I'd rather go with the ip7000 since it has better audio gear in it.
For a deskphone everything Polycom = ip450 is absolutely wideband
enough for a deskphone.
Personally I am currently a total fan of the VVX which is a video
deskphone with the same audio as a IP6000
Am 05.11.2009 um 18:45
-You could check the sofia debug for r15332 here:
http://pastebin.freeswitch.org/11008
Phone/Devices:
The caller is the DID provider's Switch. The callee (which also sends the
REFER) is Asterisk 1.4.26.
Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same
results.
fixed in 15376
On Thu, Nov 5, 2009 at 12:35 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
yes sounds like a bug.
I think i redid it and forgot to check for true still =0
On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb larc...@yahoo.com wrote:
Thanks for the help. Yes, I am using a
yes sounds like a bug.
I think i redid it and forgot to check for true still =0
On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb larc...@yahoo.com wrote:
Thanks for the help. Yes, I am using a lua script to handle inbound calls
with continue_on_fail set to true:
session:execute(set,
Hello
As an alternative to more expensive alternatives like OpenVox or Sangoma,
I'd like to order an X100P clone from www.x100p.com for use in France.
According to a PDF on the site, the reason this card gets bad reviews is
that the Silicon labs Si3012/Si3035 DAA chip used in the original
On Wed, Nov 4, 2009 at 10:22 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If
SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out
to landline call, as I provide sample below;
condition
Hi,
actually i'm trying to setup an IVR that, when the choice is done,
transfer the call to a group, really simply.
Here the dialplan in default context to handle call to group (four
extensions, one for group, from 2001 to 2004)
http://pastebin.freeswitch.org/11014
Here the output log
It cleans up after itself fine, but it is an indication of some issue
in the code we need to address. if you can reproduce this in svn
trunk, please file a bug on jira.freeswitch.org with details how to
reproduce.
mike
On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote:
Hello!
I have
This would be specific to the zaptel driver for that card, not openzap.
mike
On Nov 5, 2009, at 1:43 PM, Fred-145 wrote:
Hello
As an alternative to more expensive alternatives like OpenVox or
Sangoma,
I'd like to order an X100P clone from www.x100p.com for use in France.
According to a
Debian 5.0.3
FreeSWITCH Version 1.0.trunk (15376M)
openzap and libpri-1.4.10.2
dahdi-linux-complete-2.2.0.2+2.2.0
Digium Wildcard TE110P T1/E1 Card (running as a T1)
This was working with zaptel. I thought that I would upgrade from zaptel
to DAHDI, but it's generating no such device or address
Hello
I updated to 15376 added some build depends and still no joy. Any more
pointers. Thanks.
--FC
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I thought that I would upgrade from zaptel to DAHDI,
After I send the message, the answer comes to me. I guess that's the way
things work. :-) I had forgotten to define the channels in
/etc/dahdi/system.conf. Here are the settings, and things are working.
Thanks for listening. :-)
The internal.xml also has an ext-rtp-ip variable and in trying to
understand what this is for (my version of fs is 1) I noticed in trunks
conf file it is explained. So the available options that I have given
my setup is multihomed with a lan/wan setup where the wan interface is
dynamic would be a
Here is what is believed to be a bug found by Robert Hadley found in
Freeswitch1.0.4/scripts/gentls_cert.in build file:
Fix for gentls_cert remove to work:
[scripts]# diff gentls_cert.in gentls_cert.in~
129c129
if [ -d ${CONFDIR}/CA ]; then
---
if [ ! -d ${CONFDIR}/CA ];
auto-nat tries to use upnp/nat-pmp to figure it out... auto will just
put your IP in there.
The other values can be stun:host or an IP.
The docs in trunk show this now... its really simple to understand but
you should NEVER have to set that unless you have a nat scenario that
requires you
In the future please post issues to jira.freeswitch.org along with a
diff -u from the root freeswitch source directory.
