I deactivated all dial-rules except the one forward all digits to gateway-
Gateway is connected as I can receive incoming calls just fine.
So I tested again with DEBUG level on.
I think this is the part.
2010-07-03T09:57:54.245842Z:16482:SIP:DEBUG:voip.denckert.com:SipXProxyCse
On 2010-07-02 23:32, m...@grounded.net wrote:
Just now, *while* on a call, the phone dropped registration so it's getting
worse.
Another phone with the same number, registered didn't drop.
It's random and I'm looking at the logs now.
That is most likely just a broken phone, but there is far
On 7/2/10 7:41 PM, Tony Graziano wrote:
A lot of people are griping and moaning about this but their comments
lie in completely unrelated email threads.
Diagnostic in general, look at the FAQ's (not the written FAQ's, but
the questions and problems people post the most about). look at the
On 7/2/10 9:09 PM, Tony Graziano wrote:
Smells like an applet, but I don't see how that would make it easier.
you guys are thinking too much like programmers and engineers :-).
Think 'stupid user' or first time user.
I haven't been using sipx so long that I
Note how little actual information is in this report of a problem:
sigh This was just a quick post saying the problem had happened again and
that I was looking into it.
I then posted additional information.
___
sipx-users mailing list
If the server does not have CPU, RAM or other operational issues it's safe
to say there 'likely is an issue with the phone, or at least its
connectivity to the network.
I've never had phones lose registrations, but then I am pretty picky about
using phones that seem to work very well.
What kind
Here's my blog entry on it: http://sipxecs.blogspot.com/2010/03/using-
sipviewer-on-your-pc-for-windows.html
Thanks for the URL. I did find that and some other things in the wiki which I
had posted in the other thread I had started which lead me to the problem of
not being able to merge.
When
Are your log files in the proper location?
/var/log/sipxpbx
If not, there is something very wrong in your setup. You have said for weeks
that this is all you get when you run merge-logs, but that assumes:
your current logs are in:
/var/log/sipxpbx and that you run merge-logs from the same
I had the exacts same issue merging logs the first time I tried. I was
following a totally incorrect post/write-up on how to do it. What works
for me is copy logfiles from /var/log/sipxpbx to temp directory and then
run merge-logs
Sounds simple enough, there are some posts/write-ups out there
The hardware would be an IBM BladeCenter LS20 blade, dual duo-2.2Ghz, 4GB of
memory. I've checked for errors on the blade, haven't seen anything obvious,
everything else works, no errors in the log which usually shows CPU/memory
errors.
The dropped call was on a LinkSys PAP2 which has been
On Sat, 3 Jul 2010 09:29:01 -0400, Tony Graziano wrote:
Are your log files in the proper location?
/var/log/sipxpbx
Yup, standard, default, non modified ISO 4.0.2, updated to current via yum,
install.
/var/log/sipxpbx and that you run merge-logs from the same location and
grab the
Good call, I had also tried that because I didn't want to change anything in
the logs directory, so that if I broke the logs, I had a backup. So I copied
everything to a temp dir and ran merge-logs from there, tried different things
such as using the full path and I can't recall what other
Maybe try cleaning some older logs out of the temp dir? Just keep the newest
ones that you know capture the actual problem? Just a guess on my part. There
was something causing me to get the almost empty file you are getting. I
cleaned up all my old logs, copied only the recent logfiles to a
What i normally have to do, because I am trying to capture a particular
problem, is rotate logs (or simply delete the current logs), then make the
call happen that fails and then do merge-logs.
On Sat, Jul 3, 2010 at 11:19 AM, Matthew Kitchin (Public)
mkitchin.pub...@gmail.com wrote:
Maybe try
This is what we do when we test a known problem but in this case, we've not
found a pattern on when it happens.
So far, it's just guesses, while trying to get a pattern out of it. We'll try
another conf call test soon and see what happens with that.
