3825 FXO Port remains in off-hook after call

Hello,
I have a 3825 router with 8 FXO ports running Cisco IOS Software, 3800 Software (C3825-SPSERVICESK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2). The problem we are facing is that after a call is placed through any of the FXO ports and the call is ended by the user, the port remains in off-hook till a reset of the port is done or someone restarts the router. Only then is the port accessible again.
I am thinking of changing the cards, but i do not want to invest in replacing the cards and then find out that this doesnt solve the problem.
The wierd thing is that this issue started on its own accord not too long ago.
Comments and suggestions please!
Regards,
Femi

Hello,
I do not want to change the FXO card till I am sure that is the problem and I did state that I always have to reboot the router when the problem starts. Rebooting clears the problem but it is back immediately I attempt a call again and hang up that call.
I have timeouts call-disconnect already configured, see below:
voice-port 0/0/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 2626878
caller-id enable type 1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
Regards,
Femi

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FXO port not going back onhook when call goes to CUC Voice mail.

HI All,
Quiet new to the voice world and have just set up a home lab and have run into the following problem.
When a call comes in from the PSTN and gets transferred to voicemail CUC, when the caller has left the message and hung up the FXO port on my Cisco 3825 does not go back on hook
If a call comes in from the PSTN and is answered on a phone registered to the CUCM and the internal user hangs up the FXO port goes back on hook but if the caller hangs up the FXO port does not go back on hook.
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Hi Shaun,
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Using PCM capture we can findout the exact frequency and cadence of the disconnect tone sent by provider.

SPA400 FXO Ports are not dropt down after receving a call

Hi Team, does any one know why all FXO ports on a SPA400 are not disconnecting calls (there is not much configuration options to make sure we tier down ports)?
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I'm looking at an older firmware version, so I'm not quiet sure if this is also available on your SPA400, under Setup then Voice tab , there's some parameter at the bottom for disconnect like , battery reversal as disconnect signal , tear down fxo port when silence detected for n seconds. Have you tried tweaking this settings? Also, on the SPA9000, you can try setting Reorder delay to 255 and see if it will help.

VG224 port stuck in 'off-hook' state

Hi All,
I have strange problem with my VG224 connected to CME4.0.
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IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
============== == ============ ===== ==== ======== ======== ==
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Configuration for VG
======================
stcapp ccm-group 1
stcapp
stcapp feature access-code
prefix *#
voice-card 0
interface FastEthernet0/0
ip address 10.102.1.35 255.255.255.0
duplex auto
speed auto
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sccp local FastEthernet0/0
sccp ccm 10.102.1.12 identifier 10 version 4.0
sccp
sccp ccm group 1
associate ccm 10 priority 1
dial-peer voice 3 pots
service stcapp
port 2/2
=========================
This port work well for 2 weeks before this problem happen. Another port have same configuration and its working

FXO ports got jammed after a call is placed

Hi team.
We have a analog line connected to an FXO port, every time we make a call through this line something happens and it's required for us to go to the gateway, unplug and plug again the line, next time anybody makes a call, it shows the same behavior in the line.
Best regards
Fernando

You can refer to the many threads on the same subject
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

