...@esgw.org
Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM
Is the call an intra or inter domain call?
On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote:
The 12xx gateway has a gateway setup for 37xx and the appropriate
dial-plan.
Aaron Pursell
Network
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:55 AM
Thats not a valid answer. Your answer should be in the form of either
in error, please
immediately advise the sender by reply email
and delete all copies.
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(not as desirable, IMO), you probably forgot to add the subnets for the
other systems at the other end in each system...
if so, thats an oopsie on you.
On Tue, Apr 24, 2012 at 11:12 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
did you try calling from a UA at one end in the same form?
1234
, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 9:29 AM
is it an internal (vpn or locally connected) call?
if it is, do the subnets on each side reflect a local subnet (to not
invoke media relay)?
i am thinking if each system can see the other
have been seeing some concerns about, these arrived last week
with that firmware installed.
-Bryan Anderson
On Mon, Apr 23, 2012 at 2:00 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Really the best thing you can do is put your log with sipx (proxy) to
debug, and grab
a need for more than 4 FXO ports, the 4500 series is the way to go
(IMO).
On Tue, Apr 24, 2012 at 3:07 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I would use the model 4524 instead of 4114.
On Apr 24, 2012 2:48 PM, Bryan Anderson branderso...@msn.com wrote:
Sorry I sent this yesterday
Hello
when I send request to assp-white@ I receive an email with an empty body
message.
I have seen the problem also using assp-spam@ .
Thank you
Graziano
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Does
yum info sip page
Return anything? Is this a clean install or update?
On Apr 23, 2012 8:03 AM, Kumaran thiru.venkateshwa...@ttplservices.com
wrote:
Douglas Hubler wrote:
On Mon, Apr 23, 2012 at 6:11 AM, Kumaran
thiru.venkateshwa...@ttplservices.com wrote:
I'm getting following
/
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of their
frequency will be used to validate incoming addresses. Note:
LocalAddresses_Flat or DoLDAP or DoVRFY must be enabled.
Thank you
Graziano
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The issue is with the trunk connection. Re-read my post.
On Apr 21, 2012 8:34 PM, Tommy Laino tomla...@gmail.com wrote:
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
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@list.sipfoundry.org
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Ask about
a superior product and I hope it continues to improve and
continues to be used by more companies.
Thank you, Tony, and everyone else, for your help.
Stiles
On 04/19/2012 01:31 PM, Tony Graziano wrote:
On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wat...@datatek-net.comwrote:
On 04/18/2012 06
] *On Behalf Of *Tony Graziano
*Sent:* Friday, April 20, 2012 9:29 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] Fax Problems (Big Time)
** **
faxing into sipx only support t.38 and your audiocodes fax settings have
g711u. you need to change it to t.38
:
sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* Friday, April 20, 2012 11:00 AM
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] Fax Problems (Big Time)
** **
Sorry, am not a audiocodes guru. I have documented the Patton PRi
-users@list.sipfoundry.org
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for this.
Thanks Regards,
Chitra
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On Thu, Apr 19, 2012 at 2:20 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I do think there is room for improvement here. If you do not have
voicemail
permissions, a brief audio message stating, This account is not enabled
for messages., would certainly be more desirable.
I do
Yes I think he wants the interopserver though I am not sure if it handles
rfc4475 stuff.
On Apr 19, 2012 1:47 PM, Michael Picher mpic...@ezuce.com wrote:
Was this what you were looking for? http://interop.sipfoundry.org ?
I don't think that works either at the moment...
I'm looking at
, Apr 19, 2012 at 4:02 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Yes I think he wants the interopserver though I am not sure if it handles
rfc4475 stuff.
On Apr 19, 2012 1:47 PM, Michael Picher mpic...@ezuce.com wrote:
Was this what you were looking for? http
http://track.sipfoundry.org/browse/XX-10120
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Telephone
:
Name:3014
Registrar:voiptest.netappsid.local(the hostname of my server)
User:3014
Authentification User:3014
Password:The generated sip password
Timeout:3600
and it give me a Transport Error when I try to enable it.
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
Then I suggest you
of ekiga i don't have a Stun Server option but when I go in
the Ekiga configuration with the Configuration Editor tool on Ubuntu 11.10
i see the stun server option and it's disabled.
