The call is dropped with 404 not found.
-
This guide will concentrate on the open SIP standard based VoIP, because S40 does not have Java APIs to build closed system VoIP solutions (such as Skype, Fring, Nimbuzz ...)
Nokia maintains a list of the S40 phones that have built in SIP VoIP capability in the firmware.
Configuring the phone for VoIP usage
There are three different ways how to configure VoIP in use in S40,
Using the built in setup wizard which can be found from, menu>settings>connectivity>internet tel.
The list of VoIP service providers differ based on which country you are located in.
If you cannot find your VoIP service provider from the list, but know the SIP details for it, choose "sip settings" from the provider list and fill in the SIP details
Some operator branded firmware have crippled setup wizard so that it's empty
Configuration over the air (OTA SMS). The VoIP service provider providing the OTA SMS configuration needs to have S40 VoIP compliant configuration settings
Manual configuration requires following steps,
Download settings example (XML file) compatible to your S40 phone
Download wbxml2 tool
Modify the XML file to suit your needs
Use the xml2wbxml.exe (from wbxml2 tool) to convert the XML file to wbxml format and rename the.wbxml to .prov
Scan and connect your phone over Bluetooth to your PC
Send (OBEX PUSH) the .prov file to the connected phone.
There exists a free tool for the manual configuration called n0kVoIP which makes the process easier.
Other tips
Default call type (i.e.. What happens when green button is pressed) can be change in menu>settings>call>call type settings. Setting it to GSM only, will disable VoIP for registering to the service.
If the phone supports VoIP v72, v81 or v92 then VoIP works only over WiFi. If the phone supports VoIP v104 then VoIP works both over WiFi and 3G.
SIP URIs can be added to the contact in phonebook by adding 'internet telephony' detail into it.There is no operating system as such in the IP phones like 7960 or any other IP phone. It will have some firmware and config files which it downloads from the Callmanager while registering to it. All the call processing is done by the Callmanager. There is no special software running on the phones. You may like to have a look at the IP Telephony documents which have more information.
http://www.cisco.com/pcgi-bin/Support/browse/index.pl?i=Products&f=2695
http://www.cisco.com/pcgi-bin/Support/browse/psp_view.pl?p=Software:Cisco_Call_Manager -
Voice over ip configuration between 2600 and 1700 routers
Hi,
I have the following set up:
2 offices with full T1 to the internet; one using a cisco 2600 router with FXS cards and the other using a 1700 router with FXS cards as well. I was asked to configure VOIP between the two sites and the PBX tech gave me at each office trunks which I connected to the FXS interfaces. The goal will be that the PBX guy be able to program the PBX with extension numbers of his choice and get the calls between the offices using the internet connection. I do not have a lot of experience setting up VOIP and I wonder if someone out there might have a similar environment.
According to the PBX tech, this trunk lines are loop start.
Thanks in advance for any ideas,
Uriel Naranjo.It should be fairly straight forward. On the 2600 you will need to configure a voip dial-peer pointing to the remote site and pots dial peer to terminate a call on to the 2600 and you do the same on the 1700.
Check out the sample configurations on the following URL:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1828/products_configuration_guide_chapter09186a00800ca621.html#5593
disregard the rsvp portion and just simply configure your routers to pass voip traffic. You can do QoS later depending on the load and traffic. -
FXS and FXO configuration and design help
hi all...
i am new to voip configuration... and i have to configure the voip for our branch office... we have 2610 with the 2FXS and 2FXO prot... now i have one pulblic IP address which is used by my local lan user...and we have configured NAT/PAT on our VPN concerntrator... now we are looking to establish the VOIP phone (analogphone with the help of FXO and FXS) we have ASTRIK server at our main office and i want to register my local office analog phone with server at our main office... now what kind of configuration we need on our 2610 in order to configure voip...
connectivity:
ADSL connection form ISP---VPN 3005---D-Link nonmanagable swithch---LAN
at present we have above connectivity and now i want to add my 2610 router with analogphone connected to it... how can i connect and how can i configure it...?
regards
Devangyou need to configure pots and voip dialPeers for the incoming/outgoing legs of the calls.
pots dialPeers will be used for the FXO/FXS ports. voip dialPeers will be used for connection to ASTERISK voip pbx.
you need voice port configuration for your FXS to connect to your analogPhone.
you need voice port configuration for your FXO to connect to analog pstn. (if you have any)
see these links for more info:
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080147524.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml
http://www.cisco.com/en/US/tech/tk652/tk90/technologies_configuration_example09186a00801bc341.shtml
http://www.cisco.com/en/US/tech/tk652/tk653/tech_configuration_examples_list.html
there are plenty of examples throughout these for reference. -
I configured VoIP on a MPPP with LLQ as a QoS policy in a lab environment, it works fine.
Now I`d like to configure it over a satellite link with 128 Kbps of bandwidth and four channel.
What considerations should I to do in my configurations for it?
Are there some command configs for tunning the QoS in this environment?
I know that the delay between two points through satellite link is 500ms.
Someone has some experience with this case?. I`ll appreciate your suggestions.
Thank you.hi everybody,
i have an internet connection via sat with the following config:
1. 2Mbps down and 1Mbps up.
2. ps4500 sat modem.
3. 2600XM router.
4. AS5400 gateway.
5. its connected to PSTN.
we have about 100 users that use it for calling through PSTN.
