CME:10.5 Incoming calls transferred through auto-attendant have no audio

[Shahzad Ayub]

Hi,

We just upgraded the CME version from 9.1 to 10.5 and realized that we are having some issue with incoming calls which are transferred through auto attendant.Calling party does not hear any voice but called party can listen the calling party.

I am putting some configuration related to this issue and I have also attached log during the test call.

I called from 551540150 to 4207800(Auto attendant number) then I select 3 to transfer the call to HR(4207815), call transferred successfully but only called party can hear me and I cannot hear the called party.

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw

voice translation-rule 1
 rule 1 /011420\(78..\)/ /\1/
 rule 2 /420\(78..\)/ /\1/

dial-peer voice 1 voip
 description ****INBOUND SIP CALL LEG*****
 translation-profile incoming incoming_calls
 session protocol sipv2
 session target sip-server
 incoming called-number 01142078..
 voice-class codec 1  
 voice-class sip dtmf-relay force rtp-nte
 dtmf-relay rtp-nte
 no vad
!

Thanks

Shahzad

[Shahzad Ayub]

everybody looks busy....... :(

[nabeel asghar]

Hi Experts,

Greetings

I am calling from IP Phone to PSTN & called party is not answering my call, after some rings call is getting hangup and i am not able to hear hangup tone or beep, rather i am getting system recording like "your call can not be completed as dialed please console your directory"

Please advise me to remove the system recording & let me able to get beep tone after called party not answering call

Thanks in advance

Warm Regards:

Nabeel Asghar

[Carlo Poggiarelli]

Hi Shahzad.

Your entire config and a debug ccsip message output could be useful 

Thanks 

Regards

Carlo

[Shahzad Ayub · Accepted solution]

Hi Carlo,

Thank you for your reply.

Please find the full config & debug ccsip in attachments.

Thanks

Shahzad

[Carlo Poggiarelli]

Hi Shahzad 

Under voice service voip and than sip

please add early-offer forced 

and

midcall-signaling passthru

After these changes please post another output of debug ccsip message

HTH

Regards

Carlo

[Shahzad Ayub]

Hi Carlo,

I am unable to enter these commands as mentioned.

I would like to tell you, I have just upgraded CME version 9.0 to 10.5 and facing this issue.

Thanks

Shahzad

[Carlo Poggiarelli]

Shahzad,

you have to add those command in this way

conf t

voice service voip

sip

early-offer forced

midcall-signaling passthru

You can copy and paste on your terminal what above 

HTH

Regards

Carlo

[Shahzad Ayub]

still same issue am attaching the log.

Shahzad

[Shahzad Ayub]

I just removed g711alaw codec under voice class codec and issue is resolved.

Thanks

Shahzad

[Rolando Valenzuela]

For the time-stamp I will assume it is solved, right?

I was about to mention a coded problem, but it seems you find it.

Regards.

Rolando Valenzuela.