Hi,
I have a problem using my Siemens Openstage phones on sipxecs, latest 4.1
version.
Calls coming in to the sipxbridge and destined for my internal phones
work very well. Pickup and Hold/Resume works on SNOM and Polycom. As we
are in Germany and our guys are used to Siemens, they want to
In the HA config described at
http://sipx-wiki.calivia.com/index.php/High-Availability_Installation,
does the media server also get replicated between the two servers? If
so, in a HA config with 2 call control servers and a separate media
server, does the call control servers try to replicate voic
Hi, I just svned (svn co http://sipxecs.sipfoundry.org/rep/sipXecs/main sipx
) the build of 17289 and made build. Whereas the build was broken by
following errors.
Pls help how I fix this problem.
Thanks
Jun
--
Thanks. Verizon initially acted like nobody had ever hooked up an open
source system to their VOIP network, but they seem to be warming to the
idea as of our conference call yesterday.
Tony Graziano wrote:
> If it were me I'd use the latest patch. Then test your trunks and open
> a case in the
If it were me I'd use the latest patch. Then test your trunks and open a
case in the event it needs something specifically done in sipxbridge for
verizon.
On Wed, Nov 18, 2009 at 5:30 PM, mkitchin.pub...@gmail.com <
mkitchin.pub...@gmail.com> wrote:
> Thanks. I planned to connect to a Sip Trunk f
Thanks. I planned to connect to a Sip Trunk from Verizon on Monday, so I
guess I do have an issue. I had been following some of the posts on that
issue, and I had applied patch 21. I see it is up to patch 23 now on the
wiki. I obviously need the sip connection to Verizon to work properly. I
wou
Hi,
I have also started to use and test the Siemens phones.
Were you able to get them working correctly?
I have problems with:
*MWI
*MOH is not played to the caller put on hold
*Hold/resume if calls are coming in from sipxbridge
How do you configure the phone? Only with webinterface or with an
Until 4.0.4 is made available you can do this:
edit the repo file on your system. Where it has centos version "5.2"change
it to "5", where it has sipx "LatestStable" change it to "4.0.3".
yum clean all
yum upgrade
You might as scott what problems you might face with siptrunking on 4.0.3
though.
On Wed, 2009-11-18 at 15:59 -0600, mkitchin.pub...@gmail.com wrote:
> Sorry if this is obvious, but the information seems to be a little
> conflicting. I'm currently running 4.0.2. It is not in production. It
> was built from the SipX iso. I need to demo several different handsets
> on Monday, a
Sorry if this is obvious, but the information seems to be a little
conflicting. I'm currently running 4.0.2. It is not in production. It
was built from the SipX iso. I need to demo several different handsets
on Monday, and one of them is the Polycom SoundPoint 450. According to
this post:
http:
I've asked this several times but it was burred in a long thread so thought I
would ask
again.
I have a used Media Gateway 3200 (Mediant 2000) I purchased which I would have
loved to
use with sipx. However, I have read that there are possible problems with older
software
versions and more imp
On Wed, 2009-11-18 at 10:32 -0600, Josh Patten wrote:
> It would appear I spoke too soon. Not 10 minutes after my last post it
> happened to one of my helpdesk people. Ranga, I submitted ticket
> http://track.sipfoundry.org/browse/XX-7059 with snapshots of my servers.
I just assigned it back to
It would appear I spoke too soon. Not 10 minutes after my last post it
happened to one of my helpdesk people. Ranga, I submitted ticket
http://track.sipfoundry.org/browse/XX-7059 with snapshots of my servers.
Josh Patten wrote:
> Wow, so the proverbial crap hit the fan yesterday and my Adtran TA
On Wed, 2009-11-18 at 09:26 +, Gabor Paller wrote:
> " sipXecs does not officially support phones that do not support RFC
> 4235,
> and we do not intend to do so."
