Incoming PSTN calls dropped

Answered Question
May 26th, 2010

Hi All,

I have an interesting problem, I know what it is but can not find the correct info to fix it.  it's occur When made Call or forwarding through UC520 that time Outgoing calls are fine but Both Incoming call are droped out after few second menas when i pickuped call on VOIP Phone or any ext. but if we are Making Call or forwarding through a service provided by optus (an Australia Telco) Then All is going well. 
which is why the gateway drops the call after few second. I belive I need to set signal immediate, but am unsure,Can you please point me in the right direction for solve this issue.

Thanks

VIkrant

I have this problem too.
0 votes
Correct Answer by Md. Mijanur Rahman about 6 years 6 months ago

What do you see the 'Disconnect Cause' when debugging with 'debug voice ccapi all'?

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Md. Mijanur Rahman Fri, 05/28/2010 - 06:10

Are you using PRI? or FXO? Did you set any incoming dial-peer? are you stripping the number matching to your phonenumber?

Anything can happen bro, specify the system specs and the exact problem.

Vikrant Ambhore Fri, 05/28/2010 - 20:52

Hi bilashece,

Thanks for ur reply,

I have two analog line, You can refer below configuration, We need to use (0/1/0) for all outbound &Inbound calls at this stage, but We want to Disable All outbound call from (0/1/1) but we need Incoming call on 0/1/1, So I configured as per below but i am suffering from call drop issue means I am getting a lot of call drop out with the phone system at the moment, I had 4-5 inbound calls drop, When I pickup call on VOIP phone then after 2-3 second call are droped out, I hope u understand my issue, Pls help me

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
  no update-callerid
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
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 translation-rule 1111
!
voice translation-rule 1112
rule 1 /^0/ //
!
voice translation-rule 2222
!
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
!
voice translation-profile CallBlocking
translate called 2222
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
!
!
voice-card 0
no dspfarm
no local-bypass
!
!
application
  service queue flash:app-b-acd-2.1.2.2.tcl
  param aa-hunt3 207
  param queue-len 10
  param aa-hunt4 4444
  param aa-hunt9 2222
  param aa-hunt1 205
  param aa-hunt2 2222
  param number-of-hunt-grps 4
  param queue-manager-debugs 1
  !
  service aa flash:app-b-acd-aa-2.1.2.2.tcl
  paramspace english index 1
  param number-of-hunt-grps 2
  param menu-timeout 6
  param handoff-string aa
  param dial-by-extension-option 5
  paramspace english language en
  param max-time-vm-retry 2
  param max-extension-length 4
  param aa-pilot 1000
  paramspace english location flash:
  param second-greeting-time 60
  param welcome-prompt _coinop_welcome.au
  param call-retry-timer 15
  param voice-mail 5000
  param max-time-call-retry 600
  param service-name queue
  !
voice-port 0/0/0
timeouts ringing infinity
!
voice-port 0/0/1
timeouts ringing infinity
!
voice-port 0/0/2
timeouts ringing infinity
!
voice-port 0/0/3
timeouts ringing infinity
!
voice-port 0/1/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone TEST
supervisory dualtone-detect-params 1
output attenuation -6
cptone AU
timeouts call-disconnect 3
timeouts wait-release 3
timing sup-disconnect 300
connection plar opx 206
impedance complex1
caller-id enable
!
voice-port 0/1/1
supervisory disconnect dualtone mid-call
supervisory custom-cptone TEST
supervisory dualtone-detect-params 1
output attenuation -6
cptone AU
timeouts call-disconnect 3
timeouts wait-release 3
timing sup-disconnect 300
connection plar opx 206
impedance complex1
caller-id enable
!
voice-port 0/1/2
output attenuation -6
cptone AU
timeouts call-disconnect 5
timeouts wait-release 5
connection plar 1000
impedance complex1
caller-id enable
!
voice-port 0/1/3
output attenuation -6
cptone AU
timeouts call-disconnect 5
timeouts wait-release 5
connection plar 1000
impedance complex1
caller-id enable
!
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music
!
sccp local Loopback0
sccp ccm 10.1.1.1 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dial-peer cor custom
name internal
name local
name domestic
name international
!
!
dial-peer cor list call-internal
member internal
!
dial-peer cor list call-local
member local
!
dial-peer cor list call-domestic
member domestic
!
dial-peer cor list call-international
member international
!
dial-peer cor list user-internal
member internal
!
dial-peer cor list user-local
member internal
member local
!
dial-peer cor list user-domestic
member internal
member local
member domestic
!
dial-peer cor list user-international
member internal
member local
member domestic
member international
!
!
dial-peer voice 4 pots
service stcapp
port 0/0/3
no sip-register
!
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 400
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1000 voip
description gg gg
service aa
destination-pattern 1000
voice-class codec 1
session target ipv4:10.1.1.1
incoming called-number 1000
dtmf-relay rtp-nte
!
dial-peer voice 100 voip
dtmf-relay h245-alphanumeric
!
dial-peer voice 2 pots
service stcapp
port 0/0/1
no sip-register
!
dial-peer voice 3 pots
service stcapp
port 0/0/2
no sip-register
!
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 401
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 1 pots
service stcapp
port 0/0/0
no sip-register
!
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
!
dial-peer voice 50 pots
destination-pattern 0000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 51 pots
description Emergency dial-peer
destination-pattern 000
port 0/1/0
forward-digits 3
no sip-register
!
dial-peer voice 52 pots
destination-pattern 0T
port 0/1/0
no sip-register
!
dial-peer voice 53 pots
destination-pattern A0
port 0/1/0
no sip-register
!
dial-peer voice 58 pots
destination-pattern 0000
port 0/1/2
forward-digits 3
no sip-register
!
dial-peer voice 59 pots
description Emergency dial-peer
destination-pattern 000
port 0/1/2
forward-digits 3
no sip-register
!
dial-peer voice 60 pots
destination-pattern 0T
port 0/1/2
no sip-register
!
dial-peer voice 61 pots
destination-pattern A2
port 0/1/2
no sip-register
!
!
!
no dial-peer outbound status-check pots
sip-ua
ephone-dn  11  dual-line
number 202 no-reg primary
label 202
description India Nagpur
name India Nagpur
call-forward busy 400
call-forward noan 400 timeout 25
!
ephone-dn  15  dual-line
number 206 no-reg primary
label 206
description Lindsay Clark
name Lindsay Clark
call-forward busy 202
call-forward noan 00411069447 timeout 10

