Hi
Ah, sorry, thought I added the data needed...
- No HA system, single install (vmware though, but everything else works fine)
- SipXecs 4.2.1-018932
- Polycom Soundpoint IP 335, v3.2.1 (as delivered)
- I have two aliases set, the FQDN of the SipX server as well as the IP address
of the SipX s
Yes, but it's not very cost effective for small business from what I can
tell. Can't confirm as they aren't real fast at returning calls.
From: Martin Steinmann [mailto:mar...@ezuce.com]
Sent: Monday, August 23, 2010 2:36 PM
To: 'Todd Hodgen'; sipx-users@list.sipfoundry.org
Subject: RE: [sipx
>
> On Mon, Aug 23, 2010 at 5:30 PM, Martin Steinmann
> wrote:
> > There now is an issue on the firmware upgrade to 3.3.0
> > http://track.sipfoundry.org/browse/XX-8833
> > We will not support 3.3.0 in the upcoming fall release,
>
> ezuce may not, but I would support anyone who wants to try to
>
I'll try to login in the next few days and create a how-to.
On Mon, Aug 23, 2010 at 12:33 PM, Ali wrote:
> I see.
>
> Hmm how did you setup the SRV records for subdomains? This something that
> we need too.
>
>
> On Mon, Aug 23, 2010 at 7:21 PM, Tony Graziano <
> tgrazi...@myitdepartment.net> wr
On Mon, Aug 23, 2010 at 7:18 PM, Tony Graziano
wrote:
> Sweet. How difficult is it to edit the vm and sipxconfig templates manually
> to reflect config changes, etc.?
Most of it is updating the line.xml and phone.xml files.
http://wiki.sipfoundry.org/display/xecsdev/Configuration+Model+and+Sett
Sweet. How difficult is it to edit the vm and sipxconfig templates manually
to reflect config changes, etc.?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telep
On Mon, Aug 23, 2010 at 5:30 PM, Martin Steinmann wrote:
> There now is an issue on the firmware upgrade to 3.3.0
> http://track.sipfoundry.org/browse/XX-8833
> We will not support 3.3.0 in the upcoming fall release,
ezuce may not, but I would support anyone who wants to try to
implement it at an
You might want to also look at
NetIQ(http://www.netiq.com/solutions/ucm/default.asp) and Prognosis
(http://voicequality.com/compare)
--martin
From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
Sent: Monday, August 23, 2010 6:08 PM
To: Martin Steinmann
Cc: Todd Hodgen; sipx-users@li
Yes...
***
McIlvin, Don to me
show details 6:29 PM (0 minutes ago)
Polycom Productivity suite is what we have for the station level.
But the SIP RTCP "reports" generated at the station level need to get
collected and aggregated in a database somewhere for subsequent analysis
and reporting activi
Isn't that what Telchemy does?
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen
Sent: Monday, August 23, 2010 4:36 PM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] OT RTCP-XR monitor
Polycom has an
This has been fixed some time ago:
http://track.sipfoundry.org/browse/XX-2384
Is this still (again) a problem?
--martin
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org] On Behalf Of Heather Sanders
> Sent: Monday, Aug
There now is an issue on the firmware upgrade to 3.3.0
http://track.sipfoundry.org/browse/XX-8833
We will not support 3.3.0 in the upcoming fall release, which likely pushes
this into next year. The dynamic updating feature will allow a very elegant
hotelling feature. If you want to help wor
Thanks. tried that.
made juse one too many configuration changes and java pegged cpu at 100%.
this below fixed that. without disrupting calls.
On 8/18/10 9:44 AM, Matthew Kitchin (public/usenet) wrote:
sipxproc --restart ConfigServer
--
Michael Scheidell, CTO
o: 561-999-5000
d: 561-948-2259
so it would be good to test transfers, sending a call to voicemail, music on
hold before you get too involved.
On Mon, Aug 23, 2010 at 4:56 PM, Michael W. Burden wrote:
>
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Yeah, that works, but only temporarily. As soon as we have
a 2nd DID number with this ITSP, we will need the
Polycom has an application suite that creates RTCP-XR voice quality streams
for monitoring in a network management application. Anyone have a low cost,
or open source product they have used for monitoring this traffic reliably?
Looking for something for on customer sites, as well as for running on
In the meantime...
you can add the account number as an alias in the system or on a user and
test other aspects of the calls as a sidestep measure...
On Mon, Aug 23, 2010 at 3:52 PM, Michael W. Burden wrote:
>
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> Or
Is this a HA system?
What version sipx? What version firmware on the polycom phone?
