[OSL | CCIE_Voice] Voicemail not working for some calling users

2011-05-19 Thread Daniel Gómez
Hello experts,

I have the following scenario:

Phone A at Company A HQ(CUCM 8.0 + Unity Express 8.0) -- LAN -- ASA -
2Mbps WIRELESS LINK -- Checkpoint -- LAN -- Phone B at CompanyB HQ

_

Phone A @ Company A can call Phone B at Company B.
Phone B @ Company B can call Phone A at Company B.

PSTN users can call Phone A @ Company A.
PSTN users can leave a voicemail to user  phone A in company A.

Phone B @ Company B CAN'T leave a voicemail to user  phone A in company A.
__

At first, we were thinking about a transcoding issue but everything seems to
be OK. As we are limiting bandwidth to 128kbps between Company A and Company
B, we're using G729 codec for calls between these sites.

We have a XCODE at the gateway and it is registered correctly and seems to
be working OK.

In the past, we have had issues regarding blocked ports on the checkpoint
causing 1-way audio, or Phone B not hearing PhoneA's music on hold.
Do you think it could be part of the issue again?

We're running out of ideas as we have checked everything we think could be
causing the issue.

Thanks a lot!

Daniel G.
___
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Re: [OSL | CCIE_Voice] Voicemail not working for some calling users

2011-05-19 Thread William Affeldt
I would do a wire shark packet capture on the phone side and see if you can see 
both streams. If you can't repeat that at every hop until you find the break in 
the stream. 

Sent from my iPhone

On May 19, 2011, at 7:19 AM, Daniel Gómez danie...@gmail.com wrote:

 Hello experts,
 
 I have the following scenario:
 
 Phone A at Company A HQ(CUCM 8.0 + Unity Express 8.0) -- LAN -- ASA - 
 2Mbps WIRELESS LINK -- Checkpoint -- LAN -- Phone B at CompanyB HQ
 
 _
 
 Phone A @ Company A can call Phone B at Company B.
 Phone B @ Company B can call Phone A at Company B.
 
 PSTN users can call Phone A @ Company A.
 PSTN users can leave a voicemail to user  phone A in company A.
 
 Phone B @ Company B CAN'T leave a voicemail to user  phone A in company A.
 __
 
 At first, we were thinking about a transcoding issue but everything seems to 
 be OK. As we are limiting bandwidth to 128kbps between Company A and Company 
 B, we're using G729 codec for calls between these sites. 
 
 We have a XCODE at the gateway and it is registered correctly and seems to be 
 working OK.
 
 In the past, we have had issues regarding blocked ports on the checkpoint 
 causing 1-way audio, or Phone B not hearing PhoneA's music on hold.
 Do you think it could be part of the issue again?
 
 We're running out of ideas as we have checked everything we think could be 
 causing the issue.
 
 Thanks a lot!
 
 Daniel G.
 ___
 For more information regarding industry leading CCIE Lab training, please 
 visit www.ipexpert.com
 
 Are you a CCNP or CCIE and looking for a job? Check out 
 www.PlatinumPlacement.com
___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com

Re: [OSL | CCIE_Voice] Voicemail not working for some calling users

2011-05-19 Thread Randall Saborio
I would open a TAC case.

2011/5/19 William Affeldt william.affe...@yahoo.com

 I would do a wire shark packet capture on the phone side and see if you can
 see both streams. If you can't repeat that at every hop until you find the
 break in the stream.

 Sent from my iPhone

 On May 19, 2011, at 7:19 AM, Daniel Gómez danie...@gmail.com wrote:

  Hello experts,
 
  I have the following scenario:
 
  Phone A at Company A HQ(CUCM 8.0 + Unity Express 8.0) -- LAN -- ASA -
 2Mbps WIRELESS LINK -- Checkpoint -- LAN -- Phone B at CompanyB HQ
 
  _
 
  Phone A @ Company A can call Phone B at Company B.
  Phone B @ Company B can call Phone A at Company B.
 
