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connecting a modem on FXS;ISDN PRI

zlabovic
Level 1
Level 1

Hello,

I have 2801 with a PRI connection to PSTN and 3 4port FXS/DID cards. I have connected several modems on FXS ports ( I believe you can connect a modem to a FXS port) but when users dial in through ISDN to these modems the connection is quite bad (to many errors and it is very slow). Does anybody know a way to solve this? Is there some special configuration that has to be done on the router when you are connecting a modem on a FXS?

Thanx in advance

12 Replies 12

carenas123
Level 5
Level 5

Try to set the following on both the originating and terminating gateways

voice service voip

h323

call start slow

modem relay nse codec g711ulaw and then on the terminating gateway, configure incoming called-number under the voip dial peer.

I only have one gateway ( no originating and terminating gateway) and I only have POTS dial-peers.

I have tried using what you have suggested but the connection once established is of low speed (9600) as it was without these commands.The configuration of my router is as follows (some output has been omitted):

Current configuration : 3888 bytes

!

version 12.3

!

hostname voiceswitch

!

boot-start-marker

boot system flash c2801-ipvoice-mz.123-14.T.bin

boot system flash c2801-ipvoice-mz.123-11.T3.bin

boot-end-marker

!

voice-card 0

controller E1 0/3/0

framing NO-CRC4

pri-group timeslots 1-31

!

controller E1 0/3/1

framing NO-CRC4

clock source free-running

pri-group timeslots 1-22

!

!

interface Serial0/3/0:15

no ip address

isdn switch-type primary-net5

isdn incoming-voice modem 64

!

!

ip classless

!

voice-port 0/0/2

cptone SI

!

voice-port 0/0/3

!

voice-port 0/3/0:15

!

voice-port 0/1/0

!

voice-port 0/1/1

!

voice-port 0/1/2

!

voice-port 0/1/3

!

!

voice-port 0/2/0

!

voice-port 0/2/1

!

voice-port 0/2/2

!

voice-port 0/2/3

!

dial-peer voice 2 pots

destination-pattern 3701

incoming called-number 3701

direct-inward-dial

port 0/0/1

!

dial-peer voice 3 pots

huntstop

destination-pattern 3700

incoming called-number 3700

direct-inward-dial

port 0/0/2

!

dial-peer voice 10 pots

destination-pattern 3777

incoming called-number 3777

direct-inward-dial

port 0/0/0

!

dial-peer voice 4 pots

huntstop

destination-pattern 0T

incoming called-number 0T

direct-inward-dial

port 0/3/0:15

!

dial-peer voice 11 pots

preference 1

destination-pattern 3777

incoming called-number 3777

direct-inward-dial

port 0/1/0

!

dial-peer voice 12 pots

preference 2

destination-pattern 3777

incoming called-number 3777

direct-inward-dial

port 0/1/1

What you are doing here is called TDM hairpinning. You are taking the PCM from the ISDN B channel and directing it to the codec on the FXS port.

In order for this to work correctly, you need to ensure clock synchronisation on the PRI and the FXS port.

By default, an ISR will use the clock signal coming in on the PRI for the recieve traffic, but it will use an internal clock for the transmit traffic. Therefore there will be a difference in the clocking, and this causes clock slips on the E1 interface.

If you do a show controller e1 0/3/0 , I would not mind betting you see clock slips being registered.

There is a simple solution - add the following commands:

network-clock-participate wic 3

network-clock-select 1 e1 0/3/0

This will make the router use the clock coming in on port 0/3/0 as the master clock reference, so it will be used across the entire backplane. After the call connects, the DSP's will drop out of the connection and the PCM streams between the PRI B channel and the FXS port will be directly connected and have full synchronisation.

You may also need to set the clocking on port 0/3/1 to internal depending on what you have connected to it.

You are very right as to registering of the slip errors. I added what you have suggested and there are no slip errors and the connection is somewhat better. The only thing is, the connection speed is always 9600bps and I think that FXS can support up to 56K.

