: 1555052k total, 1514728k used,40324k free, 282008k buffers
Swap: 6289436k total, 608k used, 6288828k free, 418468k cached
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looks like I need a bigger box!
On 8/15/12 8:54 AM, Michael Picher wrote:
Well, 2 GB is the minimum recommended for 4.4.
4 GB is the minimum recommended for 4.6.
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is the minimum recommended for 4.6.
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people buy more memory, just how they wanted to make everything in
memory and less dependent on storage. It's a two-edged sword for sure
though.
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On 8/15/12 12:56 PM, Joegen Baclor wrote:
On 08/16/2012 12:42 AM, Michael Scheidell wrote:
you hear see the engineers at DOS, DISA, NASA, United Defense, laugh
when you talk about 'real time java'
Really not a place to ridicule a tool where such tool shined. (coming
from a C++ perspective
Read the archives, I found this out last month
We had to scrap all are cisco phones and put in polycom.
Anyone would like to buy some cisco phones, real cheap? 10 dollars a phone? 15
dollars for a pair?
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Michael Scheidell, CTO
|SECNAP Network Security
-Original message-
From: Ly
Correct, 4.4.0 broke it.
And it has nothing to do with obsolete 0.0 .0 .0 hold. Newer versions of the
cisco firmware don't use that anymore.. haven't, actually for several years.
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Michael Scheidell, CTO
|SECNAP Network Security
-Original message-
From: Ly Tran ly.t...@synaptyk.com
Did you do a backup of your configuration before the yum upgrade ?
If so, boot a 4.2 .1 cd on a brand new piece of hardware.
Restore your configuration and voicemail. Look at the website for the 4.2 .1
repo file,
Upgrade to latest 4.2 .1.
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Michael Scheidell, CTO
|SECNAP Network Security
The cdrom?
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Michael Scheidell, CTO
|SECNAP Network Security
-Original message-
From: Ly Tran ly.t...@synaptyk.com
To: apos;Discussion list for users of sipXecs softwareapos;
sipx-users@list.sipfoundry.org
Sent: Mon, Aug 13, 2012 22:39:11 GMT+00:00
Subject: Re: [sipx-users] Cisco Hold
updates for sipx ;-)
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.el5_8.2.i386 dhcp-3.0.5-31.el5_8.1.i386
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yum upgrade, and it said 'CDR [Failed]'
(I did a refresh, and restarted CDR), it restarted and the active call
entry that survived one crash, one CDR restart and one reboot, finally
went away.
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noop, just received a phone call, 10 mins ago, and it does not show up
in CDR at all! 'No details available'
restarting CDR syows 'Failed' [snow details]-
On 8/11/12 4:45 PM, Michael Scheidell wrote:
I had a sipx crash, while I was on a call.
The CDR still shows the call active.
(I did
when, when I rebooted? or when the server crashed?
On 8/11/12 5:10 PM, Tony Graziano wrote:
Did you send the server its profiles and let those services restart?
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,
SharedAppearanceAgent=Running,
SipXrest=Running}
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with [sip.secnap.com] on
https://sip.secnap.com:8092/RPC2;
2012-08-11T21:10:17.254000Z:40:JAVA:INFO:sip.secnap.com:P1-19::XmlRpcClientInterceptor:XML/RPC
ProcMgmtRpc.getStateAll with [sip.secnap.com] on
https://sip.secnap.com:8092/RPC2;
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INNSlocalhost.
;; Query time: 0 msec
;; SERVER: 127.0.0.1#53(127.0.0.1)
;; WHEN: Sat Aug 11 17:33:02 2012
;; MSG SIZE rcvd: 57
grep localhost /etc/hosts
127.0.0.1 localhost.localdomain localhost
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sure.. .
go check on your system, you will get exactly the same results.
On 8/11/12 5:41 PM, Tony Graziano wrote:
that would do it!! It created a zone localhost for you?
My 5 year old would say What the WHAT!
On Sat, Aug 11, 2012 at 5:33 PM, Michael Scheidell
michael.scheid...@secnap.com
=ConfigurationTestFailed,
sipXmrtg=Running,
ConfigServer=ConfigurationTestFailed,
sipXacccode=Running,
FreeSWITCH=Running,
SIPXProxy=Running,
SipXrelay=Running}
On 8/11/12 5:29 PM, Tony Graziano wrote:
sipxproc --state
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caching-nameserver.i386 30:9.3.6-20.P1.el5_8.2 dhclient.i386
12:3.0.5-31.el5_8.1 dhcp.i386 12:3.0.5-31.el5_8.1
initscripts.i386 0:8.45.42-1.el5.centos.1sudo.i386
0:1.7.2p1-14.el5_8.2
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; retry
1W ; expiry
1D ); minimum
IN NS @
IN A127.0.0.1
IN ::1
On Sat, Aug 11, 2012 at 5:49 PM, Michael Scheidell
this part scares me:
(oh, and phones are totally down, and I have a customer in amsterdam,
who is down)
On 8/11/12 5:58 PM, Tony Graziano wrote:
ResourceListServer=Running,
CallResolver=ConfigurationTestFailed,
sipXmrtg=Running,
ConfigServer=ConfigurationTestFailed,
--
Michael
ok, phones answer, and I can call out, but no gui at all anymore.
