Please tell us more about your network configuration.
Does the SipX server also create the VPN tunnel with the static IP?
SipX should only be used with one network interface enabled.
So maybe you need to establish the tunnel with another device and connect
your SipX server via LAN to the VPN.
Ren
Jim,
yes we Tested the current version of VOP and those features are working fine
with SipX 4.2.1.
The only problem we see with VOP is that the CallerID is shown on the
monitor but not on the phone that is ringing (You can use a telephone as
"audiodevice" for VOP). This is really annoying when usi
Perhaps you have a different setting in your firewall to allow it.
Asterisk traditionally uses ports 1-2 but sipx uses 3-31000.
I think it is still a firewall question and you have not provided any info
about it.
Tony Graziano, Manager
Telephone: 434.984.8
I meant to say... I can register to ANY of the 10 asterisk servers. Sorry.
Mark D. Theis
Southern California Telephone & Energy
27515 Enterprise Circle West, Temecula Ca. 92590
From: sipx-users-boun...@list.sipfoundry.org
To: 'sipx-users@list.sipfoundry.org'
Se
The crazy part is that I can register the same phone(s) to one of the 10
asterisk/switchvox servers from the same ip and it works perfectly well. I am
hoping to transition 8 of the 10 pbx's to sipxecs if I can get this working. I
do not understand why the firewall is friendly to the asterisk ser
It would mean that yuour firewall at the office has a SPI or SIP ALG enabled
that needs to be disabled.
On Mon, Sep 20, 2010 at 3:44 PM, Mark Theis wrote:
> Ah…. Sorry. I failed to mention that the sipXecs server is not in my
> office. It is at a co-lo. I am on a 192.168.*.* range both in th
Ah Sorry. I failed to mention that the sipXecs server is not in my office.
It is at a co-lo. I am on a 192.168.*.* range both in the office and at home.
With that said, I can't imagine why it would be any different for me when I try
to get the phones working from home or in the office.
Understand what that means.
NO NAT is what it should say when it is in the office, because it is not
behind NAT, so that sounds correct.
What causes the registration to add the "privcontact" is media relay, but
since the phone is not behind nat it is not passing through it. Before (when
it was),
The proxy has a plugin that auto-detects the presence of NAT. If it detects
NAT it inserts that header.
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Mark Theis
Sent: Monday, September 20, 2010 2:53 PM
To: Discussion list
Most of you will think that this is a stupid question and I bet that you can
answer this in about 10 seconds.
Well.. here it goes...
What decides what the registration string will be when a device registers with
sipXecs (or any IP PBX for that matter)? What I mean specifically is... What
caus
Yes, more IM LDAP,
When using AD for authentication, I get none of the group contacts,
including MyBuddy. When I change it back to local authentication
everything returns. Do I assume that with AD/LDAP groups must come
from AD?
Kyle
On Mon, Sep 20, 2010 at 10:11 AM, Haefner,Kyle
wrote:
> La
See response below:
Tony Graziano wrote:
> When you generate a profile, it creates the configuration files the
> phone needs.
>
> When you get a 0x error on booting a polycom it is usually because the
> profile is there or there is a firmware you are trying to send the
> phone that is not compa
Must i rely have a static ip
If i run sipxces-setupsystem i are prompted to enter a ip
But if i do it my server network die
I run a isp wich only have dynamic ip
And i run a openvpn tunnel with a static ip
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you wrote, "Is it just this hard for you?"
I'm not sure how to take that.
Anyway, I did as you said and clicked "Send Profiles." The job failed
due to timeout, but it must have generated all the files, because on the
next reboot, the phone registered - via tftp!
I know you think it's easier to
Go to the phone in sipxconfig and generate (push) profile to the phone. A
reboot of the phone is not necessary.
Is it just this hard for you?
Hopefully you don't have a version of firmware earlier than 3.2.1. A 335 is
a difficult phone to deploy remotely with a polycom r.e bug. If I were you
I'd
That is very true.
Interesting development: I pulled down the polycom ip 335 admin guide
and on the top of page 41 (3-5) there is the following note:
"Setting Option 66 to tftp://192.168.9.10 has the effect of forcing a
TFTP download. Using a TFTP URL (for example,
tftp://provserver.polycom.co
> If you have sipxecs behind a firewall you could certainly limit it there.
It's behind a firewall, lan side users need access to their controls.
In my application, I just want to use iptables to allow SIP/RTP and other
services as usual but limit 80/8443 to specific clients.
_
If you have sipxecs behind a firewall you could certainly limit it there.
On Mon, Sep 20, 2010 at 12:25 PM, Martin Steinmann wrote:
> I think all we would need is a correctly configured iptables firewall. See
> here: http://track.sipfoundry.org/browse/XX-5197
>
> You can certainly do this manua
> I think all we would need is a correctly configured iptables firewall. See
> here: http://track.sipfoundry.org/browse/XX-5197
> You can certainly do this manually.
Thanks. Wasn't sure if there might be some issues. Typically, on phone systems,
iptables aren't used but I want to allow all traff
You can follow the GRUU discussion here:
http://track.sipfoundry.org/browse/XX-8834
It is my understanding that there is one outstanding issue in the proxy, by
Staffan or Dale would have more of the details
--martin
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...