2012/4/19 Tony Graziano tgrazi...@myitdepartment.net
perhaps you have enabled STUN (not needed with sipx) on Ekiga
/
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Graziano tgrazi...@myitdepartment.net
wrote:
The profile is not negotiating t.38, it clearly shows g711u.
profile voip default
codec 1 g711ulaw64k rx-length 20 tx-length 20
codec 2 transparent-clearmode rx-length 20 tx-length 20
no dtmf-relay
rtp traffic-class local-default
no dtmf
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gateway) ?
Thanks,
Jes
On Thu, Apr 19, 2012 at 12:19 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
no *codec 2 g711alaw64k rx-length 20 tx-length 20*
On Thu, Apr 19, 2012 at 11:27 AM, Jesse Becker beck...@sunyulster.eduwrote:
Tony,
I modified my profile voip default
for the provisioning mode.
So, I've changed the section title to 'Starting Bria in provisioned mode'.
Now it's way clear.
On Thu, Apr 19, 2012 at 12:30 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
no, its not wrong, I have changed it back...
Only the Bria 3.x is capable of being auto
at 12:42 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
No. I thought I sent you a 4960 config some time ago. The most recent one
I put on the wiki does work in conjunction with the 4424 profile I already
gave you though.
I'm sure if it doesnt work after comparing, patton can fix with you
On Thu, Apr 19, 2012 at 1:06 PM, Stiles Watson wat...@datatek-net.comwrote:
On 04/18/2012 06:15 PM, Tony Graziano wrote:
You shouldn't invite violence. I have been known to clobber people with
iPads.
Why am I not surprised by this?
Um that was a cartoon reference dude, you take things
** **
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Do you mean multi tech?
On Apr 19, 2012 2:47 PM, Todd Hodgen thod...@frontier.com wrote:
Does anyone have an good or bad experience with Microtech FaxFinder
Appliance, either SIP or analog version to share. I’m looking at using it
for an outbound fax solution for a customer, and looking for
/
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to
come back on a port that isnt being sent by the router.
2012/4/19 Simon Brûlé sbr...@360-innovations.com
So your saying the problem may come from the router I have (Linksys E2500)
because it's not doing the symmetrical Nat so the RTP is getting lost?
2012/4/19 Tony Graziano tgrazi
Hello list,
is there a simple way to enable dnsmasq dhcp server on an astlinux 1.0.2
installaed on a single ethernet system?
I have a machine installed with ast1.0.2 and i'm going to activate a
dhcp server to provide ip addresses to our sip phones but I cannot find
information.
Thank
graziano
Phone: (07) 4093 3826
Fax: (07) 4093 3869
Email: andrew.ra...@yuruga.com.au
Web: www.yuruga.com.au
On 17/04/2012, at 8:04 PM, Tony Graziano wrote:
You need to explain what kind of coverage you need and what kind of
wireless infrastructure you have (if any).
Snom makes a dect phone which also
other tools
for this test?
Thanks a lot.
Derrick
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I don't get you. Counterpath has a product for the desktop that can be
centrally provisioned. Their mobile devices can't (yet).
Maybe we can create a stiles.wiki. JK.
I think the wiki might benefit from a mobile apps section, but it requires
proper layout for and could use a simple table layout
Account advanced works unchecked if your DNS is correct.
On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote:
So can the wiki be updated to reflect the differences in Bria for
iphone/ipod touch?
- Use SIP password not PIN
- No Provisioning server
- Under Account
The profile is not negotiating t.38, it clearly shows g711u.
profile voip default
codec 1 g711ulaw64k rx-length 20 tx-length 20
codec 2 transparent-clearmode rx-length 20 tx-length 20
no dtmf-relay
rtp traffic-class local-default
no dtmf-mute-encoder
response-preferred-codec
spelled D N S. If you are trying to do this
outside without a VPN, you can't sidestep it.
Stiles
On 04/18/2012 04:32 PM, Tony Graziano wrote:
Account advanced works unchecked if your DNS is correct.
On Apr 18, 2012 4:07 PM, Stiles Watson wat...@datatek-net.com wrote:
So can the wiki
/2012 04:31 PM, Tony Graziano wrote:
I don't get you. Counterpath has a product for the desktop that can be
centrally provisioned. Their mobile devices can't (yet).
