The SIP ALG in my VPN router ( netgear FVX538 ) was already disabled and the
STUN/ICE of my remote worker client, eyebeam sofphone, is just disabled.
Whereas, the issue remains as before - only single direction voice.
I cannot disable NAT in my router since it will affect other people behind
the s
The t.38 support for hylafax is highly experimental. I don't know of anyone
actually using it successfully via a siptrunk. The t.38 modem software is a
nice idea, but noone has really worked on it for a while. I certainly hope to
see it someday.
>>> "Paul Herron" 07/20/09 9:25 PM >>>
There
A lot of gateways can route certain DID numbers to another system. A patton
E1/T1 can do a sip route to a FXS device connected to a fax machine, for
example.
>>> "Paul Herron" 07/20/09 9:38 PM >>>
_
From: Todd Hodgen [mailto:t.hod...@misiusystems.com]
Sent: Monday, July 20, 2009 9:30
Yes. If the two LAN are interconnected without a firewall or via a VPN, you can
set up sip trunking between the two sipxecs servers. It's easier if they have
different extension ranges (2xx and 3xx), and use a dialing rules to route 2xx
numbers from the other end to the system that handles those
_
From: Todd Hodgen [mailto:t.hod...@misiusystems.com]
Sent: Monday, July 20, 2009 9:30 PM
To: Paul Herron
Subject: RE: [sipx-users] how to do Fax in SipXecs
Or, there is an integrated Fax Server for the M1000 if that isnt
overkill for you. Or, how about an FXS card that you can dir
There is an issue open in the tracker for this feature (XX-4989). If
you are interested in it, vote for it.
-Original Message-
From: Picher, Michael [mailto:mpic...@cmctechgroup.com]
Sent: Monday, July 20, 2009 5:47 AM
To: winson (Elabram); sipx-users@list.sipfoundry.org
Subject: R
Means SipXecs not provide any Fax service?
Because want use an E1 / T1 line to connect our VOIP and receive fax as
well.
Example : PSTN --> m1000 audiocodes gateway (E1 card) --> server
Winson
Picher, Michael wrote:
> No faxing capabilities in sipXecs.
>
> You could look to a dedicated fax de
Hi all,
i am installed two IPPBX servers in separate LAN's. Is there any way to test
SIP trunking between two local sipx servers?
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubs
this is doesn't cause the problam in my casethe problam is caused by
font directory permissions. Buy doing touch * in the directory
/usr/share/fonts solved the problam.
On Mon, Jul 20, 2009 at 1:59 PM, Keith Gearty wrote:
> naga raju wrote:
>
> Hi all,
> I am installed sipx 4.0 successful
On Mon, 2009-07-20 at 16:31 -0400, Derek Bartolo wrote:
> Hey there I am trying to dial out using a MP114 FXO Gateway. I believe
> I have entered all the correct settings etc. on the status home page
> port one is set to inactive and no longer disconnected. I have also
> created a custom dial plan
Hey there I am trying to dial out using a MP114 FXO Gateway. I believe I
have entered all the correct settings etc. on the status home page port
one is set to inactive and no longer disconnected. I have also created a
custom dial plan using the manual as a reference and have moved the dial
plan rig
Is there such a device?
I am looking for an FXO or FXS devices that can be run off the USB ports.
Thanks
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://
On Mon, 2009-07-20 at 12:04 -0400, Derek Bartolo wrote:
> Hey there, when I connect a polycom phone, after creating a user,
> creating a phone, and setting a line to a user I boot the phone up and
> when I go to pick up the phone it displays “url call is disabled” with
> no dial tone. I checked the
Hey there, when I connect a polycom phone, after creating a user,
creating a phone, and setting a line to a user I boot the phone up and
when I go to pick up the phone it displays "url call is disabled" with
no dial tone. I checked the registrations and it is not yet registered
but after some time
>Hi, How to receive fax form PSTN to Mail server in SipXecs?
