MOH is changed dramatically in 4.2, and follows your request almost verbatim
from what I have seen. I would suggest downloading and playing with it.
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis
S
How many total sites on this network?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
mkitchin.pub...@gmail.com
Sent: Monday, February 01, 2010 6:14 PM
To: Tony Graziano
Cc: sipx-users@list.sipfoundry.org
Subject:
Sorry to come to the party late. Been a busy past few weeks.
I'd like to add my $.02. Or $.05, as it were.
1) I would like to see a graphic representation of a handset's digit map
before committing the change to the handset. As much as I love to fly
by "trial and error", sometimes I just don't
Details on the test and environment are below, but the general question is:
should BLA work at all?
I'm happy to gather data and debug in detail but before I do I'm just curious
if BLA still too green to bother testing with.
I've read this page
http://sipx-wiki.calivia.com/index.php/Bridged_L
No. Per their setup requirements, I don't register. I have the option
unchecked under ITSP account.
Correct. They just route to the IP we provided them.
On 2/1/2010 8:07 PM, Tony Graziano wrote:
> Verizon isn't supporting registrations are they? I think they are just
> sending the DID toa specifi
Verizon isn't supporting registrations are they? I think they are just
sending the DID toa specific ip address.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Te
Thanks. Tools for my toolbox is what I need right now.
I guess I'm looking for that perfect mix of decentralized vs
centralized. I need to have as little as possible running at the remote
facility. This goes way beyond just telephony. With that being said I'm
going to take the points presented b
On Mon, Feb 1, 2010 at 7:35 PM, Jim Canfield wrote:
> 2010/2/1 Bryan Simmons
>
> Has anyone found and used a soft phone with a customer that utilizes
>> citrix thin clients or any other thin client device successfully?
>>
> This might be an appropriate use for application streaming. I would
2010/2/1 Bryan Simmons
> Has anyone found and used a soft phone with a customer that utilizes
> citrix thin clients or any other thin client device successfully?
>
This might be an appropriate use for application streaming. I would have
concerns with audio quality running as a traditional app
In the polycoms (and I presume pretty much any other multi-line phone) you can
register external lines that register direct from the handset to a voip
provider. SipXecs manages the phone configuration in this case but otherwise
doesn't get involved. The normal sipXecs lines continue to work norm
I have not. I would like to see a client webapp that uses jav to embed the
sipclient in the web page...
As with any sip call, you have to worry about a delay in media for it to be
worthwhile and successful. Without some ridiculous resources, its never made
sense to me.
Has anyone found and used a soft phone with a customer that utilizes
citrix thin clients or any other thin client device successfully?
Our company accepts no liability for the content of this email, or for the
consequences of any actions taken on the basis of the information
provi
>>> "mkitchin.pub...@gmail.com" 02/01/10 5:24 PM >>>
>>>By the way, can you share what flash device you have had good luck with?
We worked with an ODM to design a fan-less "appliance". We use it for all of
our SipX deployments as well other "appliance" deployments we do. I've
attached an imag
By the way, can you share what flash device you have had good luck with?
On 2/1/2010 4:09 PM, Matt White wrote:
>> It sounds to me your looking to do what SipX did before 4.0 and sipxbridge.
>>
>> That is to say, when you do NOT enable SipxBridge, Sipx does not handle the
>> media.
>>
>> You are
There are fax machines that are on POTS lines and a metered line that
can also be used with a traditional handset on it (that would also work
in a power failure).
I do understand what you are saying. Somethings are not my decision though.
On 2/1/2010 4:16 PM, Tony Graziano wrote:
Still, you hav
On 2/1/2010 4:09 PM, Matt White wrote:
> It sounds to me your looking to do what SipX did before 4.0 and sipxbridge.
>
> That is to say, when you do NOT enable SipxBridge, Sipx does not handle the
> media.
>
> You are then required to have some SBC device to handle the NAT/anchoring of
> media.