FXO port issue

Dear all,
Am using cisco 2921 CME 8.6 with ios version c2900-universalk9-mz.SPA.151-4.M4.bin..
Also am using a dedicated one fxo port for a specific user to send and receive calls,, the user always faces busy tone when initiate outgoing call, the fxo port doesnt disconnect from the previous call.
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts interdigit 3
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4015
impedance complex2
caller-id enable
Any Idea to solve the issue, please???
#sh voice port 0/3/2
Foreign Exchange Office 0/3/2 Slot is 0, Sub-unit is 3, Port is 2
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar opx
Connection Number is 4015
Initial Time Out is set to 15 s
Interdigit Time Out is set to 3 s
Call Disconnect Time Out is set to 1 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 1 s
Companding Type is u-law
Region Tone is set for SA
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to complex2 Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming):
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is disabled
Number Of Rings is set to 1
Supervisory Disconnect is dualtone mid-call
Answer Supervision is inactive
Hook Status is On 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 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
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Hi again Lan,
Strange issue occur, the voice ports seem to be ok, in on-hook status and incoming calls are fine, the strange issue is outgoing calls can`t be initiated despite the dial-peer are configured. this issue happens to me suddenly before changing any thing. As soon i dial the first digit after dialing 9, i receive a fast busy tone, like below as soon i dial 7 i receive a fast busy tone from an ip phone with no restriction...
dial-peer voice 2002 pots
trunkgroup 2
corlist outgoing CDMA
destination-pattern 977.......
forward-digits 9
voice-port 0/2/0
trunk-group 2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4446
impedance complex2
caller-id enable
I restart the router, boot from another ios, remove and configure the dial-peers, shut and unshut the voice port, nothing solve the issue. it seems it doesn`t recognize outside digits,,, local calls are working fine..
Any ideas to solve the issue please...
the below is info you requested while the ports are hanging:
VoiceGW#sh voice call status
CallID     CID ccVdb      Port        Slot/DSP:Ch Called #   Codec    MLPP Dial-peers
0x11       121D 0x2B326244 0/2/1            No DSP 7444624261 None     0/1015
0x17       121D 0x2B334ADC 0/3/3            No DSP *7444624261 None      1015/0
1 active call found
VoiceGW#sh voice call status
Enter configuration commands, one per line. End with CNTL/Z.
VoiceGW(config)#exdo sh voice port summ
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/1/0           -- fxs-ls      up    dorm on-hook idle     y
0/1/1           -- fxs-ls      up    dorm on-hook idle     y
0/1/2           -- fxs-ls      up    dorm on-hook idle     y
0/1/3           -- fxs-ls      up    dorm on-hook idle     y
0/2/0           -- fxo-ls      up    dorm idle     on-hook y
0/2/1           -- fxo-ls      up    up   idle     off-hook y
0/2/2           -- fxo-ls      up    dorm idle     on-hook y
0/2/3           -- fxo-ls      up    dorm idle     on-hook y
0/3/0           -- fxo-ls      up    dorm idle     on-hook y
0/3/1           -- fxo-ls      up    dorm idle     on-hook y
0/3/2           -- fxo-ls      up    dorm idle     on-hook y
0/3/3           -- fxo-ls      up    up   idle     off-hook y
50/0/1          1      efxs     up    up   on-hook idle     y
50/0/1          2      efxs     up    up   on-hook idle     y
VoiceGW(config-voiceport)#no sh
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, FXOLS_CONNECT, E_HTSP_IF_OOS]htspm_mgt_statehtspm_disc_ind
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
Jul 31 12:13:57.923: htsp_timer - 1000 msec
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_RELEASE_REQ]act_disc_conf
Jul 31 12:13:57.923: htsp_process_event: [0/3/3, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
QNB_VoiceGW(config-voiceport)#Jul 31 12:13:57.923: htsp_timer_stop3
Jul 31 12:13:57.923: [0/3/3] set signal state = 0x4 timestamp = 0
Jul 31 12:13:57.923: htsp_timer - 2000 msec
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_EVENT_TIMER]htspm_disc_conf
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_OOS_CONF]
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_timer_stop3
Jul 31 12:13:58.923: htsp_timer_stop_mlpp
Jul 31 12:13:58.923: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to Administrative Shutdown
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_DSP_INTERFACE_INFO]
Jul 31 12:13:59.339: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_INSERVICE]
Jul 31 12:13:59.343: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to up
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 rx_signal_map:
F F F F
5 F F F
F F F F
F F F F
Jul 31 12:13:59.343: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 tx_signal_map:
0 4 4 4
4 4 6 4
C C C C
C C C C
Jul 31 12:13:59.343: htsp_process_event: [0/2/1, S_OPEN_PEND, E_HTSP_GO_UP]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_NULL, E_HTSP_INIT]fxols_null_init
Jul 31 12:13:59.455: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_INIT, E_HTSP_INSERVE]fxols_init_inserve
Jul 31 12:13:59.455: [0/2/1] set signal state = 0x4 timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.923: htsp_process_event: [0/3/3, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Jul 31 12:13:59.927: htsp_process_event: [0/3/3, FXOLS_ONHOOK, E_DSP_SIG_0100]

Transfer called number from FSX port SPA8000 to FXO port analog PBX.

Hello all!
I have CUCM 8.6 which connecting with SPA8000 by SIP trunk.
FXS port SPA8000 connect to FXO port analog PBX.
When I call from CUCM to analog PBX through SPA8000, SPA8000 does not pass called number in line.
Anybody have ideas?

