asterisk users - Nov 2016 - Upgrading to Asterisk 13 - Strange Music On Hold Issue - Banging my head on this one

I sent this last night but it never showed up in the thread list so
I'm trying again. Please pardon me if it duplicates.

So I've been banging my head against the rack on this one and am now
turning to the group for help.

I'm in the process of bringing five Asterisk servers (all originally
built from source code by myself) from various versions
(1.6.2.x,11.6-cert13, and 13.1-cert2) up to the latest Asterisk
13.8-cert3. This has happened across multiple operating systems and
compiler versions: an old CentOS 5.9 server (GCC 4.1.2) that is
running 11.6-cert13 just fine and wanting to do an in place upgrade to
13.8-cert3, an Ubuntu 14.04.5 LTS (GCC 4.8.4) that was built from
scratch to replace an old server, and multiple Ubuntu 16.04.1 LTS (GCC
4.8, 4.9 and 5.4.0) built from scratch to replace old servers. The
servers themselves range from old to new, fast and faster, Dell and
Supermicro, production and lab. With one exception, I have had the
same Music On Hold problem on each one of the servers.

The problem is this: when a caller is placed on hold or parked,
initially the music on hold 100% of the time is garbled. It sounds
like what it did back in Asterisk 1.2 when zaptel/dahdi was screwed up
and timing was off. Within 5-20 seconds it will clear up. Once its
cleared up, you can take that caller off hold and put them back on
hold, or un-park and re-park them as many times as you want and the
Music On Hold will not glitch again.

I've tried timing with res_timing_dahdi and res_timing_timerfd. I've
tried quietmp3 and files (wav, gsm, ulaw, sln, mp3). I've tried
different codec: ULAW and G729. There is absolutely no load on any of
these machines when I'm testing... just the one phone call that I'm
testing on. I've tested with various SIP endpoints (Polycom
SoundPointIP 550, Polycom VVX 500 and 600, Zoiper, Linphone). It
doesn't matter if its two SIP endpoints or a SIP endpoint to PSTN (via
SIP of course), the behavior is always the same. Absolutely nothing
I've done has made a dent in the problem.

The only thing that is consistent is I don't have the problem on
11.6-cert13 or -cert15 (regardless of OS or processor). I have not
tried Asterisk 12 to see if the issue is present. I have tried
compiling 13.8-cert3 with GCC 4.8 and 4.9 and no change there either.
I've also tested Asterisk 14.1.1 and the behavior is the same This
behavior exists already on the in production 13.1-cert2 server, but
doesn't impact anything because nothing ever uses Music On Hold on
that box.

The two exceptions are a new Supermicro server and a Dell server that
are not having the problem at all running Asterisk 13.8-cert3. The
Supermicro is running Ubuntu 16.04.1 LTS (GCC 5.4.0) and the Dell is
running Ubuntu 14.04.5 LTS (GCC 4.8.4). Using res_timing_timerfd and
no dahdi kernel modules loaded, Music On Hold is pristine. The only
thing I can see is that both of these machines have Xeon E5
processors.

For anyone interested, my build instructions/script is posted at the link below.

Below are the processors running in the five servers. This is the only
difference I can really see in the machines. If it's a CPU issue, if I
knew what specific feature was causing it work I'd just replace the
three problem children with servers that knowingly fixed the problem.

Working fine on these processors:
Intel Xeon CPU E5-2650 @ 2.00Ghz (DELL PowerEdge R620, Currently
Ubuntu 14.04.5 LTS)
Intel Xeon CPU E5-2637 v4 @ 3.50Ghz (Supermicro, Currently Ubuntu 16.04.1 LTS)

Not working on these processors:
Intel Atom CPU 330 @ 1.60Ghz (Supermicro, Currently CentOS 5.9,
retiring but not until I figure this out!)
Intel Xeon CPU X3470 @ 2.93Ghz (Supermicro Currently Ubuntu 16.04.1 LTS)
Intel Xeon CPU E3-1230 v2 @ 3.30Ghz (DELL PowerEdge R310, Currently
Ubuntu 16.04.1 LTS)

I feel like I'm spinning my wheels on this one. Any help would be
GREATLY appreciated!!!

-- Mitch Sharp (bluecrow76)
http://files.bluecrow.net/asterisk/13/

Hi,

I am pulling my hair for days now...

I can?t get a PolyCom SoundStation IP 6000 (Conferencephone) to register 
with my Asterisk.

There are no SIP Packets arriving at my asterisk at all... and it has 
nothing to do with a firewall or similar...

Simple Question:
Does anybody have a running SoundStation IP 6000 registerd with asterisk?
If so... would you please be so kind to tell me whats wrong with my setup?

AsteriskServer: 192.168.1.211
SIP-user: 165

(the SIP-Settings on asterisk-side are OK, tested with a normal 
Softphone... registering and placing calls is no problem...)

The phone-log only says: "Registration failed User: 165, Error Code:480 
Temporarily not available"

I tried with newest firmware, resetting to factory 100 times, using a 
provisionig file (which the SoundStation correctly downloads)
but it is always the same... the SoundStation does not contact the 
asterisk for registering...