Maybe we can create a stiles.wiki. JK.
I think the wiki might benefit from a mobile apps section, but it requires
proper layout
Graziano
tgrazi...@myitdepartment.net wrote:
You shouldn't invite violence. I have been known to clobber people with
iPads.
I think the wiki could be clearer, but really I think you are the only
one to make the leap from Bria is bria is bria is bria. It ain't, but
sipfoundry doesn't
,
Kumaran T
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You need to explain what kind of coverage you need and what kind of
wireless infrastructure you have (if any).
Snom makes a dect phone which also has wireless repeaters and should work
fine. The battery life and talk time is very good and does not interfere
with wifi at all.
If you have a wifi
Web: www.yuruga.com.au
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a moot point.
Mike
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Good.
On Apr 16, 2012 3:07 PM, Mike Graham mike_gra...@hempfieldsd.org wrote:
Content-Type: text/plain;
charset=utf-8
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
camgknjwtdrrrwpgtplhomg9j8jw3tj0ke0unnakfvauouqy...@mail.gmail.com
X-FUDforum:
The problem fixed itself after today I applied an EXIM upgrade , even if it's
weird since with older
ASSP versions 1.8.1.7(0.0.00) and older it was working .
Graziano
I tried older ASSP versions and I found this
current version 1.9 does not work
...
... other tested no works
...
1.8.5.9
I would suspect ipsec would work well, we use it all the time for site to
site vpn's.
On Apr 14, 2012 10:46 AM, Nitin Mirchandani nitin_mirchand...@hotmail.com
wrote:
The only thing that can raise issue is that the sites are connected via
PPTP VPN.
Latency is sub 10ms.
Will MS PPTP(the BEST
You can simulate this inside but I think pptp is a poor choice in that the
packets use a different IP address to be native to the network.
IPSEC AND OPENVPN are more suited to point to point and can properly tested
with the same network schema before rolling the systems out and over a vpn.
Pptp
same problem with latest 1.9.6.7(0.0.06)
Graziano
Hello
using 1.9.6.5(0.0.05)
The email below (local good email) is going in black hole , never reaches MTA
and disconnects after 0 seconds (?).
.dat is allowed in GoodAttach
Apr-13-12 11:55:45 id-33433-00011 [Attachment] 127.0.0.1info
version which passes the email below is 1.8.5.9(0.0.06) .
After 1.8.5.9(0.0.06) there is NO ASSP version
which is able to pass the email below to MTA .
Graziano
also using noprocessing I can't find a way to pass this email to MTA .
i...@jancr.com and ro...@ciam.com are local email . The email
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these results.
I see the system has its certificate generated. I expect the binary
simply doesn't know how to interpret the result since its the binary
from 4.4 that I used.
Is this a known issue? Is there any other information I can provide?
Is a JIRA warranted?
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On Fri, Apr 13, 2012 at 8:36 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
All tests are failing.
2012-04-13T12:22:00.89Z:162:JAVA:ERR:sipx.dev.myitdepartment.net:pool-4-thread-1::ExternalCommand:Cannot
execute: preflight
java.io.IOException: Cannot run program /usr/bin
OK, thanks!
On Fri, Apr 13, 2012 at 8:42 AM, George Niculae geo...@ezuce.com wrote:
On Fri, Apr 13, 2012 at 3:36 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
All tests are failing.
Is this a known issue? Is there any other information I can provide?
Is a JIRA warranted?
Yep
it will be easy to track it down.
Regards,
Kumaran T
Douglas Hubler wrote:
On Fri, Apr 13, 2012 at 8:17 AM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
I added a domain alias and then found sipxconfig to be running. All
web queries were rejected. I had to manually start apache
to append to open issue
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To get some attention to it, votes would be appropriate.
On Fri, Apr 13, 2012 at 6:52 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
http://track.sipfoundry.org/browse/XX-10111
On Fri, Apr 13, 2012 at 4:56 AM, Joegen Baclor jbac...@ezuce.com wrote:
There is a current limitation
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Subject: Re: [sipx-users] Multiple line appearances
That is what I did. I have one line assigned to the phone and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You register the SAME LINE on both line appearances and set
and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You register the SAME LINE on both line appearances and set the limit to 1
for each.
