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On Wed, Dec 12, 2012 at 2:33 PM, Josh Patten jpat...@ezuce.com wrote:
Try sending them my response to the mailing list and see where that gets
you. If they balk at it, then I'll see how I can explain it better to them.
On Wed, Dec 12, 2012 at 4:23 PM, Josh Patten jpat...@ezuce.com
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(REFER) in product x. Which is exactly what Josh seems to have
asked.
Go Josh!
On Wed, Dec 12, 2012 at 3:15 PM, Josh Patten jpat...@ezuce.com wrote:
It looks like Adtran and Digium are screaming the same thing. I'm going
to post the response I sent to Digium:
According to http
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** **
In Maine tonight. .. bring beer.
On Dec 5, 2012 1:14 PM, Josh Patten jpat...@ezuce.com wrote:
Release party: when and where??? :-)
** **
On Wed, Dec 5, 2012 at 12:10 PM, Михаил Родионов ma...@siplabs.ru wrote:
CONGRATULATIONS!!!
4.6 is a huge milestone! Thank you
who understand binary and
those who don't.
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also was working on FreeSWITCH awhile back to do this exact same
thing and had the freeswitch consultants looking at this and this is
exactly the same issue i ran into with them and there was no fix.
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I ran into a dependency
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, 2012 at 7:51 AM, Melcon Moraes mel...@gmail.com wrote:
Can we simply drop in a new FS RPM package to replace the 1.0.7 version
that comes with sipXecs and all the current working stuff will continue to
work?
On Mon, Nov 5, 2012 at 1:58 AM, Josh Patten jpat...@ezuce.com wrote:
Yes, MoH, ring
PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Karoo Bridge also uses the FS libraries. What FS lacks is an admin GUI,
just like karoo bridge. Does MOH work in your tests too?
On Nov 4, 2012 4:22 PM, Josh Patten jpat...@ezuce.com wrote:
LAN/Telephony/Security and Control Systems
AM, Melcon Moraes mel...@gmail.com wrote:
Can we simply drop in a new FS RPM package to replace the 1.0.7 version
that comes with sipXecs and all the current working stuff will continue to
work?
On Mon, Nov 5, 2012 at 1:58 AM, Josh Patten jpat...@ezuce.com wrote:
Yes, MoH, ring on transfer
for.
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+Trunking+Gateway
Enjoy!
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list
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, Josh Patten jpat...@ezuce.com wrote:
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: helpdesk@voice.myitdepartment.**nethelpd...@voice.myitdepartment.net
Helpdesk Customers:
http://myhelp.myitdepartment.**nethttp://myhelp.myitdepartment.net
Blog: http
-Howard
McDonogh School Technology Department
Sent from a mobile device; please excuse typos.
On Nov 4, 2012 6:18 PM, Josh Patten jpat...@ezuce.com wrote:
How many of these sets do you have? If it's only a few it would be worth
your time just to buy new sets. Cisco 79xx phones have notoriously bad
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from failing
scenario? (Meanwhile will try to get an to replicate / debug too)
Thanks
George
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version of sipx?
Thank you,
Marco Crociani
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Interesting though, I've been on 3.4.2 for a couple of weeks now. Asked me
to upgrade for a while before that. Wonder why your users are just now
seeing that.
On Mon, Aug 13, 2012 at 7:07 PM, Josh Patten jpat...@ezuce.com wrote:
[image: Inline image 1]
IT'S A TRAP!!!
Seriously though, tell
the problem? Thanks.
~Noah
Scanned for viruses and content by the Tranet Spam Sentinel service.
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as Zultys and Epygi commercial PBXs with perfect results.
The problem is not with Aastra.
--
Regards
Mark Dutton
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) and wondering what I should expect.
--thanks
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Are you currently an eZuce customer/partner or are you just
using sipXecs? Please let me know as there are different
protocols for support on each platform. If you are an eZuce
customer/partner and have a paid support contract I will
direct you to the correct support portal. Otherwise, please
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who don't.
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image002
://list.sipfoundry.org/archive/sipx-users/
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Patten jpat...@ezuce.com wrote:
For those of you who have mixed Cisco/Polycom environments and want to
call from Polycom to Cisco, here is a temp workaround until something
better can be devised: http://track.sipfoundry.org/browse/XX-8762
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** **
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For those of you who have mixed Cisco/Polycom environments and want to call
from Polycom to Cisco, here is a temp workaround until something better can
be devised: http://track.sipfoundry.org/browse/XX-8762
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http
http://wiki.sipfoundry.org/display/sipXecs/SSL+Web+Certificates
Hope it helps someone...
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Also, if someone wants to go clean that section of the wiki up (or any
section of the wiki) please feel free to :-)
On Mon, Apr 30, 2012 at 10:32 PM, Josh Patten jpat...@ezuce.com wrote:
http://wiki.sipfoundry.org/display/sipXecs/SSL+Web+Certificates
Hope it helps someone...
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connecting to an
ITSP
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of a sipx
server in micro installs) in many different ways.
