Hi,
as we want to publish a simple free ClickToDial tool for sipx on our website
I wonder if I have to take care about a trademark or something?
Are there some simple rules that I may follow? If there are some rules it
would be a great idea to put them into the (developer-) wiki.
Of course we
There is some information about that:
http://sipx-wiki.calivia.com/index.php/How_to_configure_User_Call_Forwarding
2010/9/7 Worley, Dale R (Dale) dwor...@avaya.com
From: sipx-users-boun...@list.sipfoundry.org [
sipx-users-boun...@list.sipfoundry.org]
Hi all,
At the moment sipx UI statistics page shows only usage of / partition.
It is quite useless, because logs and voicemails are in /var.
And it is /var partition that needs to be monitored.
In some installations other partitions may exist.
There is rather old issue
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Hi Douglas,
Yes I have read the
On Tue, Sep 7, 2010 at 11:53 PM, Dave Redmore
dave.redm...@spigotsystems.com wrote:
My settings for the gateway are all default - Under Configuration, I
defined Address as den.teliax.net - Under CallerID I set the
Default Caller ID to my incoming phone number - under ITSP Account I
defined
Login to the bbox modem (yes there is a way) via telnet or ssh...
do a top and kill the process caller
tr97
tr98
sipd
See this post.
http://patrick.vande-walle.eu/category/bbox-2/
There should also be a way to
chkconfig sipd off
and keep it from turning on at bootup. Be creative. Take
further...
here's a link on a how to keep those process from ever starting...
http://wildcat.espix.org/doc/bbox2
On Wed, Sep 8, 2010 at 6:26 AM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
Login to the bbox modem (yes there is a way) via telnet or ssh...
do a top and kill the process
There is an assumption being made here the sip alg on bbox2 is made to work
with any sip provider. If this were the case it would allow you some basic
configuration. It does not.
bbox2 runs sipd
http://wildcat.espix.org/doc/bbox http://wildcat.espix.org/doc/bbox2/2
and is meant by the provider
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Hi Tony,
Yes, I did it already
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For the SIP configuration, It was
I suggest you disable sipd. The tr68/tr69 pid's are a problem, because it is
possible they can update the firmware and then lock you out from making
changes later on. I also noticed they use a considerable amount of CPU.
The sipd process is really geared towards their SIP service, so I doubt that
If your provider says IT WILL work that you can have your own sip service
using their sipd, they should make it work. If they are a public utility
they should provide assistance. IF you log in to the bbox2 and change the
SIP port to 5080, with sipd running, does that fix your audio? I suspect it
the setting is under
Advanced Settings, Telephone...
On Wed, Sep 8, 2010 at 8:45 AM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
If your provider says IT WILL work that you can have your own sip service
using their sipd, they should make it work. If they are a public utility
they should
Reading the specs for the bbox 2 it has fxs ports. Using sipd I suspect
they are sending RTP to the fxs port, resulting in one way audio. I don't
think ANY SBC CAN OVERCOME THIS with sipd running!!!
Changing the FXS SIP port to 5080 SHOULD allow both services to peacefully
co-exist.
I for one
On Wed, Sep 8, 2010 at 3:32 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Hi all,
At the moment sipx UI statistics page shows only usage of / partition.
It is quite useless, because logs and voicemails are in /var.
And it is /var partition that needs to be monitored.
In some installations other
Yes, but it would be better if there were no need to edit files manually.
May be UI statistics configuration page may ask admin which partition to
monitor?
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
What I was thinking. See the comments on XX-5271 just added.
On Wed, Sep 8, 2010 at 9:53 AM, Nikolay Kondratyev k...@nstel.ru wrote:
Yes, but it would be better if there were no need to edit files manually.
May be UI statistics configuration page may ask admin which partition to
monitor?
Does it contain any pieces of sipXecs itself ? Than you are obliged by the
LGPL. Not sure about the trademark sipXecs which belongs I think to Sipfoundry
now so Martin would be the right person to answer.
