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

Anthony Kouloglou akoul at dataways.gr
Tue Jan 24 14:34:28 EST 2012


Nope!
I have got a TAC open since the day before yesterday but i really count 
on all of you guys!!
:-)

On 24/1/2012 9:29 μμ, Ruben Montes (Europe) wrote:
> Is it working properly now?
> ------------------------------------------------------------------------
> *From:* cisco-voip-bounces at puck.nether.net 
> [cisco-voip-bounces at puck.nether.net] on behalf of Anthony Kouloglou 
> [akoul at dataways.gr]
> *Sent:* 24 January 2012 20:17
> *To:* Mike King
> *Cc:* cisco-voip at puck-nether.net; Mike
> *Subject:* Re: [cisco-voip] After upgrade to 8.6.2a one way audio for 
> some calls-No codec selected!
>
>
>
> Hi all,
> well, i have disabled any kind inspection on the ASA.Isn't that enough?
> ASA does NOT NAT. Isn't that enough?
> However, i have to check some corporate linux based vpn endpoints.
>
> Anthony
>
> On 24/1/2012 6:30 μμ, Mike King wrote:
>> Yes.
>>
>> But not just 8.6.
>>
>> https://supportforums.cisco.com/docs/DOC-8131
>>
>> (Hey Wes, can you fix the link on that to remove the partner only 
>> link ( SCCPv17 significantly changes message formats from previous 
>> versions 
>> <http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/rel_notes/7_0_1/cucm-rel_notes-701.html#wp584451> )
>>
>> It's when you upgraded the firmware on the Phones.
>>
>> The SCCP protocol has version numbers.  I'm finding references all 
>> the way up to SCCP version 20 (in 8.5.1).
>>
>> Looks like ASA version 8.3 only supports up to version 19.
>>
>> ASA version 8.4 supports SCCP v2.0  (Don't know what that means)
>>
>> Mike
>>
>> 2012/1/24 Anthony Kouloglou <akoul at dataways.gr 
>> <mailto:akoul at dataways.gr>>
>>
>>     Hi Mike,
>>     i have completely disabled inspection on an ASA that i have that
>>     does only routing.
>>     The question is: has something changed in SCCP negotiation in
>>     CUCM 8.6?
>>     The whole setup has been working for 3 years!!
>>
>>     Anthony
>>
>>
>>     On 24-Jan-12 16:34, Mike King wrote:
>>>     Having been bitten by this, Check for this.
>>>
>>>     Specifically, do you have ASA's doing site to site VPN's?  By
>>>     default they do INSPECTION, which can drop SCCP packets they
>>>     don't recoginize.
>>>
>>>     Mike
>>>
>>>     2012/1/23 Dennis Heim <Dennis.Heim at cdw.com
>>>     <mailto:Dennis.Heim at cdw.com>>
>>>
>>>         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
>>>         <mailto:cisco-voip-bounces at puck.nether.net>
>>>         [cisco-voip-bounces at puck.nether.net
>>>         <mailto:cisco-voip-bounces at puck.nether.net>] on behalf of
>>>         Ryan Ratliff [rratliff at cisco.com <mailto:rratliff at cisco.com>]
>>>         *Sent:* Monday, January 23, 2012 2:05 PM
>>>         *To:* Anthony Kouloglou
>>>         *Cc:* Mike; 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!
>>>
>>>         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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>>>>
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>>>>
>>>
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>>>
>>
>
>
>
> itevomcid

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