Install 3.10.2 stable from the ISO, then change the yum repo to the dev version
and yum update it to the dev version.
>>> "voice" <[EMAIL PROTECTED]> 09/14/08 07:09 AM >>>
We are presently testing Grandsteams IP PBX GXE-5028. These device's
firmware
have a switch that turns off registration user
Make sure URL dialing is enabled in the Polycom configuration (sipxconfig). By
default it should be on, but in preference some like it off to hide IP
addresses in regular phone calls.
Phone>Preferences>
URL dialing (Default: unchecked) Controls whether URL/name dialing is
available from a lin
Change your REPO settings...
1. Edit
/etc/yum.repos.d/sipxecs-stable-centos.repo
[sipxecs-stable]
name=SIPfoundry sipXecs pbx - latest stable version
baseurl=http://sipxecs.sipfoundry.org/pub/sipXecs/LatestStable/CentOS/$releasever/$basearch/RPM
gpgcheck=0
gpgkey=https://secure2.pingtel.com/RPM
3.10.2
>>> Ananda Teertha <[EMAIL PROTECTED]> 09/18/08 07:11 AM >>>
I do not see this option(URL dialing ) under Phones> > User
Preferences
.
The scs version is 3.11.6-013456
What is your sipx version.
Thanks,
Anand
Tony Graziano wrote:
> Make sure URL
You need to make sure you activate the dialing plan after making a change. Have
you activated it after adding the second gateway?
>>> "Anand Yogas" <[EMAIL PROTECTED]> 9/22/2008 8:09:06 AM >>>
Hi All,
I am using 10.2 of Sipxces. I am facing a problem regarding Gateway.
We have de
services of Sipxces, but still there is some problem.
On Mon, Sep 22, 2008 at 5:48 PM, Tony Graziano
<[EMAIL PROTECTED]> wrote:
> You need to make sure you activate the dialing plan after making a change.
> Have you activated it after adding the second gateway?
>
>>>&
Should it be assumed someone might be using a gateway or siptrunk and not
sipXbridge (i.e. AudiCodes gateway or a siptrunk via independent appliance)?
Should the solution be centered around sipXbridge or should it be centered
around an intelligible response from a gateway or provider to make the
In trialing firmware version 3.1 with sipx 3.10.2, I noticed some undesired
behavior with BLF. While there were some display oddities, probably due to
mismatched template files, etc., the larger functional and non-cosmetic issue
had to do with BLF.
It was interesting to see the enhanced BLF fro
ng
to properly cure it.
If I can offer any more insight, please let me know.
>>> "Paul Mossman" <[EMAIL PROTECTED]> 09/24/08 04:34PM >>>
Tony Graziano wrote:
> In trialing firmware version 3.1 with sipx 3.10.2, I noticed
> some undesired behavior with
Restart your Call Resolver from the services menu in sipxconfig. This will
typically clean that up.
Sometimes a misconfigured gateway or UA could also cause this.
>>> "Vikas Sharma" <[EMAIL PROTECTED]> 10/03/08 03:26 AM >>>
hi all,
I am using 3.11.5
Diagnostics > call detail records > active ca
>>> "Andy Spitzer" <[EMAIL PROTECTED]> 09/03/08 03:44PM >>>
Woof!
Executive summary:
/etc/sysconfig/clock is set INCORRECTLY by the Fedora GUI
system-config-date program.
It sets the file thusly:
ZONE="America/New York"
When it should be:
ZONE="America/New_York"
(There is an unders
Just one, see below:
>>> Damian Krzeminski <[EMAIL PROTECTED]> 10/16/2008 11:30:16 AM >>>
http://track.sipfoundry.org/browse/XCF-2817
***
Why not have the minor/major version differ only slightly?
"Major version upgrade" - from 4.0 to 4.2
"You are currently running sipXecs 4.0.0, there is a
Is the build on the CD 3.11.7-013745?
If not, there is a build (3.11.7-013745) dated today. If you YUM the
install you have from
http://sipxecs.sipfoundry.org/temp/sipXecs/main/CentOS/5/i386/RPM/
Does it install the Java? I see the Java RPM's are there too. I'm not
sure it will fix anything, and
Can it be assumed that FS will also store the voicemail or will the
voicemail still be stored in the same fashion it is now
(/var/sipxdata/mediaserver/data/mailstore/)?
