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Calls from Etisalat PSTN to FXO to voicemail do not disconnect

abdulbari_cisco
Level 1
Level 1

I have a tricky issue where outside caller calls in and when the call is forwareded to voicemail because of CFNA, the FXO do not  disconnect. I have a setup where a Etisalat Analog lines are directly connected to UC560 FXO ports using RJ11.

When  a call comes in over the PSTN to an FXO port on my UC560 and the call  is answered by the user and after that user goes on-hook, FXO disconnects or gets released normally . When the user does not answer the call and becuse of CFNA timeout the call is forwarded to users voicemail box , then CUE answers, a voicemail is recorded, but  when the calling party hangs up FXO doesnot disconnect instead it stays in OFFHOOK state (HANGS). Because of this no more calls are possible on that FXO line. I have to issue shut and no shut command on the FXO to get it released.

The IOS version as follows

uc500-advipservicesk9-mz.151-2.T4

and CME and CUE version are as follows 8.0.2

The follwing is the configuration on my UC560

voice class dualtone-detect-params 1

freq-max-deviation 25

freq-max-power 0

freq-min-power 13

freq-power-twist 4

cadence-variation 4

!

voice class custom-cptone UAE-CUSTOM-SIEMENS

dualtone disconnect

  frequency 425

  cadence 425 325 250 500

voice-port 0/1/0

trunk-group ALL_FXO 61

translation-profile incoming INCOMING_CallerID_PROFILE

supervisory disconnect dualtone mid-call

supervisory custom-cptone UAE-CUSTOM-SIEMENS

supervisory dualtone-detect-params 1

no battery-reversal

input gain 14

cptone AE

timeouts call-disconnect 2

timeouts wait-release 2

timing min-ring 62

connection plar opx 202

description Configured by CCA 4 FXO-0/1/0-Custom-BG

caller-id enable

!

This the dial peer for viocemail

dial-peer voice 2000 voip

description ** cue voicemail pilot number **

translation-profile outgoing XFER_TO_VM_PROFILE

destination-pattern 399

b2bua

session protocol sipv2

session target ipv4:10.1.10.1

voice-class sip outbound-proxy ipv4:10.1.10.1 

dtmf-relay rtp-nte

codec g711ulaw

no vad

39 Replies 39

I have this problem only when the call is passed to CUE and this redirect the call ( for example press 1 for Sales). This is a strange problem in the cue setting.

I have some news.

I have tried with a gateway fxo Linksys SPA 3102 connected via Sip to UC560.

If I change the parameter of Disconnect Tone in :

Whit gateway everything work fine !

This parameter is corresponding to configuration of the FXO integrated in the UC560:

voice class custom-cptone IT-CUSTOM

dualtone disconnect

  frequency 425

  cadence 200 200

!

voice-port 0/1/0

trunk-group ALL_FXO 64

supervisory disconnect dualtone mid-call

supervisory custom-cptone IT-CUSTOM

input gain 14

cptone IT

timeouts call-disconnect 2

timeouts wait-release 2

connection plar opx 396

impedance complex2

caller-id enable

!

But not working !

Where is the problem !?!?

This is the magical paramenter :

!

voice class dualtone-detect-params 1

freq-max-deviation 50

!

voice-port 0/1/0

supervisory disconnect dualtone mid-call

supervisory custom-cptone IT-CUSTOM

supervisory dualtone-detect-params 1

!

Im realy realy happy ! Maybe the CUE manipulates this frequence in some way.

Hi Paolo Moserle,

I am glad that you have resolved the issue. You should have tried this first as it is marked as the correct answer.

Just to add - the CUE is not manipulating the frequence. It just has longer disconnect timeout. That is why the CME should correctly recognize the disconnect tone to act on time without waiting to disconnect from the called side.

Best regards,

Alex

Hello Paolo,

we have the same problem that you solved 2 years ago.

Can you please post me your complete configuration?

mine is:

 

voice class dualtone-detect-params 1
 freq-max-deviation 50
!
voice class custom-cptone testy
 dualtone disconnect
  frequency 425
  cadence 200 200


and

voice-port 0/0/0
 trunk-group ALL_FXO 64
 supervisory disconnect dualtone mid-call
 supervisory custom-cptone testy
 supervisory dualtone-detect-params 1
 input gain 14
 cptone IT
 timeouts call-disconnect 2
 timeouts wait-release 2
 connection plar opx 6681
 impedance complex2
 caller-id enable

Ciao!

 

Hi Giovanni, this is my configuration:

voice-port 0/1/1
 trunk-group ALL_FXO 64
 supervisory disconnect dualtone mid-call
 supervisory custom-cptone IT-CUSTOM
 supervisory dualtone-detect-params 1
 no battery-reversal
 input gain 14
 cptone IT
 timeouts call-disconnect 2
 timeouts wait-release 2
 connection plar opx 400
 impedance complex2
 description * FISSO *
 caller-id enable

 

Some times the error still occurs, I suggest you to switch the number in a sip trunk. In my experiences I find many problems to configure FXO in Italy, where in other country there are not!

Hello Paolo,

same configuration but not works : (

yes, i would like that the customer change the connection to provider with a SIP trunk but still wants to use analog line for backup.

Another problem is that if the user with IP phone is busy the called always hears ringback :(

 

Giovanni

Really annoying problem. The only thing that I can suggest you, is to try   with a different gateway, like a patton, in front of the router.

this really works! thank you. who knows what this parameter does?

Razeen Abdulla
Level 1
Level 1

Dear All,

I solved the issue.

I am from Saudi Arabia. I use STC analog lines at my office. And this command under my voice port worked like a charm. Hope it helps

 

voice-port 1/0/0
groundstart auto-tip
cptone GB
timing sup-disconnect 85
connection plar opx 5999  ==> your AA pilot number
impedance complex5
caller-id enable
supervisory disconnect anytone
no battery-reversal
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate

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