I've cleared the logs to test but each
On Sat, 3 Jul 2010 15:19:34 +, Matthew Kitchin (Public) wrote:
Maybe try cleaning some older logs out of the temp dir? Just keep the
I even went as far as stopping sipx, clearing everything out, restarting, then
trying to get the problem.
The empty file problem, my guess is that it's
I would never use polycom firmware 3.2.x in production.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk
Think: if that question was never ever ever asked again, how much noise
would it save on the mailing list?
One more problem is definitely old information, such as wiki and other sipx
related historical info.
I can't count how many times I've searched high and low only to find old
information
On Sat, 3 Jul 2010 11:36:23 -0400, Tony Graziano wrote:
I would never use polycom firmware 3.2.x in production.
Must be working ok for someone else since it was suggested to me on the list.
Everyone has different setups so things that work well for another might not do
so good for someone
Still trying. Calling the different phones, staying on the line, hanging up,
nothing, no disconnections yet. Very random but still trying.
On Sat, 3 Jul 2010 11:36:23 -0400, Tony Graziano wrote:
I would never use polycom firmware 3.2.x in production.
Tony
3.2.x has known issues (Polycom is aware). Any latency on a connection
causes the call the drop... Use it if you wish, I've never considered 3.2.x
stable.
On Sat, Jul 3, 2010 at 12:42 PM, m...@grounded.net m...@grounded.netwrote:
On Sat, 3 Jul 2010 11:36:23 -0400, Tony Graziano wrote:
I would
An old dog on this list that has gone on to being a full time gamer taught
us this trick about 6 months ago -
logrotate -f /etc/logrotate.d/sipxchange
This will archive your logs and clean up the log directory. Run this.
Then, make a very simple call and run merge-logs from the var/log/sipxpbx
And yet very few people go to the trouble to go to that wiki page and update
the information in it when they find it wrong.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Saturday, July
I need to do some testing on a non-production sipx install. I was going to
virtualize it on my laptop (don't worry, I don't expect to make any calls/etc.
- this is more for me attempting to integrate the click to call app onto the
same box as sipXecs).
I've used most of the virtualization
I'm still waiting for access as I recall.
On Sat, 03 Jul 2010 10:48:31 -0700, Todd Hodgen wrote:
And yet very few people go to the trouble to go to that wiki page and update
the information in it when they find it wrong.
-Original Message-
From:
On Sat, 3 Jul 2010 13:41:27 -0400, Tony Graziano wrote:
3.2.x has known issues (Polycom is aware). Any latency on a connection
causes the call the drop... Use it if you wish, I've never considered 3.2.x
stable.
What do you suggest for a good version on these? The device files I have on
sipx
Mike,
Not to be critical here, but this is a key piece of information that hasn't
been in your emails as you explained this dropped call from last night -
that they were on Vonage and could have dropped on their end.
My suggestion here. Forget about all of the past history of this phone did
Not to be critical here, but this is a key piece of information that hasn't
been in your emails as you explained this dropped call from last night -
that they were on Vonage and could have dropped on their end.
That's the only vonage related call. Since it was an incoming vonage call,
that's
3.1.3RevC
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
3.1.3RevC
I can update my server with that.
For boot, I have spip_ssip_BootROM_4_2_2_release_sig.zip
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems
VMware ESXi 4.x works great except for some jittering on voicemail, IVR, etc.
-Original Message-
From: Nathaniel Watkins nwatk...@garrettcounty.org
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Sat, 3 Jul 2010 13:50:55
To: 'sipx-users'sipx-users@list.sipfoundry.org
Subject:
logrotate -f /etc/logrotate.d/sipxchange
Done.
Then, make a very simple call
Done.
and run merge-logs from the var/log/sipxpbx directory. Look again for merged.xml
Run from the sipx logs directory, Done.
