FXO & FXS Card

Hello Guys ,
I have installed VIC-2FXS and VIC-2FXO Card on my Cisco 2651, but my router is not detecting the Voice cards.
I have seen the output of Show Version and Show Diag.
If i am installing any other WIC-2T it is detecting.
Can you Please let me know what might be the problem.
Waiting for your reply,
Regards,
MAX

First, let's examine the current system:
1-you have an 2651 XM
2-using an ios that is voice capable(c2600-ipvoice-mz.123-13)
3-the platform supports one network module and two interface cards
Secondly, let's examine what you did:
1-you installed a nm-4a/s (that means four synch/asynch serial wan interface) into the NM slot of 2651XM.
2-you installed a fxo and a fxs voice interface card into two built in interface card slots.
Lastly, see what is the problem:
1-nm-4a/s works without any problem
2-system does not recognise fxo or fxs cards
So let's see what is the reason:
1-2651xm does not support fxo or fxs interface cards into the built in interface card slots.(where you try to install fxo&fxs!!!)
2-2651xm supports voice interface cards by using a voice network module(instead you use nm-4a/s to get asynch/synch wan interfaces!!!)
At the end, let's say what must be the right hardware configuration:
1-one nm-hd-2v(to install into nm slot of 2651xm)(also this nm provide enough dsp for vics)
2-one vic2-2fxo and one vic2-2fxs(this cards must be installed into nm-hd-2v)
3-two wic-2a/s (this cards will be installed into built in interface card slots of 2651xm)
So you need:
one nm-hd-2v + one vic2-2fxo + one vic2-2fxs + two wic-2a/s; on the other hand, nm-4a/s will be unnecessary
Thanks

Similar Messages

Cisco 2610xm NM2V FXO FXS

I have 2 cisco 2610xm gw one with fxs and another one with fxo i want to originate call from the fxs and terminating with the fxo connected to pstn line. But the IOS does not support Voice-port command. What IOS is recommended for voip on the 2610xm and where can i get a sample configuration for terminating calls to pstn with the fxo. Thnaks

Hi there any ios ip plus code will do, better if u can go for some latest version on 12.3(GD) train .but do remember that your box have enough DRAM and Flash space to acommodate the new ios code.
and as far as the config samples do a search using keywords like configuring fxo or fxs cards in cisco.com site that will yield lots of sample config templates ,which will be helpful to configure as per ur requirement.
regds

What sort of fxs card to buy for the cisco router 2801

Hi
I have just bought a cisco router 2801 which l want to setup for my VIOP lab. Can pls advice me on what sort of fxs card l should buy for the cisco router 2801 to connect the ip phones or the analog phones.

Hi
It depends what you want to do:
IP Phones - need nothing
FXS - needed only if you want to connect analogue handsets or faxes
FXO - needed if you want to connect the system to a POTS line
There are further interfaces (BRI, PRI etc) if you have different PSTN connectivity.
What connectivity do you plan to have in your lab?
Aaron
Please rate helpful posts...

Tieing 2 key systems together with fxo/fxs and 1760 routers

Hello,
I need some programming help from someone good on voice. I've got two offices that I'm trying to tie the phone systems together with 2x Cisco 1760 routers each with 2x PVDM-256K-4 1 DSP Modules. I've got the layout below and am basically looking to do two things.
First, I would like ext. 210 from the first site to dial a co group â1â or directly access a âCO lineâ that is connected to the Cisco and get dial tone to be able to dial the directory number for a âCO lineâ with the same setup at the second site and have it able to be answered like a normal call and be transferred.
The second connection I would like is to have ext. 210 be able to dial locally to one of site 1's analog single line extensions and have the Ciscos make a connection through to site 2 and go off-hook on one of the analog single line extensions of site 2 in order to get a site 2 dial tone and be able to dial locally @ site 2 to any extension, or dial one of site 2's co groups or directory number for one of site 2's real CO lines and place a âlocalâ call to the outside world from site 2's lines.
Obviously this process would all be reversed for site 2 accessing site 1. I've come across a couple of documents, like ID: 15405, and a section of a VoIP Configuration guide labeled OL-1070-01 and have some command structure available, but the concept of how it all takes place and should be configured is a little fuzzy.
Thank you,
Mark

