Any comments ?
_
From: jun,wen [mailto:jun@msn.com]
Sent: Wednesday, April 28, 2010 1:15 PM
To: 'sipx-users@list.sipfoundry.org'
Subject: sipXopenfire service failed
Hi, I am running sipX 4.2.0, build 18724. The sipXopenfire cannot be started
which gives me the following standard
Bingo! You have voicemail.
On Wed, Apr 28, 2010 at 9:50 PM, Nathaniel Watkins
nwatk...@garrettcounty.org wrote:
Anyone know if there is a service to test inbound sip uri dialing? I'm
assuming I still have some bugs to work out on my end. Or would anyone be
willing to try it (off-list)?
5. Which value is good to be set as Refresh time on the
phone? I know that default hardcoded minimum time is 300.
Does it make sence to put a lower value or a high value?
A hardcoded minimum of 300 is not enough. Phone is expected to be able to go
lower. SipXecs will use refresh
192.168.42.27#49299:
query: _sip._tls.devnull.net.nz IN SRV +
If there is indeed a problem here let me know what else I can provide to
try and track it down.
Thanks
Mike
sipxopenfire.log.20100429.bz2
Description: application/bzip
___
sipx-users mailing
I'm sorry, I'm actually running 4.2.0, I was just saying that since 3.10.2 that
I've had to use that workaround to get everything working. I was hoping that I
wouldn't in 4.2.0, but it's the same thing.
I've tried in every sipxecs version and polycom firmware to allow
auto-configuration and
Internal or external calls (or both)?
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:
Both, any call that's handled by the Polycom phones in question can't transfer
them to another polycom or to voicemail.
-Original Message-
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, April 29, 2010 8:18 AM
To: Lara Johnson; thod...@verizon.net;
I did not manage to get it working today, the third server got
initialized, but I could not restart any services.
Tomorrow is queens day over here (the Netherlands) and next week I am off,
but I will give it a try, just wait.
Could you give some more details about the Bria problem
Is the
The described functionality to be able to change the Caller ID to the
DID during external forwarding of a call is definitely on the wish list.
I investigated this myself, on this list, about a year ago. At that
time without success.
The main driver is to be able to identify the call a
Am I to understand that calls forwarded out a gateway ignore the user
defined caller ID settings and just forward the original number out the
gateway even though explicit caller ID settings are defined?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On
5. Which value is good to be set as Refresh time on the phone? I
know that default hardcoded minimum time is 300.
Does it make sence to put a lower value or a high value?
A hardcoded minimum of 300 is not enough. Phone is expected to be
able to go lower. SipXecs will use
Maybe, after finishing my testing, since its not that many netblocks, it
might be time to just firewall all incoming connections from amazon's
cloud until they make an announcement that they actually take network
security seriously.
Below is just an example of 'hits' on the ET (snort) sigs
I did not manage to get it working today, the third server
got initialized, but I could not restart any services.
Tomorrow is queens day over here (the Netherlands) and next
week I am off, but I will give it a try, just wait.
Keep us posted on your progress...
Could you give some more
From: jun,wen [mailto:jun@msn.com]
Sent: Wednesday, April 28, 2010 1:15 PM
To: 'sipx-users@list.sipfoundry.org'
Subject: sipXopenfire service failed
Hi, I am running sipX 4.2.0, build 18724. The sipXopenfire
cannot be started which gives me the following standard error -
How did
I am seeing what I think is some odd behavior with Openfire,
but then it may just be me.
I have installed a sipxecs 4.2 test system on Centos 5.4
x86_64 using the YUM repos. sipxecs-setup has been run to do
the base configuration.
DNS Advisor reports no issues and the only
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of JOLY, ROBERT (ROBERT)
[rj...@avaya.com]
I suggest you set your phone up so that it will propose an Expires of 3600
seconds when it registers. The sipXecs
I've attached the siptrace file. It's frame #3 (the 4th frame) that's
sent multiple times (200ms apart). I'm sure sipXecs is behaving as
it's supposed to, just not sure if I've misconfigured something.
If this looks normal I'll play around with a VM of trixbox and see if
I get the same problems.
Hi all,
sorry for a stupid question, but..
i'm trying to setup xmpp federation with google.
I've tried to find some info on teh wiki, but found only one page:
http://wiki.sipfoundry.org/display/xecsuserV4r2/Instant+Messaging+and+Presence
I connected pidgin to sipx xmpp. I can successfully see
Does it pass pre flight tests, and diagnostics tests with no issues?
-Original Message-
From: Lara Johnson [mailto:lcr...@ciscorp.biz]
Sent: Thursday, April 29, 2010 5:19 AM
To: Tony Graziano; thod...@verizon.net; mpic...@cmctechgroup.com;
sipx-users@list.sipfoundry.org
Subject: RE:
Today is your lucky day...
; SRV record for XMPP SERVER TCP voice.mydomain.com
; priority: 1 weight: 0 port: 5269 server: sipx.voice.mydomain.com
;
_xmpp-server._tcp.mydomain.com. IN SRV 1 0 5269 sipx.mydomain.com.
