Thank you!
2010/5/17 Tony Graziano
> Disable the repo
>
> 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: 434.984.84
There are several things you should do:
1. Make sure there is a filter to *allow* all GRE traffic (protocol:any, not
just tcp), on both systems.
2. I will also assume you have two different sip domains. I will also assume
they can lookup and resolve each other SRV records and resolve them to a
pr
I'm using IPSEC GRE and pfsense interfaces have private IPs. should I still
need NAT for that matter?
Thanks
On Tue, May 18, 2010 at 3:03 AM, Picher, Michael
wrote:
> It should be set to manual and yes.
>
>
>
> *From:* Rhon [mailto:c4rdi...@gmail.com]
> *Sent:* Monday, May 17, 2010 9:33 AM
> *T
I'm using IPSEC GRE and pfsense interfaces have private IPs. should I still
need NAT for that matter?
Thanks
On Tue, May 18, 2010 at 3:03 AM, Picher, Michael
wrote:
> It should be set to manual and yes.
>
>
>
> *From:* Rhon [mailto:c4rdi...@gmail.com]
> *Sent:* Monday, May 17, 2010 9:33 AM
> *T
It should be set to manual and yes.
From: Rhon [mailto:c4rdi...@gmail.com]
Sent: Monday, May 17, 2010 9:33 AM
To: Picher, Michael; sipx-users@list.sipfoundry.org
Subject: Re: [sipx-users] No Voice/IVR on Site-to-Site
Hello Michael,
I have the static NAT port set to NO on pfsense.
Also,
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis
[jballa...@alliancetechsolutions.com]
When I look at the phone, the message is displayed where the alerts are
displayed: Beneath the date and
Wipe the phone to defaults. Delete the user you created in sipx for the
phone.
make sure if your phone is local, you use 3.1.3revc or 3.2.3 9do not use
3.2.3 if phone is remote).
Create the user. Add the line to the phone and ONLY enable instant messaging
for the user at the sipx line (i.e. 200).
I promise, I did it exactly as I stated below. Sounds like I might need
to either place a call to Polycom or delete the handset and add it
again.
Jake Ballamis
Technical Support Manager
p. 801-566-TECH (8324)
f. 801-208-9317
jballa...@alliancetechsolutions.com
This e-mail is intended solely fo
> Ok, let me see if I can get more granular with my
> explanation. I was in a hurry before and didn't give the
> detail I should have. I apologize.
Not a problem. Most times, it's hard to converge on the right solution without
the full background.
>
> I am running sipXecs 4.2 with Polycom 6
I don't follow you. You can't do the instant messages on a polycom phone.
I made a wiki page on "setting an expectation" for IM "how does this work".
You need spark or pidgin on a pc as a companion to a polycom. You need the
bria im account on a bria. I think there is a footnote about NOT using th
Ok, let me see if I can get more granular with my explanation. I was in
a hurry before and didn't give the detail I should have. I apologize.
I am running sipXecs 4.2 with Polycom 650 phones.
On my phone, I went to Menu > Features > Messages > Instant Messenger >
New. I entered the extension I
I do this all the time, voip.ms too. I sincerely think if it is not
registering it is a firewall config issue.
I have a pdf that explains it well enough on http://blog.myitdepartment.net
Look at the call setup example.
Tony Graziano, Manager
Telephone: 434.984.8430
Fa
On 5/17/2010 10:30 AM, Eelco Brölman wrote:
> web_nl.properties (with some basic translations) is not taken from the
> sipxconfig_nl.jar file, but from the deployed sipxconfig.war file.
>
Eelco,
Pretty sure there maybe a bug with language pack files overwriting the
same files checked in to
I can verify this is happening as well.. It seems to happen very
suddenly, and the web interface will be very slow for about 5 minutes,
then I'll get the white screen of death. The only way to revive it is to
run the following from the command prompt:
sipxproc -r ConfigServer
Josh Patten
Assis
On 5/17/2010 6:13 AM, Tony Graziano wrote:
> Port 5080 must be symettrically nat'ed to register.
Just to confirm "Symmetric NAT" -
sipXecs sends a request to voip.ms from 10.8.158.121 port 5080;
firewall adds an entry to its NAT table and substitutes the external IP
address
216.8.158.237 for 10.
Wrong... read below...