Output is;
-rw-r--r-- 1 root root 79 Jul 3 14:16 merged.xml
Contents are;
# more
Boot is not a problem.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
tar and gzip your logs, email them directly to me, and I will see if they will
merge on one of my machines (unless someone knows a reason this wouldn't work).
What version of sipx are you running?
-Original Message-
From: m...@grounded.net m...@grounded.net
Sender:
I started a new thread a short while ago and included the file. Not sure why
it's not known up yet?
More than happy to create another set when we see all of the phones disappear
again and send them to you privately.
That however was a snapshot. If you want me to send all of the logs, I can do
The attachment may have been too big.
-Original Message-
From: m...@grounded.net m...@grounded.net
Sender: sipx-users-boun...@list.sipfoundry.org
Date: Sat, 3 Jul 2010 16:03:59
To: sipx-userssipx-users@list.sipfoundry.org
Reply-To: m...@grounded.net
Subject: Re: [sipx-users] All phones
The hardware would be an IBM BladeCenter LS20 blade, dual duo-2.2Ghz, 4GB of
memory. I've checked for errors on the blade, haven't seen anything obvious,
everything else works, no errors in the log which usually shows CPU/memory
errors.
The dropped call was on a LinkSys PAP2 which has
I know that you know what you mean by it, but what does dropped mean?
Right, I see I used that term twice in similar descriptions.
One 'drop' was a call that came in from someone using vonage. The call ended
abruptly, which we usually call a dropped call. We automatically figured it was
our
First of all, if I didn't cover all of the bases, PLEASE, do not hijack my
thread about it, either start another or email me personally.
For a couple of weeks or more now, we have started noticing a behavior which is
that of phones dropping registration.
Dropping, meaning that they disappear
I'm trying to serve another website from our sipx install. I have webmin
loaded, but it looks like sipx doesn't use the apache webserver module in
webmin. I'm a total newbie in regards to linux.
Here is my goal
Setup a listener for abc.sipxinstall.xyz and have apache serve up a website
when
On Sat, 3 Jul 2010 19:38:09 -0400, Nathaniel Watkins wrote:
Setup a listener for abc.sipxinstall.xyz and have apache serve up a website
when you visit http://abc.sipxinstall.xyz - I have the record in BIND - but
have no idea how to go about setting this up in apache...any pointers would
be
Only thing to keep in mind is that you might need to use a different port
depending on your needs and assuming this is not a problem on sipx. The
iptables filter aren't in play so other than that, I don't know why this would
not work.
http://apptools.com/phptools/virtualhost.php
On Sat, 3
Another newbie question. Any danger in loading php up on top of sipX? Anyone
done this before? I assumed that it would already be installed - doesn't look
like that is the case (unless I'm doing something incorrectly) - I found some
configurations it said to add to httpd.conf - when I tried
No php installed, it's all java/html based. If you need a bunch of custom
things, might be better to fire up a LAMP server somewhere and leave sipx
alone. It's too easy to break a well working system.
On Sat, 3 Jul 2010 23:40:34 -0400, Nathaniel Watkins wrote:
Another newbie question. Any
First of all, if I didn't cover all of the bases, PLEASE, do not hijack my
thread about it, either start another or email me personally.
For a couple of weeks or more now, we have started noticing a behavior which
is that of phones dropping registration.
Dropping, meaning that they
I had a request to port the click to call app we've written so everything could
be ran on a single box - right now it's spread across 3 servers (sipxecs/db/web
- trying to consolidate).
It is a group consensus that this isn't a good idea or would cause instability?
This message and any files
I'm not implying that it can't be done, it sure can be as there are other
integrations on the box.
I didn't know what you were after when I offered some input and as with
anything, depends on how much overall resources the end result would consume to
make it worth the effort or not.
I'm sure
In the middle of me playing with the sipx install at my house (for other
reasons entirely) - I thought I'd try setting up calling between the courthouse
and my house. Much easier than I thought it would be.
Is there a reason to not do the following (referring to my upcoming install
between 2
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