; SRV record for XMPP CLIENT TCP voice.mydomain.com
;
We upgraded to 4.2 and since then, our Polycom phones continually lose their
registration with the server.
The re-registration interval is 60 seconds and the registration period is
3600 seconds. It almost appears that the system waits for each phone to
expire before resetting the expiration
are the phones local or remote?
if they are remote and the nat refresh is empty, set it for 30.
On Thu, Apr 29, 2010 at 12:27 PM, Ken Fulmer
kenful...@icstechnologysolutions.com wrote:
We upgraded to 4.2 and since then, our Polycom phones continually lose
their registration with the server.
I already know one of these answers, but they seem related and I wanted to
make sure my logic was correct. I don;t normally cross post, but wanted a
broader audience if I can get it because it will come up be be a question
for a lot of people upgrading to 4.2, and helps with proper planning.
I
Tony,
thanks for the clarification.
One more question: i have sipx in the private network and private dns zone.
I can set up those xmpp srv records, but they will point to externally
visible name, which points to external ip address, which is nated into
internal sipx address, which is named in
They are on the same LAN segment.
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Thursday, April 29, 2010 11:33 AM
To: Ken Fulmer
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Polycom phones losing registration
are the phones local or remote?
if
That's a very good question. MY logic tells me NO (wont work) if the DNS
ZONE is a different name than the internal one. I may be wrong, but look at
my question to user/developers on EXACTLY what my logic is and asking for
some confirmation (or rejection) of my logic.
If the ZONE name (SIPDOMAIN)
Tony,
I can register from outside to external name, which is configured in the
sipx as domain alias. I works, but mwi and caller id do not work for such
remote users.
And i know this can be solved by so called split dns.
This is regarding sip... but i dont know regarding xmpp...
Any way i
The most widely used XMPP federation method is known as server
dialback, which uses public DNS. There is also a certificate-based
method, but it is not widely used.
I haven't played with sipXecs XMPP integration, so I'm not sure about
your other questions. What do you mean by register remotely?
I already know one of these answers, but they seem related
and I wanted to make sure my logic was correct. I don;t
normally cross post, but wanted a broader audience if I can
get it because it will come up be be a question for a lot of
people upgrading to 4.2, and helps with proper
Can someone explain to me how to move this from Perosnal Space to the wiki
for general availability?
http://wiki.sipfoundry.org/x/D4Fd
On Thu, Apr 29, 2010 at 1:51 PM, JOLY, ROBERT (ROBERT) rj...@avaya.comwrote:
I already know one of these answers, but they seem related
and I wanted to make
In preparing the upgrade documentation for 4.0.4 - 4.2 I presented my
upgrade plan and upgrade testing results to my bosses and they raised a
question I could not answer: How do you fully test the 4.2 upgrade by
mirroring your current setup and then running the upgrade, then test
I use a vmware instance to test with. Install a second server with the same
version, different ip and hostname, and use a different domain or subdomain.
Create a couple of phones.users, gateway, dialplan... Now test it.
Now if this is a virtual server that has added benefits...
Do the upgrade on
Already did those things you recommended with the First Test System.
That doesn't mirror my current production environment. It only makes a
separate environment that I can play around in.
What I need to be able to do is have an exact replica of my 4.0.4
production environment installed and
What firmware version?
-Original Message-
From: Ken Fulmer [kenful...@icstechnologysolutions.com]
Date: 04/29/2010 12:27 PM
To: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] 4.2 Polycom phones losing registration
We upgraded to 4.2 and since then, our Polycom phones
Come on Josh - easy peasy and willy nilly - what more could you possibly want ;)
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Thursday, April 29, 2010 3:38 PM
To: Josh Patten
Cc: sipx-users@list.sipfoundry.org
On Thu, 2010-04-29 at 13:06 -0400, Tony Graziano wrote:
That's a very good question. MY logic tells me NO (wont work) if the
DNS ZONE is a different name than the internal one. I may be wrong,
but look at my question to user/developers on EXACTLY what my logic is
and asking for some
Well...
There are more elaborate ways to do this... On a SAN? Copy the running
config. Connect to it with another machine and upgrade it, the swap the
connections.
Analogous to swapping hard drives back and forth with the same version and
different data. In this way the registrations are there
On Thu, 2010-04-29 at 14:51 -0500, Josh Patten wrote:
Already did those things you recommended with the First Test System.
That doesn't mirror my current production environment. It only makes
a separate environment that I can play around in.
What I need to be able to do is have an exact
On Thu, 2010-04-29 at 14:33 -0400, Tony Graziano wrote:
Can someone explain to me how to move this from Perosnal Space to the
wiki for general availability?
http://wiki.sipfoundry.org/x/D4Fd
Under the Tools menu in the top right of the page, there should be a
'Move' entry that will do it.
That answers my question in that it's probably not worth risking corrupt
configuration on the redundant server. The scenario you describe, Scott,
would put everything into the upgraded environment all at once which
kinda defeats the purpose of testing :-)
Unless I can think of a way to do this
Hey After Upgrade from 4.0.4 i am getting some errors in my provisioning
service.