On 8/11/12 6:02 PM, Michael Scheidell wrote:
this part scares me:
(oh, and phones are totally down, and I have a customer in amsterdam,
who is down)
On 8/11/12 5:58 PM, Tony Graziano wrote:
ResourceListServer=Running
1W ; expiry
1D ); minimum
IN NS @
IN A127.0.0.1
IN ::1
On Sat, Aug 11, 2012 at 5:49 PM, Michael Scheidell
michael.scheid
: [ OK ]
Checking localhost name is not shared: [ OK ]
Checking /tmp directory has correct permissions: [ OK ]
Starting sipXpbx:
Starting sipxsupervisor: [ OK ]
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service postgresql start
Starting postgresql service: [FAILED]
[root@sip init.d]# service postgresql stop
Stopping postgresql service: [FAILED]
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sipx-restore -c configuration.tar.gz
Restore script invoked at Sat Aug 11 22:37:30 UTC 2012, INTERACTIVE=yes.
PostgreSQL does not appear to be running.
You must start it in order to Restore.
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: [FAILED]
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. Thanks for the warning though!
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won't
it start?
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
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needs to edit the wiki
and remove the cisco 79** support from that. they will NEVER work
remote, NEVER, and there is too many strange things wrong with them to
pretend you support them)
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On 7/29/12 5:45 PM, Tony Graziano wrote:
use an ata that does t.38 instead of g711 and that problem will fix
itself.
who has a good two port t.38 TA?
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buy you 10 polycoms. Only to
have something different happened to cisco phones next upgrade.
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|SECNAP Network Security
-Original message-
From: Michael Picher mpic...@ezuce.com
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
for one person won't work for the other, due to firmware
versions.
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I tested T.38 with 'faxzero' works fine.
but, if you use 'filesanywhere', you can upload your file, and fax it
(from your desktop, android, etc)
(i sent a 63 page fax to see what would happen.. worked fine, 29 mins
later :-)
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Michael Scheidell, CTO
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anyone else, and they all heard MOH.
(ps, I tested this, and can't find the recording.. where is it? )
/var/sipxdata/? where?
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a conference call.
on ext 701289
where do I find the conference recording?
cd /var/sipxdata/mediaserver/moh
-bash: cd: /var/sipxdata/mediaserver/moh: No such file or directory
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Michael Scheidell, CTO
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On 7/30/12 2:59 PM, George Niculae wrote:
You should see it in user portal for conference owner, conference folder
cd /var/sipxdata... where?
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sipxchange 0 May 8 12:08 1175-00.sta
-rw-r--r-- 1 sipxchange sipxchange343 May 8 12:08 1175-00.xml
-rw-r--r-- 1 sipxchange sipxchange 159518 May 8 12:07 1175-00.wav
[root@sip 1289]# ls -lt saved
I made the recording this am (or so I thought)
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Michael Scheidell, CTO
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On 7/30/12 3:00 PM, Tony Graziano wrote:
/var/sipxdata/mediaserver/data/mailstore/subscriber-number/conference
cd /var/sipxdata/mediaserver/data/mailstore/1289/conference
-bash: cd: /var/sipxdata/mediaserver/data/mailstore/1289/conference: No
such file or directory
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Michael Scheidell
Yes i did.
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Michael Scheidell, CTO
|SECNAP Network Security
-Original message-
From: Tony Graziano tgrazi...@myitdepartment.net
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Sent: Mon, Jul 30, 2012 19:26:20 GMT+00:00
Subject: Re: [sipx-users] 4.4.0
authentication, no registration), and ps, they support T.38
trunking!
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Additionally, a cisco phone can't make transfers anymore,
prior, it could make normal or blind to cisco. or blind to poly.
Now it can't make any transfers.
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yes.
On 7/29/12 2:26 PM, Todd Hodgen wrote:
May have missed this detail in your earlier email -- did you perform a
Yum update on the last ISO install? There are about 17 patch
released to 4.4 that are not included in the ISO.
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.
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workarounds.
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And if wishes were horses my driveway would prolly be full of crap.
