CME FXO Card
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
Similar Messages
-
hi,
I have a 2801 CME and 2 fxo lines. Some times when both lines are seized (because of incoming and outgoing call) and after the phones are on hook the lines are still seized and are not released and when i removed the cables from fxo port, the led on each port start to blink in sequence and then when i reconnected, the port was busy again.. so i had to reboot the router to make it work. i changed the slot but dont think that it will cause any difference..do i need to upgrade the ios or some other issue..when i am calling the disconnect is fine but it happens some time when the line get stuck i have to reboot the router. one more thing when i changed the slot of the card the router didnt detected the change of slot and was showing fxo port on old slot. i rebooted the router to detect the fxo card on the right slot..sorry for the double post but no body was replying to i post it again..
Currently Being Moderated
May 12, 2012 11:49 AM (in response to Naresh Rathore)
FXO lines stuck/hang (2801 CME)
hi following is the sh run and sh version. cme output portion is removed
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.02.03 14:27:39 =~=~=~=~=~=~=~=~=~=~=~=
User Access Verification
Username: admin
Password:
Router_Home>en
Password:
Router_Home#term
Router_Home#terminal le
Router_Home#terminal length n 512
Router_Home#sh run
Building configuration...
Current configuration : 15203 bytes
! Last configuration change at 14:41:02 UTC Thu Feb 3 2011 by admin
! NVRAM config last updated at 14:38:09 UTC Thu Feb 3 2011 by admin
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router_Home
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
clock calendar-valid
crypto pki trustpoint TP-self-signed-2416845307
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2416845307
revocation-check none
rsakeypair TP-self-signed-2416845307
crypto pki certificate chain TP-self-signed-2416845307
certificate self-signed 01
30820243 308201AC A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32343136 38343533 3037301E 170D3131 30323032 31373332
34335A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313638
34353330 3730819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100DFAC E4312F46 0F2BF242 E55FE0C1 73DC95F2 B2844295 DA691CE3 D9A202B1
77B1B0C3 5AE3D936 C4D786DB 7CFA6624 024C6A82 A29B3AAC 3BA89A77 5425A97A
6D79A88A C1327171 C88AA5E0 AB52F461 87FB472E A7622955 17C8F22C 58842EF4
4DFA422A 54E6B96A FA536C59 BD93FDCD 872C0586 08117535 2D13F1E0 A53E65AB
FE470203 010001A3 6B306930 0F060355 1D130101 FF040530 030101FF 30160603
551D1104 0F300D82 0B526F75 7465725F 486F6D65 301F0603 551D2304 18301680
14E940B0 E437790B 4B825CD2 0FA9020F 63C9ED3A ED301D06 03551D0E 04160414
E940B0E4 37790B4B 825CD20F A9020F63 C9ED3AED 300D0609 2A864886 F70D0101
04050003 8181006B D6136D19 6EF4DDCD B3AF591E 57B9B831 79578799 03862FCF
4AF772DE AC72FC85 3F6B6B20 81F528F0 F7B2CBD0 E9795060 C46AB102 AE2CDF53
11C39D67 B49A7AE8 FB619A0F 525543F7 8BA1D52C CABEFFEB 9E5EC7E7 938AA602
84F1ECD2 303E8609 A9AB0699 9078051B 1853BC9A 4B45F2A2 204310D5 8B34B5DD
2FA51064 EFF5D4
quit
dot11 syslog
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 10.168.0.1
ip dhcp excluded-address 10.168.0.2
ip dhcp excluded-address 10.168.10.1
ip dhcp excluded-address 10.168.10.2
ip dhcp excluded-address 10.168.111.1 10.168.111.6
ip dhcp pool Data
network 10.168.0.0 255.255.255.0
default-router 10.168.0.1
option 150 ip 10.168.10.1
dns-server 212.72.1.186 212.72.23.4
ip dhcp pool VOICE
network 10.168.10.0 255.255.255.0
default-router 10.168.10.1
option 150 ip 10.168.10.1
ip dhcp pool WLAN
network 10.168.111.0 255.255.255.0
default-router 10.168.111.1
option 150 ip 10.168.10.1
dns-server 212.72.1.186 212.72.23.4
ip host members.dyndns.org 204.13.248.112
ip name-server 212.72.1.186
ip name-server 212.72.23.4
ip ddns update method example_dyndns
HTTP
add http://whatisthis:[email protected]/nic/update?system=dyndns&hostname=<h>&myip=<a>
interval maximum 28 0 0 0
interval minimum 28 0 0 0
multilink bundle-name authenticated
voice-card 0
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
h323
sip
registrar server expires max 600 min 60
username xxxxx password 0 xxxx
archive
log config
hidekeys
interface FastEthernet0/0
no ip address
ip nat inside
ip virtual-reassembly
speed auto
full-duplex
no keepalive
interface FastEthernet0/0.2
description ***** Voice LAN ******
encapsulation dot1Q 2
ip address 10.168.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/0.3
description ***** Data LAN ******
encapsulation dot1Q 3
ip address 10.168.0.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/0.4
description ***** Wireless LAN ******
encapsulation dot1Q 4
ip address 10.168.111.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1
description $ES_WAN$
no ip address
shutdown
duplex auto
speed auto
interface ATM0/1/0
no ip address
no atm ilmi-keepalive
dsl operating-mode auto
interface ATM0/1/0.1 point-to-point
pvc 0/35
pppoe-client dial-pool-number 1
interface Dialer0
no ip address
interface Dialer1
mtu 1492
ip address negotiated
no ip redirects
no ip unreachables
no ip proxy-arp
ip mtu 1452
ip nat outside
ip virtual-reassembly
encapsulation ppp
ip route-cache flow
ip tcp adjust-mss 1412
dialer pool 1
dialer-group 1
ppp authentication chap pap callin
ppp chap hostname xxxx
ppp chap password 0 xxxxx
ppp pap sent-username xxxx password 0 xxxxx
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 Dialer1
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip dns server
ip nat inside source list 101 interface Dialer1 overload
ip nat inside source route-map NAT_RMAP interface Dialer0 overload
access-list 101 permit ip 10.168.0.0 0.0.0.255 any
access-list 101 permit ip 10.168.111.0 0.0.0.255 any
route-map NAT_RMAP permit 10
match ip address 101
control-plane
voice-port 0/0/0
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