OK, let me simplify things. I think I'm putting way too much thought into it all. I've got site A and site B. Site A (currently for testing) has a single line extension from Site A's key system plugged into port 0 in fxo card in slot 2. Site A will have a patch from port 0 in fxs card in slot 3 to a CO line on the key system. Site B has the same setup. Both have fa 0/0 configured with IP addresses on the same network (just to simulate the connections - later I will actually move these to two separate internet feeds for more advanced testing).
Currently for testing I have disconnected the fxs patches to the phone systems and just have a regular analog phone plugged in. When my phone plugged into Site A goes off hook, I get dial-tone from the extension hooked up to Site B (which is the exact way I want it). When my analog phone is plugged into Site B (port 0 of fxs card in slot 3) and goes off hook, it will ring port 0 of fxs card in slot 3 of Site A. This I don't understand. If I can get both to behave like Site A, I'd be happy.
I need to know if this makes sense to anyone on how I want this to operate? Is it achievable?
Here's my base config on it (Site A first, then Site B):
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Simmering
boot-start-marker
boot-end-marker
enable secret 5 $1$F/AM$ige2qFh9lVD6uNubE.qm80
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.30 255.255.255.0
speed auto
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 290
voice-port 2/1
connection plar opx 291
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 190
voice-port 3/1
connection plar 191
voice-port 3/2
voice-port 3/3
dial-peer voice 280 pots
destination-pattern 280
port 2/0
dial-peer voice 281 pots
destination-pattern 281
port 2/1
dial-peer voice 290 voip
destination-pattern 29
session target ipv4:192.168.254.40
dial-peer voice 180 pots
destination-pattern 180
port 3/0
dial-peer voice 181 pots
destination-pattern 181
port 3/1
dial-peer voice 190 voip
destination-pattern 19
session target ipv4:192.168.254.40
line con 0
logging synchronous
line aux 0
line vty 0 4
password Corazon64789
logging synchronous
login
transport input telnet
end
SITE B:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Sigma
boot-start-marker
boot-end-marker
enable secret 5 $1$1H58$6.VsVieyi4srcC7v4Lndv0
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.40 255.255.255.0
speed auto
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 280
voice-port 2/1
connection plar opx 281
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 180
voice-port 3/1
connection plar 181
voice-port 3/2
voice-port 3/3
dial-peer voice 290 pots
destination-pattern 290
port 3/0
dial-peer voice 291 pots
destination-pattern 291
port 3/1
dial-peer voice 280 voip
destination-pattern 28
session target ipv4:192.168.254.30
dial-peer voice 190 pots
destination-pattern 190
port 2/0
dial-peer voice 191 pots
destination-pattern 191
port 2/1
dial-peer voice 180 voip
destination-pattern 18
session target ipv4:192.168.254.30
line con 0
logging synchronous
line aux 0
line vty 0 4
password Corazon64789
logging synchronous
login
transport input telnet
end

CSS in CUCM for H.323 GW with FXS card

I have been working on getting FXS attached devices to be able to make calls (mostly faxing) from a remote site thru our corporate PRI. I have setup the GW in CUCM as H.323 and given it a CSS of "PSTNGateway_CSS" which contains my Internal_PT, VM_Pilot_PT, and Blocked_PT. I have found that I am not able to make outbound calls from FXS connected devices since I do not have any external patterns in those partitions which would be tied to a route list.
My question is:
What is the best practice for H.323 GW configurations in CUCM when you have an FXS card in the router?
Generally, the best practice for a GW is to only give it access to internal patterns (extensions or pilots) so as not to risk toll fraud. However, that is counter productive when you have FXS ports that need to be able to dial the outside world.
H.323 rtr (FXS ports) ----> 50Mb Metro-E ---> CUCM (Corp) ---> H.323 GW (PSTN Access)
As a measure of just making it work, I have added a special partition with local and long distance patterns to the current "PSTNGateway_CSS" so the calls will work.

You need to set whatever CSS is required for that GW to make the calls it needs to, whichever it may be in your deployment, otherwise youll be stuck with internal dialing only
Sent from Cisco Technical Support iPad App

How to config CME with FXS card for analog phones

I want to config FXS cards in CME router with SCCP protocol for transfer, pickup ... feature. I can find information about the sccp (the command about stcapp), but I can not find the information about how to config CME, e.g. the phone no.(something like ephone-dn .... )
Can anyone help me ? or send me the link
Thx
Dennis