P s, there is an rfc that says the pigs will fly given enough thrust. -:)
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-Original message-
From: Tony Graziano tgrazi...@myitdepartment.net
To: Discussion list
from the cli
sipx-restore -c configuration.tar.gz -v voicemail.tar.gz
chown -R sipxchange:sipxchange moh fixes it.
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/and or use port natting on my firewall?
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Picher wrote on Thu, 12 January 2012 16:40
It'll be in 4.6
you forgot the smiley face.
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On 1/25/11 5:29 PM, Tony Graziano wrote:
Realistically as an admin you have to ask yourself how many ancient
phones am I willing to support though
cisco's all work fine!
only problem is when you have cisco's and polycom's with those buggy
polycom firmware versions. :-)
--
Michael
Interesting. This could solve the port 5060 issue with most tier 1 itsps.
Maybe even the remote issues.
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-Original message-
From: Joegen Baclor jbac...@ezuce.com
To: Discussion list for users of sipXecs software
sipx-users
2011
20:29]Try karoo bridge instead. See
http://wiki.sipfoundry.org/display/sipXecs/Karoo+Bridge+SBC
can that be installed on the same computer as sipx? or do we
need a second computer?
--
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Michael Scheidell, CTO
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___
sipx
are crap, along with the 'buckets'
you need to constantly fill with cash.
BUT, based on ALL the problems sipx has with gremlins in
firewalls, natting, 3g, 4g, and call quality over cell
phones, running skype on the cell phone to talk to a office
sip box would be great.
--
--
Michael Scheidell, CTO
.
that way, you can enforce HIPAA/GLBA/ EU privacy
requirements on account access, but give the luser a simpler
voicemail pin code.
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. then FWD
the call there.
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=Michael Picher wrote on Fri, 26 August 2011
16:39]wrong list?
unless we now have voicemail spam assassin...
On Fri, Aug 26, 2011 at 4:32 PM, Michael Scheidell
mailto:michael.scheid...@secnap.com[/email] wrote:
would be an interesting project.
blacklists for attacks, voice to txt for voicemail
want to block
outbound caller id.
I think you can block it on sipx, even if user dials 911.
Other issues include the fact that if the user has a cheap
home firewall, its likely to muck up sip. Your support
calls will go through the roof.
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: update tmp directory $UPDTmp);
}
elsif (!clean_update_dir($UPDTmp)) {
die channel: attempt to clean update dir failed, aborting;
}
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sorry, wrong list.
On 8/26/11 4:32 PM, Michael Scheidell wrote:
found a bug in sa-update
https://issues.apache.org/SpamAssassin/show_bug.cgi?id=6655
if TMPDIR is not writable, sa-update will continue to attempt to
rotate through mirrors, over and over and over.
if (!$UPDTmp
at first about
an unknown browser) but works just fine.
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isn't picky, since a simple firewall that will
PAT, inbound udp 5060, to the sipx box on port 5080, seems
to work just fine.
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List
(might need to delete
gateway), ask them to send trunking to 5080, and see what
happens.
we do like the features of sipx, and they really can't be
beat, even if you needed to pay for it!.
--
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Michael Scheidell, CTO
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tickets every month to get credits for
strange things on the bill, incorrectly billed calls, etc.
pps, did I say that these are the best sip trunks and fiber
I have ever used?
--
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Michael Scheidell, CTO
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___
sipx-users mailing
fragmented packets, and if your firewall
blocks fragmented packets, you can lose audio.
SOUNDS like one of the problems with one certain version of
polycom firmware, and may be unrelated.
so, why, after sipx selected PCMU, does the AA outplay L16?
--
--
Michael Scheidell, CTO
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meetings.
This one glaring issue alone will prevent us from upgrading to 4.4.0,
and we might as well wait till 4.6 or beyond.
4.4.0 added for us T.38 (but we don't really need it). didn't add port
5060 trunking (which we needed), and breaks end users use of the
conferencing system.
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the nightly cronjob.
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they will get it right.
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http://www.foxnews.com/scitech/2011/04/21/amazon-server-outage-knocks-major-websites
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.
a good simlink will fix that.
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On 4/20/11 11:24 AM, Michael Scheidell wrote:
looks like (almost random) decisions on the sipx 4.2.0 dial plans on
how it matches 'near match' plans.
It is not like you can 'stack' dial plans in priority order.
it looks like its more of a 'best match first' type of thing. (so you
would need
On 4/20/11 11:24 AM, Michael Scheidell wrote:
looks like (almost random) decisions on the sipx 4.2.0 dial plans on
how it matches 'near match' plans.
It is not like you can 'stack' dial plans in priority order.
Looking to (better) implement E.164 dial translations on sipx 4.2.0
without
something like this might work. .
http://cpansearch.perl.org/src/SULLR/Net-SIP-0.55/bin/stateless_proxy.pl
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?
