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FXO Disconnect Problem ;)

rgeorgiou
Level 1
Level 1

Hi guys,

I have the classic disconnect problem and with all the terrible bytes on the internet have be unable to find the appropriate solution.

If i ring the PSTN line connected to the FXO port i have PLAR directing the call to 1000 my Reception phone. Im in Australia and the cptone is set to AU.

I have tried different voice class's but i am not getting anywhere. i recorded the disconnect tone from CIPC and i believe the frequency is 400, and the cadence is 380. I inserted a custom class and linked i to my voice port 1/1/0 but with no joy.....

Anyone got any ideas.....

Please advise

24 Replies 24

12.4.15T hmm ok ill try that...

i let you know how it go

so far my conf looks like this

voice class custom-cptone TEST

dualtone disconnect

frequency 400

cadence 380 380 380

voice-port 1/1/0

supervisory disconnect dualtone mid-call

supervisory custom-cptone TEST

supervispry dualtone-detect-params 1

cptone au

timeouts call-disconnect 3

timeoute wait-release 3

connection plar 4000

????

HELP!!!

I was speaking with some ccvp's and ccie's and i have been told that loopstart has a known problem and if i get the telco company to change it to ground start, ill be fine.

I'm not sure what they mean by a 'known problem'. Loop start operation has been around in one way or another for over a hundred years, and any specific problems would have been sorted out a long time ago. It was designed for connecting telephone handsets and the idea has always been that the people say 'goodbye' and then physically hang up towards the network. In this sense the disconnect supervision is controlled by the people on the phone call. The telephone network NEVER hangs up towards the users - it signals the far end disconnect by battery reversal or disconnect tones.

There is no reason why this should not work with disconnect tones. The next suggestion would be to get a wireshark trace of the call and then hang up the PSTN side. This will capture all the call signaling and the audio from the gateway

Groundstart is unheard of outside of North America. Telstra/Optus can supply an analogue loopstart trunk or ISDN BRI for low density connections.

our issue here is the same with the loopstart issue. Normally, to resolve the issue we do two things.

1. Request to the carrier that the line be put in a trunk group and enable disconnect on thier side ( to enable disconnect it must be controlled hence the reason for being placed in a trunk group) im no guru so dont ask me to expand :)

second, i noticed on some sites that although i did that i still had to enter the following command:

#timing sup-disconnect (and i normally set it as 300 ms) - This is under the voice port itself

it wont hurt to try.

BTW ground start should work too

rgeorgiou
Level 1
Level 1

What fixed it....

opx int the plar command

and

freq-max-deviation 20

cadence-variation 50

int the voice class attached to the fxo port....

voice class dualtone-detect-params 1

freq-max-deviation 20

cadence-variation 50

!

!

voice class custom-cptone TEST

dualtone disconnect

frequency 425

cadence 350 350

!

!

voice-port 1/1/0

supervisory disconnect dualtone mid-call

supervisory custom-cptone TEST

supervisory dualtone-detect-params 1

cptone AU

timeouts call-disconnect 5

timeouts wait-release 5

connection plar opx 1000

impedance complex1

!

!

dial-peer voice 4000 pots

destination-pattern 4000

port 1/0/0

Thanks to Paul Cameron - CISCO SYSTEMS, CCIE

Regards

Rick

Hi,

I have same Issue, Did you get any Proper solution about above Issue, or anyone is there whose can sort out this issue or anyone please help me

my IOS version is 12.4(11) My configuration as per below:

voice class custom-cptone test
dualtone disconnect
  frequency 476
  cadence 488 488 488 488 487 487 487 487

!

!

voice-port 0/1/0
supervisory disconnect dualtone pre-connect
supervisory answer dualtone sensitivity high
supervisory custom-cptone test
no battery-reversal
output attenuation -6
cptone AU
timeouts call-disconnect 3
timeouts wait-release 3
connection plar opx 202
impedance complex1
caller-id enable
!
voice-port 0/1/1
supervisory disconnect dualtone pre-connect
supervisory answer dualtone sensitivity high
supervisory custom-cptone test
no battery-reversal
output attenuation -6
cptone AU
timeouts call-disconnect 3
timeouts wait-release 3
connection plar opx 206
impedance complex1
caller-id enable

!

!
dial-peer voice 1 pots
destination-pattern .T
port 0/1/0

!
ephone-dn  11  dual-line
number 202
label 202
description India Nagpur
name India Nagpur
call-forward busy 206
call-forward noan 206 timeout 20

!
ephone-dn  15  dual-line
number 206 no-reg primary
label 206
description Lindsay Clark
name Lindsay Clark
call-forward busy 0411069447
call-forward noan 0411069447 timeout 10

Regards

Vikrant

Hi all!