Still, you have no real "survivability" from the phones. You should ask your
"boss" to consider the health and human safety factor in this type of
decision. You could use a local gateway and program the phones to use it to
dial "911". You could place a sipx system, even what Matt is suggesting, at
On Mon, Feb 1, 2010 at 5:03 PM, mkitchin.pub...@gmail.com <
mkitchin.pub...@gmail.com> wrote:
> Thank you. So, it this completely a function of the SBC? If I used an SBC
> that supported this feature, should it work from a Sipx point of view then?
> I'm not saying I would go this route, but I jus
>>
>>
>>
>> He wants sipx in the HQ with the remote branch phones registering to
>> it, using sipXbridge as the gateway for the trunks, but with the media
>> anchored at the handsets in the branches to the Verizon Gateway (not
>> to sipXbridge).
>
> Right... the feature
Thank you. So, it this completely a function of the SBC? If I used an
SBC that supported this feature, should it work from a Sipx point of
view then?
I'm not saying I would go this route, but I just want to see what my
options are. If I'm going to consider putting a device at each facility,
th
On Mon, 2010-02-01 at 13:38 -0800, Nathan Nieblas wrote:
> I am running into some oddities with my IVR. A few weeks ago we
> noticed that our Business Hours IVR is picking up calls outside normal
> business hours. Looking through the logs on the system, it somehow
> figured that Saturday during 7:3
I had the exact same issue. I had Saturday and Sunday unchecked under
working hours. They had the default times in the from and to. I noticed
the working time attendant was picking up on Saturday. I left them
unchecked, but tried putting in 12:00 AM for both the From and To. I
applied it and ev
On Mon, 2010-02-01 at 16:09 -0500, Tony Graziano wrote:
>
>
> On Mon, Feb 1, 2010 at 3:49 PM, Scott Lawrence
> wrote:
> On Mon, 2010-02-01 at 14:25 -0600, mkitchin.pub...@gmail.com
> wrote:
> > Does anyone know if this is even a common feature among SIP
> servers?
I'm not sure that is exactly it. The last email I sent had a lot more info.
I believe the first answer in this link is along the lines of what I'm
looking for.
http://www.voip-info.org/wiki/view/Asterisk+sip+directrtpsetup
On 2/1/2010 3:37 PM, Todd Hodgen wrote:
In other words, they want phon
I am running into some oddities with my IVR. A few weeks ago we noticed
that our Business Hours IVR is picking up calls outside normal business
hours. Looking through the logs on the system, it somehow figured that
Saturday during 7:30a to 5:00p is in my M-F schedule that is configured
on the AA an
In other words, they want phones to use a non-existent SBC instead of the
SipXbridge, right?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Monday, February 01, 2010 1:10 PM
To: Scott Lawrence
Cc: sipx-users@list.si
On 2/1/2010 2:49 PM, Scott Lawrence wrote:
> On Mon, 2010-02-01 at 14:25 -0600, mkitchin.pub...@gmail.com wrote:
>
>> Does anyone know if this is even a common feature among SIP servers?
>> Verizon is indicating they have never worked with a client setup that
>> didn't support this. Maybe somet
On Mon, Feb 1, 2010 at 3:49 PM, Scott Lawrence wrote:
> On Mon, 2010-02-01 at 14:25 -0600, mkitchin.pub...@gmail.com wrote:
> > Does anyone know if this is even a common feature among SIP servers?
> > Verizon is indicating they have never worked with a client setup that
> > didn't support this. Ma
What version sipx? What version polycom firmware?
On Mon, Feb 1, 2010 at 3:54 PM, thomas peterseil wrote:
> the phone is a polycom 450 and the gateway is a portech MV-372. for me
> it's strange, because i can make outbound calls directly from the sipx
> (from the conference bridge) but not from
the phone is a polycom 450 and the gateway is a portech MV-372. for me it's
strange, because i can make outbound calls directly from the sipx (from the
conference bridge) but not from the polycom, even when all devices are in the
same network.
thank you very much!
thomas
Original-N
On Mon, 2010-02-01 at 14:25 -0600, mkitchin.pub...@gmail.com wrote:
> Does anyone know if this is even a common feature among SIP servers?
