Hi All,
Good Answer Scott. )
But at my end when I spell by name I got the message that spelled name is
not in the directory.
Still I have the entry for that.
Do any setting I have to enable to avail the dial by name facility.
I am spelling 'JOY'
extension : 806
Regards,
Vikas
On Wed, Jun 4, 2
On Tue, 2008-06-03 at 13:38 -0700, IT Services wrote:
> Hi all:
>
> The dial by name feature works great, but I would prefer if Dial by Name
> is by FIRST NAME rather than last name.
>
> How do I modify this feature?
Well, the easy way would be to just put last names in the first name
field and
Great News!
Using trixbox in-line actually re-originates the call, and I am able to
successfully complete an outgoing call. Now I have to fight with direct
media, and If I can accomplish that, I will have a temporary solution in
place to begin rolling out sipx in production.
A suggestion for the
I just installed the sipx, and like to have some basic test drive.
Although the document of sipx is full of details, but it lack some
kind of "quick start guide", what I am looking for is:
1. quick start guide of how to set up couple of users, and make the
testing call.
2. how to make the simpl
I tried with 3.10 and this works fine. You just create a user, but do
not assign a phone to that user. Calling the extension transfers the
call to VM immediately.
Is that what you wanted to do?
--martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ton
Hi all:
The dial by name feature works great, but I would prefer if Dial by Name
is by FIRST NAME rather than last name.
How do I modify this feature?
THANKS!
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We are concentrated on this list right now. There are both a users and a
dev list. The main reason is that for most developers it is too much
work to stay current in too many different places.
--martin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim
The PBX is on a protected private network, and only listens from trusted
addresses. My primary goal is to get it working initially, and I'll add
permissions later.
My SBC is set up as an unmanaged "SIP Trunk" Gateway, and I have a
custom dial plan which looks for 9 and 11 digits, then forwards the
Hi,
I have a setup with multiple ACD queues and multiple agents in more
then one queue at a time.
I see no way of prioritizing queues as such but there is an option
of moving queues up and down in context of an agent. What exactly
does that option do ?
I've made several tests
On Tue, 2008-06-03 at 15:43 -0400, Hegner, Travis wrote:
> I stopped appending the + to no avail, I still get three via headers. I
> changed my dial plan from custom to the default "Long Distance" plan,
> and that actually added a 220+ byte "Proxy Authentication:" header, so I
> switched back to a
Which version are you running?
>>> Tim Booth <[EMAIL PROTECTED]> 06/03/08 03:55PM >>>
Is it possible to make ghost extensions with sipx?
Extensions without phones so it acts a just a voice mail box.
I have tried and you cannot ring the extension at all, I thought the
call
would go straight to voic
Is it possible to make ghost extensions with sipx?
Extensions without phones so it acts a just a voice mail box.
I have tried and you cannot ring the extension at all, I thought the call
would go straight to voice mail.
thanks
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I stopped appending the + to no avail, I still get three via headers. I
changed my dial plan from custom to the default "Long Distance" plan,
and that actually added a 220+ byte "Proxy Authentication:" header, so I
switched back to a custom dial plan.
If I can eliminate even one of the three heade
Are the any know forums out there for sipx similar to something like the
trixbox forums or is it just this list?
Thanks
Tim Booth
VisionCom
MaineVoIP Systems
/(207)321-2789/
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OK, I am in fact checking for a 9 as a prefix, dropping that and
appending a + before sending out the number. My provider claims to
require the plus, but I think it will work without it. I can try
removing that. In the mean time, here is my mappingrules.xml.in:
http://www.sipfoundry.org/sipX/sch
On Tue, 2008-06-03 at 14:51 -0400, Hegner, Travis wrote:
> Using a combination of an OpenSER Proxy as an SBC, and cisco routers at
> my branch offices, I can completely traverse all NATS in my network. I
> have tested and proven this theory successfully.
>
> My branch office phones register to my
Using a combination of an OpenSER Proxy as an SBC, and cisco routers at
my branch offices, I can completely traverse all NATS in my network. I
have tested and proven this theory successfully.