This already seems to be fixed in svn trunk can you verify.
Thanks,
Brian
On Nov 5, 2009, at 5:00 PM, Jerry Richards wrote:
Here is what is believed to be a bug found
auto-nat tries to use upnp/nat-pmp to figure it out... auto will just
put your IP in there.
The other values can be stun:host or an IP.
The docs in trunk show this now... its really simple to understand but
you should NEVER have to set that unless you have a nat scenario that
requires you to lie
If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip
REMOVE them. If your multi homed then you'll need to set them.. we
don't listen on 0.0.0.0 you'll have to start a profile for each IP you
wish to listen on.
/b
On Nov 5, 2009, at 5:20 PM, Joseph L. Casale wrote:
FYI,
the latest Windows SVN build now includes the option to configure FusionPBX,
a port of the pfsense/FreeSWITCH gui: http://fusionpbx.com/index.php
If you plan to install it someplace other than the default location of
C:/FreeSWITCH just make sure to update the paths in Admin, System Settings
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
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global_getvar local_ip_v4
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 5-Nov-09, at 3:40 PM, Joseph L. Casale wrote:
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have
I am trying to make a call through a Gateway that sends the INVITE with no
SDP and ONLY wants the 200 OK w/SDP when the callee answers.
For some reason, Freeswitch answers the call with 200 OK w/SDP even before
the callee answers the phone. Is this to provide ringback? Can I disable
that
If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip
REMOVE them. If your multi homed then you'll need to set them.. we
don't listen on 0.0.0.0 you'll have to start a profile for each IP you
wish to listen on.
I am multihomed, and the wan nic is dynamic. Is there any way for me
Just use ${local_ip_v4} then.
and enable auto-restart on the sofia.conf.xml
/b
On Nov 5, 2009, at 5:45 PM, Joseph L. Casale wrote:
I am multihomed, and the wan nic is dynamic. Is there any way for me
to
control how it guesses the IP of a `specific` interface without the
use
of a third
This all depends on what you're doing in your dialplan if you do stuff
like record it requires media and will trigger it.
A sip trace or some such debug would be more helpful then a terse
description of a problem.
/b
On Nov 5, 2009, at 5:49 PM, Jerry Richards wrote:
I am trying to make a
vars.xml
but watch out the core will trump local_ip_v4 if it happens to change.
/b
On Nov 5, 2009, at 5:40 PM, Joseph L. Casale wrote:
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
__
2009/11/6 Giovanni Maruzzelli gmar...@celliax.org
On Thu, Nov 5, 2009 at 6:57 PM, Seven Du dujinf...@gmail.com wrote:
Ciao Giovanni,
Do you still plan to merge this?
Sorry Seven,
I've lost track of this, and now I'm so sick I'm completely un-useful ;).
That's OK, we all have a lot of
If you need a really cheap entry-level phone that does Polycom's HD Siren
codecs then check out the IP 335 that just came out. It's very basic but I'm
hearing good things from people who've used them.
-MC
On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman rob4manh...@gmail.com wrote:
Hey all,
Looking
Just use ${local_ip_v4} then.
and enable auto-restart on the sofia.conf.xml
Cool, it seems to always use the public ip, quite reliably.
That is what I am after (why), is there something in the code that
forces it to favor for example, non RFC 1918 addresses?
It works, I just want to understand
Screenshots for the FusionPBX graphical interface
http://fusionpbx.com/files/fusionpbx_com/screenshots/index.php
--- On Thu, 11/5/09, Carlos Talbot carlos.tal...@gmail.com wrote:
From: Carlos Talbot carlos.tal...@gmail.com
Subject: [Freeswitch-users] FusionPBX
To:
Hi all,
Is there any way to receive events while running a perl program with the
help of mod_perl??
I've seen some functions related to sending and receiving events in the
mod_perl wiki. But I don't know how to use that.
Any help!!!
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