Do you have an IP alias on your sipxsystem (not exactly applicable if a HA
system)? Do you have a user alias of staffan entered in on your account?
On Mon, Aug 23, 2010 at 4:01 PM, Staffan Kerker wrote:
> Hi
>
>
As Far As I Recall
I would assume that if it is a NOTIFY message being sent by the phone, it is
either coming from the proxy or registrar (since it is being sent to the
first registered line on the phone).
On Mon, Aug 23, 2010 at 4:08 PM, McIlvin, Don
wrote:
> Does the "Notify" to Restart come
Does the "Notify" to Restart come from SIP Proxy?
What is "AFAIR"?
-Original Message-
From: Douglas Hubler [mailto:dhub...@ezuce.com]
Sent: Sunday, August 22, 2010 9:08 AM
To: McIlvin, Don
Cc: sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] Job Failures on Restart after projecti
Hi
I've seen this problem before and thought it was GRUU related (see
http://track.sipfoundry.org/browse/XTRN-970) but I now have a similar issue
with another phone.
My Tandberg E20 video terminal fails to establish inbound calls because the ACK
is "stuck" in the SipXproxy. The GRUU parts loo
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PS: Oh, and thanks!!:)
--
Mike Burden
Lynk Systems, Inc
e-mail: mailto:m...@lynk.com[/email][/ema
Hi,
i'm quite new to sipexcs but i'm very interested in support for the 8.x
firmware and to get this resource-list bug fixed.
let me know if i can do some testing work etc...
Otto
Am 17.08.2010 17:24, schrieb Martin Steinmann:
> Rene
>
>
>
> Thanks. I think from all the various report
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I expect that it will. This ITSP has been very willing to
wor
I can't read the trace on my blackberry. I suspected as much.
There is an "unstable" 4.3 version that might allow using the "to" header.
Its "typical" with a lot of providers to not be flexible enough to send the
DID in the INVITE. As Carriers go, it "behooves" them to be able to adhere
to standa
Thanks Martin!
Is it too early to set an expectation of what will be in the fall release or
have a short term roadmap outlined?
I saw a prioritization put on an issue I opened (knowing it was something
fundamentally needed), and I don't expect all of it to be in one release,
but prioritized. It w
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No need to ask them, it was in the sipxtrace I posted.
That i
Wanted to give you the good news that we almost completed the reorganization
of the SIPfoundry tracker and are now ready to drive development again out
of this tracker. You might have noticed quite a few issues being merged in
over the last few days and we did quite a bit of cleanup to get ready.
This is what is in sipxconfig.log at the time in question.. 10:53am
8/20/2010
"2010-08-20T14:53:46.057000Z":4257:JAVA:WARNING::pool-1-thread-1::sipXconfig:"using default tls security
policy"
"2010-08-20T14:53:46.363000Z":4258:JAVA:WARNING::pool-1-thread-1::SipxService:"Returning fi
Your itsp might be sending the did in the to header instead.
Ask them what they send in the invite and what they send in the "to".
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control
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1. 4.0.4-017289 (Yes, I know we're overdue for an upgrade.
T
On Mon, Aug 23, 2010 at 12:20 PM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
> Is there a jira on this?
>
>
http://track.sipfoundry.org/browse/XX-8484
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List
1. What version sipx?
2. Does sipx and the registrars agree that sipx has registered with them?
3. Who is the ITSP ( 2nd one)?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Syste
Is there a jira on this?
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:
http://www.myitde
(Yay!)
It always helps to know what the ITSP expects to see for those settings!
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.
Well - I found the problem. It wasn't a pfSense issue at all - it was the
fact that I had enabled SIP Keepalive timer's on the ITSP advanced settings.
Every 20 seconds it was sending an empty ISP packet that would cause the
'SINGLE:NO_TRAFFIC' state and that would lock me out of inbound and outboun
I haven't seen this mentioned yet, but I've been dealing with it recently so
I thought I'd pass it along.
Music on hold is broken by default for the VVX-1500 for the 4.2.x releases,
but the fix is fairly simple. All you have to do is add sip: to the MOH uri
under line settings. Doing so will als
Good afternoon,
We have two ITSPs, set up as Gateways in our sipXecs server.
The two Gateways are configured identically, except for the
IP Address, username, and password.
We have six DID Numbers. I have users (DID01, DID02, DID03,
..., DID06) configured for each DID Number, with the actual
ph
Do you mean their advanced functions for export and import records or you
mean another way to do that?
When i try to import NAPTR records in godaddy panel, then i get blank page
in script execution. I suppose this happens because their scripts and dns
systems does not support and recognize these r
We are hosting our own DNS - the ISPs/hosting companies we've utilized in the
past were very difficult to express what we wanted/needed. I think off-loading
DNS hosting to someone that actually knows sipXecs and SRV records would be a
smart move.
