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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'>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. <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'>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.<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'>Incoming:<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'>PRI -> Cisco 3640 –SIP-->
Main_Asterisk (10/26/04) –IAX--> Our_Asterisk (1/06/05) –SIP-->
Cisco 7960 IP Phone<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'>Local Outgoing:<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'>Cisco 7960 –SIP--> Our_Asterisk
(1/06/05) –IAX--> Main_Asterisk –SIP--> Cisco 3640 --> PRI<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'>Long Distance Outgoing:<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'>Cisco 7960 –SIP--> Our_Asterisk
(1/06/05) –IAX--> Main_Asterisk –IAX--> NuFone<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'>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. <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'><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'>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?<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'>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. <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'>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. <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'>I don’t/have not used any extra-curricular
Asterisk patches. Most of them confuse or concern me anyway.<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'><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
12:13 PM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> RE: [Asterisk-Users]
Static/Breaking up after I upgraded Asteriskaswell 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 class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I am also running a pretty recent version
albeit not today’s CVS, but CVS-HEAD-11/20/04-11:29:52. <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'>D you have problems b/w the Ciscos or only
when going out to the PSTN??<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'>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.<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'>Can you tell us a little more about your
setup???<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'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Tahoma><span
style='font-size:10.0pt;font-family:Tahoma'>-----Original Message-----<br>
<b><span style='font-weight:bold'>From:</span></b>
asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]
<b><span style='font-weight:bold'>On Behalf Of </span></b>Paul Rodan<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, January 10, 2005
11:58 AM<br>
<b><span style='font-weight:bold'>To:</span></b> '<st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName>'<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Asterisk-Users]
Static/Breaking up after I upgraded Asterisk aswell as a crash - Can't trace
bug</span></font><o:p></o:p></p>
<p class=MsoNormal style='margin-left:.5in'><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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 style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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 style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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 style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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 style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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 style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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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