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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>We use Cisco 7960’s with the P0S3-7-3-00 firmware,
which was the latest as of a few months ago.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>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. <o:p></o:p></span></font></p>
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