Thx Johncaston,
Your config is for H323, it doesn't support features, something like transfer funtion .......
I have found the config for SCCP, let's share my finding
====================
ephone-dn 178
number 9933
description Board Room
name Board Room
hold-alert 30 originator
ephone-dn 179 dual-line
number 9990
description Meeting Room 3
name Meeting Room 3
hold-alert 30 originator
ephone 95
mac-address 9556.60FA.0200
type anl
button 1:178
ephone 96
mac-address 9556.60FA.0201
type anl
button 1:179
GigabitEthernet0/0 is up, line protocol is up
Hardware is MV96340 Ethernet, address is 0019.5566.0fa0 (bia 0019.5566.0fa0)
================================
stcapp ccm-group 1
stcapp
sccp local GigabitEthernet0/0
sccp ccm 172.16.136.125 identifier 1
sccp
sccp ccm group 1
associate ccm 1 priority 1
signaling dscp af31
audio dscp ef
voice-port 1/0/0
cptone HK
caller-id enable
voice-port 1/0/1
cptone HK
caller-id enable
dial-peer voice 21 pots
service stcapp
port 1/0/0
dial-peer voice 22 pots
service stcapp
port 1/0/1
=========================
Thanks !!
Dennis

FXO - FXS PSTN emulation for testing lab

Hi netpros,
For testing purposes. I am wondering whether it is possible to simulate PSTN by connecting routers FX0<->FXS ports as per the diagram. I will be purchasing E1 cards in several weeks but for now I am trying to get PSTN emulation using what I have got. Also ..do I need to have FXO ports on the 'PSTN emulator' device as well or having FXS connected to the edges router's FXO will be fine? Do I need to use PLAR to redirect the incoming calls from PSTN to the destination analog phone ?
Your help is much appreciated !!!
NOTE: I have tried getting some help on the Training forum without any luck

Hi,
Yes It will work.But you have to give connection plar under FXO ports.
HTH.
Please Rate if it helps

US 500 Series FXO/FXS

Hello all. I just need to clarify/ask a question. From my discussions with Cisco tech support, the maximum number of FXO and FXS ports I can have on a UC 500 series is 8 of each, using a VIC card for 4 more FXO and a SPA8800 for 4 more FXS.
Does anyone know of anyway to increase this? I have a client who needs to run with 12 FXO and 12 FXS to make a functioning system.
Does anyone have any other solutions I can deliver? (And please do not go into BRI or SIP; it's not going to happen.)
Thanks all.

Hello Kris,
If you are using UC560-FXO and add two additional FXO VICs - VIC2-4FXO you will have a total of 12 FXO ports.
Then add SPA8000 will give you additional 8 FXS ports - total 12.
This is how you add SPA8000 to the UC.
https://supportforums.cisco.com/docs/DOC-9465
For FXS also you may use ATA186.
HTH,
Alex
*Please rate helpful posts

FXO/FXS IOS COMMAND ISSUES

Hello,
I've got 2 1760 routers with each a VIC2-4FX0 and VIC-FXS/DID cards. I've got 128mb of system RAM in them, running IOS c1700-advsecurityk9-mz.124-3.bin, which is supposed to be a voice ios with advanced feat. pack according to Cisco's website. I've also got 2x of the needed dsp's (can't remember model number, but its only 1 dsp chip on it - again cisco said 1 dsp would handle 4 channels, unless we're using the high-end codecs. My issue is that I cannot issue any of the voice-port commands. I don't think the hardware is somehow installed correctly. When I issue a "show hardware", they are not listed, however, when I capture a boot, they are listed as "card in slot 2/ card in slot 3" where the router allocates memory to devices.
Can anyone please help? And by the way, these are default no config, all I have done on the one is give a hostname, fastethernet ip and some basic password sec. and that's it.
HERE'S THE BOOT:
System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1) TAC Support: http://www.cisco.com/tac Copyright (c) 2003 by cisco Systems, Inc.
C1700 platform with 131072 Kbytes of main memory
program load complete, entry point: 0x80008000, size: 0xcbd968 Self decompressing the image : #################################################
# [OK]
Smart Init is enabled
smart init is sizing iomem
ID MEMORY_REQ TYPE
MainBoard 0X00027A80 1760
0X000F3BB0 public buffer pools
0X00211000 public particle pools
0X0003B100 DSP Buffers
0X004C 0X00000000 Card in slot 2
0X003A 0X00000000 Card in slot 3
TOTAL: 0X00367730
If any of the above Memory Requirements are "UNKNOWN", you may be using an unsupported configuration or there is a software problem and system operation may be compromised.
AND HERE'S SHOW HARDWARE:
Simmering#sh hardware
Cisco IOS Software, C1700 Software (C1700-ADVSECURITYK9-M), Version 12.4(3), REL EASE SOFTWARE (fc2) Technical Support: http://www.cisco.com/techsupport Copyright (c) 1986-2005 by Cisco Systems, Inc.
Compiled Fri 22-Jul-05 12:04 by hqluong
ROM: System Bootstrap, Version 12.2(7r)XM2, RELEASE SOFTWARE (fc1)
Simmering uptime is 4 days, 16 hours, 17 minutes System returned to ROM by reload System image file is "flash:c1700-advsecurityk9-mz.124-3.bin"
This product contains cryptographic features and is subject to United States and local country laws governing import, export, transfer and use. Delivery of Cisco cryptographic products does not imply third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for compliance with U.S. and local country laws. By using this product you agree to comply with applicable laws and regulations. If you are unable to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to [email protected].
Cisco 1760 (MPC860P) processor (revision 0x500) with 114688K/16384K bytes of mem ory.
Processor board ID FOC08252LAR (617609820), with hardware revision 0000 MPC860P processor: part number 5, mask 2
1 FastEthernet interface
32K bytes of NVRAM.
32768K bytes of processor board System flash (Read/Write)
Configuration register is 0x2102
Simmering#
Thank you,
Mark

