e or issues I would like that self assurance that our
main line can still get to our answering service if it can't get to us.
Thanks
From: Warren K. Wiemer [mailto:wwie...@warrentek.com]
Sent: Friday, June 25, 2010 8:05 AM
To: Nathan Nieblas
Cc: sipx-users@list.sipfoundry.org
Anyone having a problem with Bandwidth.com right now?
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sipXecs IP PB
Y, Dale R (Dale) [mailto:dwor...@avaya.com]
Sent: Thursday, May 27, 2010 1:03 PM
To: Nathan Nieblas; keccles
Cc: sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] message queue is over half full
From: sipx-users-boun...@list.sipfoundry.org
[
em is currently running 4.2.0-018575
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of WORLEY,
Dale R (Dale)
Sent: Thursday, May 27, 2010 6:17 AM
To: Nathan Nieblas; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-
-boun...@list.sipfoundry.org] On Behalf Of keccles
Sent: Thursday, May 27, 2010 6:49 AM
To: Nathan Nieblas
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] message queue is over half full
On Wed, 2010-05-26 at 13:27 -0700, Nathan Nieblas wrote:
> Still getting this mad scrolling in the logs wit
Still getting this mad scrolling in the logs with parked calls on 4.2.
Any ideas? Is there a way to increase the queue size?
"2010-05-26T20:22:54.495213Z":9384181:KERNEL:NOTICE:pbx01.um.sacatech.co
m:WAVDecoder-556:B27EFB90:sipxpark:"OsMsgQShared::doSendCore message
queue '' is over half full -
Anything special with the Bria soft phone to get it to pick up parked
calls? The error when dialing is "call cannot be established". I've
tried with *4xxx and also created a dial plan that rewrites *4xxx when
dialing xxx. At least the presence is working now though J
Thanks
__
It appears that the presence function in the IM portion does not
recognize line status for users on X-lite 3 or 4 and any Cisco phone. Is
there something specific that is missing or can be configured on these
phones to point to? I assumed it was a backend function between openfire
and sipx directly
hanks
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Wednesday, April 28, 2010 6:11 AM
To: Nathan Nieblas
Cc: Tony Graziano; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] call park dropped calls - 4.2
On Tue, Apr 27, 2010 at 5:53 PM, Nathan Nieblas
wrote:
>
Anyone see a problem with this and what it should be? This was an
upgrade from 4.0 to 4.2
[r...@pbx01 openfire]# ls -l
lrwxrwxrwx 1 sipxchange root 26 Apr 20 15:41 openfire ->
/var/log/sipxpbx/openfire/
[r...@pbx01 openfire]# pwd
/var/log/sipxpbx/openfire/openfire/openfire/openfire
__
while it was happening though.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Tuesday, April 27, 2010 2:48 PM
To: Nathan Nieblas; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] call park dropped calls - 4.2
What is your default logging level
acatech.c
om:WAVDecoder-38:B1FF6B90:sipxpark:"OsMsgQShared::doSendCore message
queue '' is over half full - count = 1599, max = 1600"
"2010-04-27T21:24:04.252011Z":38166292:KERNEL:NOTICE:pbx01.um.sacatech.c
om:WAVDecoder-585:B12E9B90:sipxpark:"OsMsgQShared::doSen
ess I will just replace my Cisco's with Polycom's before
asking for any more help.
Regards
Nathan Nieblas
SACA Technologies, Inc.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael
Scheidell
Sent: Friday, April
e call (if
that even makes sense).
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@lis
Sent: Tuesday, April 20, 2010 6:01 AM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Cisco and sipX 4.2
>>> On 4/19/2010 at 08:44 PM, in message
,
"Nathan Nieblas" wrote:
> Ran into some firmware compatibility issues after upgrading from 4.0
to
> 4.2
>
Polycom calling the Cisco phone issues again.
Just FYI... If anyone has anything to share Cisco related I'm all ears J
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan
upgrading as simple as changing the yum repo's?
Regards
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
-Original Message-
From: M. Ranganathan [mailt
nected.
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
-Original Message-
From: M. Ranganathan [mailto:mra...@gmail.com]
Sent: Tuesday, March 09, 2010 12:28 PM
To:
This happens with and without hold music enabled on sipXbridge with both
Cisco and Polycom phones. Any ideas? Using bandwidth.com as my ITSP.
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
f the box functionality with Cisco products, mainly exhaust all
possible matches until there is an exact match (think ACL's).
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sa
as enabled
- then it would show up with everything else when editing a user (if
that's even possible).
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.c
All I have is "Extension NNN will ring first." I remember seeing of this
feature in the upcoming 4.1.x release but not 4.0.4 - is there something
in the backend that can be changed, even at a global level for all
extensions?
Nathan Nieblas
SACA Technologies, Inc.
1260
Using 4.0.4, is there any way to change the forward no answer duration
on an extension?
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com <mailto:nathan.n
I noticed that the records created when running the DNS server on sipX
does not create an A record for the sip domain itself, once I added that
my phones would stay registered.