Am 08.09.2010 um 09:18 schrieb Rene Pankratz:
Hi,
as we want to publish a simple free
That was my initial sipX setup as well (except I had Auth User set equal
to User).
On the Teliax side under device settings did you do either of the following?
* enable DNIS so they send the number instead of the user in the SIP
INVITE?
* enter your pubilc IP
The reason I ask is
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Many thanks for the infos Tony,
I think you need to disable the sip alg on the sonicwall.
On Wed, Sep 8, 2010 at 10:19 AM, Stiles Watson wat...@datatek-net.comwrote:
That was my initial sipX setup as well (except I had Auth User set equal
to User).
On the Teliax side under device settings did you do either of the
your firewall is a REALLY important pice of the puzzle. Thanks for finally
telling us what it is.
In the sonicwall:
1. Open web administration interface
2. Select VoIP from the left menu
3. Check/uncheck Enable SIP Transformations
4. Click Accept
Then try your call again and see
Yes, but what I want to hear is that you changed the sip port on the bbox2
to 5080 and now you have two way audio working when sipd is running and
using ANY soft or hard phone registered to sipx.
On Wed, Sep 8, 2010 at 10:20 AM, Smith mb11...@telenet.be wrote:
Content-Type: text/plain;
Tony,
Why do you think that it could be difficult to parse results of a mount
comand?
[r...@beaver sipxpbx]# mount
/dev/sda2 on / type ext3 (rw)
proc on /proc type proc (rw)
sysfs on /sys type sysfs (rw)
devpts on /dev/pts type devpts (rw,gid=5,mode=620)
/dev/sda5 on /var type ext3 (rw)
I think in sonicwall it is called consistent nat.
To enable Consistent NAT, select the Enable Consistent NAT setting and click
Apply. This checkbox
is disabled by default.
On Wed, Sep 8, 2010 at 10:31 AM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
your firewall is a REALLY important
Yes, I do have DNIS checked on the Teliax side. No, I do not have an IP
entered.
Teliax/FreeSwitch will send the Invite to whatever port you have registered
from - assuming FreeSwitch has recognized that you are behind NAT. In my case,
at the time, this was port 37678. I have not opened any
Hello Michal,
no the tool has been written from scratch without any external components.
My question is just related to the name sipXecs. Of course we want to use
the name in context with the tool and link to sipfoundry and so on.
Therefore I just wanted to ask if we might get into trouble when
On Wed, Sep 8, 2010 at 10:44 AM, Dave Redmore
dave.redm...@spigotsystems.com wrote:
Yes, I do have DNIS checked on the Teliax side. No, I do not have an IP
entered.
Teliax/FreeSwitch will send the Invite to whatever port you have registered
from - assuming FreeSwitch has recognized that
Thanks Tony!
I already had consistent NAT enabled, but not SIP Transformations
(turning that on with Trixbox resulted in the Sonicwall being pegged at
100% - which is why I had not tried it with sipX). I enabled
transformations and made my call again and after 2min it was still up so
that
right. proper firewall configuration...
On Wed, Sep 8, 2010 at 11:01 AM, Stiles Watson wat...@datatek-net.comwrote:
Thanks Tony!
I already had consistent NAT enabled, but not SIP Transformations (turning
that on with Trixbox resulted in the Sonicwall being pegged at 100% - which
is why I
Premature rejoicing - problem is not solved. Right after I made the
change, I voluntarily ended the next call at 2min. But every call after
that has disconnected at 1min 29sec as before.
Other interesting things to note. If I make a call through sipx/teliax
to a cell phone the call sounds
Thanks Todd,
I did search the archives before sending the question to the list. There
is not much discussion.
Stiles
Todd Hodgen wrote:
There have been some discussions about this ITSP on the list in the past.
I did find this one.
From: Melcon Moraes [mel...@gmail.com]
[regarding
https://wiki.openscs.org/display/xecsuserV4r0/Manually+Configuring+Phone+BLF]
Is this wiki still closed? I couldn't access it.
We're still having problems getting
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Danny Shay
[ds...@norlemtc.com]
BLF worked when we first installed 4.2.0, but failed shortly thereafter. I
think it failed after I set the phones to
I am about to build a 64bit replacement for a 32bit system.
