It is on the tracker. I think the GUI needs to be adjusted to allow it. I
also know 4.6 has major cert rework, maybe it can be posted to the dev list
whether it is being addressed in 4.6 or not.
http://track.sipfoundry.org/browse/XX-9390
All of the above (intermediate and changing the script to 2
I also had to drop the Certificate Authorities CRT files for GoDaddy of
gd-class2-root.crt, gd_intermediate.crt & gdroot-g2.crt into the(
/etc/sipxpbx/ssl/authorities ) directory. I restarted the sipxecs service and
then proceeded to add the web certificate downloaded from GoDaddy.
sipXecs Sy
Rob,
I'd appreciate it if you would not publish my name.
Thanks,
jim
> Yeps, no luck in the search.
>
>
>
> However Jim Nolen of IIPS was a great help and gave me the following
> information to solve the problem.
>
>
>
> Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
>
> ServerKeyBits=1024
Yeps, no luck in the search.
However Jim Nolen of IIPS was a great help and gave me the following
information to solve the problem.
Edit: /usr/bin/ssl-cert/gen-ssl-keys.sh:
ServerKeyBits=1024[change to 2048]
If I knew how to add this info to the wiki I would. Perhaps a feature could be
I seem to recall the script may need to be or was it already modified to
handle 2048 bit certificates?
Besides that I think it had to be done manually AND noone updated the wiki
or the list as to whether or not it worked.
On Fri, Apr 20, 2012 at 4:34 PM, Michael Picher wrote:
> did you check th
did you check the wiki?
On Fri, Apr 20, 2012 at 4:21 PM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:
> How do I change the configuration for the certificates area to generate a
> 2048 bit key instead of a 1024? I have changed the openssl.cnf file in
> /etc/pki/tls/ locatio
So far I have had no reports of problems. I am waiting for a bit to know if
this issue is resolved completely before posting details of the AC
configuration.
So Far So Good,
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Grazi
How do I change the configuration for the certificates area to generate a 2048
bit key instead of a 1024? I have changed the openssl.cnf file in /etc/pki/tls/
location and selected the generate button and still no 2048 key is generated.
I am sure this is an educational issue on my part.
Yes I
Dave,
Thanks! I turned it off, not sure how it got started. I updated the
wiki page to reflect this:
http://wiki.sipfoundry.org/display/sipXecs/Roles%2C+Services+and+Processes
Kyle
On Fri, Apr 20, 2012 at 12:24 PM, Dave Deutschman
wrote:
> Kyle,
>
> The SharedAppearance Agent is not redundan
Kyle,
The SharedAppearance Agent is not redundant and should only be running on
the master. It will run on the secondary and Polycom phones will send
subscriptions to it and the process on the master. However, it confuses the
phones and they stop handling shared line calls properly.
There
have you compared the rpm versions?
On Fri, Apr 20, 2012 at 1:49 PM, Kyle Haefner wrote:
> Hi Users,
>
> I'm running Sipx 4.4 in a three node cluster with the most recent
> updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.
>
> In one of the redundant nodes when I get this:
>
> sudo sipxp
Hi Users,
I'm running Sipx 4.4 in a three node cluster with the most recent
updates for 4.4.0- 2012-02-08EST09:10:08 ip-10-72-10-163.
In one of the redundant nodes when I get this:
sudo sipxproc | grep SharedAppearanceAgent
"SharedAppearanceAgent"=>"Disabled",
In the other redundant node i ge
Please test the following.
Calling from PSTN to AA and try to transfer to an extension that is not
forwarded.
You have to ensure your ITSP is sending the INVITE to port 5080 in order
for any transfers to succeed.
If not, what Gerald said... how to get around that is to create a dialplan
(56+10 d
On 4/20/2012 1:08 PM, Tommy Laino wrote:
> Content-Type: text/plain;
>charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8<67729>
> Message-ID:<10891.4f919...@forum.sipfoundry.org>
>
>
>
> I have 3 IP trunks on my test sy
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67729>
Message-ID: <10891.4f919...@forum.sipfoundry.org>
I have 3 IP trunks on my test system. I am trying to have an
option from the auto attend
did you do that? if so, did it fix it?