I'm newbie on Cisco VoIP and I want to try fixing this at my work, but I don't know how to find the values for:

     frequency

     cadence

And based on what adjust the values for these ones:

     freq-max-deviation 20

     cadence-variation 50

Also I-m not using plar, I have an autoattendant script for it for the incoming calls.

I did a debug vpm signal but I only see the events and values for the timers not for frequency.

I appreciate your valuable help!

Regards



Ok ... so the background to all this is that the 'Plain Old Telephone System' (POTS) was always intended to have people on either side of the connection and they would be able to determine when the conversation was finished and they could then hang up the telephone handset to clear the call.

Since the majority of telephone signaling was originally tone based (DTMF to make the call, ringbackl/busy/fastbusy/number unavailable etc) , people quickly learnt what all the tones indicated and what actions should be taken when particular tones were heard. Another fundamental design feature of the legacy telephone network is that it always expects the users to disconnect towards the network - the network has no control of the users actually placing the telephone back on hook.

When 'intelligent' (this point could be argued) telephony devices such as IVR's and voice mail systems began to be placed across telephone connections, some very basic problems became apparent. The major issue relates to disconnect of the call when the other party hangs up.

If the voicemail or IVR decides to clear the call, then it only has to open the physical loop and current stops flowing and the call is cleared. However, if the other party hangs up, the connection from them to their local exchange is cleared, but the connection from the IVR's local exchange is maintained until the IVR decides to hang up.

Unless the IVR gets some kind of signal that the call has ended, it will stay off hook and the connection to the local exchange will also stay up. This is the cause of common disconnect issues - the endpoint is not aware the other party has cleared the call so it keeps the connection up.

Even though the telephone network does not physically disconnect the connection to the IVR, it will still play a busy or disconnect tone to indicate the call has cleared.

Therefore, the telephony endpoint (IVR, voicemail, router etc...) needs to pick up these tones and interpret them in the same way a human being can.

On a router, these tones are called the CPtones (Call Progress Tones). They tell the DSP to generate or listen for particular tone combinations (frequencies) and tone durations (cadences) - these are then defined for different call states.

When you configure a specific CPtone on the voice port, you define a table of different frequencies and cadences. Following is the definition for the AU (Australia) and US tone plans -

UC_520#test voice tone AU show
Code: AU   Country: Australia
DTMF freq.(Hz) Row / col:  697,  770,  852,  941 / 1209, 1336, 1477, 1633
Pulse dial: normal, Percent make: 35%, DTMF low Amp. = 65424,  high Amp. = 65446,   Pcm: a-Law
     Tone      NF   FOF  FOS  AOF_FXS AOF_FXO AOF_EM AOS_FXS AOS_FXO AOS_EM  ONTF  OFTF  ONTS  OFTS  ONTT  OFTT  ONT4  OFT4
BUSY           1   400     0    -120    -120    -120      0      0      0   375   375     0     0     0     0     0     0
RING_BACK      2   400   450    -120    -120    -120    -120    -120    -120   400   200   400  2000     0     0     0     0
CONGESTION     1   400     0    -150    -150    -150      0      0      0   375   375     0     0     0     0     0     0
NUM_UNOBTAINAB 1   425     0    -120    -120    -120      0      0      0  2500   500     0     0     0     0     0     0
DIALTONE       2   400   450    -150    -150    -150    -150    -150    -150 65535     0     0     0     0     0     0     0
DIAL_TONE2     2   400   450    -150    -150    -150    -150    -150    -150 65535     0     0     0     0     0     0     0
OUT_OF_SERVICE 1   950     0    -150    -150    -150      0      0      0   330   330     0     0     0     0     0     0
ADDR_ACK       1   600     0    -240    -240    -240      0      0      0   125   125   125 65535     0     0     0     0
DISCONNECT     1   425     0    -150    -150    -150      0      0      0   375   375     0     0     0     0     0     0
OFFHOOK_NOTICE 2  1400  2040    -240    -240    -240    -240    -240    -240   100   100     0     0     0     0     0     0
OFFHOOK_ALERT  2  1400  2040    -150    -150    -185    -150    -150    -185   100   100     0     0     0     0     0     0