My branch office phones register to my pbx with an internal private
address over an ipsec vpn tunnel. Open
On Tue, Jun 3, 2008 at 2:12 PM, M. Ranganathan <[EMAIL PROTECTED]> wrote:
> On Tue, Jun 3, 2008 at 12:28 PM, Hegner, Travis <[EMAIL PROTECTED]> wrote:
>> My provider is bandwidth.com. Unfortunately my environment also requires the
>> use of a SBC, which is adding even more with record-route and vi
On Tue, Jun 3, 2008 at 12:28 PM, Hegner, Travis <[EMAIL PROTECTED]> wrote:
> My provider is bandwidth.com. Unfortunately my environment also requires the
> use of a SBC, which is adding even more with record-route and via headers.
>
> My first plan of attack is to set up a trixbox server to proxy
M. Ranganathan wrote:
> On Tue, Jun 3, 2008 at 11:30 AM, Tony Graziano
> <[EMAIL PROTECTED]> wrote:
>> Unless you hve a device that can do NAT traversal with SIP and do
>> siptrunking too, it is not available in 3.10. Most people use an ingate
>> siparator for this now.
>>
>> In the upcoming 4.0 ve
1) Moving further discussion to sipx-dev
2) Ranga: Thanks, I'll look for the next build.
3) I upgraded to 3.11 to do some testing with sipXbridge in my
environment. I understand it is alpha.
Sorry for starting this in the wrong group though.
Mike
M. Ranganathan wrote:
> On Tue, Jun 3, 2008 a
On Tue, Jun 3, 2008 at 11:30 AM, Tony Graziano
<[EMAIL PROTECTED]> wrote:
> Unless you hve a device that can do NAT traversal with SIP and do
> siptrunking too, it is not available in 3.10. Most people use an ingate
> siparator for this now.
>
> In the upcoming 4.0 version (development version, uns
On Tue, Jun 3, 2008 at 11:30 AM, Luis F Urrea <[EMAIL PROTECTED]> wrote:
> Hi all,
>
> We want to test sipxbridge and I wanted to get some input from your general
> experience on what a good ITSP would be.
>
> Basically using good support and good quality/availability as the main
> criteria.
I ha
My provider is bandwidth.com. Unfortunately my environment also requires the
use of a SBC, which is adding even more with record-route and via headers.
My first plan of attack is to set up a trixbox server to proxy through as a
b2bua. If it re-originates the call, then I think it may work as a t
Sergiy Genyuk wrote:
> Is it possible to get it bigger?
>
> Thank you,
Which version are you using?
sipx-config --version
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Hi all,
We want to test sipxbridge and I wanted to get some input from your general
experience on what a good ITSP would be.
Basically using good support and good quality/availability as the main
criteria.
Thanks for all your input.
Regards,
Luis
__
On Tue, Jun 3, 2008 at 11:25 AM, Scott Lawrence <[EMAIL PROTECTED]> wrote:
>
> On Tue, 2008-06-03 at 10:03 -0400, Michael Cote wrote:
>> I have been running sipX under CentOS 5. I started with a 3.8
>> installation, upgraded to 3.10 and
>> had everything working fine. Yesterday, I upgraded (via
On Tue, 2008-06-03 at 10:03 -0400, Michael Cote wrote:
> I have been running sipX under CentOS 5. I started with a 3.8
> installation, upgraded to 3.10 and
> had everything working fine. Yesterday, I upgraded (via the RPMs) to
> 3.11.3
Why?
>From the download page:
The builds in
Unless you hve a device that can do NAT traversal with SIP and do
siptrunking too, it is not available in 3.10. Most people use an ingate
siparator for this now.
In the upcoming 4.0 version (development version, unstable), there is a
project called sipXbridge which has this type of functionality,
On Tue, 2008-06-03 at 09:46 -0400, Hegner, Travis wrote:
> Hello All,
>
>
>
> I am having an issue with sipx where my udp invite packet is too
> large. While everything locally appears to be handled correctly with a
> fragmented udp packet, my provider is refusing to support it. They
> also ar
New to the list so I hope I'm posting in the correct area.
