FXS versus FXO cards in 2821
I am new to Cisco CME and VOIP and am trying to configure a new 2821 with CME 4.1. I have two 4 port FXS cards. The intent was to use the ports on these cards to connect to PSTN POTS lines for inbounce and outbound cards. Can I use these FXS cards for this purpose or do they need to be FXO cards? According to the description they do as the FXS is intended to have phones connected and supplies dial-tone and the FXO does not supply dial-tone and is intended to be connected to PSTN. The reason that I am a little confused is that I have configured the system to dial out but was having issues where it would not hang up the line.
Hi Michael,
Here is some background info to add to Paolos always Great info (hey P.);
Analog Telephony Protocols
Analog telephony signaling, the original signaling protocol, provides the method for connecting or disconnecting calls on analog trunks. By using direct current (DC) over two-wire or four-wire circuits to signal on-hook and off-hook conditions, each analog trunk connects analog endpoints or devices such as a PBX or analog phone.
To provide connections to legacy analog central offices and PBXs, Cisco CallManager uses analog signaling protocols over analog trunks that connect voice gateways to analog endpoints and devices . Cisco CallManager supports these types of analog trunk interfaces:
Foreign Exchange Office (FXO) Analog trunks that connect a gateway to a central office (CO) or private branch exchange (PBX).
Foreign Exchange Station (FXS) Analog trunks that connect a gateway to plain old telephone service (POTS) device such as analog phones, fax machines, and legacy voice-mail systems.
From this good doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00801ec5cc.html#1134121
FXS and FXO Interfaces
An FXS interface connects the router or access server to end-user equipment such as telephones, fax machines, or modems. The FXS interface supplies ring, voltage, and dial tone to the station and includes an RJ-11 connector for basic telephone equipment, keysets, and PBXs.
An FXO interface is used for trunk, or tie line, connections to a PSTN CO or to a PBX that does not support E&M signaling (when local telecommunications authority permits). This interface is of value for off-premise station applications. A standard RJ-11 modular telephone cable connects the FXO voice interface card to the PSTN or PBX through a telephone wall outlet.
FXO and FXS interfaces indicate on-hook or off-hook status and the seizure of telephone lines by one of two access signaling methods: loop start or ground start. The type of access signaling is determined by the type of service from the CO; standard home telephone lines use loop start, but business telephones can order ground start lines instead.
Loop-start is the more common of the access signaling techniques. When a handset is picked up (the telephone goes off-hook), this action closes the circuit that draws current from the telephone company CO and indicates a change in status, which signals the CO to provide dial tone. An incoming call is signaled from the CO to the handset by sending a signal in a standard on/off pattern, which causes the telephone to ring.
Loop-start has two disadvantages, however, that usually are not a problem on residential telephones but that become significant with the higher call volume experienced on business telephones. Loop-start signaling has no means of preventing two sides from seizing the same line simultaneously, a condition known as glare. Also, loop start signaling does not provide switch-side disconnect supervision for FXO calls. The telephony switch (the connection in the PSTN, another PBX, or key system) expects the router's FXO interface, which looks like a telephone to the switch, to hang up the calls it receives through its FXO port. However, this function is not built into the router for received calls; it only operates for calls originating from the FXO port.
Another access signaling method used by FXO and FXS interfaces to indicate on-hook or off-hook status to the CO is ground start signaling. It works by using ground and current detectors that allow the network to indicate off-hook or seizure of an incoming call independent of the ringing signal and allow for positive recognition of connects and disconnects.
From this very descriptive doc;
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_configuration_guide_chapter09186a0080080afd.html
Hope this helps!
Rob
Similar Messages
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Connecting FAX to PSTN line using FXS and FXO ports
I have 2 cisco 1760 routers with FXS and FXO card installed in each. I have to transport 2 PSTN lines from Head Office to remote loaction using FXS and FXO cards and these lines will be used for voice calls and FAX( one line each for FAX and voice). I have configured the router for voice lines and it is working fine. Now I am using the same config for running fax machine but it is not working. Can anyone help me out how to configure FAX in this scenario also if anyone can share any sample config. I am attaching my config for both routers. Right now both routers are connected with a cross-over cable for lab test but we will connect them later using satellite connection.
