Problem - Unable to Hear the Unity Voicemail Prompts

[Louis Niel Panganiban]

Hi Cisco Support Community,

I'm Louis an IT Voice Analyst. Kindly see details of my issue below:

I'm having a problem with one of our center, wherein if an outside caller want to be transfered from a phone to another internal phone the caller cannot hear any rings but take not the one at the desk hears the rings. So the caller cannot also hear the voicemail prompt and the VM became useless. I have founded a Cisco Document which is:

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c11.shtmlhttp://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c11.shtmlhttp://www.cisco.com/en/US/products/sw/voicesw/ps2237/products...

Please share your insights with regards to my issue.

Platforms:

CUCM

7.1.3.32900-4

Unity

4.2(1.0)

Thank you very much.

[Gajanan Pande]

Louis,

Is there an issue of low volume on PSTN calls for that site ? i.e. the PSTN callers hear very low volume ( as if no volume at all ) from the user phones, VM etc ? If that's the case then input gain & output attenuation values for the trunk port can be tweaked to increase the overall volume heard for PSTN callers.

Btw, your document link is broken.

GP.

Pls rate helpful posts by clicking on stars below the post !!

[Louis Niel Panganiban]

@Gajanan: Outside calls directly to a phone is okay even the volumes and Voicemail prompts of the extension. We are encountering the issue after the receptionist transfer the outside call going to an extension no.. An outside caller can hear a dead air but the user where the call should be transferred hears his phone rings but the outside caller cannot.

Reposting the Document:

Cisco Unity with Cisco CallManager: One Way Audio (search via google)

Thank you very much Gajanan.

[Gajanan Pande]

So, are you saying the MoH for PSTN callers is not working ?

About the link, which exact problem mentioned there, represents your issue ?

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c11.shtml

GP.

[Louis Niel Panganiban]

@Gajanan: Something like that. And also we cannot hear the Voicemail Prompt if the user is busy even the rings to the phone. Thank you.

[Louis Niel Panganiban]

@Gajanan: same as the title of this post thanks again.

[Gajanan Pande]

Louis,

If the call is ringing on phone then Voicemail doesnt come in picture. VM is invoked only when the phone doesnt respond to call & ultimately reaches to VM of that phone or the phone user diverts the call to VM manually.

Further, if you are saying the phone user is already busy on a call & a PSTN caller calls him, then yes caller will only hear ringing, no announcement of that user being busy on another call ( if you meant that as voicemail prompt ).

So if the MoH is not working for PSTN callers is it working for internal callers ?

GP.

[Louis Niel Panganiban]

@Gajanan: Yes the internal MOH is working. I'm suspecting its a firewall issue with regards to the RTP packet. Whats your insights for this one? Thanks

[Gajanan Pande]

Louis,

Please enable MTP ( check " MTP required " option ) & see if it makes the difference. Is that an MGCP gateway or what it is ?

GP.

[Louis Niel Panganiban]

Its a SIP gateway.

[Rob Huffman]

Hi Louis,

I would also have a look at these docs

http://www.cisco.com/en/US/tech/tk652/tk698/technologies_tech_note09186a008009484b.shtml#verifymtp

http://docwiki.cisco.com/wiki/Cisco_Unified_Communications_--_One-Way_Audio

Cheers!

Rob

[acampbell]

Hi,

Are you hitting this bug

CSCtb01167

After blind transferring a call the transfer destination gets one-way voice
and eventually the call disconnects

!

!

Conditions:

This issue is first observed with CUCM 7.1(2) with following topology:

Calling Party A --- SIP trunk --- CUCM/Called Party B --- SIP trunk Transfer destination Party C

A calls B and B blind transfers to C

Workaround:

Instead of blind transferring the call perform a consult transfer

Further Problem Description:

Once the transfer is completed and part C answers the call, SDP/media will be re-negotiated.
The final SIP 200 OK from PartyA/SIP trunk will not be ACK'ed by CUCM resulting in observed behaviors

NOTE:
This will also cause the Calling party name not displayed when the call comes to the final transferred number In this call flow.

Regards

Alex

[Louis Niel Panganiban]

Hi acampbell,

I'll try the consult transfer first. When we did a blind transfer I (caller) cannot hear any rings or music on hold the transferred party just answered the call but they confirm that they are hearing rings from their end.

Please advise.

Thank you very much guys.

Louis

[Louis Niel Panganiban]

Topology

Sent from Cisco Technical Support iPhone App

[Louis Niel Panganiban]

Topology the router of 1503 is having the issue, a one way audio...

Sent from Cisco Technical Support iPhone App