I think all we would need is a correctly configured iptables firewall. See
here: http://track.sipfoundry.org/browse/XX-5197
You can certainly do this manually.
--martin
> -Original Message-
> From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-
> boun...@list.sipfoundry.org
You changes the sip password in sipxconfig since the phone loaded its
profile last.
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: 4
Laurentiu,
Thanks for the reply. I think I figured it out.
With Active Directory the default fieldname that user lookups are
performed on is sAMAccountName. Even though I set this in the IM
userid field when I originally set up LDAP, that field is not being
used by openfire.
I added this field
The registration request is getting to sipx. I turned the logging level
to DEBUG, restarted the services and executed following:
tail -f /var/log/sipxpbx/sipXproxy.log | grep "REGISTER sip" | grep
"1...@datatek-net.com" > regdebug141.log
After the polycom reboot completed, I executed
wc -l re
This is huge...anyone tested it yet?
* VOP now supports sipXecs parking extension with monitoring.*
You can now transfer a call to a parking extension by a simple drag and
drop.
You can retrieve a parked call by dialing *4X (where X is the parking
extension number). The caller/ee ID will be
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On 09/20/2010 11:24 AM, Douglas Hubler wrote:
> Re:changes to 4.2
> Enabling erlang mod is one, may be others
>
> Re:rake errors
> Can you post a link to a pastebin?
http://pastebin.com/RDc08d6U
joe
>
> On Sep 20, 2010 11:16 AM, "Joe Micciche" wr
Re:changes to 4.2
Enabling erlang mod is one, may be others
Re:rake errors
Can you post a link to a pastebin?
On Sep 20, 2010 11:16 AM, "Joe Micciche" wrote:
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On 09/19/2010 09:51 PM, Douglas Hubler wrote:
> There were minor tweaks to 4.3 to make th
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On 09/19/2010 09:51 PM, Douglas Hubler wrote:
> There were minor tweaks to 4.3 to make this setup possible
> http://wiki.sipfoundry.org/display/xecsuser/sipXecs+and+OpenACD+Call+Center
> but it does work. There are fairly significant load tests bei
I've noticed that iptables on sipx is always disabled when I install a server.
I was wondering if there are any problems with using iptables? I have a
requirement where I wish only certain clients to have access to port 80/8443
and want to use iptables to allow/deny access to these ports.
Thank
If it has a valid config file and the remoe firewall has spi and sip alg off
and your sonicwall is not getting in thje way, yes.
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: tgrazi...@myitdepartment.net
LAN/Telephony/Security and Control Sys
Still working through the options you've given me, but the Polysom should
be able to register remotely without ftp if everything is configed
correctly, right?
Stiles
On Fri, 17 Sep 2010 20:43:24 -0400, Tony Graziano
wrote:
> By the way... the first sentence in this thread is:
> "OK Tony, shoot m
Hi,
I've tested ldap openfire (using openLdap server) and it works fine.
I configured the ldap conection, I checked 'Settings / Instant Messaging
Authentication' check box and I restarted 'Instant Messaging'
service. After that, I was able to register two LDAP users and I had a chat
between those
you need to provide information. noone could actively assist you with that
description and detail.
On Mon, Sep 20, 2010 at 6:57 AM, mattias wrote:
> How to solve it
>
> ___
> sipx-users mailing list
> sipx-users@list.sipfoundry.org
> List Archive: http
How to solve it
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
ok thanks
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Monday, September 20, 2010 12:14 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] The http port
80 and
Nothing specific. We are planning to install the latest SipX server in our lab.
One of the prime
motive is to test GRUU since there seems to be no open SIP Servers that
supports GRUU.
While interoping with SIP-SW server, we came to know that it is really SipX
server and it supports
GRUU.
Best
i see you are in business now, congrats.
On Mon, Sep 20, 2010 at 6:17 AM, Tony Graziano wrote:
> Not to be confusing, but if the 80 redirect worked it would send you to
>
> (example) https://88.80.28.43:8443/sipxconfig/
>
> Which doesn't seem to be working because it can't find it in your case.
Not to be confusing, but if the 80 redirect worked it would send you to
(example) https://88.80.28.43:8443/sipxconfig/
Which doesn't seem to be working because it can't find it in your case. Are
you running other services (webmail) on the same box? If you recently did an
update perhaps you need t
80 and 8443. http:80 redirects to https:8443. If you connect via http it
will connect you to 8443.
On Mon, Sep 20, 2010 at 5:38 AM, mattias wrote:
> I have forgot the http port number
> Wich are it
>
> ___
> sipx-users mailing list
> sipx-users@list.si
I have forgot the http port number
Wich are it
___
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sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
It should. There is an issue with snom that would force you to turn off GRUU
in snom. Do you have a specific question about a particular UA or a use case
issue?
On Mon, Sep 20, 2010 at 2:37 AM, hanifa.mohammed <
hanifa.moham...@globaledgesoft.com> wrote:
> Hi all,
>Pl clarify whether the late
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