If not, is this going to be dictated by FS or is this going to be
written into sipx in a different fashion?
_
PROTECTED]> 10/27/08 09:28 AM >>>
Woof!
On Sat, 25 Oct 2008 18:22:47 -0400, Tony Graziano
<[EMAIL PROTECTED]> wrote:
> Can it be assumed that FS will also store the voicemail or will the
> voicemail still be stored in the same fashion it is now
> (/var/sipxdata/med
The intent is for the IMAP4 server to use the UI store itself. So a
delete from IMAP will delete the VM in the user store WITHOUT having to
change the way MWI works.
>>> "Andy Spitzer" <[EMAIL PROTECTED]> 10/27/08 09:55 AM >>>
Woof!
On Mon, 27 Oct 2008 09:45
M >>>
On Mon, 2008-10-27 at 10:27 -0400, Tony Graziano wrote:
> The intent is for the IMAP4 server to use the UI store itself. So a
> delete from IMAP will delete the VM in the user store WITHOUT having
to
> change the way MWI works.
But unless it does that deletion through primiti
Agreed. I found myself on the end of that once when a 3.10 patch was posted
that didn't support FC6 but it applied anyway. In my case the DB had updated,
so I did the backup and wiped the system, and was able to restore.
Should there be a mechanism in the restore to determine the DB schema versi
Perhaps every time a dial plan is activated, the previous one gets archived
with a date/time stamp that it used when it was created. Allowing a "time
machine" to restore dial plans of sorts? The archives could be viewed or
restored?
>>>
It would be good to have some status indication in the "
Is it possible to get Voicemail via SOAP?
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OK, I see SOAP is used for configserver. The reason I ask this is to
determine the nest method to access the voicemail for a true email integration.
Our mail system is fully SOAP enabled, and SOAP is how we integrate our
Blackberry Server into it, which is very slick.
Other than writing a ga
I see there are no open issues for release 3.10.3. Does this mean this is going
to be released? I think a lot of people are waiting for the fix to test moh and
a few other things.
Thanks,
Tony
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L
Would this have anything to do with the 3.10.3-4 versions not showing anywhere
yet?
I am in a perfect position to install 3.10.2 and do an upgrade to 3.104 for
testing over the next several days in a non-production environment (sound like
a "Mother may I?" doesn't it?, and no Scott I am NOT ref
Andrei,
For what it's worth, I have seen this type of behavior when there is an issue
with the phone sharing an IP network with the phone system, and the two have
different gateways, or their VLAN does not directly connect or route to each
other.
I would also verify the config or the server and
I was wondering if anyone knew the timing to release 3.10.4 becoming available.
There doesn't seem to be a repo for 3.10.3 and there is no build version
apparent but all of the issues have been marked closed for a while.
Thanks,
Tony
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I think this would install when installing sipviewer. If there are problems
installing it, you could try doing a manual install of the RPM I think.
java-1.5.0-sun-fonts-1.5.0.10-2jpp.i586.rpm
>>> "Picher, Michael" 12/20/08 6:07 AM >>>
Recently while trying to do a few things for a non-profit under our wing, I
found myself looking for a feature or two that was not in sipxecs.
Notification email length - It would be good to be able to pick a "short" email
message type, that would be "under" 150 characters so it is suitable for
Polycom announced support for their EFK (Enhanced Functionality Key) macros in
firmware 3.1. Polycom in firmware 3.1 also removed EFK from the productivity
suite, so it is available to all phones supporting it without the additional
license. I searched for this in the JIRA and found this closed
Thanks for responding Paul. I see now why it was closed, just not a lot of
detail until you made more clear to me, and I appreciate it.
I am thinking of the efficiency for someone at a receptionists position.
