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

[Dennis Heim]

This may have already been mentioned but building on what Ryan said... probably between 6.1(2) and 8.6.x you had a firmware change, probably from around 8.4ish to 9.x. The sccp version changes, and it sounds like you might have some firewall/security device in the way that is not opening the ports because it is used to the older version of skinny.



-Dennis-

________________________________
From: cisco-voip-bounces at puck.nether.net [cisco-voip-bounces at puck.nether.net] on behalf of Ryan Ratliff [rratliff at cisco.com]
Sent: Monday, January 23, 2012 2:05 PM
To: Anthony Kouloglou
Cc: Mike; 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!

If the phone don't show a codec when the call is set up then this isn't a typical routing issue.  The most obvious reason for the phone not sending audio is it isn't getting the skinny StartMediaTransmission message from CUCM.
Have you looked at ccm traces for one of these calls?   When you do look at the messages going to and from the phones in the call. Compare/contrast what you see there to a working call and call out what's different.

You can get a packet capture at the phone as well to see what it is being told to send to from CUCM.   I'd also double check there's nothing in the network doing sccp inspection.   You can get a simultaneous packet capture at the phone and cucm to make sure every packet leaving the server gets to the phone (intact).

-Ryan

On Jan 23, 2012, at 1:48 PM, Anthony Kouloglou wrote:

There is no way that this is the problem.
In one remote site i had only one 7911 working fine with CUCM 6.1.2.
After the upgrade to 8.6.2a, even this old phone is having the same issue!
I keep having on the phone status: failed to update itl .

On 23/1/2012 8:09 μμ, Peter Slow wrote:
I think what MIke meant was "Check the routing path between the two phones."

-Peter


On Mon, Jan 23, 2012 at 12:41 PM, Mike <mikeeo at msn.com<mailto:mikeeo at msn.com>> wrote:
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> [mailto: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<mailto: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> [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<mailto: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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