[cisco-voip] After upgrade to 8.6.2a one way audio for some calls-No codec selected!

Mike mikeeo at msn.com
Mon Jan 23 12:41:47 EST 2012


Your key statement is this:

 

Then, we moved it to another subnet.
It got registered but not audio in one way!

 

Check your routing path to the CM.

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony Kouloglou
Sent: Monday, January 23, 2012 10:15 AM
To: Nate VanMaren
Cc: cisco-voip at puck-nether.net
Subject: Re: [cisco-voip] After upgrade to 8.6.2a one way audio for some
calls-No codec selected!

 

Yes!
Everything seems to be as it supposed to be!
One Phone got registered at the main site. Worked fine.
Then, we moved it to another subnet.
It got registered but not audio in one way!

Can't this ITL/CTL feature/bug be disabled?

On 20-Jan-12 17:26, Nate VanMaren wrote: 

Are your phones running firmware you expect them to be?

 

From: cisco-voip-bounces at puck.nether.net
[mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Anthony Kouloglou
Sent: Friday, January 20, 2012 1:33 AM
To: cisco-voip at puck-nether.net
Subject: [cisco-voip] After upgrade to 8.6.2a one way audio for some
calls-No codec selected!

 

Hi all,
here is a tough one! 
I recently upgraded my 6.1 cluster to 8.6.2a.
Since my Hardware was 7825H3 typically it was not an upgrade rather than a
fresh install using a usb drive (cisco has this procedure for these type of
servers)
The upgrade was smooth for pub and one sub.
All phones reregistered and upgraded.
In the main site there are 20 devices (7975, 7961, 7911) and at 2 remote
sites 2 devices (one at each site).
After the upgrade:
all phones in the main site can talk to each other.
The two remote phones can talk to each other.
Each of the remote phones when talking to main site have one way audio!
The remote site does not hear the main site always.
There is no firewall/NAT  between the sites.
I noticed that there is no codec selected for the audio stream that has the
problems and so no transmit (or received packets for the other).
And i explain: in an active call between the main site and a remote i
checked the send/received codecs and statistics.
the main site had g711 as received codec and of course the received packets
augmented
but there was none as send codec and of course no packets transmited.
In the remote site the findings were inversed (no receive codec and no
receive packets

lease advise

BR
Anthony









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