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Incoming call on FXO

Vibha Verma
Level 1
Level 1

Hi

We have CUCM 9.0 on HQ and H323 Branch router with FXO and IP Phones registered on CUCM.

FXO voice port config is below on every FXO port. Also outbound call works fine.

voice-port 0/1/2
 no battery-reversal
 cptone ID
 timeouts call-disconnect 3
 timeouts wait-release 3
 timing hookflash-out 500
 connection plar 23947
 description * line out Telkom *
 station-id name TELKOM
 caller-id enable
 

 

Also incoming dial peer is configured as under

dial-peer voice 2000 pots
 incoming called-number .T
 direct-inward-dial

 

VOIP dial peer towards CUCM is also configured as under

dial-peer voice 2001 voip
 description *** dial peer to CUCM ***
 destination-pattern 123..
 session target ipv4:A.B.C.D
 incoming called-number .
 dtmf-relay rtp-nte
 no vad
 

where A.B.C.D is IP of CUMC subscriber and 123.. is pattern of Branch numbers.


Now incoming call from PSTN reaches IP phone at branch and rings. But as soon as we answer it, there is silence for about 3-5 seconds and then call disconnects.

 

Please suggest if any chnages required in FXO voice port config.

Thanks and regards

Vibha

11 Replies 11

Please share the output of debug voice vtsp

Hi

please find the debug output attached.

Waiting for response.

Regards

vibha

I can see that you tried to call 23923. However, the call is disconnected with cause value 34 (no circuit available).

Please share your router configuration and H323 trunk configuration. Also, try to reset the h323 trunk and test

 

Hi

 

I reset H323 gateway. We do not have h323 trunk.

But still no luck

Thanks

Hello All,

 

I suggest to collect the following:

 

1. show run

2. debug voip ccapi inout

3. debug h225 asn1

4. debug h245 asn1

 

Thank you,

Shadi

Hi Vibha,

from the description of the issue, it appears to be an issue of codec mismatch or mtp unavailablity. 

Could you please update us if the issue has been resolved?

 

I will update once the issue gets resolved. Possibly with the solution

 

saif musa
Level 4
Level 4

Vibha,

Could you please upload the debug from (( debug voip ccapi inout)). set-up the call from POTS leg to VOIP leg after issuing this command to verify voice call control.

 

Regards

Hi saif Musa

Please see the debug output requested by you.

thanks and regards

Vibha

Vibha,

   I'm aware of PSTN network temporary failure. did you try to disconnect PSTN line from FXO port and connect it to analog phone device and try to call it from another line. verify its operation wither it was normally or not.

 

Regards

Nadeem Ahmed
Cisco Employee
Cisco Employee

Can  you apply the voice class codec  on dial-peer terminating to CUCM with G729 ,G711a &u.

 

can you  please make another test call with capturing below debug

 

debug h225 asn1

debug h245 asn1

debug voice ccapi inout

debug vpm signal

debug vtsp all

 

Br,

Nadeem

Br, Nadeem Please rate all useful post.