Here is what I would recomend, I have set this up at several installations and it works well. The only difference would be that the ones that I have setup had more than 64K between sites but that is still doable so long as you keep the traffic on that link well regulated.
Anyhow. At each site install a VIC-2FXO and a VIC-2FXS into your 1760. You can use the higer density modules if you need more paths. Tie the FXOs to analog station ports on your PBX and the FXSs to analog CO ports. This may seem backwards but it is correct. Cisco's FXS is expecting to provide dialtone and the analog CO ports on the PBX are expecting to receive dial tone. Depending on what you are using for call processing ITS/CCME/CCM configure your gateway and your route pattern, etc. You will also need to be able to somehow access those fxs ports for outbound calls. You could have direct line access buttons on your phones and instruct users that when calling the remote site the must use lines 7 and 8 as an example. You could also setup a line pool code for these new lines.
Now when site A wants to call Site B, they dial whatever you wish them to dial. You do whatever translations you may wish prior to sending the call. Here is how the call flow works. User at site A accesses the line either by hitting a line button or dialing the line pool code. This grabs the analog line tied to one of the FXs ports on the router. The call is then sent over the 64K link. The router at Site B then uses one of the FXO ports to pass the call to an analog station port on the other PBX. To the PBX at site B it will appear as though someone pickedu picked up an analog station and dialed a local extension.
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How do FXO ports work?

I am trying to understand how FXO ports work. I have little experience with this so I am trying to learn how it works at our office. We have a Cisco 2801 with CME 8.1 and CUE 3.2. The 2801 has a VIC2-4FXO card in it. It has 3 analog phone lines attached to it and are setup in a hunt group by the carrier. There is an Auto Attendant setup that prompts a caller with various options for redirecting calls.
1. Question: When a call comes into the Auto Attendant, I assume this ties up a single analog line. If there is a second call into the main number it also goes to the Auto Attendant and is routed to the number or appropriate voicemail. Would this tie up a second analog line?
2. Question: If a phone in the office is used to make an outbound call or a person in the office calling someone outside of the office at the same time as above, then would this tie up the 3rd analog line?
3. Question: If there is another person calling the AA at the same time as the previous 2 questions. We get a busy signal. Does this mean that all 3 analog lines are currently busy and no other call can come in or be made out due to the 3 lines being currently used?
When I run the following commands on the
router# sh voice call sum
port 0/0/0 S_Connect FXOLS_Connect
port 0/0/1 S_Connect FXOLS_Connect
port 0/0/2 S_Connect FXOLS_Connect
Also when a port is used and the caller hangs up the phone it takes between 30 and120 seconds for the call to release. Is this normal for analog lines to take time to release in the phone system?
Are all these things normal or if I am having busy calls then do we have something configured improperly?
Thanks for any help

Hi,
As your router has 3 analog lines connected to FXO port , so at a time 3 call maximum you can receive incoming and call outgoing (if you dont have any other lines or trunk).
1. Would this tie up a second analog line? - Yes
2. then would this tie up the 3rd analog line? - Yes
3. Does this mean that all 3 analog lines are currently busy and no other call can come in or be made out due to the 3 lines being currently used? - Yes
Now apply these two commands under voice port and then check the call release time.
timeouts call-disconnect 1
timeouts wait-release 1
Rate all helpful post
Thanks
Manish

Why do ports remain open after closing Firefox?

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Scrren remains black/off after switched off to save power.

I have this problem on two different Porteges with
windows 98 second edition.
They are both obtained second handed, and all drivers are
as found on the internett. And the windows 98 is
standard CDs (not Toshiba CD). Both have the latest BIOS.
Everything else is working fine, this is the only
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This works just fine. The screen goes black, and then I
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Hi,
which Portege models do you own? Because as you explained you use the MS Power Management which is not the best one!
You can look at http://eu.computers.toshiba-europe.com - Driver downloads if the Toshiba Power Saver is available for your units and W98!
It can be that the old W98 Power management has problems with the complex hardware of the portege series therefore is the power saver availale at Toshiba site!
Good luck
Rudi

Issue connecting SPA3102 FXO port to paging interface

Hello,
I was referred to this community after a call to small business support, we were unable to come to a solution.
SPA3102 v5.2.13 FW
Viking Electronics PI-1 Paging Interface
We were told from Viking that the interface only supports an FXO port, all the connections were verified with their support and we're able to page by connecting an analog phone straight to the interface. Plugging into the FXS port on the SPA3102 also results in a dial tone playing over the paging system.
All settings on the SPA3102 are default minus SIP registration information, we see the unit as registered within the web interface as well as within our Asterisk CLI.
Trying to ring the extension results in constant ringing if voicemail is disabled, and straight to voicemail if enabled. Within the Asterisk CLI we see a SIP response 503 "Service Unavailable."
I can get the SPA3102 to respond by lowering the "Line In Use Voltage" down from its default value of 30. However all I hear is a hum and I'm not able to speak over the paging system.
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Thanks!

Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE. The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.
I would try the following:
Viking Pl-1:
Talk Battery Switch (PT): On
Audio In: Cabled to FXO port of SPA3102
SPA3102:
Interface Analog Phone attached to SPA3102:
Line 1 Tab
Line Enable: YES
Dial Plan: (S0<:@gw0>)
PSTN Line Tab
Line Enable: Yes
VoIP-to-PSTN Gateway Enable: Yes
One Stage Dialing: Yes
Line 1 VoIP Caller DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: xx
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If you lift the phone with the above settings you should be connected to the paging system.
Interface to an asterisk pbx system:
PSTN Line Tab
Setup Registration on PSTN Line Tab to Asterisk system.
Register: Yes
Proxy: xxx
UserID: xxx
Password: xxx
VoIP-to-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: None
One Stage Dialing: Yes
VoIP Caller Default DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: same comments as above
If you call the extension on the asterisk PBX you should be attached to the Viking unit.

How can I pass a # sign to my MGCP FXO port?

CUCM 7.1.5 and a 2951 with IOS 15
Goal: I need to be able to dial 50# and have that passed out the FXO port, including the # symbol.
**I cannot use and FXS card or ATA because the intercom system cannot "answer" the call. It stays off-hook at all times.
I have an MGCP-controlled FXO port tied to an intercom system. All I need is for the FXO port to pick up and pass "50#" down the line to the paging system. No matter what I do, I cannot get CUCM to pass the # to the gateway, only the "50". So the user then has to manually hit # after connecting to the paging system. This is for emergencies so we really don't want to require them to remember this. I have verified with CCAPI debugs that the called number does not contain a # when it arrives at the gateway.

There is a CM service parameter that controls whether CUCM will leave or strip the "#" from the called party string. By default this parameter is set to "true". Which means the "#" is stripped. Go to CM service parameters and search for 'strip'.
Note that this is a system-wide setting and could affect other call flows. You want to pay particular attention to any route patterns (or translations) where you rely/allow inter-digit timeout.
HTH.
-Bill (@ucguerrilla)
http://ucguerrilla.com

"Phone off-hook" HP Officejet Pro 8500 Wireless, Product CB023A

I have a HP Officejet Pro 8500 wireless printer/fax/copier all in one.
The message "phone off-hook" appears on my display.
I ran the Fax Test and received a PASS for all results.
A few minutes later, a message appeared on the printer/fax display: "Turn power off, then on again". I did so.
This message, "Turn power off, then on again" continues to repeat and after I turn power off then on, the "phone off-hook" displays again. It is an endless loop of receiving these two messages.
I also did the following:
1. Unplugged the telephone fax line from LINE of the Officejet Pro.
2. Unplugged the power cord to the printer.
3. Replugged the power cord to the printer.
4. Powered up the printer
5. After powering up, the message, "phone off-hook" reappears.
I ran the fax test
Run Fax Test:
1. select Tools from HP Officejet Pro 8500 wireless display panel.
2. Select Run Fax Test
Results:
Fax Hardware Test ........................................................ PASS
Fax Connected to Active Telephone Wall Jack ......... PASS
Phone Cord Connected to Correct Port on Fax ........ PASS
Using Correct Type of Phone Cord with Fax ............. PASS
Dial Tone Detection ....................................................... PASS
Fax Line Condition Test ................................................ PASS
3. The message still appears.
My printer description is as follows:
- Model No. : A909G (look inside of the ink access door); Regulatory Model No. : SNPRC-0703-02
- Product No. : CB023A
- Serial No. CN0737R2DH
Is the printer defective? What should I do to correct this problem?
Thanks,
ArtLaw

Hello ArtLaw,
If you have followed all the troubleshooting steps outlined in this troubleshooting article then the only other solution is to have the printer serviced as it may be a printer hardware issue.
Troubleshooting 'Phone Off Hook' Front Panel Messages
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3825 FXO Port remains in off-hook after call

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