I'm not that of an in-depth cisco programmer yet, I'm still working on my ccna - 1/2 way there, but this is what show diag gave me:
Simmering#show diag
Slot 0:
C1760 1FE VE 4SLOT DV Mainboard Port adapter, 1 port
Port adapter is analyzed
Port adapter insertion time unknown
EEPROM contents at hardware discovery:
Hardware Revision : 5.0
PCB Serial Number : FOC08252LAR
Part Number : 73-7167-05
Board Revision : B0
Fab Version : 04
Product (FRU) Number : CISCO1760
EEPROM format version 4
EEPROM contents (hex):
0x00: 04 FF 40 03 16 41 05 00 C1 8B 46 4F 43 30 38 32
0x10: 35 32 4C 41 52 82 49 1B FF 05 42 42 30 02 04 FF
0x20: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x30: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x40: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x50: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x60: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
0x70: FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF FF
Simmering#
I don't understand that because there is nothing in slot 0, and there are cards in slots 2 and 3. According to the cisco ios selector or whatever their cross-reference is said this ios has the voice/voip features, but if you have another ios version you think I should be using, please by all means let me know.
Thanks,
Mark

- FXO- FXS- call transfer not working...

We have cisco2811 with c2800nm-adventerprisek9-mz.124-16 and 1 FXO + 2 FXS ports.
at this moment i use CME B-ACD to auto answering and forward call to internal fxs port.
problem - this call, terminated to fxs, cann't be transferred by using flash button on analog phone(to transfer call from fxs to fxs port i use app-h450-transfer.2.0.0.9.tcl)
Can somebody tall me WHAT to use to make transfer feature working??
my config in attachment

Configuring Feature Mode for SCCP Analog Phones
This task configures feature mode for SCCP analog phones on the voice gateway and enables SCCP analog phones to invoke features using feature access codes after hook flash.
Prerequisites
You must first configure full consultation transfer on Cisco CallManager using the transfer-system full-consult command in order for feature mode to work on the voice gateway.
SUMMARY STEPS
1. enable
2. configure terminal
3. stcapp call-control mode feature
my IOS says that didn't know "stcapp" command....

Use of Digium 24-port FXO analog cards for FreePBX system

Does anyone have experience with the installation of Digium 1AEX2400 Series cards in a FreePBX distro? I am researching a new system and want to see if anyone has successfully used this hardware before, and if there was any trouble installing the hardware that I should know about. It would probably be installed in a Dell R310 1U server.
I have used 8-port Digium FXO cards for many years now. And yet I find very little on discussion groups for either the 8-port cards or their 24-port equivalent.
Any information would be helpful.
This topic first appeared in the Spiceworks Community

Right now I can get a 570 for 175 bucks... R4yMoNd. When I look at the 570 in comparison to what I have it doesnt look as though itis going to make too much difference. http://www.hwcompare.com/9048/geforce-gtx-570-vs-radeon-hd-5870/ I'm sure however the CUDA cores will speed it right up. I'm no good when it comes to all that so thanks to both you for the help. I have tracked down a 670 in stock (seems tough) for $440 then I need a 700 w power supply for about 70. If the 570 should take care of the problems listed above thats what I'm going for.. Just wasnt sure if it would make the needed difference

FXS Card and Fax Line.