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p
Make sure the phones can resolve your sip domain. I am also running on 8.5.x
code with no issues other than not being able to receive calls from Polycoms
(which is Polycom's problem IMO).
Sent via BlackBerry by AT&T
-Original Message-
From: Ben Wannan
Date: Fri, 26 Feb 2010 17:32:45
T
What version firmware are you running?
Sent via BlackBerry by AT&T
-Original Message-
From: Ben Wannan
Date: Fri, 26 Feb 2010 16:38:55
To:
Subject: [sipx-users] Cisco 7941G 7970G drops registration
___
sipx-users mailing list sipx-users@lis
x27;t want to work the way my Cisco mind does it in Unity.
So is it just me or is scheduling simply broke across the board?
Thanks
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: n
Is sipX on it's way to being multi-tenant friendly? Any ETA on this? I
want to replace Cisco Callmanager someday.
Thanks
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan
And I don't know how to paste I guess... Unmangled version:
http://www.pastebin.ca/1743041
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
-Original Message-
bal_policy
class inspection_default
no inspect sip
!
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
-Original Message-
From: sipx-users-boun...@list.sipf
in to everyone who got me going in the right direction.
Nathan Nieblas
SACA Technologies, Inc.
1260 N. Hancock Street, Suite 102
Anaheim Hills, CA 92807
p: 714-777-3222 x113
f: 888-603-9033
e: nathan.nieb...@sacatech.com
-Original Message-
From: Picher, Michael [mailto
Cool, I've disabled it - I am really looking forward to no problems come
morning. I'm assuming sipXproxy/sipXbridge behaves nicely with their
built in NAT functionality then with remote clients, etc?
Nathan Nieblas
SACA Technologies, Inc.
-Original Message-
From: Piche
It's a Cisco ASA 5510. I have now disabled port randomization for the NAT,
looks like the media control ports 3-31000 are pretty important and need to
stay the same. I attempted to turn off the NAT functions within sipX (based on
an archived post you made) but it appears to break DTMF on the
I have been all over the wiki. Pieced a bunch of stuff together, none seem to
fit my scenario just right.
For example, I am delivering calls from bandwidth.com to sipXbridge on port
5080. sipX is sitting with a private ip behind a sip aware firewall. My phones
can communicate with sipX without
Anyone willing to let me pick their brain so I can get my setup working 100%?
Thanks
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Hey Tony, I'm curious when Internet calling should/should not be enabled
Can you shed some light on this?
Thanks
-Original Message-
From: Tony Graziano
Date: Wed, 6 Jan 2010 10:08:04
To:
Cc:
Subject: Re: [sipx-users] No audio problem in both sides doing outgoing
calls usi
way audio, unable to pickup a
ringing call (on a Cisco).
-Original Message-
From: Dale Worley [mailto:dwor...@avaya.com]
Sent: Tuesday, January 05, 2010 8:27 AM
To: Nathan Nieblas
Cc: Picher, Michael; Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] multi site deployme
Confirmed 8.2.2SR4 resolves the 406 issue. 3491672850*# is my friend.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathan
Nieblas
Sent: Tuesday, January 05, 2010 2:01 AM
To: Todd Hodgen; sipx-users@list.sipfoundry.org
Subject: Re
p.com]
Sent: Tuesday, January 05, 2010 1:54 AM
To: Todd Hodgen; Nathan Nieblas; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] multi site deployment
"My guess is that you do not have a single problem here, but several
small ones."
Not the least of which could be running si
...@verizon.net]
Sent: Tuesday, January 05, 2010 12:01 AM
To: Nathan Nieblas; sipx-users@list.sipfoundry.org
Subject: RE: [sipx-users] multi site deployment
Nathan,
The best thing to do right now is to get a complete trace of these
failed calls. Clear your logs, regenerate the failed calls
ld not go past as well? I
have the phones on 8.5.2 right now. This page is obviously a little
outdated:
http://sipx-wiki.calivia.com/index.php/HowTo_configure_Cisco_SIP_phone_w
ith_sipX
Thanks
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Monday, January 04, 2010 9:07 PM
To: N
on is that the firewall is essentially handling the SIP
protocol for translation sanity (since it's being NAT'd) and security.
-Original Message-
From: Josh Patten [mailto:jpat...@co.brazos.tx.us]
Sent: Monday, January 04, 2010 8:21 PM
To: Nathan Nieblas
Cc: sipx-users@list.sipfoun
ocal sipXbridge
and I have SIP inspection also being done over that port with the ASA. I
have nothing special configured for inspection.
What could I be doing wrong here?
Nathan Nieblas
SACA Technologies, Inc.
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Are you using the combined or split files? Are the new files world readable in
the directory?
--Original Message--
From: dan
Sender: sipx-users-boun...@list.sipfoundry.org
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] Polycom firmware upgrade
Sent: Jan 3, 2010 9:53 AM
I'm try
e now.
Thanks for the help.
Nathan Nieblas
SACA Technologies, Inc.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Saturday, January 02, 2010 5:22 PM
To: gabriel
Cc: Nathan Nieblas; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-
pX generated config files and TFTP.
Thanks
Nathan Nieblas
SACA Technologies, Inc.
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