Message from sipXecs
Alarm: SPX00032
Reported on: plsipx01.franklinamerican.com
Reported at: 2010-04-29T20:10:21.746582Z
Severity: WARNING
Alarm Text: The configuration data for process 'sipXprovision' ()
A small note we patched the earlier version to support the polycom 450
this may be the reason the rpm upgrade did not complete.
post rpm upgrade the following files were not implemented
locate *.rpmnew
/etc/issue.rpmnew
/etc/localtime.rpmnew
/etc/pam.d/system-auth.rpmnew
Seems like you are answering your own question.
Also, should not be running 3.2.2 on either 4.0.4 or 4.2.0. 3.1.3 on either
or 3.2.3 on 4.2.0.
So now you need to compare those files and replace the ones that are
different. The last three probably are your issue.
Tony
You did verify you don't have any 4.0.4 rpms still installed didn't you?
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:
Brings me another thought... On upgrade the phones are pushed a new
profile...
In an HA environment or not.
So the phones reboot and re-register to their primary.
Right? Things that make you go hmmm.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax:
On 4/29/10 12:27 PM, Ken Fulmer wrote:
We upgraded to 4.2 and since then, our Polycom phones continually lose
their registration with the server.
The re-registration interval is 60 seconds and the registration period
is 3600 seconds. It almost appears that the system waits for each
phone
http://wiki.sipfoundry.org/display/xecsuserV4r2/Upgrade+or+Install+Planning+for+4.2+and+XMPP
On Thu, Apr 29, 2010 at 4:20 PM, Scott Lawrence xmlsc...@gmail.com wrote:
On Thu, 2010-04-29 at 14:33 -0400, Tony Graziano wrote:
Can someone explain to me how to move this from Perosnal Space to the
Now that I reread your post Scott, it make much more sense, and this is
what we ultimately decided to do. One question that may or may not be
worth asking, is there an easy way to copy registrations over so that
the mock upgrade system will immediately have all registrations so
actually
And if you do an upgrade and the server comes back online, would it matter
since it will oush profiles and send reboot messages to the phones?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security
Perhaps what I'll do is reboot all the phones and while they're
rebooting I'll swap the servers out. sounds like a plan, ehh?
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/29/2010 6:26 PM, Tony Graziano wrote:
And if you do an upgrade and the server
Makes me ask myself if a HA upgrade should have a special bootup
script/procedure to prevent the profile push but allow for a certificate
handshake/sunc to the other server before assuming any interactive roles
with users/gateway.
Tony Graziano, Manager
Telephone:
Proper planning prevents *iss poor performance
From: sipx-dev-boun...@list.sipfoundry.org
[mailto:sipx-dev-boun...@list.sipfoundry.org] On Behalf Of Nikolay
Kondratyev
Sent: Thursday, April 29, 2010 1:06 PM
To: 'Tony Graziano'; 'Sipx-users list'; 'Sipx-dev list'
Subject: Re: [sipX-dev]
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Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 45840
Message-ID: b310.4bda2...@forum.sipfoundry.org
PSTN and FXS extensions work from C1760V gateway to internal
extensions 1000-1999. When a FXS
I have the snapshot but was unable to sign up for an account at
track.sipfoundry.org - the site returned a system error I set everything to
debug for the snapshot, the file is 1.4mb - I doubt that is appropriate to
email to the list. Let me know how you'd like me to proceed.
Thanks
Please mail me the snapshot.
Thanks
Ranga
On Thu, Apr 29, 2010 at 8:17 PM, Nathan Nieblas
nathan.nieb...@sacatech.com wrote:
I have the snapshot but was unable to sign up for an account at
track.sipfoundry.org - the site returned a system error I set everything
to debug for the
No i tried to remove the older rpm files and replace them first, I am posting
to the list because sadly that did not help those service start.
The firmware and app versions don't affect the services either. I have sent
profiles on servers after all my changes
any idea how i can get these
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Organization: SipXecs Forum
In-Reply-To: b310.4bda2...@forum.sipfoundry.org
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 45850
Message-ID: b31a.4bda4...@forum.sipfoundry.org
To clarify there still is a problem with the
depending on the service that is failing...
post
sipxproc --state
and then the output of:
rpm -qa | grep 'sipx*'
so we know what's failing and what versions of which packages are installed
(or not) along with what is running and what is not.
Also, I assume you can get into sipxconfig and
Next thing you know you're going to gather monthly for tea with matching
black and blue hats (as opposed to the red hats the old ladies wear).
On Fri, Apr 30, 2010 at 12:23 AM, J Coatline jcoatl...@gmail.com wrote:
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Well, how about providing some information?
What the heck is a C1760V gateway? Who makes it? The FXS or SIP UA's don't
dial the gateway. They should DIAL the proxy. The proxy should then send the
call out to the gateway.
WHO the provider is DOES matter.
Stop being mysterious and maybe you can
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