I have a remote location that wants to add an additional fax machine. They currently have one configured like the following and all works just fine. dial-peer voice 6811 pots destination-pattern 6811 port 0/2/0 They called today and want to add a second, so I did the following. dial-peer voice 6841 pots destination-pattern 6841 port 0/2/1 We do have a voice translation rule that states all number from 6810 to 6899 if not configured should ring back to Receptionist. voice translation-rule 1 rule 1 /^68../ /9649110/ ! voice translation-rule 2 rule 1 // /0/ ! voice translation-rule 6 rule 1 /68../ /6810/ ! voice translation-rule 10 rule 1 /964910/ /6800/ rule 2 /96491/ /68/ ! voice translation-rule 11 rule 1 /^8/ /9/ ! ! voice translation-profile 6 translate called 6 ! voice translation-profile IncomingMain translate calling 2 translate called 10 ! voice translation-profile USA translate called 11 When you call 6811 it rings the Fax machine, and when you call 6841 it rings the receptionist line. What am I missing to allow this to work. If you need more of the config let me know. Dan

FrustratedSusan wrote:
If I do not catch my phone within the first 4 rings, it will ring through to our fax line. We do not have a dedicated fax line, just pay an extra fee per month to have to ability to send/received faxes ... People try to call me and cannot even leave a message - the fax line buzzes ...
You have not stated which specific services you have, and this makes it difficult to provide specific suggestions. If I assume you have VZ Digital Voice, then there is no need to have an additional "fax line." See these references:
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/How-do-I-order-a-fax-line/m-p/577521#M10...
http://forums.verizon.com/t5/Home-Phone-Landline-or-Digital/Verizon-Fax-Service/m-p/562659#M10655
It may help if I describe my system. I have an HP "all-in-one" device that uses a standard VZ Digital Voice line to send and receive fax. The procedure for setting up this device to do everything in the proper sequence is to set fax options on the device itself (depending obviously on your specific machine), and FiOS Digital Voice voicemail options on the Verizon pages or your phone itself (whichever is easier for you).
Provide more information and we will see if more specific answers are available.

Best way to extend FXO to a remote site???

Team,
I have a customer who is looking for the best way to simply extend 4 fxo lines from a location to their location. There is an ethernet connection to their location from the location containing the FXO lines. I know we can do this by putting in voice aware routers and fxo/fxs cards at both locations but that turns out to be quite expensive. I am curious if there is a simple way to do this through an appliance/gateway or any other creative way someone has tried.
Any thoughts and suggestions are welcome.
Thanks
Paul

Hi Paul -
If I understand you correctlly, I've done this is in the past but with one drawback.
What I did was to use an SPA-3102 at the office (which I had connected to a local PBX, which happened to have 4 outgoing PSTN lines - FXO), then at the remote end - I simplly used a number of spare 2100 ATA for each workstation.
The 3102 was setup in gateway mode, while each 2100 had thier own number and universal dial plan.
The dial plan, simplly allowed the users to dial in-house extensions hanging off the PBX, and by pressing 0 to get an outside line, this would allow you to call external parties through the FXO lines at the main office site.
Kind of what your after, I think?
Drawbacks:
I found I had major echo issues when using the 3102 as a PSTN gateway, the echo was not prevalent when dialling an internal PBX extension though.
Never really got the echo issue sorted out, just minimised as best I could.

Please Can Someone Help Me??????

Please, I've been working on this for a few weeks now, and have had to learn everything pretty much on my own, because no one can seem to give me an answer and help me. No one has even replied to 3 different posts about this subject, so I'm trying it in here.
Here's the situation. I have 2 1760 routers with DSPs, FXO, & FXS cards. They both have the on-board Fa 0/0 interface. I'm tieing two key systems together via VoIP trunks with these routers. I've completed the VoIP programming and it is working as expected. I'm now trying to implement a site-to-site VPN tunnel for these two routers and the VoIP traffic to go through.
I'm not sure if it can be done with just the onboard fa 0/0 interface, and no real LAN behind it. These routers will be on an existing network, however, they will be obtaining(static) a 2nd WAN IP to connect to the outside world only for VoIP traffic between the two.
Please...I'm not Cisco genius and I need some help. I've got probably 20 docs on configuring VPN, but I need some help and questions answered. I'll send my config if anyone needs to look at it.
Mark