With the assumption the user has a 6xx series phone and sidecars, they would
need Voic
I am looking
at:XTRN-229 http://track.sipfoundry.org/browse/XTRN-229 and XCF-2981
http://track.sipfoundry.org/browse/XCF-2981I have noticed issues in
doing what I call a consultative transfer with Bootrom 4.0-4.1x and firmware
3.0-3.1x. I consider a consultative transfer answering a cal
In looking at XCF-2798, I have some usability questions about pening "feature
codes" into speed dial.
Does speed dial in development versions allow certain user speeddials with the
ability to "anchor" themselves into a certain position on the phone? In other
words, when the RLS server loads any
EFK - Enhanced Functionality Keys
Example: A call in progress shows a "BlxFer" or "VMxFer" option. This would be
a macro with the polycom configuration file. The macro would do the invite as a
blind transfer (one button push) and wait for "x" digits (settable) to transfer
a call before sending
Thanks. At the same time "crap".
So when 4.0 is released this will help. Living with it until this time is
painful in any environment.
Did they give you an indication what other proxying system is a problem? I am
using this with an Ingate and not sure there is a way for that to manage the
As a side note to this, from a provisioning standpoint, i am of the
understanding that the tftp server in sipx does not actively hand out
bootrom/firmware until a device has been configured the first time, perhaps
even sending the profile. So I am wondering if there should not be a template
"wi
I recently encountered an experience where HR had to terminate someone for
cause. In this instance the users voicemail on the system was needed by HR in
the event there were incriminating voicemails in the inbox/saved/deleted
folders, and there was a desire to "cover the bases" so to speak.
Cha
This question is best asked in the sipxtapi-dev list.
https://list.sipfoundry.org/mailman/listinfo/sipxtapi-dev
>>> Mert K¹r 01/19/09 7:37 AM >>>
I had a question about memory last year. I think Scott answered it for me. When
looking at MRTG on a system in a stellite office, I would think the system was
consuming memory over the first few days after a reboot. I assumed since the
memory was not freed up, it was a problem, but this was not
I noticed the tftp server wont actually activate in production builds until the
first profile is actually sent. I'm not sure you could configure tftpd manually
if the rpm is not there. Did it (tftp, ftpd, dhcpd, named) install?
>>> "Nikolay Kondratyev" 01/30/09 12:05 PM >>>
I normally just do "yum clean all".
>>> On 2/5/2009 at 8:34 AM, in message
<20090205133407.4619611...@mail.nstel.ru>, "Nikolay Kondratyev"
wrote:
Thanks for the reply.
I tried removing all under /var/cache/yum/…
But alas, now I’ getting the following error:
[r...@sipx3 yum]# yum list availabl
It's on the wiki, look under roadmap. The current version is 3.10.3, the
development version is 3.11.x (unstable), which will become 4.0 and i think it
is being said it might be available in the next month or so.
>>> On 2/5/2009 at 8:32 AM, in message
>>> <1233840734.3214.8.ca...@hugo.iguanait.
do
rm -fr /var/cache/yum/*
yum clean all
yum makecache
then try adding
http_caching=packages
to the repo and see if that bypasses the problem
>>> On 2/5/2009 at 8:58 AM, in message
<20090205135830.6604e11...@mail.nstel.ru>, "Nikolay Kondratyev"
wrote:
I tried ‘yum clean all’ but after
I would only add it to the repo you are having a problem with.
>>> On 2/5/2009 at 9:20 AM, in message
<20090205142011.20ad111...@mail.nstel.ru>, "Nikolay Kondratyev"
wrote:
Can you please clarify, shoud I add http_caching=packages to
sipxecs-unstable-centos.repo file or to yum.conf ?
Sorry for
If this is the case is there a way to make automatic activation of a
dialing plan pause until all active calls have completed?
>>> On 2/5/2009 at 9:24 AM, in message
, Melcon
Moraes wrote:
On Tue, Feb 3, 2009 at 10:25 AM, Damian Krzeminski
wrote:
Nikolay Kondratyev wrote:
> Just imho:
>
> W
+2, maybe just uncover the option to manually control it in an advanced
link?
>>> On 2/5/2009 at 1:16 PM, in message
, Melcon
Moraes wrote:
+1 to visual cues and big fat warnings.
