No sound When redirect by Auto-Attend

[Toss Leey]

Hi

I have situation if anyone faced this issue before which when I dial internal extension during Auto-Attend, it is ringing to internal extension phone but when answer no sound can hear in both end, but when i dialed 0 during Auto-Attend for operator phone or call directly DID it is working fine.

Thank you

[Charles Hill]

Hi,

When you say Auto-Attend are you referring to a Call-Handler?

Call manager version?

Unity version? or Unity Connection version?

Thanks.

[Toss Leey]

Hi

Yes, call-handler, and Call Manager/Unity connection Version: 10.5

Thank you

[Charles Hill]

Any difference if you call call handler from an internal phone versus calling the call handler from an outside phone line?

Take a look at the posting below.  Sounds like exact same symptoms.

https://supportforums.cisco.com/discussion/11975636/no-audio-both-ends-when-you-enter-extension-during-greetings-call-handler

"No audio for both ends when you enter an extension during greetings on Call Handler"

"The only way I was able to get this solved was checked the MTP on trunk in the CUCM. I have never seen this before with no two-way audio between CUC and CUCM."

Hope this helps,

Please rate helpful answers,

Thanks.

[Toss Leey]

Hi

From internal works fine, only from outside

Thank you

[Charles Hill]

Is MTP required checked on the trunks?

SIP trunks?

[Toss Leey]

Hi

I tried both check/Uncheck sip trunk CUCM to CUC MTP, but same thing, after 15 sec, it is disconnect by default without hear sound

Thank you

[Charles Hill]

Sounds like a codec mismatch.

Trying to understand your setup a little better.

SIP trunk from the phone company to the Call manager?

SIP trunk integration between Call manager and Unity Connection?

[Toss Leey]

Hi

Yes,

-IP Phone---CUCM---SIP---CUBE----SIP---ITSP

-CUCM---SIP---CUC

Thank you

T

[Charles Hill]

Do a "debug ccsip mess" on your gateway.

Make a test call and upload the debug. 

I'm no SIP expert by no means, but if we don't see the problem, maybe a SIP guru will chime in and help us out. 

Another question, do you have a MTP configured?
 

[Toss Leey]

Hi

Please check debug in attachment,

@do you have a MTP configured? you means Voice class-Codec

Thank you

[Charles Hill]

Please do a debug ccsip all

and upload the debug.  If I'm looking at the debug correctly, i see the codecs that are supported, but not what is negotiated. 

On a successful internal call, what is the negotiated codec when a call is transferred from a call handler to an internal DN. 

[Toss Leey]

Hi Charles

Issue has been resolved, just add Payload 97, and works (:

Thank you

[Charles Hill]

Very good!

Thanks for the update.

[A.HASEKE1]

Hi All,

How to Configure Auto Attendance on C2911-CME-SRST/K9 ?

we have this :-

1- C2911-CME-SRST/K9       2911 Voice Bundle w/PVDM3-16,FL-CME-SRST-25, UC License PAK

2- VIC2-4FXO

3- CP-7945G=      QTY1

4- CP-7942G=      QTY3

5- CP-7931G        QTY1

we did all the basic configuration , we can call each other extensions and we can call out lines

and also we can receive incoming calls direct to operator on extension 400 .

now we want to add Auto Attendance , any one can help us ?

kindly see the attached configuration and also the flash files