Scott, Dale,
Thanks a lot for the help.
Nikolay.
-Original Message-
From: Scott Lawrence [mailto:xmlsc...@gmail.com]
Sent: Wednesday, April 21, 2010 10:13 PM
To: Nikolay Kondratyev
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] sipxbridge and late media
On Wed,
Hi,
After upgrading to 4.2 the Bria Pro 2.5 softphones we are using are
getting a complete extension-directory.xml file that contains all users,
including superadmin.
There was a phonebook defined on the system, but that did not include the
administrators group, so superadmin should not be
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Hi All, I am still trying to find answers for
O
I have 7940's and 60's running P0S3-08-12-00 (latest)
against SipX 4.0.3 and 4.2 (test deployment), in both
deployments, the MWI works fine.
You only have to enter the external MWI number, where the
phone is on a remote network or non sipx network.
mine are all local, and would not
Hi all,
A bit earlier in the list i found a command for manual log rotating.
I tried
logrotate -f /etc/logrotate.d/sipxchange
And all logs were rotated except sipxcallresolver.log.
And i noticed that it was never rotated... and is 200M long now.
I believe it is due to the following lines in
This does not work here.
None of my Gateways have ITSP Account settings and I can't find any way
to activate them.
Back to 4.0.4 I guess?
-- Robert
On 4/21/2010 7:37 PM, mkitchin.pub...@gmail.com wrote:
I deleted and re-added my gateway and it showed up. Not the whole server
config, just
Regarding point 1:
It may happen that you did not setup who is getting mail for root (alias
for root) in /etc/aliases file.
Here is mine:
# Person who should get root's mail
#root: marc
root:k...@nstel.ru
Don' forget to issue newaliases command after editing /etc/aliases file.
Some extra info:
The extension-directory.xml file can be seen under DevicesPhones,
then selecting an existing phone, non-Bria phones don't have all the
contacts in the file.
Also, in the mean time I found a file that seems to be responsible for
creating the directory.xml
in
As far as i understand, this is intended behaviour.
Proxy Authentication Required message is used to verify that calling user
has ehough permissions to make particular call.
And both polycom and audiocodes are capable of doing digest authorization.
A dialplan rule poses a requirement for the user
Tony,
Is there something other than opening the Add new gateway... drop-down
menu and then selecting SIP trunk that is different for adding a SIP
trunk with an ITSP account?
I've already deleted and re-added the SIP trunk and there's no ITSP
account settings. I am not getting any means to add
Hi Robert!
Had the same issue with 4.2. I installed from iso.
It was strange, no account settings and i could not add more gateways without
internal errors.
I had to stop sipx/postgresql and do a:
/usr/bin/sipxconfig.sh --database drop create
After that it worked.
Account settings link showed.
Correct.
You need to use an SBC if you are sending to a siptrunk. The provider
templates will not appear otherwise.
If you are not using sipxbridge, it could be a problem.
One way to overcome it (Disclaimer: Not sure of how this will work), is to
create an unmanaged SBC. Then create your
Tony,
Okay -- so you're right, but here's the issue and apparently a bug...
When I restored my configuration, it did not bring over the SBC Route.
Nor, once a SIP trunk is created, can I change the SBC Route. That
option does not appear.
I have quite a few trunks configured, so re-adding them
What was the SBC device? An unmanaged one? If so, recreate it manually and
see if you can assign the gateway to it and see if the settings are still
there.
On Thu, Apr 22, 2010 at 8:22 AM, Robert B d...@spudland.com wrote:
Tony,
Okay -- so you're right, but here's the issue and apparently a
http://www.trixbox.org/forums/trixbox-forums/open-discussion/sipxecs-version-42-released
: ]
This message and any files transmitted with it are intended only for the
individual(s) or entity named. If you are not the intended individual(s) or
entity named you are hereby notified that any
I am curious as to whether the following should work in 4.2 under the
following:
sipxecs serverfirewall---|Internet|---firewall---remote branch,
users: thing1 and thing2
Should thing1 and thing2 be able to call each other? Setting the media relay
to aggressive did not seem to help.
