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<title>Re: [Asterisk-Users] Static/Breaking up after I upgradedAsteriskaswell
as a crash - Can't trace bug</title>
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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>About 15 of the 20 phones are plugged into
a Cisco Powered Switch. About 5 of the phones are using Power Cubes. The
specific users having problems, their phones are powered off the powered
switch. The powered switch is plugged into a UPS. The 24 port powered switch
has about 75% phones and 25% regular computers connected to it, and it is using
VLAN’s to separate the 4 LAN segments I mentioned earlier. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks for the suggestion though, I’ll
try and keep an eye out for customers reporting quality issues that do use
Power Cubes. I noticed there are 2 different types of Power Cubes, official
Cisco ones, and aftermarket ones, which were you using?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Alexander Lopez<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, January 10, 2005
1:18 PM<br>
<b><span style='font-weight:bold'>To:</span></b>
asterisk-users@lists.digium.com<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users]
Static/Breaking up after IupgradedAsteriskaswell as a crash - Can't trace bug</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p style='margin-bottom:12.0pt'><font size=2 face="Times New Roman"><span
style='font-size:10.0pt'>If your remotes are not reporting any trouble. This
may be farfetched but power may be to blame. I have had the ciscos 'freak out'
with unstable power. It looks like the load on the power cubes cannot keep the
caps loaded to deal with fluctuations. Or you may have a ground loop somewhere.<br>
<br>
Are the phones plugged into UPSs?<br>
<br>
I had flaky 7960 work fine after pluged into a Cisco POE switch.<br>
<br>
Keep me in the loop I would like to are how this turns out.<br>
<br>
<br>
<br>
-----Original Message-----<br>
From: asterisk-users-bounces@lists.digium.com
<asterisk-users-bounces@lists.digium.com><br>
To: '<st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial
Discussion</st1:PersonName>' <asterisk-users@lists.digium.com><br>
Sent: Mon Jan 10 12:49:40 2005<br>
Subject: RE: [Asterisk-Users] Static/Breaking up after I upgradedAsteriskaswell
as a crash - Can't trace bug<br>
<br>
Sure. I have yet to test inter-office calls, I’m not sure if it’s
out to the PSTN or not. But I would assume if it’s Asterisk, then
inter-office calls should suffer as well? Will have to test that further,
as well as phones connected to Sipura converters, see if they have issues. But
every minute I’m running this version I’m afraid it’ll up and
crash on me again.<br>
<br>
<br>
<br>
We have 4 PRI’s plugged into a Cisco 3640 Router, which then converts an
incoming call to SIP and sends it to the Main Asterisk server. They’re
connected via a Cisco Catalyst SmartSwitch (not PoE), but with no VLAN’s
or QOS or anything special configured on it. It’s really only got the
Cisco Router, the Main Asterisk Server, the Backup Asterisk Server, and our
Companies Asterisk server off of it. An incoming call then goes from the Main
Asterisk Server to our Companies Asterisk server, via IAX and then from there
our Asterisk server sends it to our phones via SIP. It’s all local. There
are several of our partner companies registering with the Main Server, for now
I’ve left that on the 10/26/04 version. No severe quality issues reported
on their end.<br>
<br>
<br>
<br>
Incoming:<br>
<br>
PRI -> Cisco 3640 –SIP--> Main_Asterisk (10/26/04) –IAX-->
Our_Asterisk (1/06/05) –SIP--> Cisco 7960 IP Phone<br>
<br>
<br>
<br>
Local Outgoing:<br>
<br>
Cisco 7960 –SIP--> Our_Asterisk (1/06/05) –IAX-->
Main_Asterisk –SIP--> Cisco 3640 --> PRI<br>
<br>
<br>
<br>
Long Distance Outgoing:<br>
<br>
Cisco 7960 –SIP--> Our_Asterisk (1/06/05) –IAX-->
Main_Asterisk –IAX--> NuFone<br>
<br>
<br>
<br>
At first I thought it was NuFone, but I’ve heard this reported on local
phone calls as well, and I think (but am not positive) I’ve heard of this
happening on incoming calls as well.<br>
<br>
<br>
<br>
<br>
<br>
I think the first thing I need to do is isolate whether inter-office calls are
garbled, this will isolate it to CVS 1/06/05; I think my next step is to try a
Sipura SPA-2000 converter, see if it experiences quality issues inter-office or
out to the PSTN; Are there known issues with IAX linking between older and
newer versions?<br>
<br>
<br>
<br>
I have IAX trunking turned off, since it’s all local anyway, but I am
using the ZapRTC hack on our Asterisk server, the Real-Time-Clock replacement
to give me Zaptel timing, so I do have zaptel timing. I have the modules zaprtc
and zaptel loaded, and I have rtcsetup program running in the background on our
Asterisk server. I have used meetme with Our_Asterisk and it works great.<br>
<br>
<br>
<br>
I had issues with quality when I tried to change the kernel on the
Main_Asterisk server, I tried to upgrade to the latest stock kernel, but with
RTC not compiled in, this way I could also use zaprtc. I’m not why
Asterisk liked the standard Redhat kernel much more than the stock kernel.