On Thu, Feb 5, 2009 at 1:41 PM, Damian Krzeminski
wrote:
Damian Krzeminski wrote:
> Scott Lawrence wrote:
>> On
Great. It probably marks an issue with an upstream transparent proxy
between you and the repo. I have read about that problem before but
never had it happen to me. In your case, lightning may strike twice,
assuming there is a problematic proxy sitting between you and some other
part of the world fo
As this relates to ISN dialing:
I am wondering if sipXbridge is capable of handling ISN dialing?
If so, there does not seem to be a way to protect against that without removing
the function. Since ISN losts are sometimes published, there seems to be an
inherent way of filling up anyone's mailb
Maybe I don't fully understand your request here, but the E.164 format relates
to calls from "outside". One would not give an E.164 number to internal calls,
rather you would assign a DID to this to achieve the same result.
We use bandwidth.com with an ingate and our dial plan strips the "+" on
You might get quicker answers posting to the sipx tapi dev list.
http://list.sipfoundry.org/archive/sipxtapi-dev/
>>> Daniel Thornhill 02/15/09 2:58 PM >>>
This is "typical' behavior with ITSP's.
Because the call has gone unanswered, the ITSP cancels the call. There is a
default timer (usually between 30-45 seconds) with your ITSP before they cancel
the call. Some ITSP's can adjust this per account, and some per number.
I find that also the TELCO
Agreed, it should be easier to configure and find. In my case I have one remote
worker with a phone registered via two different sipx systems. It is enormously
helpful to have this easier to understand, though I think the help text
provided actually needs to alert the admin to the possibility of
I spoke with Bandwidth.com at length, assuming this is BANDWIDTH.COM, they say
the incoming/outgoing call timer for both their LEVEL3 and their own CLEC are
120 seconds.
The issue you might be facing is what the cancel timer is by the carrier
(destination number you are forwarding to) is, whic
It's on the wiki.
http://sipx-wiki.calivia.com/index.php/SipX_on_Different_Platforms
You should start with a minimal install on FC4 (no apache, etc. or just start
with the ISO installer.
If you add the REPO to a FC4 install, you should be able to install via YUM.
>>> "chengwu8" 03/02/09 8:43
I opened a ticket with bandwidth.com and they told me the default timer was 120
seconds. If so, I suspect they can change the account to that length.
===
Tony Graziano, Operations Manager
Telephone: 434.984.8430
Fax: 434.984.8431
sip:4...@cavalierbroadband.net
I for one would like to see aliases for hunt groups, ACD's and conferences if
it is possible. I would also like to see an ALIAS for voicemail ("101") too.
Currently the methods used can be quite frustrating to make the DID's flow down
to those by creating "phantom users", adding the DID alias
Separately I think it would be awesome to have a feature for the "receptionist"
to turn on/off the AA and have the calls routed to his/her line. It would be
good if the boss calls a meeting they could hit a button and go into AA mode
and back again without too much fuss. Something that can be do
a good practice to leave your major routing
decisions up to a phone. Probably OK on a smaller system though...
This was all done before there was such a thing as time based forwarding
so you could use a phantom to route for hours that are definitely night
hours and make it a little more reliable.
Then if the phone reboots or has a problem the routing goes away unless someone
comes in and touches the phone again.
>>> "Scott Lawrence" 03/17/09 9:09 AM >>>
On Mon, 2009-03-16 at 16:55 -0400, Picher, Michael wrote:
> I think it should be on any item as Tony mentioned.
>
> While we're at it a
> As part of our roadmap I would like to see us define and deliver unified
> communications functionality that reinfoces the value of openness *and*
> delivers a suite of user capability that is anchored in the UC
> categories above and accessible from the user portal or softclient.
> (note: wher
some answer inline below...
>>> سهر احمد 03/21/09 2:13 AM >>>
I re-write the massage in better way , so you can understand what i looking
for...
Some questions in Sipxesc:
- can sipx work in windows OS.
**Noone has ported sipxecs to Windows yet***
- Can sipx work in Ubuntn Linu
Similar problem here, but ConfigAgent was disabled. I dropped my database and
recreated, still no joy.