Has
that is how I tried to configure one of my clients before. can get the phones
to ring, but no audio. I tried different firewalls, pfsense and endian. I
couldn't get any to work. Although a bit different is my other firewall is
actually a linksys router.
sipxecs server
I am curious as to whether the following should work in 4.2
under the following:
sipxecs
serverfirewall---|Internet|---firewall---remote
branch, users: thing1 and thing2
This is definitely a scenario that is working. Support for this kind of
deployment was one of the major
On Thu, 2010-04-22 at 15:13 +0400, Nikolay Kondratyev wrote:
Hi all,
A bit earlier in the list i found a command for manual log rotating.
I tried
logrotate -f /etc/logrotate.d/sipxchange
And all logs were rotated except sipxcallresolver.log.
And i noticed that it was never rotated... and
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When we pick up an incoming call, we hear two beeps.
They don't seem to cause any problem if
On Thu, 2010-04-22 at 07:22 -0500, Robert B wrote:
Tony,
Okay -- so you're right, but here's the issue and apparently a bug...
When I restored my configuration, it did not bring over the SBC Route.
Nor, once a SIP trunk is created, can I change the SBC Route. That
option does not
Have you reviewed your polycom config carefully?
Make sure you don't have things like call waiting tone or message
waiting tone turned on.
On Thu, Apr 22, 2010 at 9:23 AM, Michael W. Burden m...@lynk.com wrote:
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So apparently I'm a sipX evangelist.How ironic
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/22/2010 7:35 AM, Nathaniel Watkins wrote:
http://www.trixbox.org/forums/trixbox-forums/open-discussion/sipxecs-version-42-released
: ]
This message
On Thu, Apr 22, 2010 at 9:09 AM, JOLY, ROBERT (ROBERT) rj...@avaya.comwrote:
I am curious as to whether the following should work in 4.2
under the following:
sipxecs
serverfirewall---|Internet|---firewall---remote
branch, users: thing1 and thing2
This is definitely a scenario
Are the phones in question monitoring any lines with BLF? I get those
beeps sometimes using firmware 3.1.3c. I haven't noticed them in 3.2.3
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/22/2010 8:23 AM, Michael W.Burden wrote:
Content-Type:
More info:
Apparently in 4.2 WEBDAV is enabled by default for Bria Pro
(Resource list method under System in the Bria Pro phone template)
and sipx is configured to be a WEBDAV server.
Probably via the script file below an extension-directory.xml file is
created automatically with ALL users in
We have a customer that's using the 4.0.4 version. When we enable call
forwarding for the user, nothing happens. I perform a capture and see the
conversation dies between processes on the system.
I also notice on the Polycom phone / line / diversion screen, there are
fields for call forward
Hi All, I am still trying to find answers for these issues.
Update:
On point 3 - I came across a setting on CUCM which enables
the sending of RDNIS across the SIP trunk. The impact of this
now is when a call is diverted to the voicemail pilot number
the voicemail system provides
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer
[kenful...@icstechnologysolutions.com]
We have a customer that’s using the 4.0.4 version. When we enable call
forwarding for the user, nothing
http://news.softpedia.com/news/Red-Hat-Enterprise-Linux-6-Beta-Is-Here-140348.shtml
Finally. RHEL/CentOS 5 was getting a bit dated. Good to see 6 is around
the corner.
--
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
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results of yum and current repos file below
I modified the repos file, changing 5.2 to 5 as
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Hi Dave,
I am still missing something and this new feature doesn't have any
documentation that I can find.
The file does exist and I've edited it to suit but the system continues to use
the default notifications.
Is there something else I need to edit in order to tell the system to use the
new
On Thu, 2010-04-22 at 10:35 -0400, Gary wrote:
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results of yum and current repos file
How the heck do you guys figure these things out without documentation unless
you are part of the dev team as well?