HO Router (FXO card)
Current configuration : 1718 bytes
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
resource policy
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
ip subnet-zero
ip cef
no ip dhcp use vrf connected
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.1 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.1
ip classless
no ip http server
control-plane
voice-port 2/0
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2001
description Remote PSTN#:35296913
music-threshold -70
voice-port 2/1
output attenuation 0
echo-cancel coverage 32
no vad
no comfort-noise
timeouts interdigit 3
timeouts call-disconnect 3
connection plar opx 2002
description Remote PSTN#:35296914
music-threshold -70
voice-port 3/0
voice-port 3/1
dial-peer voice 2000 voip
destination-pattern 200.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.2
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
dial-peer voice 1321 pots
description line 1
huntstop
destination-pattern 1321
port 2/0
dial-peer voice 1322 pots
description line 2
huntstop
destination-pattern 1322
port 2/1
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
end
Remote Router (FXS Card):
version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
enable password cisco
mmi polling-interval 60
no mmi auto-configure
no mmi pvc
mmi snmp-timeout 180
voice-card 2
voice-card 3
no aaa new-model
ip subnet-zero
ip cef
no ip domain lookup
no ftp-server write-enable
voice class codec 10
interface FastEthernet0/0
ip address 10.10.10.2 255.255.255.248
speed auto
h323-gateway voip bind srcaddr 10.10.10.2
interface Ethernet1/0
no ip address
shutdown
half-duplex
ip classless
no ip http server
control-plane
voice-port 2/0
description PSTN#:
voice-port 2/1
description PSTN#:
voice-port 3/0
voice-port 3/1
dial-peer voice 2001 pots
description Remote
huntstop
destination-pattern 2001
port 2/0
dial-peer voice 2002 pots
description Remote
huntstop
destination-pattern 2002
port 2/1
dial-peer voice 1320 voip
destination-pattern 132.
no modem passthrough
voice-class codec 10
session target ipv4:10.10.10.1
incoming called-number .
dtmf-relay cisco-rtp h245-signal h245-alphanumeric
fax-relay ecm disable
fax rate 7200
fax nsf 000000
no vad
line con 0
password cisco
line aux 0
line vty 0 4
password cisco
login
endIn your voice class codec 10 there aren't any codecs declared.
Add G.711 codec in this way:
voice class codec 10
codec preference 1 g711alaw
If the fax communication fails again try to disable T.38 and try fax passthrough mode:
no fax rate
modem passthrough nse codec g711alaw
fax protocol pass-through g711alaw
Regards. -
UC 540 VIC 4 FXO CARD ADDITION IN VIC EXPANSION SLOT
Dear all,
I would like to add an additional VIC 4 FXO card in UC 540 expansion slot. Can I add additional VIC 4 FXO card without adding DSP modules.?
Same device I have to create auto attendant system. What are the licence required for auto attendant feature ?
Products details are given below
UC 540 VIC 4 FXO CARD ADDITION IN VIC EXPANSION SLOT
Cisco UC540W-FXO-K9 (MPC8358) processor (revision 0x100) with 235520K/26624K bytes of memory.
MPC8358 CPU Rev: Part Number 0x804A, Revision ID 0x20
30 User Licenses
10 FastEthernet interfaces
2 terminal lines
1 Virtual Private Network (VPN) Module
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Voice MoH interface
1 802.11 Radio
1 cisco service engine(s)
128K bytes of non-volatile configuration memory.
250880K bytes of ATA CompactFlash (Read/Write)
Regards,
Ajay Jose KHi Emmanuel Valdez,
Thank you for your support.
We are not using conference and Transcoding in our network.We have built in VIC 4 FXO and VIC 4 FXS in our system.When I go through show dsp farm
Dspfarm Profile Configuration
Profile ID = 1, Service = CONFERENCING, Resource ID = 1
Profile Description : DO NOT MODIFY, active CCA conference profile - CCA2.0 codec711
Profile Service Mode : Non Secure
Profile Admin State : UP
Profile Operation State : ACTIVE
Application : SCCP Status : ASSOCIATED
Resource Provider : FLEX_DSPRM Status : UP
Number of Resource Configured : 4
Number of Resource Available : 4
Maximum conference participants : 32
Codec Configuration: num_of_codecs:2
Codec : g711alaw, Maximum Packetization Period : 30 , Transcoder: Not Required
Codec : g711ulaw, Maximum Packetization Period : 30 , Transcoder: Not Required
SLOT DSP VERSION STATUS CHNL USE TYPE RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED
0 3 28.3.5 UP N/A FREE conf 1 - - -
0 3 28.3.5 UP N/A FREE conf 1 - - -
0 4 28.3.5 UP N/A FREE conf 1 - - -
0 4 28.3.5 UP N/A FREE conf 1 - - -
Total number of DSPFARM DSP channel(s) 4
Can I use auto attendant system with out CUE ?
Thanks&Regards,
Ajay Jose K -
Need to test a bunch of FXO cards.