Ah Nick...thank you for responding. Honestly, I don't understand a lot of what your saying. But, maybe seeing my config would help both of us. I honestly don't believe the H323 is being used. I think something more along the lines of a G721 codec or something like that. These are just strictly dial-peers to each other. I've since add ssh access, but here it is:
SITE 1:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname SITE 1
boot-start-marker
boot-end-marker
enable secret XXX
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.30 255.255.255.0
speed auto
no shutdown
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 290
voice-port 2/1
connection plar opx 291
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 190
voice-port 3/1
connection plar 191
voice-port 3/2
voice-port 3/3
dial-peer voice 180 pots
destination-pattern 180
port 2/0
dial-peer voice 181 pots
destination-pattern 181
port 2/1
dial-peer voice 190 voip
destination-pattern 19
session target ipv4:192.168.254.40
line con 0
logging synchronous
line aux 0
line vty 0 4
password xxx
logging synchronous
login
transport input telnet
end
AND SITE 2:
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname SITE 2
boot-start-marker
boot-end-marker
enable secret XXX
no aaa new-model
voice-card 2
voice-card 3
ip cef
interface FastEthernet0/0
ip address 192.168.254.40 255.255.255.0
speed auto
no ip http server
no ip http secure-server
control-plane
voice-port 2/0
connection plar opx 280
voice-port 2/1
connection plar opx 281
voice-port 2/2
voice-port 2/3
voice-port 3/0
connection plar 180
voice-port 3/1
connection plar 181
voice-port 3/2
voice-port 3/3
dial-peer voice 190 pots
destination-pattern 190
port 2/0
dial-peer voice 191 pots
destination-pattern 191
port 2/1
dial-peer voice 180 voip
destination-pattern 18
session target ipv4:192.168.254.30
line con 0
logging synchronous
line aux 0
line vty 0 4
password xxx
logging synchronous
login
transport input telnet
end
Thank you,
Mark

"frags delayed" counter incrementing for Voice PVC

Hi,
We are using VoFR between two Cisco 2610 using FXO\FXS Cards. It is a point-point link with two PVCs, one for Voice and one for Data.
I have implemented Traffic-Shaping and FRF. However when i do a "show frame pvc " command, i can see "frags delayed"counter incrementing for the Voice PVC, indiciating delay in sending packets and thus compromising Voice Quality.
1. Is it normal to have this counter increasing ? What is the acceptable percentage i.e "frags delayed \ total frags" ?
2. Is there anything i can do ? Would PVC Priority Queuing help ?
I need to be sure if PVC Priority is the solution, as we would have to do a Flash Upgrade to install the new software with this feature.
++++++++++++++++++++++++++
show frame pvc 103
PVC Statistics for interface Serial0/0 (Frame Relay DTE)
DLCI = 103, DLCI USAGE = LOCAL, PVC STATUS = ACTIVE, INTERFACE = Serial0/0.3
input pkts 373951 output pkts 374604 in bytes 11542352
out bytes 12245392 dropped pkts 0 in FECN pkts 0
in BECN pkts 0 out FECN pkts 0 out BECN pkts 0
in DE pkts 0 out DE pkts 0
out bcast pkts 5474 out bcast bytes 1571038
pvc create time 11w3d, last time pvc status changed 04:06:31
Service type VoFR-cisco
Voice Queueing Stats: 0/100/0 (size/max/dropped)
Current fair queue configuration:
Discard Dynamic Reserved
threshold queue count queue count
64 16 2
Output queue size 0/max total 600/drops 0
configured voice bandwidth 30000, used voice bandwidth 0
fragment type VoFR-cisco fragment size 320
cir 32000 bc 320 be 0 limit 40 interval 10
mincir 32000 byte increment 40 BECN response no
frags 374604 bytes 12261814 frags delayed 6501 bytes delayed 1609296
shaping inactive
traffic shaping drops 0
+++++++++++++++++++++++++++++++++++

The following links explains the delay in voice traffic and gow to do traffic policing
VoIP over Frame Relay with QoS (Fragmentation, Traffic Shaping, LLQ / IP RTP Priority)
http://www.cisco.com/warp/public/788/voice-qos/voip-ov-fr-qos.html#15
Troubleshooting Output Drops with Priority Queueing
http://www.cisco.com/warp/public/105/priorityqueuedrops.html
Understanding Delay in Packet Voice Networks
http://www.cisco.com/warp/public/788/voip/delay-details.html
Voice QoS: ToS-CoS Mapping Via LLQ
http://www.cisco.com/warp/public/788/voice-qos/tos-cos.html
Frame Relay Traffic Shaping for VoIP and VoFR
http://www.cisco.com/warp/public/788/voip/fr_traffic.html

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