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Outbound call -busy tone

Toss Leey
Level 1
Level 1

Hi

I have following call flow: CUCM-----H323-----CME---FXO, when initiate call give me busy tone, any help plz..

 

Thank you

1 Accepted Solution

Accepted Solutions

Can you please post the show run from GW? Its seems issue to be with TECLO where we are getting power denial event.

 

As call hitting the GW and matching incoming and outgoing dial-peer , when call traverse over the fXO port its power denial.

 

In order to test the FXO port or teclo issue, can you please connect analog phone to that port and make test call.

 

 

 

Jun 29 12:17:17.486: htsp_process_event: [0/0/1, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected
Jun 29 12:17:17.486: htsp_timer2 - 1000 msec
Jun 29 12:17:17.486: htsp_timer_stop
Jun 29 12:17:18.486: htsp_process_event: [0/0/1, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc

 

Br,

Nadeem

Br, Nadeem Please rate all useful post.

View solution in original post

7 Replies 7

Shawn Guertin
Level 1
Level 1

Look in your CUCM ig there is an IP in your gateway's page, if not, Its not registered, the registration status does not tell you the truth, the IP does.

If there is no IP address reset/restart the gateway in the CUCM.

 

There is IP For G.W in CUCM-Gateway Page

Hi Manish

cucm 9.1.1.20000-5
phone: 7942
busy tone after dial No.

 

I can see FXO 0/0/1 move to Up/OffHook for sec and back to dorm:onhook in CME

 

+Check Debug/Logs in attachment maybe will be helpful..

The error in the debugs is similar to the following post

https://supportforums.cisco.com/discussion/11353441/srst881-fxo-port-uk-phone-keeps-disconnecting-during-ringing

Try the suggestion as per the post and let us know how it goes.

 

Manish

Can you please post the show run from GW? Its seems issue to be with TECLO where we are getting power denial event.

 

As call hitting the GW and matching incoming and outgoing dial-peer , when call traverse over the fXO port its power denial.

 

In order to test the FXO port or teclo issue, can you please connect analog phone to that port and make test call.

 

 

 

Jun 29 12:17:17.486: htsp_process_event: [0/0/1, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_1100]fxols_power_denial_detected
Jun 29 12:17:17.486: htsp_timer2 - 1000 msec
Jun 29 12:17:17.486: htsp_timer_stop
Jun 29 12:17:18.486: htsp_process_event: [0/0/1, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER2]fxols_power_den_disc

 

Br,

Nadeem

Br, Nadeem Please rate all useful post.

Manish Gogna
Cisco Employee
Cisco Employee

Hi Faziz,

Please provide some more details.

CUCM version, phone model being used, when do you get fast busy ( after going off hook or after the called party responds ? etc?

Which codec is being configured between ip phone and h323 gateway and also on CME dial-peers? if the two calls legs are using diff codecs then is there a  transcoder configured?

 

HTH

Manish

 

Nadeem Ahmed
Cisco Employee
Cisco Employee

What is the DNA result shows?Is the call hitting the GW?

 

Can you please debug the GW atleast see whether call hitting the GW if not then we can troubleshoot on the first node.

 

Debug voice ccapin inout

debug h225 asn1

debug h245 asn1

debug vpm signal

debug vtsp all

 

 

Br,

nadeem

Br, Nadeem Please rate all useful post.