I am tasked with testing a bunch of FXO cards (VIC-2FXO) and I have a 1760 router with an FXS card in it. I configured the router so that when I call from an IP phone (ext 4000) to the FXS card port 1/0 (ext 4015) it calls the FXO card on port 0/0 which uses plar to ext 4444 which is port 1/1 on the FXS card.
I also tried to plar calls to ext 4001 an IP phone.
When I call 4015 from the IP phone, it rings the 4444 extension once then the FXS port 1/0 (ext 4015) hangs up. A call remains between the FXO port 0/0 and the FXS port 1/1 but since the FXS port 1/0 dropped off you can not talk to the IP phone.
4001 calls ---> 4015 FXS 1/0 which rings ---> FXO 0/0 using PLAR rings ---> 4444 which is on FXS port 1/0
The port light for FXS 1/0 comes on during the call proces but then goes out. The port lights for the FXO 0/0 and FXS 1/1 stay light until I unplug th RJ11 cable between them.
Any ideas how I can keep FSX 1/0 from dropping one the call is answered on FXS 1/1?
Current configuration : 1917 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 15
voice-card 0
voice-card 1
interface FastEthernet0/0
ip address 192.168.69.200 255.255.255.0
speed auto
voice-port 0/0
connection plar 4444
voice-port 0/1
voice-port 1/0
station-id number 4015
voice-port 1/1
station-id number 4444
dial-peer voice 400 pots
destination-pattern 4015
port 1/0
dial-peer voice 200 pots
incoming called-number .
port 0/0
dial-peer voice 444 pots
destination-pattern 4444
port 1/1
telephony-service
max-ephones 6
max-dn 6
ip source-address 192.168.69.200 port 2000
auto assign 1 to 6
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 4000
ephone-dn 2 dual-lin
number 4001
ephone 1
no multicast-moh
mac-address 000A.8AF0.3016
type 7960
button 1:1
ephone 2
no multicast-moh
mac-address 001B.535C.CABD
type 7940
button 1:2
line con 0
line aux 0
line vty 0 4
login
end
ThanksMichael,
I have seen issues like this and its all dow to actual physical wiring.
Your 2 port FXS & FXO cards have 2 lines available on each port.
Can you ensure that you wire everything as a single 2 wire.
FXS 1/0 RJ11 PIN3 ------------------------------------------------- FXO 0/0 PIN 3
FXS 1/0 RJ11 PIN4 ------------------------------------------------- FXO 0/0 PIN 4
FXS 1/1 RJ11 PIN3 ------------------------------------------------- EXT 4444
FXS 1/1 RJ11 PIN4 ------------------------------------------------- EXT 4444
Do not put through any other wires
Regards,
Alex.
Please rate useful posts. -
I have a 2621xm router. Can someone tell me which Network Module I need to get to support either 2 VIC-2FXO cards or 1 VIC-4FXO card. And these hardwares will be use in India so I also need to know which VIC-FXO card I need to get for India. Thank you in advance !!!
I understand the VIC-4FXO is universal but not sure of the VIC-2FXO.
Any helps will be greatly appreciated !!
DannyThose cards work with NM-1V and NM-2V
(Currently EoS)
Customers are encouraged to migrate to the Cisco IP Communications Voice/Fax Network Modules (NM-HD-xx) and associated VICs. The Cisco IP Communications Voice/Fax Network Modules (NM-HD-1V, NM-HD-2V and NM-HD-2VE) support all the features of the current low density voice/fax network modules (NM-xV) while offering significant advantages such as higher port and channel densities and a new generation of digital signal processors (DSPs) with greater memory and processing power
This link contains the info you want.
http://www.cisco.com/en/US/products/hw/modules/ps5365/products_data_sheet0900aecd801c595e.html -
We have an fxo card that used to work with an intercom. It is no longer working. No configs changed or anything. Is there any command or method of testing an FXO card in a CME>
You can open a TAC case and collect some details debugs, if you have a contract.
You can also try changing ports (and the configuration for the related port) to see if it happens on another port.
May also want to check deeper into your network to see if anything else has changed.
For instance, upgrading from CUCM 6 to CUCM 7 would change your phone loads, and the phone load in CUCM 7 by default has a nasty duplicate DTMF bug. This duplicate DTMF bug may be causing users to dial the wrong number when they dial in. (If your IP phones are 8.4.1 or 8.4.1 sr 1 this is the problem).
You can check show diag to make sure the board is 'analyzed' and doesn't have an error message related to it.
But there is no 'check hardware' command or anything like that.
hth,
nick -
I have 2620 router with fxo card. The call is not disconnecting at the fxo side after the call is complete. This issue is happening only with PBX. With PSTN no problem.
I tested all options of our discconecting FXO card from cisco site. It didn't help.