I've received input from the list saying that it should not be a problem doing
this.
However, what about the ssl cert, do I need to keep the same name that is on
the current system in order for everything to go smoothly and if not,
Perhaps you also need to look at the refresh time too... How many seconds
left in the current registration before attemtping to register again.
Perhaps slightly increasing that value with registration time back to 3600
will present a better offer that the phone won't see its registration
expire.
Everything should be the same: ip, sipdomain, hostname.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk
On Wed, 8 Sep 2010 12:56:01 -0400, Tony Graziano wrote:
Everything should be the same: ip, sipdomain, hostname.
That's what I thought but hoped there might be another way.
Thanks, that's what I'll do then.
Mike
___
sipx-users mailing list
Tony,
I've looked at the sipx server and cpu usage never gets above 27%
utilization and the sonicwall (SW) never gets above 10% utilization so
neither of those are an issue.
The SW is able to prioritize ports and traffic as well as assign
individual ports and port groups to separate subnets.
If you have an internal-internal call where sipx is not inbetween (no remote
user, no internet, aa, voicemail or siptrunk) you can unplug the ethernet
to sipx with an internal-internal call as long as media is established. All
other bets are off.
Tony Graziano, Manager
The externals calls don't register in cdr? Definitely a config issue. Are
you sure your sip alg is off in the sonicwall?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Systems
external calls register, internal calls do not.
Tony Graziano wrote:
The externals calls don't register in cdr? Definitely a config issue. Are
you sure your sip alg is off in the sonicwall?
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
It has been asked a few times, but most of the recent questions seem
to be related to trouble with a particular kind.
Can anyone tell me about a cordless/wireless setup that has worked well?
It is a new system that will be 4.2.1. Handsets will be Polycom 450 and
550. The Polycoms will use
From: Staffan Kerker [ietf-li...@kerker.se]
Snapshot linked below:
http://www.kerker.se/files/sipx-snapshot-sipx.kerker.se-patch2.tar.gz
I'm getting a 404 on that URL (at Wed Sep 8 19:25:14 UTC 2010).
Dale
Just installed sipXconfig (4.2.1-018971.dhubler 2010-08-21T04:59:23 build37)
and noticed a small issue.
There is no refresh amount in the Active Registrations page. It shows Refresh
everyseconds with no seconds, not even hidden or invisible.
Mike
Does sipx support any type of pre-authentication method or the passing of
security credentials, such as SSO.
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List Archive: http://list.sipfoundry.org/archive/sipx-users/
On Wed, Sep 8, 2010 at 3:32 PM, m...@grounded.net m...@grounded.net wrote:
Just installed sipXconfig (4.2.1-018971.dhubler 2010-08-21T04:59:23 build37)
and noticed a small issue.
There is no refresh amount in the Active Registrations page. It shows
Refresh every seconds with no seconds,
From: Staffan Kerker [ietf-li...@kerker.se]
Seems like a new minor issue happends after this patch is applied. The ACK is
now lost
in the sipxregistrar again.
Yes, the Registrar is not handling Route headers in
I have tested with the polycom kirk wireless server 300 and a kirk 5020 handset
and it works really well. They can even be monitored by the BLF service and can
perform transfers.
Sent from my Samsung Moment™ only on the Now Network™
- Original Message -
From:Matthew Kitchin
Staffan
New RPM attached with the third patch from Dale:
http://track.sipfoundry.org/secure/attachment/26634/sipxregistry-4.3.0-01902
2-patch3.sipxbuild.i386.rpm
--martin
-Original Message-
From: Worley, Dale R (Dale) [mailto:dwor...@avaya.com]
Sent: Wednesday, September 08, 2010
I'm only looking at 1 or at most 2 phones per site, and the Cisco WAP
is already there.
I'm thinking about the Specrtalink 8020.
http://www.polycom.com/products/voice/comparison/wifi_matrix.html
I have only used the excellent Polycom Soundpoint provisioning built
into Sipx so far. Does anyone
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