> Verizon is indicating they have never worked with a client setup that
> didn't support this. Maybe something is getting lost in translation
> here.
I'm not c
On Mon, 2010-02-01 at 19:13 +0100, thomas peterseil wrote:
> hello,
>
> thank you very much for your help, now the phone is in the same
> network as the sipx and the gateway and i still have the same problem:
>
> 697.858001 10.1.1.169 -> 10.1.1.85SIP/SDP Request: INVITE
> sip:069910696...@
Does anyone know if this is even a common feature among SIP servers?
Verizon is indicating they have never worked with a client setup that
didn't support this. Maybe something is getting lost in translation here.
On 2/1/2010 1:01 PM, Tony Graziano wrote:
I don't think you can do what you are tr
Thanks for the response Scott,
> Well, let's start with that - first, what version (exactly) of
> sipx-trace are you using? Where did you get it (you didn't install on a
> Mac from an RPM :-) ).
I copied it (the shell script) over to the Mac from the sipx server on Centos.
Are there other thin
maybe the wrong way around, but if I call a number, and that number is
on a sip based system (broadsoft, others) and the number I call is
forwarded, 99% of the time, I get reorder.
example: I call my direct number from a sipx system, to a broadsoft
voip system,and it has call fwd no answer.
I understand. Some things are not my call. I don't always get to do
everything the way I think is best.
On 2/1/2010 1:01 PM, Tony Graziano wrote:
I don't think you can do what you are trying to do with sipx.
sipxbridge is an anchor. If you need to "reanchor" it to a different
location you need
I don't think you can do what you are trying to do with sipx. sipxbridge is
an anchor. If you need to "reanchor" it to a different location you need
another installation of it.
While you might centralize it, some things don't make sense from a failsafe
or operational standpoint to centralize. To m
Or you can use an analog gateway for 911 so all the phones can use it,
connecting from their local sipx installation.
On Mon, Feb 1, 2010 at 1:52 PM, mkitchin.pub...@gmail.com <
mkitchin.pub...@gmail.com> wrote:
> There will be a single POTS phone on a metered line for emergency at each
> facili
Thanks. Our entire IT model is around centralization, so we wanted to
keep as much of this centralized as possible. These are small offices
with no IT staff to assist in any hardware failure. Even if I have to
put something running sixbridge local, I would still likely set the
phones to registe
There will be a single POTS phone on a metered line for emergency at
each facility. We already have backups to the actual MPLS circuits. If
the MPLS circuit is down, we have biggere issues that the phones being
down, so that risk is OK, and there are backup phones.
Our users are remote as in t
No. There needs to be a system running sipxbridge at the remote site to
anchor the media. At the same time, this would be used to register the local
phones and be the voicemail system for it. They should be "planned" to be
able to dial each other, etc.
On Mon, Feb 1, 2010 at 1:43 PM, mkitchin.pub.
after thinking on it some more, clarification might help: because none of
this is behind NAT your remote users are not "remote", so anchoring the
media is not an option because i'm not sure it is necessary.
Enables NAT traversal capabilities in support of remote workers and remote
servers behind N
Thanks. The primary focus isn't for dialing from site to site. It is
trying to keep from having to come through our server at corporate when
a remote facility is calling or receiving a call to/from the PSTN. We
don't want to have remote sipx systems if we can help it. These are
small remote fac
sipXbridge is a media anchor.
You would connect remote sipx systems and connect them to verizon as you
have your central system. You would then create a dialplan to allow them to
route calls directly between each other (sipx to sipx).
On Mon, Feb 1, 2010 at 1:27 PM, mkitchin.pub...@gmail.com <
m
Excuse my ignorance here if I'm butchering some terminology. After a
pretty successful roll out at our corporate office, we are looking
possible implementations at some of our remote facilities that have old
dying key systems. All our facilities are connected to our corporate
office by Verizon
hello,
thank you very much for your help, now the phone is in the same network as the
sipx and the gateway and i still have the same problem:
697.858001 10.1.1.169 -> 10.1.1.85SIP/SDP Request: INVITE
sip:069910696...@xxx.xxx.cc;user=phone, with session description
697.86124210.1.1.85
On Mon, 2010-02-01 at 10:53 -0500, Matt White wrote:
>
>
> >> Does anyone have any better ideas how to make this task more
> automatable?