  PSTN users can call Phone A @ Company A.
  PSTN users can leave a voicemail to user  phone A in company A.
 
  Phone B @ Company B CAN'T leave a voicemail to user  phone A in company
 A.
  __
 
  At first, we were thinking about a transcoding issue but everything seems
 to be OK. As we are limiting bandwidth to 128kbps between Company A and
 Company B, we're using G729 codec for calls between these sites.
 
  We have a XCODE at the gateway and it is registered correctly and seems
 to be working OK.
 
  In the past, we have had issues regarding blocked ports on the checkpoint
 causing 1-way audio, or Phone B not hearing PhoneA's music on hold.
  Do you think it could be part of the issue again?
 
  We're running out of ideas as we have checked everything we think could
 be causing the issue.
 
  Thanks a lot!
 
  Daniel G.
  ___
  For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com
 
  Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com
 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com




-- 
Randall da ill Saborio
CCIE Voice Wannabe #10054675811
___
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Re: [OSL | CCIE_Voice] Voicemail not working for some calling users

2011-05-19 Thread George Goglidze
Hi Daniel,

Can you give us the following information please?

0) Describe the problem better, when the call is forwarded to voicemail what
happens?
a) Call does not establish? and after some timeout it disconnects?
b) call establishes but there is no audio? phone in company B does not
hear anything from UC?
c) call establishes, and company B can hear UC announcements, but can
not leave voicemail?
1) What region are the HQ Phones in.
2) What region is the UC in.
3) What region is the transcoder in.
4) What region is the company B telephony system in.
5) What protocol is the company B using? SIP/h323 ?
6) What are the inter-region codecs between these?
7) on the transcoder:
a) show sccp connections (when the call goes to voicemail.)
b) show call active voice compact
c) show call active voice brief
8) if company B is h323 trunk/gateway - enable inbound fast-start, and
select appropriate codec for this. You might need MTP depending on the
following:
a) In the company B phone system, enable outbound fast start. You will
need MTP in the company B side for this.
b) If you can't enable outbound fast start on company B side, then check
- MTP required box on your call manager, and configure MTP resource.
9) if it's SIP integration, do same as in 8b - check MTP requiered, and
configure MTP resource.

Hope this helps,


2011/5/19 Daniel Gómez danie...@gmail.com

 Hello experts,

 I have the following scenario:

 Phone A at Company A HQ(CUCM 8.0 + Unity Express 8.0) -- LAN -- ASA -
 2Mbps WIRELESS LINK -- Checkpoint -- LAN -- Phone B at CompanyB HQ

 _

 Phone A @ Company A can call Phone B at Company B.
 Phone B @ Company B can call Phone A at Company B.

 PSTN users can call Phone A @ Company A.
 PSTN users can leave a voicemail to user  phone A in company A.

 Phone B @ Company B CAN'T leave a voicemail to user  phone A in company A.
 __

  At first, we were thinking about a transcoding issue but everything seems
 to be OK. As we are limiting bandwidth to 128kbps between Company A and
 Company B, we're using G729 codec for calls between these sites.

 We have a XCODE at the gateway and it is registered correctly and seems to
 be working OK.

 In the past, we have had issues regarding blocked ports on the checkpoint
 causing 1-way audio, or Phone B not hearing PhoneA's music on hold.
 Do you think it could be part of the issue again?

 We're running out of ideas as we have checked everything we think could be
 causing the issue.

 Thanks a lot!

 Daniel G.

 ___
 For more information regarding industry leading CCIE Lab training, please
 visit www.ipexpert.com

 Are you a CCNP or CCIE and looking for a job? Check out
 www.PlatinumPlacement.com

___
For more information regarding industry leading CCIE Lab training, please visit 
www.ipexpert.com

Are you a CCNP or CCIE and looking for a job? Check out 
www.PlatinumPlacement.com