I ran sipxproc --start ConfigAgent and was able to log in but ConfigAgent
status changed to "ConfigurationMismatch", I upgraded again via YUM, same
issue. I confirmed the DNS works in all case
>>> On 3/23/2009 at 10:55 PM, in message
>>> ,
"wtuben" wrote:
> Hello everyone,
>
> when I use sipxezphone place a p2p call to the eyebeam , and their codec
> only use ILBC,
>
> and the result is: eyebeam CANNOT hear sipxezphone, but the sipxezphone CAN
> hear the eyebeam!!
>
> AND when
hould
say "disabled"? I get no errors at stopping/starting services.
It would be problematic if people had to drop/create their databases when
upgrades won't working properly.
>>> On 3/24/2009 at 11:16 AM, in message , Damian
Krzeminski wrote:
> Tony Graziano wrote:
> >
all setup, and sipxezphone
here
> the eyebeam, but eyebeam cannot hear sipxezphone!
>Will the codec iLBC used by the sipxezphone and eyebeam is NOT the
same??
> And I hope somebody will do the same operation to watch the "BUG" as
above?
>
>
> Thx
>
+1
I can see where the Internet is working and the ITSP is also, but if someone
else's stun server is not working, I'd certainly want my internal users to be
able to use the system anyway. That's kinda scary to me.
At the same time, does it make sense to be able to input
primary/secondary/tert
Shouldn't there also be something in place to alarm if the ITSP is unavailable
in some other way (don;t require registrations, like Bandwidth.com).
>>> "M. Ranganathan" 03/25/09 10:35 AM >>>
I am wondering whether to send an alarm for failed Registrations.
These can happen during run-time if the
I'm planning to test the CMC plugin soon and have read up on XCF-2022 (add
provisioning support for counterpath) and XCF-2898 (RLS support for
Counterpath).
This looks great. To use the auto provisioning feature does it matter is the
end user has a certain level of softphone from Counterpath?
The last two times I run an update and restarted services in
sipx 3.11, I logged into sipxconfig and was told some services needed to be
restarted.
(registrar, proxy, aa).
After the update ran I manually stopped and restarted services
and then saw these needed to be restarted. I
>>> "Andy Spitzer" 03/26/09 10:27 AM >>>
Woof!
Robert wrote:
> I introduced the dependency to avoid leaving the system where some basic
> calls work and some don't. My own personal opinion is that this kind
> of hit-and-miss behavior is worse than the system not running at all
> because can giv
I'm in the beginning stages of testing a new server to understand native
trunking with sipxbridge and remote worker setups.
My server has the role assigned. I've sent the sipxbridge profile to the
system. I still see an issue (known) to restart several services and have made
sure that is done b
Forgot the image.
>>> On 3/30/2009 at 10:59 AM, in message
<49d097ff025a4...@mail.myitdepartment.net>, "Tony Graziano"
wrote:
> I'm in the beginning stages of testing a new server to understand native
> trunking with sipxbridge and remote worker
Can only send calls to AA, not to registered users. RFC2833 (DTMF/inband) won't
work on inbound calls, remote phones cannot register except through VPN, and
even then are only reachable via AA.
>>> "M. Ranganathan" 03/30/09 12:25 PM >>>
On Mon, Mar 30, 2009
66.xx.xxx
Public Port: 5060
Start RTP: 3 End
RTP: 31000
>>> "M. Ranganathan" 03/30/09 12:25 PM >>>
On Mon, Mar 30, 2009 at 10:59 AM, Tony Graziano
wrote:
> I'm in the beginning stages of testing a new server to understand native
> trunking with sipxbri
n message
<5c7eebdd0903310726ra38bf64g21ecba8e33347...@mail.gmail.com>, "M. Ranganathan"
wrote:
> On Tue, Mar 31, 2009 at 5:37 AM, Tony Graziano
> wrote:
> > Trying to tackle this one piece at a time. 3 problems exist.
> >
> > 1. DTMF not passed.
> > 2. inbound calls will on
I'm trying to understand the current method used to send a register request.