Phoneversion:
Telefoninformationen
Telefonmodell 	SoundStation IP 6000
Teilenummer 	3111-15600-001 Rev:W
MAC-Adresse 	00:04:F2:07:0C:D3
IP-Adresse 	192.168.0.13
UC-Softwareversion 	4.0.11.0583
BootROM-Softwareversion 	5.0.5.2324


I can ping the phone from the asterisk, the phone can reach the asterisk 
server (as it downloads the tftp files, if used with
a provisioning profile), so the route and everything is correct... I 
even connected another Hardphone on the same cable
that stuck in the Polycom... no problem... the other phone can register 
and works, so there is really no cable or firewall
related problem here... it must be a setting!

thank you



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2016-12-19 16:26 GMT+01:00 Yves <yves030 at gmx.de>:
> Hi,
>
> I am pulling my hair for days now...
>
> I can?t get a PolyCom SoundStation IP 6000 (Conferencephone) to register
> with my Asterisk.
>
> There are no SIP Packets arriving at my asterisk at all... and it has
> nothing to do with a firewall or similar...
>
> Simple Question:
> Does anybody have a running SoundStation IP 6000 registerd with asterisk?
>
yes, I've got several of them running.

> If so... would you please be so kind to tell me whats wrong with my setup?
>
> AsteriskServer: 192.168.1.211
> SIP-user: 165
>
> (the SIP-Settings on asterisk-side are OK, tested with a normal
> Softphone... registering and placing calls is no problem...)
>
> The phone-log only says: "Registration failed User: 165, Error
Code:480
> Temporarily not available"
>
> I tried with newest firmware, resetting to factory 100 times, using a
> provisionig file (which the SoundStation correctly downloads)
> but it is always the same... the SoundStation does not contact the
> asterisk for registering...
>
1. Do you have any switch able to mirror traffic sent and received by
Polycom phone ?
Capturing such traffic would help to understand what's happening.

2. Some phones support zero touch config with which they download their
config files from the Internet.
Are you sure this doesn't happen ?

3. Is SNTP/NTP correctly configured on the phone ?




>
> Phoneversion:
> Telefoninformationen
> Telefonmodell SoundStation IP 6000
> Teilenummer 3111-15600-001 Rev:W
> MAC-Adresse 00:04:F2:07:0C:D3
> IP-Adresse 192.168.0.13
> UC-Softwareversion 4.0.11.0583
> BootROM-Softwareversion 5.0.5.2324
> I can ping the phone from the asterisk, the phone can reach the asterisk
> server (as it downloads the tftp files, if used with
> a provisioning profile), so the route and everything is correct... I even
> connected another Hardphone on the same cable
> that stuck in the Polycom... no problem... the other phone can register
> and works, so there is really no cable or firewall
> related problem here... it must be a setting!
>
> thank you
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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On 12/19/2016 10:26 AM, Yves wrote:
> There are no SIP Packets arriving at my asterisk at all... and it has 
> nothing to do with a firewall or similar...
<snip>
> I can ping the phone from the asterisk,
If both of these items are true, then I'd look at the phone 
configurations. Does the provisioning file contain an address for the 
phone to contact?

Mine has voIpProt.server.1.address, but I think you can also use a 
reg.x.address in the provisioning files too.

Mark

Yves,

Didn't you say that
> AsteriskServer: 192.168.1.211
> SIP-user: 165
?

On 12/21/2016 4:24 AM, Yves wrote:
> . It is sure for 100% that there is no firewall or something else 
> mangeling
> in between... another Hardphone works as expected using the same 
> Netzworkcable on the same Networkplug with UDP on Port 5060...
This other hardphone, what IP does it have?
>
> 000050.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet 
> mask 255.255.255.0
>
The line above suggests to me that your phone and your asterisk server 
are on a different network, there has to be something that routes 
between those two networks. Often what routes, can firewall.
> 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 
> Temporarily not available
>
>
Mark
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Hi Mark,

yes, you are right... these are different VLANs
I configured the other phone to use the same IP (192.168.1.13)... and it 
worked flawlessly... on the SAME Networkcable in the same plug...
so it must have something to do with the polycom phone config... 
remember... when I use tcp the phone tries to register, but does not 
even try with udp...

thank you,
yves


Am 21.12.2016 um 13:34 schrieb Mark Wiater:
> Yves,
>
> Didn't you say that
>
>> AsteriskServer: 192.168.1.211
>> SIP-user: 165
> ?
>
> On 12/21/2016 4:24 AM, Yves wrote:
>> . It is sure for 100% that there is no firewall or something else 
>> mangeling
>> in between... another Hardphone works as expected using the same 
>> Netzworkcable on the same Networkplug with UDP on Port 5060...
>
> This other hardphone, what IP does it have?
>
>>
>> 000050.848|cfg  |*|03|RT|Primary IP changed to 192.168.0.13 subnet 
>> mask 255.255.255.0
>>
> The line above suggests to me that your phone and your asterisk server 
> are on a different network, there has to be something that routes 
> between those two networks. Often what routes, can firewall.
>
>> 000122.941|sip |4|03|Registration failed User: 165, Error Code:480 
>> Temporarily not available
>>
>>
>
> Mark
>
>
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