David,
Flat out, it's not going to work. From my understanding the Cisco 79xx
will not do SLA with our commercial product. I don't think they even do
SLA if they are running SIP and talking to Call Manager.
Sounds like you may have made a poor gamble. Not sure what else to tell
you.
You have
Can we simply drop in a new FS RPM package to replace the 1.0.7 version
that comes with sipXecs and all the current working stuff will continue to
work?
On Mon, Nov 5, 2012 at 1:58 AM, Josh Patten jpat...@ezuce.com wrote:
Yes, MoH, ring on transfer, etc. all works in my tests.
On Sun, Nov
Not likely.
At some point we'll probably look to use RPMs now that they are available.
You are of course free to try ;-)
Mike
On Mon, Nov 5, 2012 at 8:51 AM, Melcon Moraes mel...@gmail.com wrote:
Can we simply drop in a new FS RPM package to replace the 1.0.7 version
that comes with
On Mon, Nov 5, 2012 at 8:51 AM, Melcon Moraes mel...@gmail.com wrote:
Can we simply drop in a new FS RPM package to replace the 1.0.7 version that
comes with sipXecs and all the current working stuff will continue to work?
There's a good chance it will, and in fact someone reported it worked
How about for remote workers?
- Original Message -
From: Josh Patten jpat...@ezuce.com
To: Discussion list for users of sipXecs software
sipx-users@list.sipfoundry.org
Sent: Sunday, November 4, 2012 10:58:51 PM
Subject: Re: [sipx-users] FreeSWITCH for SIP Trunking
Yes, MoH, ring
It's not quite that simple. There are a couple of dependency issues with
spidermonkey and js that prevent direct installation, the init script must
be pointed to the new binaries, etc. and when you update all of that breaks.
I believe the newest FS will be integrated into 4.6.1
On Mon, Nov 5,
Douglas Hubler wrote on Mon, 05 November 2012 10:19
On Mon, Nov 5, 2012 at 8:51 AM, Melcon Moraes
mel...@gmail.com wrote:
Can we simply drop in a new FS RPM package to
replace the 1.0.7 version that
comes with sipXecs and all the current working
stuff will continue to work?
As FreeSWITCH is not a SBC, this only works for SIP trunking. For remote
workers you will need something like Karoo, OpenSIPS, or a commercial SBC.
On Mon, Nov 5, 2012 at 9:30 AM, andrewpit...@comcast.net wrote:
How about for remote workers?
--
*From: *Josh
Why do you say it's unlikely Michael and Josh?
I went out of my way to integrate with standard fs package. We do ignore
all the configuration files that get installed with fs.
On Nov 5, 2012 10:50 AM, Michael Picher mpic...@ezuce.com wrote:
Not likely.
At some point we'll probably look to
Almost forgot, you do need to specifically add the
freeswitch-format-local-stream RPM, but this could be made a dependency for the
sipxfreeswitch RPM.
- Original Message -
From: Josh Patten jpat...@ezuce.com
To: Discussion list for users of sipXecs software
Kind of stumped with this one...
I have Vitelity as my SIP trunk, which is configured in Asterisk to answer with
an IVR and perform some other functions.
If a call is passed from my AA in Asterisk or the DID is configured to call an
extension that belongs to my sipX box (Polycom phones), I
I ran into a dependency issue with js
spidermonkey required a different version of js than what we provide.
On Mon, Nov 5, 2012 at 10:33 AM, Douglas Hubler dhub...@ezuce.com wrote:
Why do you say it's unlikely Michael and Josh?
I went out of my way to integrate with standard fs package. We
An unmanaged gateway is just that. Can I assume that the address for both
systems are on the same subnet? Unmanaged gateways would assume that the
other and knows how to handle the SIP REFER method.
Asterisk as a sip trunking system is not exactly compliant.
If REFER is not supported, then the
The biggest red flags are Asterisk and CUCM
The SIP stacks on these platforms aren't complete and REFER support is
usually lacking. Trunking with these platforms requires a session border
controller to interface with these platforms. How many active calls do you
expect between these systems? If
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 11/05/2012 12:01 PM, sipx-users-requ...@list.sipfoundry.org wrote:
Any ideas on how to get the Cisco 7961G phone working with sipXecs?
I have it working on an Asterisk installation, but when registered
on the sipXecs server, whatever I do to the
On Fri, Nov 2, 2012 at 11:17 PM, Steve Beaudry
steve.beau...@royalroads.cawrote:
https://sipXserver:8085/mailbox/xxx/preferences/activemessages results
in the message ‘context not understood’.
** **
Looking at the current java file from here (
On 11/5/2012 2:16 PM, Tony Graziano
wrote:
I am looking at a strange issue with a system which
had a drive failure. We replaced the drive and reloaded (did not
restore) the system, then updated it to the latest update. We see
the proxy staying steady at
I am seeing the following message within the rls logs:
sipxrls:OsSSLServerSocket SSL_accept SSL handshake error:\n SSL error: 1
'error:0001:lib(0):func(0):reason(1)'
sipxrls:OsSSLServerSocket SSL_accept SSL handshake error:\n SSL error:
336027900 'error:140760FC:SSL
On 11/5/2012 4:50 PM, Tony Graziano wrote:
I am seeing the following message within the rls logs:
sipxrls:OsSSLServerSocket SSL_accept SSL handshake error:\n SSL
error: 1 'error:0001:lib(0):func(0):reason(1)'
sipxrls:OsSSLServerSocket SSL_accept SSL handshake error:\n SSL
error:
On Mon, Nov 5, 2012 at 11:50 PM, Tony Graziano tgrazi...@myitdepartment.net
wrote:
I am seeing the following message within the rls logs:
sipxrls:OsSSLServerSocket SSL_accept SSL handshake error:\n SSL error:
1 'error:0001:lib(0):func(0):reason(1)'
sipxrls:OsSSLServerSocket SSL_accept
Two things can cause this.
1. You have a port scanner scanning your TCP ports.
2. You have a remote connection attempting to connect as using an
unsupported version of SSL (TLSv1, SSLV2, SSLV3). I have checked
sipXportLib ssl implementation and we are configured to support all
three so
The SSL error in the RLS logs is actually benign, I'm not so sure about the
no container found error.
On Mon, Nov 5, 2012 at 3:50 PM, Tony Graziano
tgrazi...@myitdepartment.netwrote:
I am seeing the following message within the rls logs:
sipxrls:OsSSLServerSocket SSL_accept SSL handshake
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