Whether this is bug !!!!!!!!!!! help out guys!!!!!!!!!!!
Anybody faced similiar issues
Thankxhi
we are facing the same issue : also be careful of the fxo as a source of crosstalk
I will let you know when we have these issues resolved -
Show ringing status on FXO card.
Hi all,
This might sound a bit nit pickey, but you know that when you typically dial out on a T1 or other digital line, it show on the phone the it is ringing. That is basically what I want to do with my FXO card. I would like the CM or Gateway to hear the tones, and send the Cisco Call Manger ringback status, out-of-service, and end-call. So that when I, or anyone, dials a number out, it shows it ringing, instead of connected, while it is still ringing.
Thank you for all your help.
Sincerly,
JZciHi Jonathan,
Just as an update, i made the changes on the config of the FXO card but the port still doesnt go to on-hook after a call is dropped. I however observed that this only happens when it is an incoming call. All incoming calls land on the IVR. So when I call in, I disconnect the call just after the attendant starts talking. When I check the status of the port, it remains off-hook till I shut down the port or disconnect the telephone line physically from the FXO card.
After applying echo-cancel enable and supervisory disconnect anytone commands, here is the config I have on the router now:
voice-port 2/0/0
supervisory disconnect anytone
cptone NG
timeouts call-disconnect 3
timeouts wait-release 3
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
I removed the following commands:
compand-type a-law
impedance complex2
Would appreciate any further suggestions.
Regards,
Femi -
Fxo card is not connecting dtmf calls
I have an analogue line that is being set up from GSM modem and connected to fxo card on VG router .The fxo card is configured to go to an automated message with option of pressing digits to take one to the right operator.i.e press 1 for....,2 for......,etc. Unfortunately,this is not working when i callled the number.Its looking like a DTMF issue,but i ve adjusted all the seeting that has to do with dtmf on the fxo card......Has anyone encounter any familiar issue?
Check to see if you have the voice license bundle for the router.
show license
Index 3 Feature: uck9
Period left: Life time
License Type: Permanent
License State: Active, In Use
License Count: Non-Counted
License Priority: Medium
Without the voice license, no voice ports can be configured on 2900 series routers. -
Differences between FXS and FXO ...?
Could someone let me know the difference between FXS and FXO with respect to a router, a PBX, a CO Switch, an analog phone, an ISDN phone?
Here are a couple of URLs explaining the same:
http://www.qtelnet.com/downloads/FXO-vs-FXS-v3.pdf
http://www.patton.com/technotes/fxs_fxo.pdf -
I have a 2811 with one four-port FXO card; 4 POTS lines are connected to it. Syslog is configured on the router, sending traps (debug level) to a host. For some reasons, I do not get any info in syslog form the card; so making or receiving a call sends nothing. Any ideas?
Thanks, S.This is the normal, making and receiving calls do not generate syslog output. If you want output, either enable some debug permanently, or use a radius server for exact accounting.
Hope this helps, please rate post if it does! -
Router Model 2801 with fxs/DID and FXO card.
Greetings.I am Yeoh here I just configured DID direct inward dial on the fxs port.RJ11 connection to pbx. dunno wheter it will function or not. should i configure the ring and voltage for the connection to run? Here i have router 2800 . Not really sure what is require to configure but the manager here says we need this running with pbx and only simple flat network. no other routing function used. the important part here is the voip configuration. I have two router one with fxs card another one with fxo. For what i understand is PBX connection plug to fx0 then from router a to router b using fe0/0 to fe 0/0 then from router b i connect it to the phone using FXS/DID is this correct? what is the configs to make my phone running. i plug it in but it seems not responding. no sound at all. just some small "TUK TUK TUK" voice in it.
Anyone can help me. Please.
-
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. -
Hi I have a VIC2-4FXO card in my Cisco 2901 router and its not responding. i replaced it with another workiing card and still the new card didnt respond.
any advice anyone?
Thanks,
Surendrahi,
yes, i can see it in my inventory and i am also seeing my PDVm in my inventory.
i dont have anything else on the router that is using dsp resources. -
Hi all,
I want to know if there is some design guide or reference to follow about WAAS implementation with IP telephony.
I've implemented WAAS in a customer's network with a central site and 20 remote offices and after that there are some problems with the telephony, specifically with call signaling. There are Cisco IP Phones in the branch offices, with a UCM in the central site and there are too FXS and FXO cards with analogic phones in the remote office.
Thanks for your comments.
Regards,
GuzmánHi Guzman,
What version of WAAS did you deploy? the images below are form 4.2.3...
By default, call signaling in the Ip telephony world, is not optimized by WAAS according to the default policy entry in the default policy template.
Maybe you are looking for
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