I've looked at every file in /etc/sipxpbx and see plenty of email and
notification options but none which seem to affect which notice I can send out,
meaning using the file I
Ken Fulmer [kenful...@icstechnologysolutions.com]:
We have a customer that’s using the 4.0.4 version. When we enable call
forwarding for the user, nothing happens. I perform a capture and see
the conversation dies between processes on the system.
Worley, Dale R (Dale) wrote:
You'll have to
I can confirm that I had the same issue when I did my update to 4.2
initially. However, I wasn't sure I had a clean system that I was updating,
so I did not report it.
Essentially, everything came up and was working except my trunks. I went in
and added the branch setting - in all areas. I
Did you look under User profiles?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
m...@grounded.net
Sent: Thursday, April 22, 2010 8:36 AM
To: sipx-users
Subject: Re: [sipx-users] Custom Notifications
I am still
If this is happening when updating, it makes sense (to me) to grep your
packages and compare that with the repo list to make sure everything:
1. Got installed (there are new packages that are installing and not
updating)
2. Got updated
If something didn't get installed or updated it should be
Hi Dave,
Thanks for your comments, the way you describe seems to be
the method most voicemail platforms work, but either I have
set something up wrong or there may be a bug in the platform.
So, I have the following:
SipXecs - 4.2.0
CUCM 6.1.3
The CUCM has a SIP trunk
Do you mean in the Users section of the GUI, in Unified Messaging, the Full,
Medium or Brief options?
I see that and the Full, Medium and Brief options in the
EmailFormats.Properties file as well.
However, placing that file in /etc/sipxpbx alone doesn't seem to call it up.
I see nothing with
I never restarted any services after putting the file in there, is that a
requirement for sipx to find it?
On Thu, 22 Apr 2010 13:40:42 -0500, m...@grounded.net wrote:
Do you mean in the Users section of the GUI, in Unified Messaging, the
Full, Medium or Brief options?
I see that and the
Place the file in /etc/sipxpbx/sipxivr
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
On 4/22/2010 1:40 PM, m...@grounded.net wrote:
Do you mean in the Users section of the GUI, in Unified Messaging, the Full,
Medium or Brief options?
I see that and the
Cisco follows IETF standards for SIP, what does Polycom and sipX follow?
I'm not convinced it's Cisco's issue, the phone itself is far superior
to any Polycom I've ever used. The one thing I did notice the last time
I looked at a trace was some attempt at TLS negotiation for the call (if
that even
Place the file in /etc/sipxpbx/sipxivr Josh Patten Assistant Network
Yup, I have, and permissions/ownership are fine too.
sipXivr will look for {prefix}/etc/sipxpbx/sipxivr/EmailFormats.properties
(including all EmailFormats_{locale}.properties variants) first, and if
not found will use the
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OK, I'm trying to set up
OK, I'm trying to set up Pidgin.
I have a sandbox DMZ for testing. On this DMZ are the
sipXecs server (4.2), A Windows PC (for admin and to run
Pidgin), and a phone (Polycom IP550).
I did a fresh install of 4.2, told sipXecs to be the DNS and
DHCP server, and imported the user and
On Thu, Apr 22, 2010 at 4:09 PM, Michael W. Burden m...@lynk.com wrote:
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Robert For the Win!
I didn't
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OK, now I'm in the same situation as the
Um. The buddy needs to get your IM and authorize you. It stays
unauthorized until then. If that buddy is out the web, you need to make
sure your SRV records are publicly available, and that ports 5222 and 5269
are natted to your sipx server.
On Thu, Apr 22, 2010 at 4:57 PM, Michael W. Burden
When I call from user1 to user2 and they both have IM accounts logged in and
enabled, I can see user1 shows on the phone to user two. I do not see any
changes for user2.
Is IM presence tied to phone events or simply the presence server in sipx?
--
==
Tony Graziano, Manager
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Actually, the
On Thu, 2010-04-22 at 12:25 -0700, Nathan Nieblas wrote:
Cisco follows IETF standards for SIP
With all due respect to Cisco, that statement doesn't mean very much.
There are lots of documents that make up 'standards for SIP', and many
ways in which implementations can be incompatible while
if they are on the same server it should not be necessary, just the IM id
(12) . openfire supports searching of users too.