On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson wat...@datatek-net.com
wrote:
According to the sipX book (p 133
releases see
http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0
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sip: tgrazi...@voice.myitdepartment.net
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sip: tgrazi...@voice.myitdepartment.net
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
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Ask about our Internet
Hello
today I found a localdomain (domain listed in localdomains) listed in
DomainBoxLimit (probably added because the domain sent spoofed email ?).
Could be avoided this behavior (having a localdomain added in DomainBoxLimit) ?
Thank you
Graziano
in
whiteListedIPs, noProcessingIPs ?
Thank you
Graziano
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also denySMTPConnectionsFromAlways .
using ASSP 1.9.6.x
Graziano
Mit freundlichen Grüßen
Could it be possible to have the behavior above also for
denySMTPConnectionsFromAlways , or at least for ips listed in
whiteListedIPs, noProcessingIPs ?
It is possible to change the code to do
- sent DATA size: 48.04
kByte;
Apr-13-12 11:55:45 id-33433-00011 127.0.0.1 info@jancr to: ro...@ciam.com
disconnected (0 seconds);
Thank you
Graziano
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** Also affects: eucalyptus/2.0
Importance: Undecided
Status: New
** Changed in: eucalyptus
Status: Fix Committed = Invalid
** Changed in: eucalyptus
Assignee: graziano obertelli (graziano.obertelli) = (unassigned)
** Changed in: eucalyptus
Milestone: 2.0.4 = None
after a chat on #eucalyptus-devel, this is not present (the mis-handling
of /) in eucal3 (marking invalid for -devel). It is targeted for 2.0.4,
but the IRC chat seems to imply that more work may be needed to
understand the right behavior here.
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You received this bug notification because you
** Also affects: eucalyptus/2.0
Importance: Undecided
Status: New
** Changed in: eucalyptus
Status: Fix Committed = Invalid
** Changed in: eucalyptus
Assignee: graziano obertelli (graziano.obertelli) = (unassigned)
** Changed in: eucalyptus
Milestone: 2.0.4 = None
after a chat on #eucalyptus-devel, this is not present (the mis-handling
of /) in eucal3 (marking invalid for -devel). It is targeted for 2.0.4,
but the IRC chat seems to imply that more work may be needed to
understand the right behavior here.
--
You received this bug notification because you
sipxecs.netappsid.voip.
and the 192.168.177.1 :
server 192.168.177.1
Default server: 192.168.177.1
Address: 192.168.177.1#53
set type=srv
_sip._udp.netappsid.voip
;; connection timed out; no servers could be reached
2012/4/11 Tony Graziano tgrazi...@myitdepartment.net
IF sipx was told
/router as the Hardphone for exemple and
everything is going to be fine?
2012/4/12 Tony Graziano tgrazi...@myitdepartment.net
sipx does not support multiple network adapters yet. If it were me, I
would reinstall with a single activated NIC and go from there. Give it a
static IP and configure
-users/
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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Archive: http://list.sipfoundry.org/archive/sipx-users/
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Tony Graziano, Manager
Telephone: 434.984.8430
sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
~~
Linked-In Profile:
http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
Ask about our
and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You register the SAME LINE on both line appearances and set the limit to
1 for each.
On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson
wat...@datatek-net.comwrote:
According to the sipX book (p 133
and the
registration settings as below.
Stiles
On 04/12/2012 11:54 AM, Tony Graziano wrote:
You register the SAME LINE on both line appearances and set the limit to
1 for each.
On Thu, Apr 12, 2012 at 11:51 AM, Stiles Watson
wat...@datatek-net.comwrote:
According to the sipX book (p
.
** **
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Tony Graziano, Manager
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sip: tgrazi...@voice.myitdepartment.net
Fax: 434.465.6833
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...@list.sipfoundry.org] *On Behalf Of *Tony Graziano
*Sent:* woensdag 11 april 2012 14:43
*To:* Discussion list for users of sipXecs software
*Subject:* Re: [sipx-users] voicemail greeting does not change
** **
if it does not change via the webgui, sipxconfig should be throwing and
error in its
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Telephone
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