Anyways, I don’t have zaptel timing working on the Main_Asterisk server,
but since trunking is off and I don’t use meetme or anything, it’s
never been an issue.<br>
<br>
<br>
<br>
I don’t/have not used any extra-curricular Asterisk patches. Most of them
confuse or concern me anyway.<br>
<br>
<br>
<br>
<br>
<br>
<br>
<br>
_____ <br>
<br>
From: asterisk-users-bounces@lists.digium.com [<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>]
On Behalf Of Alexander Lopez<br>
Sent: Monday, January 10, 2005 12:13 PM<br>
To: <st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial
Discussion</st1:PersonName><br>
Subject: RE: [Asterisk-Users] Static/Breaking up after I upgraded
Asteriskaswell as a crash - Can't trace bug<br>
<br>
<br>
<br>
I am also running a pretty recent version albeit not today’s CVS, but
CVS-HEAD-11/20/04-11:29:52.<br>
<br>
<br>
<br>
D you have problems b/w the Ciscos or only when going out to the PSTN??<br>
<br>
<br>
<br>
I have 35 7960s with a PRI and no problems that you speak of. I do get an
occational dropped call but that may be the DHC server lease running out on the
phones.<br>
<br>
<br>
<br>
Can you tell us a little more about your setup???<br>
<br>
<br>
<br>
<br>
<br>
-----Original Message-----<br>
From: asterisk-users-bounces@lists.digium.com [<a
href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>]
On Behalf Of Paul Rodan<br>
Sent: Monday, January 10, 2005 11:58 AM<br>
To: '<st1:PersonName w:st="on">Asterisk Users Mailing List - Non-Commercial
Discussion</st1:PersonName>'<br>
Subject: [Asterisk-Users] Static/Breaking up after I upgraded Asterisk aswell
as a crash - Can't trace bug<br>
<br>
<br>
<br>
We use Cisco 7960’s with the P0S3-7-3-00 firmware, which was the latest
as of a few months ago.<br>
<br>
<br>
<br>
I’ve so far found CVS-v1-0-10/26/04-07:28:01 to be the best version of
Asterisk I’ve found. I’ve upgraded regularly in the past, like
every other week. I upgraded to this version and also encountered no issues.
About a month later I tried upgrading, to some version in November, and
that’s when all the phones in my office started experiencing quality
issues, breaking up, garbled voice, maybe static. One person reported they had
issues transferring calls, but I could not verify. So I immediately downgraded
back to 10-26-04 and stayed on that version up until several days ago, when I
hoped whatever bug was introduced had been repaired. So I upgraded to
CVS-v1-0-01/06/05-01:53:07 and for a time I thought everything was fine, but
now I get the occasional report of quality issues again, phones breaking
up/garble. It’s not as bad as it was before, but I myself have started
experiencing quality issues on my phone and I’ve never experienced these
issues before. So whatever bug existed, still exists. It might not be a bug,
but maybe some modification to chan_sip has broken compatibility with the Cisco
79xx phones. Unfortunately I am not a developer and am not able to take apart
and put back together again the source, to adapt it to my own needs, so
I’m at the mercy of the Asterisk developers.<br>
<br>
<br>
<br>
Nothing has changed in our network topology, no new phones added, no computers
share the subnets with the phones. They’re part of the same physical
network as our computers, but do have their own separate subnet. I
haven’t tried other phones, or converters, but I can if anybody wants me
to do further analysis. What I have here is a unique situation to single out a
quality issue, a bug, and I’d like to help by testing different versions
of chan_sip.c to see which option/modification in fact created the quality
issue. I would stay on this version of Asterisk longer (despite the occasional
quality issues) if it weren’t for the fact that yesterday evening the
Asterisk daemon crashed for no apparent reason. Until this time, the 6 months
I’ve been running Asterisk, Asterisk has never crashed on me. All the
phones in one of my Call Groups started ringing for no reason, when I answered,
nobody was there. I had to go and answer each phone individually, there
wasn’t anybody on any phone. After the last was answered and hung up, the
phones were quiet. But when I tried to access voicemail or dial out, nothing,
that’s how I’d know Asterisk had crashed, the /var/log/asterisk/messages
file revealed nothing, absolutely nothing, neither did /var/log/messages<br>
<br>
<br>
<br>
So tonight I’m going back down to 10-26 and I’d bet money the
quality issues disappear, it’s happened before. I hate downgrading, I
feel like I’m now stuck at a certain version and am unable to proceed
safely. The security and bug fixes I keep seeing hit the stable version are all
now no longer available to me, which sucks. Can anybody suggest how I can trace
this issue? During one of the phones conference calls the quality was really
horrible, so I started a constant ping on the phone to see if there were jumps
in latency or packet loss, as their very sensitive to this, and I didn’t
see any.<br>
<br>
<br>
<br>
<br>
<br>
The only other thing I can think of is bad phones? One sales guy had continuous
phone quality issues since the upgrade, so I traded my phone for his, switched
the config files. And I made his mine. And when I started placing calls, I
started having quality issues in which I didn’t have any before. So I
thought the phone must be defective. So then I grabbed another phone, a Cisco
7940 and made it mine, and the quality is better, but I still here the
occasional robotic sound when I place calls, and this is a completely different
phone. I’m thinking I just didn’t notice the problems as severely
as others have. There’s also 4 or 5 other employees (about 20 of us in
total) reporting quality issues, I don’t think it’s possible for
that many phones to fail when they’ve been doing good for so long.<br>
<br>
<br>
<br>
Any help or advice would be helpful. However, a couple of my friends companies
run asterisk and I’ve already seen the “I use CVS Version so and so
(newer than mine) and I don’t have any problems, etc. etc.” lines,
so I know that for most it must work, but I’m one of the ones that it
does not work for, and I’d really be interested in finding out why.
I’ve pretty much eliminated the network possibility. They’re all
local, we have 4 physical segments each with 2 subnets, 1 for the computer and
1 for the phones, the subnets are all linked by a Linux firewall with multiple
interfaces. This firewall is set to give the VOIP Subnets full access to one
another, and has never interfered with the phones ability to communicate with
the Asterisk server before.</span></font><o:p></o:p></p>
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