>
> Are you aware of any soft phone that supports it? I found just one
> Counterpath version (but not all). Counterpath has not re
Wow, so the proverbial crap hit the fan yesterday and my Adtran TA908e
started doing crazy things, like dropping about 25% of the voice packets
(making faxing impossible and making voice calls very stuttery) and
randomly hanging calls up. A firmware update and a reboot didn't fix it,
so I imple
Thanks for the fast answers.
I will think on the things that you told me.
Currently i'm running both apache instances.
The one from OS listen on apache standart ports (80, 443).
Sipxecs one listen on different then 80, 443 ports. So i don't get
conflict and everything is ok. But it will be real
Hi,
That template corresponds to Registration. Can you set it up on les.net to
be a registrar. If you set it up that way it ought to work fine. I have not
tested the other configuration.
Regards
Ranga
On Wed, Nov 18, 2009 at 9:32 AM, Derek Bartolo <
dbart...@comsourcetechnologies.com> wrote:
>
Tim Byng wrote:
> Lot's of useful information, thanks. I'll see what I can come up with
> and will share after some testing.
>
If you come up with a template, send it to the list and I'll check it in.
Check out etc/sipxpbx/region_XX directories.
D.
___
Our team has been recently researching opensource projects to use in a new
unified communications project and the information on the wiki about sipXecs
is very compelling. I do like the architectural approach that has been
taken and it looks like it would be a good starting point for our project
(
On Wed, 2009-11-18 at 09:27 -0500, Roman Gelfand wrote:
> In one of the recent post it seems that Tony is saying that Jabber is
> not xmpp. I was under the impression that Jabber utilizes xmpp
> protocol.
Jabber is the original name - XMPP is the new name. They are the same.
___
Lot's of useful information, thanks. I'll see what I can come up with and
will share after some testing.
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfou
Sorry should've been more specific. We have setup for public IP which
requires no password. The only thing I'm unsure of is about the ITSP
settings in sipx, and yep I'm using the les.net template in sipx but I'm
still getting that error.
Thanks for any help J
Derek Bartolo
Technical Suppor
On Wed, Nov 18, 2009 at 9:15 AM, Derek Bartolo <
dbart...@comsourcetechnologies.com> wrote:
> Has anyone else successfully configured a sip trunk using Les.net? Still
> getting a “peer not configured correctly” message when trying to dial out.
>
>
>
Les.net allows you to configure for direct IP
In one of the recent post it seems that Tony is saying that Jabber is
not xmpp. I was under the impression that Jabber utilizes xmpp
protocol.
Could somebody explain.
Thanks in advance
___
sipx-users mailing list sipx-users@list.sipfoundry.org
List Arc
Has anyone else successfully configured a sip trunk using Les.net? Still
getting a "peer not configured correctly" message when trying to dial
out.
If I could see a sample template I'd greatly appreciate it
Thanks J
Derek Bartolo
Technical Support
Comsource Technologies
3585 Laird
" sipXecs does not officially support phones that do not support RFC
4235,
and we do not intend to do so."
Are you aware of any soft phone that supports it? I found just one
Counterpath version (but not all). Counterpath has not responded yet.
Regards,
Gabor
-Original Message-
From: Dale
UNLESS you are very familiar with apache configurations, you will most
likely break access to some pieces sipx needs to run properly. Right now
voicemail, media server and replication framework use apache.
With the upcoming 4.2 release, you might have a more difficult time of
upgrading if you have
sipXecs come with standard apache. I think you can change the apache
config file ("httpd.conf" etc in etc/sipxpbx) as you like.
On Wed, Nov 18, 2009 at 9:21 AM, an...@iguanait.com wrote:
> Hi,
>
> i have installed apache server from OS repositories. Is there a way to
> use sipx with this apache s
Hi,
i have installed apache server from OS repositories. Is there a way to
use sipx with this apache server and to not use the one that comes with
sipxecs?
What i mean is to use apache alias to access sipxecs "/sipxecs", or to
use virtual host?
We don't have chance to use a separate machine for
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