Thanks

VIkrant

heathrw Mon, 05/31/2010 - 04:27

Hi,

When the call comes into your IP Phone is it coming in direct or via your AA that is setup?

Also does your PSTN lines have any other services setup such as call waiting, fax duet, message bank? Make sure these are not in use.

Heath

Vikrant Ambhore Mon, 05/31/2010 - 09:15

When the call comes into IP Phone it is coming in direct from router

Doesn't active any service in my PSTN lines such as call waiting, fax duet, message bank. look like it's signaling or dial peer issue,

Please suggest ASAP, i need help on urgent basis.

Thanks

Md. Mijanur Rahman Tue, 06/01/2010 - 00:59

I was going through your configuration, I have a question for you. Are you sure about the below?

!
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
!
!

It's completely depends on PSTN switch, I mean you have to get these info from PSTN technicians. The best practice would be contact with any of your local vendor/service integrators and get the exact information, you will get such abnormal behaviours as you are getting if these parameters are not correct. Anyway, did you try with default? I couldn't found anything else which might cause you dropping the call.

Vikrant Ambhore Tue, 06/01/2010 - 02:48

Hi,

I used this because I expalin you We have 2 incoming line, I did configuration without above ,We Have two Incoming Line 1st is 08 9377 0540 (0/1/0) & 2nd is 08 9377 3097 (0/1/1), We need to use 08 9377 0540 for all outbound calls at this stage, as we don’t get charged any extra for calls on this number, but we do on the 08 9377 3097 number, so When someone is  making calls they should leave 08 93773097 as the number to call back on, So we did All incoming call these are comes on 08 9377 3097 (0/1/1) to forward my mobile from 08 93770540 (0/1/0), but after forwarding FXO port went to Off Hook,

then i got above conf from cisco forum https://supportforums.cisco.com/thread/300615 & he is also from Australia & also i'm so i tried it & saw Everything is going fine, but we have only one issue , inbound & outbound call drop out after some time, I hope you understand, although if i'm wrong or you have any good solution for this scenario please let me know i will implement it...

Also http://www.3amsystems.com/wireline/tone-search.htm?start=0&kCountry=11&kTone=2 as per this site it's Correct I think

Thanks

VIkrant

Correct Answer
Md. Mijanur Rahman Tue, 06/01/2010 - 07:09

What do you see the 'Disconnect Cause' when debugging with 'debug voice ccapi all'?

Md. Mijanur Rahman Sat, 06/05/2010 - 00:31

I can see your disconnect cause=19, i.e. 'no answer from user'. The problem you are facing is your router is accepting the call but it  doesn't know where to route.

Also I found your CME phone number are 20X, please confirm. Also, what do you dial from PSTN to reach your CME phones? I can see only 20X as the called number. I think it should be E.164/PSTN number. Please follow the below:

> Check what's your incoming called-number on your gateway while dialing from PSTN (with the same ccapi command), let's assume it 112233XXX, where XXX is like 202, 206 etc.

> Create an incoming dial-peer for all of these calls like below:

!

dial-peer voice x pots

incoming called-number 112233...

direct-inward-dial

translation-profile incoming StripTo3

!

voice translation-rule

rule 1 /.*\(2..$\)/ /\1/

!

voice translation-profile StripTo3

translate called

!

Modify this as per your requirement.

HTH.

Vikrant Ambhore Sat, 06/05/2010 - 01:39

Hi,

Thanks for your reply,

When anyone make call on VOIP system 1st it ring on 202 if anyone are unable ti pickup call within 10 sec. then call forwarded to 206 after 15 second call goes to my cell, when call divert from 202 to 206 that time we are getting cause is 19 although call is connected But when call divert to my cell from 206 & i answered call on cell but all calls have dropped after a while that time we are getting cause value=86, so I think we need to troubleshoot for value 86,

Please help me if i'm wrong let me know

Regards

VIkrant

Attachment: 
Vikrant Ambhore Sun, 06/06/2010 - 20:27

Hi,

Thanks for giving reply, I Have this document but i can't understand, How to solve this issue

Can you suggest me & help me how to solve it ?

Thanks

Vikrant

Md. Mijanur Rahman Mon, 06/07/2010 - 05:55

I am sorry your problem is not solved yet. Try dialing any PSTN line (not mobile), if you face the same, try connecting any PBX port and dial PBX number.

This happens for other end (PSTN, PBX) switch synchronization mismatch.

Vikrant Ambhore Mon, 06/07/2010 - 21:36

Hi,

Thanks for giving good suggestion, as per your guidence, I used Landline no. instead of Mobile for call Forwarding but still have same issue, we have issue not only inbound call but also outbound call, As per your suggestion i removed analog line from UC520, & plugged in Analog phone if we are Making Call or forwarding through a service provided by optus (an Australia Telco) Then All is going well. so i am 100 % sure issue is not from service provider.

Please i realy need your help I never forget this favor of yours, Please help me, what should we do now at this stage

Thanks 

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