On Fri, Apr 20, 2012 at 11:53 AM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:
> Tony,
>
> ** **
>
> Thank you for the suggestion and for your help.
>
> ** **
>
> Rob
>
> ** **
>
> *From:* sipx-users-boun...@list.sipfoundry
Tony,
Thank you for the suggestion and for your help.
Rob
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 11:00 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users]
Thanks for this. We were thinking of doing it at one of our locations
but doing the evaluation is difficult without having the equipment.
Geoff Van Brunt
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Philippe
Laurent
Sent: Thursday, Ap
+1
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Joegen Baclor
Sent: Friday, April 20, 2012 2:29 AM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Nat Problem
I forgive all of you for not noticing that
Thanks, Mike
Yes, I am setting up the environment for solving the issue: Aastra phone
doesn't accept SUBSCRIBE dialog or presence.
Derrick
From: Michael Picher [mailto:mpic...@ezuce.com]
Sent: Friday, April 20, 2012 10:54 AM
To: Discussion list for user
Sorry, am not a audiocodes guru. I have documented the Patton PRi gateways
for use with t.38 in the wiki. Perhaps someone else would be so kind as to
do the same for an AC PRI gateway instead of keeping it a big secret and
make sure it is also documented on the wiki.
Try:
Configuration > Protocol
I'd grab a capture of this and talk to your engineering folks. This is
what we refer to as BLA functionality (bridged line appearance).
I would think this would be different from directed call pickup which is
*78ext .
If you can make this work and help get the Aastra template sorted and new
phon
Please understand that sipxecs is meant for enterprise deployments. If it
were meant for smaller deployments as its market segment things would
probably be designed very differently, hence the prerequisite reading and
architectural understanding.
It is not aimed at users either, it's aimed at dens
Tony,
The original intent of the email was to offer some constructive
criticism and explain a little of my confusion. I never intended to make
it personal. I realize that my "never let go of that bone" comment made
it personal, and for that, I apologize.
You guys have a superior product and
Hi All,
I use SipXecs as proxy server, and register an 2 Aastra phone and
Polycom phone.
I can monitor Polycom phone(phone number 203) from Aastra phone(phone
number 200). However when I do "directed call pickup", Aastra phone send
"INVITE *78203" to SipXecs, SipXec only responds 100 Trying, t
Tony:
I did find a Fax Signaling Method of T.38 Relay under the Coders And Profile
Definitions - IP Profile Settings - Gateway Parameters - Fax Signaling Method.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: F
Tony,
Thank you for the suggestion. Are you referring to the G711Mulaw setting under
Bypass Settings? The only other option is G711Alaw_64.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Friday, April 20, 2012 9:
On Fri, Apr 20, 2012 at 4:38 PM, Robert Schroeder
wrote:
> George,
>
It's Voicemail and AA, however pay attention to Tony's reply: sipx
only support t.38
George
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoun
faxing into sipx only support t.38 and your audiocodes fax settings have
g711u. you need to change it to t.38.
On Fri, Apr 20, 2012 at 9:17 AM, Robert Schroeder <
robert.schroe...@memberfirstmortgage.com> wrote:
> Has anyone had problems with faxing into sipXecs? We are having huge
> problems wit
On Fri, Apr 20, 2012 at 4:17 PM, Robert Schroeder
wrote:
> Has anyone had problems with faxing into sipXecs? We are having huge
> problems with faxes being delivered to the users emails as zero page tiff’s.
> It is not an every fax occurrence however it is at a rate that I am no
> longer able to r
Has anyone had problems with faxing into sipXecs? We are having huge problems
with faxes being delivered to the users emails as zero page tiff's. It is not
an every fax occurrence however it is at a rate that I am no longer able to
rely upon the faxing delivery capabilities of sipXecs. I am tryi
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