I'm really interested in getting Pawel ACD board up and running. We are
transitioning from a direct dial in to a queue based call center. I think it
would set our customers at ease If the caller could see how many agents are
logged in and how long the average wait would be. We are currently
On Wed, Sep 22, 2010 at 7:28 AM, Jason Mitchell mitch...@ceicmh.org wrote:
I'm really interested in getting Pawel ACD board up and running. We are
transitioning from a direct dial in to a queue based call center. I think it
would set our customers at ease If the caller could see how many
I find how to add my sip trunk to spix
But not where to enter login settings
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sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users/
http://wiki.sipfoundry.org/display/xecsuserV4r2/SIP+Trunking
On Wed, Sep 22, 2010 at 8:02 AM, mattias m...@mjw.se wrote:
I find how to add my sip trunk to spix
But not where to enter login settings
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sipx-users mailing list
thanks but only a litle isue
i don't find the sbc option in the menu
only sbc route
and the bridge app are installed
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, September 22,
Then you are not following the instructions.
Add gatewaysiptrunk, give it a name, choose the sbc route sipxbridge-1,
choose the provider template (if applicable), then APPLY. After that there
is an option (ON THE LEFT) for ITSP Account where this information is
entered.
On Wed, Sep 22, 2010 at
aha ok i missed it
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, September 22, 2010 2:39 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Sip trunk
PHP works fine with the Centos ISO of SipX 4.2.1
Install php using yum:
# yum install php php-xmlrpc
(Maybe you need some more php packages for the Pawel Board?)
In /etc/httpd/conf/httpd.conf and in /etc/httpd/conf.d/ssl.conf you must
change the Listening ports for apache.
After that apache
I've talked to the ISP and they say that they do not block any ports.
Stiles
Stiles Watson wrote:
Yes. In this case the service is not used.
Stiles
Tony Graziano wrote:
Does the cable modem provider offer a voice service?
Tony Graziano, Manager
Then you need to do a sip trace.
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:
now the gateway are configured
but where th set settings for outgoing calls?
sorry for all questions
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, September 22, 2010 2:39 PM
To:
Dial Plans. Its also in the wiki link earlier in the thread.
On Wed, Sep 22, 2010 at 9:04 AM, mattias m...@mjw.se wrote:
now the gateway are configured
but where th set settings for outgoing calls?
sorry for all questions
-Original Message-
*From:*
i reed on this page
http://sipx-wiki.calivia.com/index.php/HowTo_Configure_the_sipX_Voicemai
l_Service
how to configure voicemail
but the general tab are not there
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf
You really need to use the wiki at sipfoundry.
http://wiki.sipfoundry.org
On Wed, Sep 22, 2010 at 9:10 AM, mattias m...@mjw.se wrote:
i reed on this page
http://sipx-wiki.calivia.com/index.php/HowTo_Configure_the_sipX_Voicemail_Service
how to configure voicemail
but the general tab
thanks
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony
Graziano
Sent: Wednesday, September 22, 2010 3:19 PM
To: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] voicemail
You really
Hello,
We have a regular occurrence with our current SIP trunk where now and
then the network responds with a TRYING event after the sipXecs server
times out the call.
Does anyone know of any timer that can be adjusted to delay sipXecs from
timing out the call?
We see the following in
I rebooted the sipX server and everything started working. I'm glad its
working, but I hate spending three or more days chasing a phantom.
Thanks for all your help and patience.
One step I was consistently missing was clicking Send Profiles after
making changes to both phones and the gateway.
I think VOP uses the same method (RFC4235), so it monitors the park
extension via the same method, via the RLS server. It is the same method you
use on a polycom phone today with the current implementation of park server.
On Tue, Sep 21, 2010 at 9:24 AM, Jim Canfield jcanfi...@emstar.com wrote:
Apparently there is a nasty bug in the 64 Bit Linux kernel
http://www.zdnet.co.uk/news/security-threats/2010/09/21/linux-kernel-exploit-roots-64-bit-machines-40090177/?s_cid=116tag=mantle_skin;content
Looks like CentOS has released their updated kernel
http://bugs.centos.org/view.php?id=4518
I
Yeah. I think when they build the ISO it would be nice to have it already on
there. In the meantime, it's nice to be able to do a base OS update without
someone saying that's not supported anymore.
On Wed, Sep 22, 2010 at 4:04 PM, Matthew Kitchin (public/usenet)
mkitchin.pub...@gmail.com wrote:
Martin,
This is good info, I will be attempting this. Just for my own curiosity, is
there a way to provide the authentication without resorting to TLS?
Danny
-Original Message-
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of
Hi all.
In further pursuing a 'Production Rollout' ready implementation of sipXecs,
I had some final questions, mostly in regards to what can and cannot be
Virtualized. Currently it is looking like anything that actually handles
voice traffic (whether recorded messages or ongoing
On Wed, Sep 22, 2010 at 4:18 PM, Tony Graziano
tgrazi...@myitdepartment.net wrote:
Yeah. I think when they build the ISO it would be nice to have it already on
there. In the meantime, it's nice to be able to do a base OS update without
someone saying that's not supported anymore.
;) Yeah, the
Fortunately the stunnel fiasco had a fairly straightforward workaround in that
the older rpm from the avaya repo would do the job.
For those of you that follow other open source VoIP products, trixbox is one of
those that cannot have OS upgrades done due to the static nature of their
I'm doing a 4.0.4 - 4.2.1 upgrade tonight. It is a 64 Bit machine. I
was contemplating updating the kernel.
Kernel is currently 2.6.18-194.8.1.el5
The patched kernel version is 2.6.18-194.11.4.el5
It doesn't seem like too big of a change, but I haven't tested anything
at all with
Ok, noone said anything on this yet, don't take this in any way except free
advice or just a perspective on what I've seen and done.
In reading on some of the other lists (like FreeSwitch users), there is no
guarantee that FS will operate properly in a Virtual environment. Resource
planning is
Tony,
What would you suggest for the proxy role in a case like this? I would like to
do a similar project and I still can't get any semblance of failover to work
when the Primary server is taken offline. I am looking for this solution,
whether it is in sipXecs, another hardware unit, or
Peter,
I like your plan. I am not an expert (maybe because I have only been working
with this for a month or so) yet, but What were you thinking to use as your
SBC1 and SBC2? Each of your other devices' names hinted at (or explicitly
said) sipXecs... but the SBC has me wondering. Are
Hi,
Has anyone here was able to integrate SipXecs 4.2 and Bigbluebutton 0.7?
My plan is whenever a videoconference is conducted using Bigbluebutton, I
can make a conference call to SipXecs where our remote offices are
connected.
This way we can use Videoconf via Bigbluebutton and have voice conf
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