As far as controller 0/3/1 is concerned it is planned that that controller gets connected to another router's controller and it (0/3/1) should give the other controller a clock, so I am not sure that I should put internal clocking?

Once the call has connected, the DSP's should drop out of the connection and you should have a pure TDM connection between the ports, so I don't know why the modem speed is so low. Can you paste you SH VER and SH RUN so we can have a closer look at how you have set things up.

OK will do. I have to mention that I have done the hardware loopback test of the E1 0/3/0 controller and there were no errors.

sh run is in the attachment,

sh ver :

Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.3(14)T, RELEASE

SOFTWARE (fc1)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2005 by Cisco Systems, Inc.

Compiled Fri 25-Mar-05 23:16 by yiyan

ROM: System Bootstrap, Version 12.3(8r)T8, RELEASE SOFTWARE (fc1)

ROM: Cisco IOS Software, 2801 Software (C2801-IPVOICE-M), Version 12.3(11)T2, RE

LEASE SOFTWARE (fc1)

voiceswitch uptime is 1 day, 6 hours, 30 minutes

System returned to ROM by reload at 05:44:22 UTC Thu Apr 7 2005

System image file is "flash:c2801-ipvoice-mz.123-14.T.bin"

Cisco 2801 (revision 4.1) with 115712K/15360K bytes of memory.

Processor board ID FHK0852322B

2 FastEthernet interfaces

53 Serial interfaces

2 Channelized E1/PRI ports

12 Voice FXS interfaces

4 DSPs, 64 Voice resources

DRAM configuration is 64 bits wide with parity disabled.

191K bytes of NVRAM.

62592K bytes of ATA CompactFlash (Read/Write)

Configuration register is 0x2102

voiceswitch#

I have a theory here ...

The DSP's are not dropping from the connection because the default codec on the analogue ports is ulaw, while the default codec on the E1 interface is Alaw. When a call passes between the ports, a DSP has to be allocated for the E1 and the analogue connection to do the transcoding.

Try the following :

Under the FXS voice port/s, change the companding to Alaw -

voice-port 0/0/1

compand-type a-law

under the POTS dial peers, remove the command 'direct-inward-dial' - this is not needed for analogue ports.

When a call is made between the E1 and the FXS port, the router will do a TDM connection and should drop the DSPs out of the connection - the DSP's could be causing extra delay that may be affecting the modem calls. You would need to change the companding on all ports that would be connecting to the E1.

When there is a call up between the E1 and the FXS port, do a SH VOICE DSP to confirm if there is anything between the ports.

Well I have tried with what you have suggested but it's always ulaw used on both the controller and fxs port

As far as direct-inward-dial command goes, I had to put it because otherwise the connection fails to establish (when I dial the number I get the free dial tone from the router). To be more specific, I have ommited the direct-inward -dial command from the POTS and for a short while everything worked (a couple of seconds, long enough to test it once) and then it stopped so I had to put back the command to resume normal operation

Hi,

I have a similar issue, the calls caming from the ISDN to the FXS are most of the times dropped. The difference here is that i have a BRI in a point to point configuration. Is it possible that i have a synchronizatio issue? Can i issue the network-clock-participate and network-clock-select for a BRI interface? How can i see if i have slips on the BRI???

I should use the A-law (in Europe) codec on the BRI and FXS ports right?

Regards

Almeida

juan.montaner
Level 1
Level 1

Hello,

I've got an issue like this.

I need to use my analogue modem to use a remote application, but our Cisco IPC solution is been rolled-out with 7960 for users and ATA-186 for FAX and door-relay.

As I've already checked, the ATA-186 doesn't support dialup calls.

I've been thinking to stick a FXS interface in our gateway (1760-V with 4 BRI) and enable dialup call going into the FXS and routed out trough any BRI.

Will this solution work out with errorfree?

What is the stadard solution that Cisco propose for a CallManager users who has a modem dialup application?

Thanks very much

Juan Montaner

Thank you