Would be nice to see this as an 'official' ezuc
We could do that here. It becomes burdensome for many companies to move dns.
However, when PUSH comes to SHOVE, not every hosting company has the right
parts.
There are a LOT of hosting companies that DO NOT support SRV.
Tony Graziano, Manager
Telephone: 434.984.8430
F
Is anyone here willing to host/manage external DNS for other sipXecs users? It
seems like it would greatly simplify the process for those wishing to fully
implement sipXecs if the person managing the DNS servers knows what is going
on...I imagine people would be willing to offset any expenses i
I see.
Hmm how did you setup the SRV records for subdomains? This something that we
need too.
On Mon, Aug 23, 2010 at 7:21 PM, Tony Graziano wrote:
> Though I did finally determine how to create and use srv records for a
> subdomain at godaddy (would be needed for openfire chatroom/conferencing
Though I did finally determine how to create and use srv records for a
subdomain at godaddy (would be needed for openfire chatroom/conferencing
services).
After 12 email and 6 online chats with godaddy (who can't their heads around
the concept I am not doing web site or web site redirecting), I ma
I host lots of dns at hostgator. They support every type of record.
Naptr is optional, and helps when deploying tls, as well as specifiying a
protocol preference to the UA. I like using them with Polycom remotes when
deployed through an sbc, because it helps us traverse remote firewalls and
switch
Yes, but the park is being proposed to be replaced using valet parking...
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: 434.984.842
On Mon, Aug 23, 2010 at 10:55 AM, Tony Graziano <
tgrazi...@myitdepartment.net> wrote:
>
>
>> I'm very interested in the some of the new features like dynamic updates
>> w/o rebooting the phones.
>>
>> hopefully!
>
> i also think distinctive ring could help with parked calls, but once the
> park s
On 8/23/10 12:01 PM, Ali wrote:
Hi,
we would like to move our dns records to godaddy dns servers, but they
do not support NAPTR records. Is it possible to use sipxecs without
NAPTR records and how important is this NAPTR record?
yes
they are optional unless you need them.
Older MS windows ser
Hi,
we would like to move our dns records to godaddy dns servers, but they do
not support NAPTR records. Is it possible to use sipxecs without NAPTR
records and how important is this NAPTR record?
Or if we need such record, is there a way to use additional SRV or TXT
records as workaround for DNS
On Mon, Aug 23, 2010 at 11:11 AM, Jim Canfield wrote:
> I've been reading up on the 3.3.0 firmware changes and it brings up several
> questions:
>
> What version(s) of SipXecs will support the new configs?
>
> i am under the impression it will be post 4.3 (5.0).
> Is the current development vers
Hey Jan,
can we chat about that on monday ? As in combining that afford ? We translated
the cd installer and some other stuff is in the works and duplicated work seems
not very effective to me ;)
cheers
Michal
Am 21.08.2010 um 15:11 schrieb Martin Steinmann:
> Jan
>
> What branch of the code
I've been reading up on the 3.3.0 firmware changes and it brings up several
questions:
What version(s) of SipXecs will support the new configs?
Is the current development version being tested against the new configs?
Is there a migration plan?
I noticed there is also a configuration file migrat
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In the AudioCodes Gateway, change the following setting to
FALSE:
Disconnect o
rpm install from the Fedora repo originally noted.
SELinux is disabled.
I am referring to the real root user on the server, not the superadmin
user in sipX. This occurs after running sipxecs-setup as root.
I am going to do another install today via CentOS or RHEL repo and will
report what happen
I think MOH needs to be on by default in sipxbridge. Some ITSP's have a MOH
setting option. On trunks this should be "off" at the provider.
Is MOH "configured" (moh uri) in the polycom? It should not matter as a
trunk call if MOH is enabled at sipxbridge.
I think if MOH is off at sipxbridge, the
When an internal caller (Snom m3, same for other phones) is put on hold
by callee (Polycom IP 650, 3.2.2) the call is _not_ dropped. RTCP
reports are being sent every 5 seconds. Music on Hold is turned off.
Calls routed through ITSP - Sipxecs - Polycom are dropped after 1 minute
if put "on hold
I did a password reset on a 4.0.4 system (via shell script).
After logging into sipxconfig (was not presented with a "set PIN page),
resetting the superadmin password failed with the following errors...
"2010-08-23T09:28:19.613000Z":755:JAVA:WARNING:sipx.sipdomain.com:P1-16::RequestExcept
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