dial-peer cor custom
name local
name longdistance
dial-peer cor list call-local
member local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list manager
member local
member longdistance
dial-peer cor list other
member local
dial-peer voice 1 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/0
dial-peer voice 2 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/1
dial-peer voice 3 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/2
dial-peer voice 4 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/3
dial-peer voice 5 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/0
dial-peer voice 6 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/1
dial-peer voice 7 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/2
dial-peer voice 8 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/3
dial-peer voice 9 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/0
prefix 800
dial-peer voice 10 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/1
prefix 800
dial-peer voice 11 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/2
prefix 800
dial-peer voice 12 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/3
prefix 800
telephony-service
no auto-reg-ephone
max-ephones 25
max-dn 25
ip source-address 10.168.10.1 port 2000
auto assign 1 to 20
timeouts interdigit 5
system message Home
max-conferences 8 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult dss
transfer-pattern .T
ephone-dn 1 dual-line
number 101
label M.M Office (101)
description M.M Office (101)
name M.M Office
ephone 1
device-security-mode none
video
mac-address 1C17.D3C3.7A31
speed-dial 1 102 label "abc"
speed-dial 2 103 label "def"
speed-dial 3 104 label "hij"
speed-dial 4 113 label "klm"
speed-dial 5 104 label "nop"
speed-dial 6 105 label "qrst"
speed-dial 7 105 label "uvw"
type 7975
button 1:1 2:21
ephone-hunt 1 sequential
pilot 150
list 121
line con 0
exec-timeout 0 0
logging synchronous
login local
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
ntp master 2
ntp update-calendar
end
Router_Home#sh di
Router_Home#sh dia
% Ambiguous command: "sh dia"
Router_Home#sh ds
Router_Home#sh dsv p
Router_Home#sh dspfarm
% Incomplete command.
Router_Home#sh dspfarm ?
all Display all DSPFARM global info
dsp Display DSPFARM DSPs information
profile Display DSPFARM profiles
Router_Home#sh dspfarm ds
Router_Home#sh dspfarm dsp
% Incomplete command.
Router_Home#sh dspfarm dsp all
Total number of DSPFARM DSP channel(s) 0
Router_Home#sh ver
Router_Home#sh version
Cisco IOS Software, 2801 Software (C2801-ADVIPSERVICESK9-M), Version 12.4(15)T10, RELEASE SOFTWARE (fc3)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2009 by Cisco Systems, Inc.
Compiled Mon 14-Sep-09 14:51 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T9, RELEASE SOFTWARE (fc1)
Router_Home uptime is 21 hours, 11 minutes
System returned to ROM by power-on
System restarted at 17:31:27 UTC Wed Feb 2 2011
System image file is "flash:c2801-advipservicesk9-mz.124-15.T10.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 2801 (revision 6.0) with 237568K/24576K bytes of memory.
Processor board ID FCZ10491208
2 FastEthernet interfaces
1 ATM interface
2 Virtual Private Network (VPN) Modules
4 Voice FXO interfaces
1 DSP, 8 Voice resources
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
Router_Home#exit
RegardsActually this has been answered already:
https://supportforums.cisco.com/thread/2148571?tstart=0
please do not open duplicate threads. You can delete you post using the Actions panel on the right. -
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 -
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 -
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 -
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. -
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. -
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! -
Hi there,
I have an install of a CME with FXO ports going live this week and I was wondering if someone could check over my sound extract and confirm for me that the frequency is 415 and cadence is 510 280. I can't test until the lines go over but just want to make sure as best as I can that it will be a smooth changeover.
Also here is my cptone and voice port config:
voice class dualtone-detect-params 1
freq-max-deviation 30
cadence-variation 20
voice class custom-cptone DISCONNECT_CP
dualtone disconnect
frequency 415
cadence 510 280
voice-port 0/1/0
trunk-group PSTN_TRUNKS 1
supervisory disconnect dualtone mid-call
supervisory answer dualtone
supervisory custom-cptone DISCONNECT_CP
supervisory dualtone-detect-params 1
cptone FR
connection plar opx 100
caller-id enable
Thanks
CarlHello Carl,
The parameters are correct but the sum of cadences must be greater than "timeout wait-release" under the voice-port. You can check the timer with the command "show voice port 0/1/0".
eg if the timer is 1 sec , 510+280=790ms < 1s , hence change the cadence value to '510 280 510 280".
Regards,
Harsh -
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.
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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. -
I have a 2801 router with cme and a four ports FXO card, I have a 800 toll free with three lines, any idea on how to configure this type of service?
Is just one telephone number asocciate with 3 physical lines on three port int he cisco fxo card.Yes, this function is configure I make test with all the options in the web page but continues working worng:
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is enabled
Number Of Rings is set to 1
Supervisory Disconnect is signal
Answer Supervision is inactive
Hook Status is Off Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 1 -
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 CommunityRight 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
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