UC_520#

UC_520#test voice tone US show
Code: US   Country: United States
DTMF freq.(Hz) Row / col:  697,  770,  852,  941 / 1209, 1336, 1477, 1633
Pulse dial: normal, Percent make: 40%, DTMF low Amp. = 65446,  high Amp. = 65467,   Pcm: u-Law
     Tone      NF   FOF  FOS  AOF_FXS AOF_FXO AOF_EM AOS_FXS AOS_FXO AOS_EM  ONTF  OFTF  ONTS  OFTS  ONTT  OFTT  ONT4  OFT4
BUSY           2   480   620    -170    -170    -240    -170    -170    -240   500   500     0     0     0     0     0     0
RING_BACK      2   440   480    -160    -160    -190    -160    -160    -190  2000  4000     0     0     0     0     0     0
CONGESTION     2   480   620    -170    -170    -190    -170    -170    -190   250   250     0     0     0     0     0     0
NUM_UNOBTAINAB 2   480   620    -170    -170    -190    -170    -170    -240   250   250     0     0     0     0     0     0
DIALTONE       2   350   440    -165    -165    -185    -165    -165    -185 65535     0     0     0     0     0     0     0
DIAL_TONE2     2   350   440    -165    -165    -185    -165    -165    -185 65535     0     0     0     0     0     0     0
OUT_OF_SERVICE 1   950     0    -150    -150    -185      0      0      0   330   330     0     0     0     0     0     0
ADDR_ACK       1   600     0    -240    -240    -240      0      0      0   125   125   125 65535     0     0     0     0
DISCONNECT     1   600     0    -150    -150    -185      0      0      0   330   330   330 65535     0     0     0     0
OFFHOOK_NOTICE 2  1400  2040    -240    -240    -240    -240    -240    -240   100   100     0     0     0     0     0     0
OFFHOOK_ALERT  2  1400  2040    -150    -150    -185    -150    -150    -185   100   100     0     0     0     0     0     0

UC_520#

NF          = number of frequencies

FOF        = Freqency of first tone 

FOS       = Frequency of second tone

ONTF     = On time (duration) of first tone

OFTF     = Off time of first tone

ONTS     = On time of second tone

OFTS     = Off time of second tone

What all this means in theory is that if the voice port DSP picks up particular tone frequency/s and tone timings (cadence) as defined in the tone tables, it will be able to determine that the call has been disconnected and then clear the voice port towards the telephone network.

We all know that theory is wonderful. We also know that in practice, things are often very different.

Even though many country CPtones are based on relevant and publicly available specifications, what we are expecting and what other telephone systems are actually using can be very different.

For this reason, IOS gives the ability to customise specific tone settings to accomodate these frequency and cadence variations.

To determine these special tone settings, you need to capture and analyse the audio from the telephone network towards the router. The easiest way to do this is to obtain a wireshark trace of the network traffic when the call is cleared from the PSTN direction. Rather than hang up the IP phone, you continue to capture the media stream that contains the continual 'beep beep beep' tones.

Provided the traffic is using G711 as the codec, it is simple to save the RTP payload in wireshark as an .AU aduio file.

You can then use a shareware application like Cooledit96 to load up this captured audio file and display it's properties. Cooledit96 makes it easy to select the bursts of tone and measure the on time and off time - this determines the cadence. You then use the frequency analysis option to pick out the tone component frequencies. Once you have the cadence and frequency, you can use these to set the values of the custom tone class.

The tone/frequency deviation values allow the DSP to handle larger variations in the frequency and timing of the tones being monitored.  In the case of the values you quoted, the frequency deviation would allow for around 20Hz (+/- 10hz) around the centre frequency, and the cadence deviation would allow 50msec (+/- 25 msec) around the tone timing.  

Hello,

 

Wanted to reply to this thread because followed almost all recommendations given here but its still not working reliable. Below config works only a couple of days/weeks! Port goes again off-hook.

 

Have tried two different configs so far as below: a and b

 

a)
!
voice class custom-cptone MYTONE
 dualtone disconnect
  frequency 425
  cadence 350 350
!
voice-port 0/2/1
 trunk-group FXO
 supervisory disconnect dualtone mid-call
 supervisory custom-cptone MYTONE
 connection plar opx 445
 description FXO Group
 caller-id enable
!

b)
!
voice class custom-cptone MYTONE
 dualtone disconnect
  frequency 425
  cadence 375 375
!
voice-port 0/2/1
 trunk-group FXO
 supervisory disconnect dualtone mid-call
 supervisory custom-cptone MYTONE
 timeouts call-disconnect 3
 timeouts wait-release 3
 connection plar opx 445
 description FXO Group
 caller-id enable
!

 

Any additonal config I can try to make this work?

R0g22
Cisco Employee
Cisco Employee
Open a new thread please.

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