On Mon, May 17, 2010 at 10:22 AM, Norman Branitsky <
nor...@cherniaksoftware.com> wrote:
> On 10-05-17 6:13 AM, Tony Graziano wrote:
>
>> Port 5080 must be symettrically nat'ed to register.
>>
>>
> Just to confirm "Symmetric NAT" -
> sipXecs sends a request to voip.ms from
On 10-05-17 6:13 AM, Tony Graziano wrote:
> Port 5080 must be symettrically nat'ed to register.
>
Just to confirm "Symmetric NAT" -
sipXecs sends a request to voip.ms from 10.8.158.121 port 5080;
firewall adds an entry to its NAT table and substitutes the external IP
address
216.8.158.237 for
Hi all,
In short: sipXconfig-web_nl.properties is not read from uploaded localization
package, but from deployed sipxconfig.war file.
Longer story:
We are working on the Dutch translations of the webinterface. I followed the
following guides:
- http://sipx-wiki.calivia.com/index.php/Install
On Mon, May 17, 2010 at 9:46 AM, JOLY, ROBERT (ROBERT) wrote:
>>
>> On Sun, May 16, 2010 at 1:19 PM, Gabe Casey
>> wrote:
>> > I am seeing a strange issue periodically in 4.2 where a call is
>> > getting stuck in the system and recalling the outside
>> number every 30
>> > min. From the Logs each
Well, it showed up that my previous complications were caused by
broken source. After re-downloading the compilation error I mentioned
seams not be a problem any more.
But in this place I have another one, I think at very end of building.
I'm realy new to sipxecs, and output sais nothing to me. I
>
> On Sun, May 16, 2010 at 1:19 PM, Gabe Casey
> wrote:
> > I am seeing a strange issue periodically in 4.2 where a call is
> > getting stuck in the system and recalling the outside
> number every 30
> > min. From the Logs each call has the same Call-ID as the
> original. Anyone else seen t
Also, I set the pfsense *to Manual Outbound NAT rule generation (Advanced
Outbound NAT (AON))*
My NAT rules below for my voice VLAN:
WAN172.16.3.0/24 * * * * * NO
Thanks again for your help.
Brgds,
Rhon
On Mon, May 17, 2010 at 3:33 PM, Rhon wrote:
> Hello Michael,
>
> I have the st
Hello Michael,
I have the static NAT port set to NO on pfsense.
Also, to I have to enable NAT traversal on sipx?
Thanks
On Mon, May 17, 2010 at 3:20 PM, Picher, Michael
wrote:
> Static NAT port on the pfSense?
>
>
>
> *From:* sipx-users-boun...@list.sipfoundry.org [mailto:
> sipx-users-boun..
> When I upgraded to 4.2, I did a quick test of the IM system.
> I sent a message from my phone to my office manager's phone.
> The problem, we can't get rid of that message. I've tried
> rebooting the phone and we've cleared all the messages on the phone.
>
>
>
> Suggestions?
Yes, pleas
Could you provide the sequence of DTMF key strokes entered (starting from
The main menu) and voice prompts heard?
I've gone the through sequence of recording, listening to and setting a
Standard greeting and the flow/options seem ok so want to determine the
specific place the prompt you mentio
Static NAT port on the pfSense?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon
Sent: Monday, May 17, 2010 9:14 AM
To: sipx-users@list.sipfoundry.org
Subject: [sipx-users] No Voice/IVR on Site-to-Site
Hi,
I have a problem with
Hi,
I have a problem with our deployment with SipXecs 4.2 which was installed
fresh using ISO build.
We cannot hear anything on both sides but are able to connect and can ring
the other end. Calling the IVR is ok but no audio as well.
SITE A:
100 - 199
SITE B:
200 - 299
Everything passed using
Hi,
Haven't read the whole thread through yet, but to me it sounds like a
garbage collection issue.
It can be, that the config application is using almost all the heap
space provided by the JVM when idle, and when a change is done to the
settings, the JVM has to do constant garbage collection in
Disable the repo
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 Contract Customers:
http://www.myitdepartment
Port 5080 must be symettrically nat'ed to register.
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 Contract
You can rename your yum repo file to sipxecs.repo.old... then it won't check.
Mike
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz
Sent: Monday, May 17, 2010 2:44 AM
To: sipx-users
Subject: [sipx-users] Remove "New so
31 matches
Mail list logo