Now we like to conffigure QoS/Diffserv for Voice connection and connect about 200 users for normal internet connection.
can anybody tell me what the consumming B/W for 100 calls and what is the best configuration for this senario.
best redards
eng thaar
oil ministry -
Next year we are planning a VOIP migration for some of our in house users. We dont have any WAN connections and bandwidth between floors are gigabit.
What I am looking to do is buy a small setup for our lab this year so I can begin to learn about VOIP etc. So far I have researched and this is what I am considering buying.
- Cisco 2801-V/K9
- VIC2-2FX0 voice interface card
- 3560 24 port 10/100 POE standard image
- 5 7940G ip phones.
Is this a decent setup for the lab ?
Would it allow me to configure a basic voip setup and learn the fundamentals of configuring VOIP ?
Any help would be appreciated.
Cheers
DaveHi Dave,
You can probably get away with this for even less.
You could probably get something like a 1751-V with PVDM-256k and VIC-2FXS with the IP phones. You may need the power bricks in this scenario.
Alternatively,
CISCO2621XM with NM-2V with VIC-2FXS and the phones/bricks
But yes, your current setup will work. You would need to add something like a PVDM2-8, PVDM2-16, or PVDM2-32 for DSP resources, with the number depending on whether or not you want to play with transcoding/conferencing.
hth,
nick -
hi. ive set up voip profile in menu-connectivity-admin settings. I'm looking now for the "Internet Call" applicaton to register the profile and to use it for voip calls, but i can't find it nowhere!!
i've read symbian^3 have built-in voip support, but i can't use it. Can you help me? In my old nokia E70 under applications i have "Internet Call" icon to start voip profiles but i can't find it in my c7...If your voip is set up correctly, you should see both "voice call" and "internet call" as options in your contact directory. There is not separate "internet call" application. Internet calling is integrated to the contact directory.
You can sign in and out of your voip accounts using the menu bar at the top when you are in contact directory (home page/call/contacts).
Make sure you install this first:
http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
See also this:
/t5/Nseries-and-S60-Smartphones/configuring-VoIP-on-N8/td-p/772361
You need an account with www.sipgate.com or other sip provider. -
Hi All,
We are planning to have a VOIP solution in our organization with in coming months. We have 4 offices. Each office has E1 links between them. Our HQ has Cisco 3745 series router with 12.3 IOS running and other location has Cisco 3725 series router with 12.3 IOS.
I have some doubts, can any one help me.
1) Do I require any new hardware? If yes please specify.
2) Does my IOS support VOIP?
3) Is there any extra cost involved?
4) Can I use soft phone?
RegardsAt all locations you probably have a PBX or Keysystem to support local office phones as well as the HQ site. Depending on the available interfaces on the PBX you would need hardware on the gateway so you can connect your 37xx routers to it. Normally connections are made through EM, FXS, FXO or T1 digital cards. These are VIC cards and you would need NM modules as well. For T1 you would need NM-HDV and VWIC T1 cards.
Your IOS does support the voip so you should not have any problem configuring voip.
Cost will be related to the hw that you would need to integrate your PBXs to the routers.
IP Softphone would need CiscoCallManager(CCM). Since you did not mention that you are expecting to install IP Phones at locations the above mentioned recommendations are tradition voip. If you are planning for IP Telephony support then you will have to look into:
CCM Server
CCM
IP Phones
And this will allow you to support softphone.
I hope that helps.
Javed -
VLAN translation for VoIP config.
I have an Cisco 1751-v4 as the core router. Also a Cisco Cat 1900 switch with ports 17-24 configured for VLAN membership VLAN 2 (VLAN 1 is data).
My goal is to configure VoIP, but I want the VLAN to be on a 172.x.x.x subnet.
Can I create a second dhcp pool I can include "option 150" for VoIP only?
my next question is I have a fast ethernet interface (LAN) and an ethernet interface (WAN, which interface do I configure the VLAN encapsulation on?
I have included my current config below.
thanks for the help.Ankur,
I am still having issues configuring VLAN's.
During some reading, I understand I should not have to create a trunk as I am only using a single cisco 1900 switch.I should not have to configure VTP on the switch as well for the same purpose.
Here is what I am looking at:
Router#>int fasteth0/0
ip address 192.168.2.1 255.255.255.0
ip helper-address 192.168.2.0
ip nat inside
From what I have read, I need to configure the DHCP DATA pool (192.168.2.0) as vlan1 native. But when I attempt to configure the IP address for the native VLAN1 on fastethernet0/0.1 (subinterface) I recieve IP address overlap with int fasteth0/0 interface error.
Question, can I configure interface fastethernet0/0 without an IP address,but configure the subinterfaces 0/0.1 and 0/0.2 with ip address?
ip dhcp pool Data
network 192.168.2.0 255.255.255.0
domain-name 5thborocs.com
default-router 192.168.2.1
dns-server 63.162.197.69 208.33.149.39
ip dhcp pool VoIP
network 192.168.3.0 255.255.255.0
domain-name 5thborocs.com
default-router 192.168.2.1
option 150 ip 192.168.2.1
interface FastEthernet0/0
description Cisco 1900sw
ip helper-address 192.168.2.0 (or without a helper address)
ip nat inside
ip virtual-reassembly
speed auto
full-duplex
interface FastEthernet0/0.1
description Data
encapsulation dot1Q 1 native
ip address 192.168.2.1 255.255.255.0
ip helper-address 192.168.2.0
interface FastEthernet0/0.2
description VoIP
encapsulation dot1Q 2
ip address 192.168.3.1 255.255.255.0
ip helper-address 192.168.3.0
no snmp trap link-status
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