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sip:4.89.241.150:5080
2011-04-19 11:49:06.095206192.168.0.2 - 1.1.1.1 SIP Status: 406
Not acceptable
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On 4/19/11 11:53 AM, Michael Scheidell wrote:
On 4/19/11 11:39 AM, Tony Graziano wrote:
what you you think of implementing an (optional) 'ping' (sip method
'OPTIONS') at specified intervals?
There is an assumption that all itsp's support that... it would be
better to send it a keepalive
' every 30 seconds while up, then
email you if down, and ping every second while down till up, then email
you if up.
(is 'alarm' a api i can use via shell?)
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not allowed is a good sign.
(36.221 ms) SIP/2.0 405 Method Not Allowed
its a timeout you don't want.
maybe if this is implemented, you can have a selection 'valid RC's'
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On 4/19/11 2:09 PM, Michael Scheidell wrote:
On 4/19/11 1:58 PM, Matt White wrote:
Yeah, that would work too but still requires the itsp to support responding to
the options. Ultimalty I think thats the more elegant solution is to support
that type of sip options and response.
The other
List Archive: http://list.sipfoundry.org/archive/sipx-users/
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on port 5050 and
go available from there to take calls.
On thing I did notice: once you do that, the user loses his ability to
see his voicemail...
(on sipx 4.2.0 anyway)
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
ports.yet. The rate limiting feature will be the longer term fix.
-M
and, even at that, if this is just trunking calls, firewall off everyone
but your itsp's.
do it now!
why wait for the rush?
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network
, adding users,
phones, etc), java gets borked.
I do reset this nightly (in cron)
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
* Certified SNORT Integrator
abnormally with
[$EXITVALUE]
fi
exit 0
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
* Certified SNORT Integrator
* Hot Company Award, World Executive Alliance
to handle the cisco phones.
ps, I have 48 cisco 7960 phones for sale, cheap.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
* Certified SNORT Integrator
* Hot
should we add one to PAT inbound port 5060 to port 5080 on trunks? :-)?
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
* Certified SNORT Integrator
* Hot Company
behind pfsense, and used a different
firewall to do the PAT. but, still use pfsense for all 'non sipx' stuff.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
I seem to remember a 'tweak/hack' that could be done to the itsptrunkxml
file to keep the rtp on the itsp side if both inbound and outbound of
the call were to/from the same itsp.
is this something I remember wrong?
if not, what is it, and where is the file I hack?
--
Michael Scheidell, CTO
to port 5060, and sipx wants it on 5080.
you will need to 'pat' or port translate inbound from their ip:5060 to
sipx:5080.
(outbound xlation not needed)
seems to work, despite the layer7 (application) issues. just seems to work.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259
range set to
5060 and redirect target port set to 5080.
can't use pfsense.. it actually doesn't do what you want it to do.
sipx 4.6? might be able to listen to on 5060 for trunks (4.4 can if you
use a public ip for sipx)
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259
On 4/13/11 4:02 PM, Tony Graziano wrote:
I remember it very well because I was rolling around laughing so hard
Trying to understand how someone Would feel want a found out It was
not really a person doing that. It was just too funny!
anyone here remember bofbot?
(tinc...)
--
Michael
the 1.
for international, yes, put 00 or 01.
or, see about what type of support is needed for e.164 numbering.
I can't seem to figure out how to OUTPULSE the +1. but am working on it.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
network in some cases.
access to one of several siphack ip reputation lists., mostly rsync'd
cidr lists.
ability to feed BACK to public siphack ip reputation lists if the sipv
type user agents hit you.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network
resulting call:
dial 12 + matching suffix: ie 1259
in CDR, I see to: 5619482259, but RECIPIENT: 1259.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention Product, Networks Product Guide
* Certified SNORT
On 4/12/11 9:26 AM, Michael Scheidell wrote:
without e.164 inbound (+1), the 'recipient' in CDR is the user field,
with e.164 inbound, the CDR is the whole called number:
maybe its just totally inconsistant, all by itself.
I deleted user 1315, and added 1315 as an alias to 1259.
new call
, and they send sdp: 200
ok, ack it, then sipx sends another invite, and I see several more sdp
200ok's from them along with another invite.
yes, I can do packet traces.
any ideas till then?
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
.. it works.. it was just my firewall.
so, I send invite to itsp.1:5060.. it sends back ok, and tells me media
server is on itsp.2.
bingo.. it works.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
* Best Intrusion Prevention
yet, and itsp has not called back to fix E.164 yet)
is it the permissions?
I noticed that our inbound did call xlation custom rule doesn't have any
permissions on it.
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
ISN: 1259*1300
*| *SECNAP Network Security Corporation
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