>--Winson
In Italy I got good results with evolve appliance + audiocode , but it's
commercial: www.empixevolve.com
Don't know if there is anyone that successfully did it with hylafax.
Heros
_
>
> Yes I observed x-sipx-nonat of remote worker in the contact
> of registration menu. Let me check the settings of STUN/ICE
> of remote worker. Do I also need to turn off STUN/ICE of
> called party as well ?
The remote NAT traversal feature is a complete end-to-end one and does
not require a
On Sun, 2009-07-19 at 16:24 -0500, robertNGN wrote:
> Hi
>
> I reinstalled fresh 4.10 ver and sipx proxy fails. sipx-dns show
> this. Can anyone point to solution?
You did not configure your DNS with the various records it needs (or did
not put them where the sipXecs system is sending the quer
On Sun, 2009-07-19 at 16:53 -0500, robertNGN wrote:
> Has anyone tried running sipXbridge with inGate firewall providing NAT
> between eth0 and eth1 gateways?
>
> Our ITSP needs to orginate traffice to public inGate (eth1) port:5080
> to accomodate sipXbridge but then terminates back to ITSP on
>
Yes I observed x-sipx-nonat of remote worker in the contact of registration
menu. Let me check the settings of STUN/ICE of remote worker. Do I also need
to turn off STUN/ICE of called party as well ?
-Original Message-
From: Robert Joly [mailto:rj...@nortel.com]
Sent: Monday, July 20, 200
On Sat, 2009-07-18 at 16:56 -0500, arda savran wrote:
> I have been trying to cluster a couple of SIPx server. When I check
> the master unit, I can see the secondary unit registered to it with
> the following services up and running:
>
> 1)CDR HA tunnel
> 2)Media Services
> 3)SIP Trunking
> 4)Me
Ok, next thing to check then is whether or not there is some SIP-aware
equipment at the remote worker site. The simplest way to tell is to
check the registration of the remote user under
Diagnostics->Registrations. If you see a X-sipX-nonat in the contact,
this indicates that the phone or remote
Yes all of my three intranet subnets, 192.168.2.0/24, 192.168.8.0/24 and
192.168.10.0/24, were already set in the "Intranet Calling".
-Original Message-
From: Robert Joly [mailto:rj...@nortel.com]
Sent: Monday, July 20, 2009 9:57 PM
To: jun,wen; sipx-users@list.sipfoundry.org
Subject: RE:
Please make sure that your 192.168.8.0/24 and 192.168.10.0/24 appear in
your intranet subnet list under System->Intranet Calling. The rule of
thumb here is that the Intranet Subnets section should contain all the
subnets that describe your intranet.
>
> Hi, I have three intranet subnets by a cen
I once tried to disable SIP features in InGate to turn it into a
non-SIP-aware vanilla Firewall but I found that I couldn't find the
recipe that would totally kill everything that is SIP-related inside the
box - disabling the SIP module it surely not enough. Perhaps it can be
done but I would sugg
> Subject: [sipx-users] Upgrade from 3.10.2 to 4.x latest via YUM
>
>
> Dear All,
>
> I am planning to upgrade a few installations running on
> 3.10.2 to latest 4.x (what is the
> latest-known-to-be-running-well version btw?) via yum.
>
> I've tried installing a test system with the latest 4
No faxing capabilities in sipXecs.
You could look to a dedicated fax device like the FaxFinder from
Multitech.
Mike
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of winson
(Elabram)
Sent: Sunday, July 19, 2009 9:
naga raju wrote:
Hi all,
I am installed sipx 4.0 successfully. Then i tried to add some users
by using web ui. After clicking apply button by entering user details
i am getting error like " An internal /error/ has occurred please
click here to continue". if i click the continue button it's lo
Hi, I have three intranet subnets by a central site with sipx server
(192.168.2.0/24 ) and two of branch sites (192.168.8.0/24 and
192.168.10.0/24) with VPN tunnel interconnections of hub-and-spoke.
The sip phones inside these three intranet subnets can make call to each
other without problem, and
28 matches
Mail list logo