I just installed sipx 3.10, looks great.
I have 2 snom's registered without any issues at all.
I have not figured out how to register a trunk with and istp,
specifically Varphonex.
Any one have a Varphonex trunk registered? Can you share y
Thanks everyone for the advice. I think I'll try my luck with FC. I'd
prefer to start development on the most supported platform, and then
move into other platforms as I get more comfortable with it.
Thanks,
Travis
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On B
Hegner, Travis wrote:
> Hi All,
>
>
>
> I just loaded a CentOS 5.1 box and I am following the “setting up a
> development environment” directions at the sipx wiki. For some reason, I
> can’t seem to install cppunit or cppunit-devel (perhaps other packages
> as well) from the repositories.
>
cppunit used to build as part of our lib builds. Look at main/lib
However, looking at
http://sipxecs.sipfoundry.org/ViewVC/sipXecs/main/lib/Makefile?view=mark
up cppunit no longer gets built for any platforms. Does someone know
why? Should this be added back into the targets rhel5, centos5 and
Thats a dev version, and probably needs to be posted to that list. In
the meantime, here's an answer, a fix should be forthcoming as it was
already discovered and a workaround posted on sipx-dev.
___
>>> Damian Krzeminski <[EMAIL PROTECTED]> 05/30/08 11:09 AM >>>
Tony Graziano wrote:
> I just
I have been running sipX under CentOS 5. I started with a 3.8
installation, upgraded to 3.10 and
had everything working fine. Yesterday, I upgraded (via the RPMs) to
3.11.3 and now the sipXconfig
UI will not work. The services all appear to be running and my actual
proxy and auto-attendan
Nikolay,
Thanks for the response. I browsed through the main repo on
sipxecs.sipfoundry.org and never found the cppunit rpm available, so I
assumed that it would be available in the standard/default centos repos.
After browsing around in the 3.8 repo, as you mentioned, I did see the
cppunit rpm
Hello All,
I am having an issue with sipx where my udp invite packet is too large.
While everything locally appears to be handled correctly with a
fragmented udp packet, my provider is refusing to support it. They also
are not ready for sip over tcp yet. Upon examination of my invite
packets, I
>
>Subject: Re: [sipx-users] Grandstream Handytone 386
>
>I checked that this was true with sipx 3.8.1. As far as I understand
the
>problem is that sipx generates wrong configuration file for ht386.
>I did not check it with 3.10, but release notes for sipx 3.10 do not
say
>that grandstream plugins
Hi Travis,
I had the same problem some time before while installing dev. environment.
I'm not a developer, but dev. environment was needed for UI translation.
I installed cppunit and cppunit-devel from 3.8 repository. It worked for me
(I needed dev. environment only for translation). I understand
Hi All,
I just loaded a CentOS 5.1 box and I am following the "setting up a
development environment" directions at the sipx wiki. For some reason, I
can't seem to install cppunit or cppunit-devel (perhaps other packages
as well) from the repositories.
[EMAIL PROTECTED] yum.repos.d]# yum sea
Take a look at the System -> General -> Presence Server page at the UI.
These are sign in/out codes for acd agents.
HTH,
Nikolay.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of VG
Sent: Tuesday, June 03, 2008 3:20 PM
To: sipx-users@list.sipfoundry.org
Subject: [sip
Hello All,
I like to know what is the significance of *86 and *88 in Sipx.
When i opened my alias.xml i found these two entry there.
Do these are special msg used in sipx
Regards,
Vikas
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Yes, it seems to generate an incorrect file (outbondproxy on 2nd line is
always incorrect), and manually only dials internally on either line.
>>> "Nikolay Kondratyev" <[EMAIL PROTECTED]> 06/03/08 03:55 AM >>>
I checked that this was true with sipx 3.8.1. As far as I understand the
problem is tha
I checked that this was true with sipx 3.8.1. As far as I understand the
problem is that sipx generates wrong configuration file for ht386.
I did not check it with 3.10, but release notes for sipx 3.10 do not say
that grandstream plugins were updated...
HTH,
Nikolay.
> -Original Message-
>
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