What I think i'd like to see long term is a 2nd ip address bound simply to the
remote register function, and see the register requests forward to that
(private) ip and allow the traffic between trunking and remote users
>>> On 4/2/2009 at 1:39 PM, in message
<1238693961.4906.6.ca...@victoria-pingtel-com.us.nortel.com>, "Dale Worley"
wrote:
> On Thu, 2009-04-02 at 13:03 -0500, Tony Graziano wrote:
> > I'm trying to understand the current method used to send a register
>
>> >>> On 4/2/2009 at 1:42 PM, in
> message
<0bdfff51dc89434fa33f8b37fce363d516aad...@zcarhxm2.corp.nortel.co
> m>, "Robert
Joly" wrote:
> > I'm trying to understand the current method used to send a
> > register request.
> >
> > What I think i'd like to see long term is a 2nd ip address
>
>> >>> On 4/2/2009 at 2:17 PM, in
> message
<0bdfff51dc89434fa33f8b37fce363d516aad...@zcarhxm2.corp.nortel.co
> m>, "Robert
Joly" wrote:
>> >>> On 4/2/2009 at 1:42 PM, in
> > > message
> > <0bdfff51dc89434fa33f8b37fce363d516aad...@zcarhxm2.corp.nortel.co
> > > m>, "Robert
> > Joly" wrote:
> >
I have a customer who calls and goes through the process of the dial by name to
reach an associate.
He goes through the AA and dials by name. When he gets to the instructions
"Press 1 for Average Joe", he interpreted is at press "14".
Get it? Press "1 for" or press "14". Strangely or sadly, it
While I may have misunderstood forwarding registrations, the questions asked
were valid, and led to more questions. After asking some questions and
realizing it needs more discussion, Robert graciously opened
http://track.sipfoundry.org/browse/XECS-2432 (thanks Robert) which will be
addressed
>>> Akshata 04/06/09 9:15 AM >>>
Thanks for the clarification Woof.
IMHO it will be nice to have "press #" prompt as you said.
Could we modify the prompt as, "Dial the name of the person last name
first. _Then press #_. Press 7 for Q and 9 for Z"
Thanks,
Akshata
Andy Spitzer wrote:
> Woof!
>
>
>>> "Mark Gertsvolf" 04/06/09 12:27 PM >>>
Scott Lawrence wrote:
> As I've said before, an ITSP that can't cope with a different
> port number is one that you'll have other problems with - get
> a better one - the good ones have no trouble at all with
> configuring a port other that 5060.
>
using latest 3.11.12... I created a profile for a softphone (added a line) for
CMC Enterprise, sent the profile and confirmed it was in the tftproot. I
installed Bria professional and entered the key, exited the application so I
could get the initial login/provisioning screen back up.
Where wou
I see the messages as to
why it failed? If I delete it and create it manually it registers fine.
>>> "Alfred Campbell" 04/06/09 1:33 PM >>>
> -Original Message-
> From: sipx-dev-boun...@list.sipfoundry.org [mailto:sipx-dev-
> boun...@list.sipfoundry.
>>> On 4/6/2009 at 2:25 PM, in message
<1d9361c130b11c4db300adbfca36527004ae7...@zrtphxm0.corp.nortel.com>, "Alfred
Campbell" wrote:
>
> > -----Original Message-
> > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> > Sent: Monday, A
>>> On 4/6/2009 at 3:07 PM, in message
<1d9361c130b11c4db300adbfca36527004ae7...@zrtphxm0.corp.nortel.com>, "Alfred
Campbell" wrote:
>
> > -----Original Message-
> > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]
> > Sent: Monday, A
I see the AOR configured in the INI file. I also se nothing in the sipxpresence
log either. I'll do a wireshark. Where do i send it?
>>> "Alfred Campbell" 04/07/09 2:21 PM >>>
> Today I installed a brand new instance of 3.11.12 and then proceeded
> toupdate it to the newest version. After getti
Today I installed a brand new instance of 3.11.12 and then proceeded toupdate
it to the newest version. After getting the basic system setupand working, I am
able to use the CMC plugin to provision a BriaProfessional softphone.
I created a speedial entry for the testuser account and sent it the
Can I assume ResourceListServer handles BLF and speed dial and is independent
of PresenceServer which only handles ACD?