On Thu, Apr 22, 2010 at 5:30 PM, Michael W. Burden m...@lynk.com wrote:
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When I didn't add
There was an article to update your repos.
Kindly check this
Linkhttp://sipxecs.blogspot.com/2009/12/how-to-fix-yum-repos-file-for-sipxecs.html
Hope this helps.
Rhon
On Fri, Apr 23, 2010 at 12:12 AM, Scott Lawrence xmlsc...@gmail.com wrote:
On Thu, 2010-04-22 at 10:35 -0400, Gary wrote:
Hi Tony,
Just a few question about your implementation.
Seeing your scenario means you only have 1 sipxecs server? Is the IPSEC
tunnel created via pfsense or from your router?
Best regards,
Rhon
On Thu, Apr 22, 2010 at 9:18 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
On Thu,
Hi Scott,
Our problem here is a transition issue (from CME to SipXecs). For some
company like us who can't always afford to replace equipments,
it's not always easy to say just replace your phones with polycoms.
We will greatly appreciate if the development team start looking into
As there is no sipXconfig support for configuring this yet, the default
format for the main email address is Full, and for the alternate email
address is Brief at the moment.
I thought perhaps this actually meant that alternate email was the alternate
notification but it's not.
At a loss,
Eric,
Can you explain how you got this feature working on your Polycoms?
Using sipx 4.2.0 here with Polycom 550's fw 3.2.3 and br 4.2.2.
To test I have two handsets, each configured to see the same line. I
don't see any indication on the station that the line is in use... Or
maybe I have the
Scott,
That's putting it quite diplomatically. Here's how I'd say it...
Cisco infects... They take emerging standards, dump millions into making
their own version of it, wait for everyone else to use an open standard,
do a half-assed implementation of said open standard in their own
product,
+1
Look at SCCP, CDP, Pre-standard PoE, etc. etc. etc.
This is why I avoid proprietary hardware: it never fails to cause
problems later because you're stuck with it and you either shell out
for all new stuff or keep running on the planned obsolescence/upgrade
hamster wheel. Case in point:
Josh,
Totally where I was headed with your examples, I just didn't want to get
too deep into it.
Recently I've been testing Cisco's UCS platform. They hype it all as
open standards because it's x86... But just wait until they start
infesting your environment with their custom vSwitches that
Cisco UCS: Cisco Unbridled Collection Software, I mean Cisco UCS: Cisco
Unified Computing System.
er, uh, I guess it depends on whether you are selling it or buying it.
On Thu, Apr 22, 2010 at 8:15 PM, Robert B d...@spudland.com wrote:
Josh,
Totally where I was headed with your examples, I
http://basecomputer.org/home/index.php
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sipXecs IP PBX --
SPAM.
On Thu, Apr 22, 2010 at 8:52 PM, Richard Alan McAlexander
alanmac1...@gmail.com wrote:
http://basecomputer.org/home/index.php
___
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List Archive:
Anyone else get this spam from a sipx addy?
Alan\ alanmac1...@gmail.com
On Thu, 22 Apr 2010 18:52:05 -0600, Alan\ wrote:
http://basecomputer.org/home/index.php
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Which development team? The one that created the current interoperability
and donated it, or a new team that is interested in making them work better
and will put together the necessary work to get it done? These seem to be
distinctly different teams than the one that is working on the current
Hi,
Is there a way I can configure a user to dial an extension directly from our
remote office?
Here's our ext.
HQ = 200 - 299
RO = 300 - 399
Say user 210 want to directly dial the extension 311 how can I accomplish
this?
I was able to use a Phantom User to direct a PSTN call to a live
The wiki shows you how to configure a site to site dial plan.
You can also program speedials with the sip uri directly into your phone
(I.e., polycom supports it).
Dial 2 and 2 digits, send entire number to gateway (other sipx server) and
vice versa.
Tony Graziano,
http://wiki.sipfoundry.org/display/xecsuserV4r0/sipXecs+to+sipXecs+Calling
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:
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