SN's have a sip call limit based on their DSP allowance. The smaller the
model, the lower the supported call volume.
Does REFER work though?
On Fri, Apr 27, 2012 at 5:08 PM, Josh Patten jpat...@ezuce.com wrote:
Apparently
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this message in error, please
immediately advise the sender by reply email
and delete all copies.
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per say because if there is no answer it goes
straight to v-mail and is not forwarded.
Stiles
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have no issues with that. I do have pesky issues with
call park and call park return, but not sure that's a gateway issue.
On Sun, Mar 25, 2012 at 9:45 PM, Josh Patten jpat...@ezuce.com wrote:
The last firmware I tested in early 2011 didn't have *78xxx directed call
pickup working (INVITE
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and ni1 and ni2 can all be done on patton gateways at the same time
(ie each t1 interface using different protocol).
I don't know if it matters though.
On Mar 26, 2012 3:53 PM, Josh Patten jpat...@ezuce.com wrote:
Tony I was referring to Adtran gateways. Audiocodes GW's don't have those
issues
with the Audio Codes
reporting a bug in the firmware with the equipment M1K-D1 firmware version
6.00A.054.002, after i enter in web interface, menu (Advanced
Applications) and select the menu (LDAP Settings) equipment simply performs
a reset.
Thanks.
2012/3/26 Josh Patten jpat...@ezuce.com
I have
They're pretty tasty to me :-P
On Mon, Mar 26, 2012 at 3:09 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I hate ribs.
On Mar 26, 2012 4:04 PM, Josh Patten jpat...@ezuce.com wrote:
Yeah the Adtran TA90xe voice T1 ports can do either PRI or RBS.
On Mon, Mar 26, 2012 at 3:00 PM
companies edge already) or ADTRAN TA.
Regards,
Mark
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questions by visiting the Knowledge
Base https://kb.sunyulster.edu/
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Look up answers to frequently asked questions by visiting the Knowledge
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Hope to see you at the sipX CoLab! http
in advance
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and SIPX or should we be looking
into another standard such as NI2?
Thanks,
Jes
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to somewhat explain how it works.
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...@list.sipfoundry.org] *On Behalf Of *Josh Patten
*Sent:* Tuesday, November 22, 2011 2:01 PM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] Dial By Name Improvement
** **
http://track.sipfoundry.org/browse/XX-9975 as well. I think this one
would help eliminate
All.
I've created an improvement request for dial by name that I think would be
useful to many here: http://track.sipfoundry.org/browse/XX-9974
Please vote for it if you think it's worthwhile.
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http://track.sipfoundry.org/browse/XX-9975 as well. I think this one would
help eliminate caller confusion.
On Tue, Nov 22, 2011 at 3:55 PM, Josh Patten jpat...@ezuce.com wrote:
All.
I've created an improvement request for dial by name that I think would be
useful to many here: http
://kb.sunyulster.eduhttp://kb.sunyulster.edu/
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!
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…
** **
Does anyone else find this delay annoying enough to ask for it to be
modified?
** **
** **
*From:* sipx-users-boun...@list.sipfoundry.org [mailto:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Josh Patten
*Sent:* Monday, October 17, 2011 9:35 AM
*To:* Discussion list
that you are using a phone that has firmware
that can interact with these requests.
On Mon, Oct 17, 2011 at 11:19 AM, Josh Patten jpat...@ezuce.com wrote:
It's a registrar function. The registrar sends a request to the proxy
which then forks it to all UA's registered as whatever user ID
X2015
M.207-956-0262
@mpicher http://twitter.com/mpicher
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://kb.sunyulster.edu/
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, the suspect phone still is blink red. I will try
downgrading the firmware to 3.2.4 and will report back.
Thanks,
Jes
On Mon, Sep 12, 2011 at 12:15 PM, Josh Patten jpat...@ezuce.com wrote:
First of all firmware 3.2.5 has some strange issues. Move back to 3.2.4
The only way to get voicemail
?
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-graziano/14/4a6/7a4
Ask about our Internet faxservices!
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M.979
, Josh Patten jpat...@ezuce.com wrote:
I was never able to get Verizon to provision an ISDN line properly. They
would only provide the service in a very basic way (2 DID's, one per
channel) and wouldn't set up anything remotely advanced on them.
DID wink should work on audiocodes but don't
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transferring a call to a different SIP domain?
Cheers, and thanks,
...Steve...
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with Exchange...
On Mon, Aug 22, 2011 at 11:01 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
me either. I think the way to do this perhaps is with an api 4 specific
imap implementations like the ones you mentioned.
On Aug 22, 2011 9:58 AM, Josh Patten jpat...@ezuce.com wrote:
I lol'd
tgrazi...@myitdepartment.net wrote:
me either. I think the way to do this perhaps is with an api 4 specific
imap implementations like the ones you mentioned.
On Aug 22, 2011 9:58 AM, Josh Patten jpat...@ezuce.com wrote:
I lol'd...stupid autocorrect!
In any case I'm pretty sure MS
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