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New Member

No Ringback for Calls transfered from Unity Connecion ( AA) Call Handler

When People from PSTN Calls ,  they hear  IVR from AA configured in Unity Connection via a CTI Route Point.

AA - IVR asks to dial any Internal Extension . Then Calls are transfered to that particular Extention and it rings

and calls get connected.

PROBLEM : The Person on the PSTN side do not hear Ringback or MoH while AA transfers Call to Extensions.

Anyways, Everything is working fine internaly. Problem exist only when call is from outside , PSTN.

Here is the Call Flow >

PSTN------- > SIP Service Provider------> CUBE -----sip---- > CUCM ----SCCP-- > Unity Conx (AA) -----> transfer to Internal Extension(SCCP Phone)

Message was edited by: Mohammed Bineesh E.K.

3 REPLIES
New Member

No Ringback for Calls transfered from Unity Connecion ( AA) Call

Hi Mohammed,

Can you give the output of the CUBE SIP Dial Peer and explain how the SIP trunk is configured on CUCM?

What version of CUCM are you running? 

Is this a new implementation?

Was it ever working or did it just start happening?

What codecs are you using?

Do you have MTP checked on your SIP trunk?  I have seen similar problems requiring MTP on the CUCM SIP Trunk.  I would suggest to configure a software MTP on the CUBE. 

Thanks,

GS

New Member

Re: No Ringback for Calls transfered from Unity Connecion ( AA)

Thanks for your response.

Can you give the output of the CUBE SIP Dial Peer and explain how the SIP trunk is configured on CUCM?

!

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

!

!

dial-peer voice 1 voip

description ## Incoming SIP Calls ##

translation-profile incoming STRIP012

rtp payload-type cisco-codec-fax-ack 105

rtp payload-type cisco-codec-fax-ind 106

rtp payload-type nte 97

session protocol sipv2

incoming called-number 0127599[123]..

codec g711ulaw

no vad

!

!

dial-peer voice 2 voip

description ## To CUCM SUB ##

destination-pattern 7599[123]..

rtp payload-type cisco-codec-fax-ack 105

rtp payload-type cisco-codec-fax-ind 106

rtp payload-type nte 97

session protocol sipv2

session target ipv4:10.1.21.30

voice-class codec 1

dtmf-relay rtp-nte

no vad

!

What version of CUCM are you running? 

------Iam Running CUCM 8.6.2

Is this a new implementation?

------ Yes this is a new Implementation.

Was it ever working or did it just start happening?

---- New Impelementation. Was no woriking

What codecs are you using?

----g711alaw on service provider side.

----g711ulaw on cube,cucm side.

Do you have MTP checked on your SIP trunk?

----Is already Checked.

I already have IOS Software MTP running on CUBE and its added in SIP TRUNK MRGL.

STILL ITS THE SAME

Message was edited by: Mohammed Bineesh E.K.

Re: No Ringback for Calls transfered from Unity Connecion ( AA)

Since there is no Ringback heard on the pstn end . Have you verified if there is an annuciator present in MRGL of the sip trunk pointing to the cube .  Could you add this to the MRGL of the sip trunk if not already present?

Regards,

Karthik Sivaram

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