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>>> On 4/9/2009 at 10:08 AM, in message
<1239286134.13289.8.ca...@scott.skrb.local>, "Scott Lawrence"
wrote:
> On Thu, 2009-04-09 at 06:18 -0400, Tony Graziano wrote:
> > Can I assume ResourceListServer handles BLF and speed dial and is
> > independent
nfig
file and let it grab the edited one, I still have no workgroup subscription,
but I do get the window (had it either way). So if I have the workgroup windows
with or without their saying to use the above syntax, should it not matter, as
long as I have it?
>>> "Tony Graziano&
Harump! That new build worked! I'm not crazy! Should I reference anything to
Counterpath to confirm the issue stille exists with the Bria?
>>> "Alfred Campbell" 04/15/09 9:56 AM >>>
> -----Original Message-
> From: Tony Graziano [mailto:tgrazi...@myitd
Good news! Does anyone know what happened to the 4.0 version?
http://sipxecs.sipfoundry.org/temp/sipXecs/4.0/
I could have sworn it was there yesterday.
>>> "Scott Lawrence" 04/21/09 9:39 PM >>>
On Mon, 2009-04-20 at 10:28 -0400, Scott Lawrence wrote:
> I'm planning to move the sipXecs projec
like the current Skype Beta has? Knowing Jingle is not sip, or at least as i
understand it. Thought it was a good question to ask.
Thanks,
Tony
--
===
Tony Graziano, Operations Manager
Telephone: 434.984.8430
Fax: 434.984.8431
sip:4...@cavalierbroadband.
Too bad Joe isn't here. This was right up his alley.
I may have something. I'll look when I get into the office later, but it's a
phone receiver with a hand under it.
>>> Cristi Starasciuc 04/24/09 5:22 AM >>>
___
sipx-dev mailing list
sipx-dev@list.
>> >>> On 4/28/2009 at 1:49 PM, in
> message
<1d9361c130b11c4db300adbfca365270051be...@zrtphxm0.corp.nortel.co
> m>, "Alfred
Campbell" wrote:
Subject: [sipX-dev] sipXecs 4.0.0 is
> released!
> >
> > The build 4.0.0-015321 has been promoted to 'stable' status, and is
> now
> > available in the
>> >>> On 4/28/2009 at 3:14 PM, in
> message
<1d9361c130b11c4db300adbfca365270051bf...@zrtphxm0.corp.nortel.co
> m>, "Alfred
Campbell" wrote:
>> >>> On 4/28/2009 at 1:49 PM, in
> > > message
> > <1d9361c130b11c4db300adbfca365270051be...@zrtphxm0.corp.nortel.co
> > > m>, "Alfred
> > Campbell"
>>> On 4/28/2009 at 3:35 PM, in message
<1240947313.28273.7.ca...@tanager.pingtel.com>, Kevin Thorley
wrote:
> On Tue, 2009-04-28 at 15:24 -0400, Tony Graziano wrote:
> >
> >
> > Centos 5.2 originally from 3.10 ISO, upgrading via rpm/yum.
> >
>
>>> On 4/28/2009 at 4:22 PM, in message
<1240950120.28273.10.ca...@tanager.pingtel.com>, Kevin Thorley
wrote:
> On Tue, 2009-04-28 at 15:53 -0400, Tony Graziano wrote:
> > >>> On 4/28/2009 at 3:35 PM, in message
> > <1240947313.28273.7.ca...@tana
>>> On 4/28/2009 at 5:23 PM, in message <1240953809.3478.183.ca...@scott>,
>>> "Scott
Lawrence" wrote:
> On Tue, 2009-04-28 at 16:29 -0400, Tny Graziano wrote:
> >
> >
> > No I haven't tried to start it by itself. I just reformatted and am
> > reinstalling. Is it possible to restore a 3.10 co
This link is broken. http://sipxecs.sipfoundry.org/pub/sipXecs/ISO/
r
- Original Message -
From: "Scott Lawrence"
To: "Tony Graziano"
Cc:
Sent: Tuesday, April 28, 2009 4:23 PM
Subject: Re: [sipX-dev] sipXecs 4.0.0 is released!
> On Tue, 2009-04-28 at 16:29 -0400, Tony
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