>
> >Several :-) ... and while I don't think they'll make it into the
> first
> >release of 4.2, they are in progress...
>
>
> The SipX interop site has an
>> Does anyone have any better ideas how to make this task more automatable?
>Several :-) ... and while I don't think they'll make it into the first
>release of 4.2, they are in progress...
The SipX interop site has an experimental web based trace option. Has there
been any thoughts on rol
On Feb 1, 2010, at 6:57 AM, Scott Lawrence wrote:
> On Mon, 2010-02-01 at 09:49 -0500, Jeff Gilmore wrote:
>> This works, but has the minor problem that sipviewer does not accept a
>> filename parameter, so I have to manually navigate to and open the
>> trace file.
>
> It should accept a file name
if you set it in your modem then your modem is not in a bridged mode. That
also means you have 2 active routing/nat and port aware elements, which is
not a good idea.
I'd suggest setting your modem to bridged mode and putting your public ip
address on your firewall directly.
==
Last 2 day my colleague bring one Sonicwall back.. i know my nightmare
come true..
i base on this http://sipx-wiki.calivia.com/index.php/Firewall_Configuration
will know more about the port setting
but i still have some issue can ask for advice?
basically my network diagram is :
internet -
On Mon, 2010-02-01 at 09:49 -0500, Jeff Gilmore wrote:
> Hi folks,
> I am trying to set up a simple way to use sipviewer from an external
> system (in this case MACOS OSX) to conveniently look at calls. I read
> the wiki page on this, but was not able to get a fully satisfactory
> solution.
I've
Hi folks,
I am trying to set up a simple way to use sipviewer from an external system (in
this case MACOS OSX) to conveniently look at calls. I read the wiki page on
this, but was not able to get a fully satisfactory solution.
I have configured ssh with a key so that I can execute commands remo
> Unless you specify an SBC other than sipXbridge for Internet
> calling, enabling it using sipXbridge breaks remotes users in
> some functions (cant receive calls, but can make calls, for example).
This is definitely not how it is supposed to work. We use such a
configuration every day here w
>
> I've seen a bunch of traffic on the list implying that to use
> an ITSP you have to turn off 'Internet calling'.
Correct, when sipXbridge is used, Internet callign should not be used.
>
> Should I be able to follow the cheat sheet on the wiki wrt
> remote workers and have it not break m
On Mon, 2010-02-01 at 07:00 -0500, Michael Scheidell wrote:
> On 2/1/10 4:49 AM, thomas peterseil wrote:
> > Dear SipX-Users,
> >
> > i am still "playing" around with the Portech GSM Gateway and i have a
> > interesting problem. i can make outbound calls via the gateway from the
> > conferencin
On 2/1/10 4:49 AM, thomas peterseil wrote:
Dear SipX-Users,
i am still "playing" around with the Portech GSM Gateway and i have a
interesting problem. i can make outbound calls via the gateway from the conferencing
bridge, but i cant make outbound calls with my polycoms. i made a tshark while
Dear SipX-Users,
i am still "playing" around with the Portech GSM Gateway and i have a
interesting problem. i can make outbound calls via the gateway from the
conferencing bridge, but i cant make outbound calls with my polycoms. i made a
tshark while both calls and here is the result:
147.4480
>
> The media path drops after 30 seconds because the call "probably" never
> acknowledges correctly back to the provider so it is disconnected. Your
> firewall needs to have any ALG turned off and static port NAT turned on.
In this test there is no provider, I'm outside the firewall on my MAC
Thanks for the reply!
> If you are using sipXbridge to connect to your ITSP, then you should
> turn off the 'Internet calling' checkbox - it is for configuring an
> external SBC (we're trying to work out a clearer set of configuration
> screens for all this).
That's the primary data point I was a
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