E1 - for voice and data

[rmv72]

ISP offer to connect E1 to 1760 router. I want to use E1 for voice and data. 1760 will also have FXS module for connect to local PBX.

How separate E1 into two parts - for data and for voice. May be i should use two V.35 modules - one for voice and other for data?

[a.awan]

First of all an FXS module is used to connect to end stations like Analog phones and fax machines. In order to connect to a PBX you will require either a FXO module or a module capable of talking ISDN PRI or CAS signaling to the PBX.

If your intention is to transparently pass some E1 channels to the PBX and run TDM voice from the PBX directly to the provider (through the router) then you should look at the VWIC-2MFT-E1-D1 module. The 'D1' om the part number denots Drop and Insert capability that basically allows you to transparently patch some E1 channels on one port to another port. Read the "Multiplexed Voice/Data Solutions: Dual-Port T1/E1 Drop-and-Insert Multiplexer with Integrated DSU/CSU" section at the following link for more information and to see if this is indeed what you are looking for:

http://www.cisco.com/en/US/products/hw/routers/ps221/products_data_sheet09186a0080088715.html

[rmv72]

Yes, it what i need. But next question - how to configure it. I think so:

controller e1 0/0

channel-group 1 timeslots 1-20

channel-group 2 timeslots 21-30

!

Int s0/0:1

Ip address …

!

Int s0/0:2

Ip address ….

!

And for connecting FXO modules i should use DIAL-PEERs.

Is't right?

[a.awan]

The following article has some example configurations for the Drop and Insert functionality:

http://www.cisco.com/en/US/products/sw/iosswrel/ps1830/products_feature_guide09186a0080087b21.html#wp46783

Yes dial-peers are used to reach FXO ports. Keep in mind that using FXO you will not be transparently using the E1 for voice and data.

[rmv72]

Can you correct me if i'm wrong ? ->

!

controller E1 1/0

framing crc4

linecoding hdb3

clock source line primary

tdm-group 1 timeslots 1-15

channel-group 0 timeslots 17-31 speed 64 (but if i want have more than 64 k for data?)

!

The tdm-group is define voice traffic from ISP ( they tell me number DS0 channels used for voice).

The channel-group is used for data delivery.

And for data i also configure serial interface -

!

interface serial 1/0:0

ip address ..................

!

Can you explain what does it mean - "using FXO you will not be transparently using the E1 for voice and data."?

[sricardi]

Hi, I have another doubt about this subject too. Here we have a 7500 router with several E1 interfaces.

All of them are presenting a lot of input CRC errors. What´s happening ?

[a.awan]

You are right that tdm-group is used to define voice traffic from the ISP (not VoIP but TDM voice). You have to have another VWIC interface connected to the PBX and transparently pass these voice channels to the PBX. In this particular scenario your router is acting as a TDM multiplexer along with a normal router.

The speed 64 keyword just denotes that each channel is capable of carrying 64 kbps of information. Do not think that it is limiting your data bandwidth to 64 Kbps. Your data throughput will be limited by the total number of channels you can allocate to data using the channel-group command and of course by any hard limit set by your ISP.

What i meant by the statement "using FXO you will not be transparently using the E1 for voice and data."? is that you cannot transparently switch voice between the VWIC-MFT tdm group and the FXO interface. If you explain how you intend to use the FXO we can tell you whether it will work or not.

[rmv72]

We've local PBX with connected non-IP phones and several channels from input E1 we want to connect to local non-IP phones (local users must have possibility to made and recive calls from PSTN). Other part of E1's input voice channels we want to send over IP (VoIP) to central office.

What is nessecary for switching voice between the VWIC-MFT tdm group and PBX?

[a.awan]

For transparently switching TDM channels between your input E1 and your PBX you need to connect the input E1 to one port of the VWIC-MFT module and the PBX E1 to another port and configure your router to transparently map the channels together.

What you are trying to achieve is definitely doable using the VWIC-MFTs. I will try to attach a PDF with this post that is specific to the 1700 routers and the VWIC-MFT module. Take a look at the Drop and Insert configuration which is basically the same thing you are trying to achieve.

[rmv72]

Thank you! It's very helpful.

And my last question - about VWIC-MFT - i did not catch from documents -is't nessecary to add PVDMs into router to support up to 30 voice channels or VWIC-MFT already have it?

[a.awan]

PVDMs come separate. The DSPs on the PVDM will only be used for channels being terminated on the router. For channels that you are transparently passing on to the PBX you do not need any DSP resources.

[rmv72]

Is't possible tha transparently passing incoming voice channels into analog phones connected to router through FXS module or to IP=phones? If yes than how?

[a.awan]

No. You will have to terminate the channels on the router and the FXS or IP Phones will use the router as a PSTN gateway to make PSTN calls. All calls will utilize DSP resources on the router.

[rmv72]

Sorry but how to terminate the channels on the router?

[a.awan · Accepted solution]

Configuring the appropriate channels as a ds0 channel group and then defining an associated dial-peer should do it. Go through the PDF i posted earlier, it has such examples.