Is this a new install of the FreeSWITCH package or is it an upgrade from and
earlier package?
Mark J Crane
mc...@yahoo.com
--- On Tue, 12/8/09, Nandy Dagondon nandy1...@gmail.com wrote:
From: Nandy Dagondon nandy1...@gmail.com
Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch -
Hello Nandy,
thanks for your hint, but it's a bit more than that.
In our application which is handled via XML-Curl, the user can define
it's forwards on a web interface. He can enter mixed local or external
numbers which are called sequentially or in parallel.
Best regards
Peter
Nandy Dagondon
Hello,
I can't get call transfer to work with a SPA2102 adapter.
I don't think it has something to do with FS, but I'm hoping someone here
could help me.
I do not get a new line in the phone (by pressing the R button), all DTMF
tones are sent as audio to the other connected phone.
Anyone got it
On Tue, Dec 8, 2009 at 5:42 AM, DJB djbin...@yahoo.com wrote:
One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
calls with load balancing from another switch. Thus, the traffic type are
pretty much identical and both FSs have exactly the same on configuration.
Any
I'm still working on this issue, and decided to take a look at the
openzap code.
First, I figured out that the parameter name for callerid
is enable_callerid rather than enable-callerid.
I also figured out that
this parameter defaults to TRUE (which is coherent with the observed
behaviour on
Thought I'd send this little hurrah! As there seems to have been a lot
of negativity on this list lately.
From my point of view, having looked at many solutions out there, FS is
still number one with regards to flexibility and performance. I cannot
imagine doing what I'm using FS for, with
No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time. It's needle in
the haystack stuff.
Here's what I know.
I have an external process listening for DTMF events. If I detect '*' I
do a kill uuid on the B leg. On a number of
Michael Jerris wrote:
Our plan for 1.0.5 is that we will also have rpm and deb packages for many
distros on our own repo. Stay tuned. This has been another major reason
for the delay in 1.0.5.
Great news. I also prefer to use packages whenever possible, so as to know
what software is
I recall implementing that back when we released openzap, it should be in there
unless someone chopped it out for some reason. Look for
zap_channel_send_fsk_data
Mike
On Dec 9, 2009, at 6:01 AM, François Legal wrote:
I'm still working on this issue, and decided to take a look at the openzap
I have the same problem with a HandyTone 502 adapter.
Anyone got any hints to get the flash button to work?
On Wed, Dec 9, 2009 at 11:25 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
Hello,
I can't get call transfer to work with a SPA2102 adapter.
I don't think it has something to do with
src/switch_ivr_bridge.c
This could just as well be a glare condition when the call is in process of
tearing down.
Mike
On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote:
No doubt, but that’s a little difficult as this only happens occasionally and
I have 200 calls going on at the time.
Hey There,
I came in not seeing any former posts of yours, so if this one is unhelpful,
just delete.
I did my FS install using PFsense as well. Its been working famously for a
few months now. Very happy.
I wrote what I discovered for FS on PFsense in this wiki:
I would have tended to agree with the glare, however, before I killed
both sides, I was back to my issue of the call not clearing down at all.
(rtp timeout eventually does it)
Thanks for the pointer to the source.
Regards,
From:
It worked!
Tnx!
Em 08/12/2009, às 16:51, Brian West escreveu:
Best option for you is to use 96 in the sofia profile you're using to
talk to these broken devices.
/b
On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote:
Dear list,
Some Nec phones sends DTMF RFC2833 with
In my FreeSWITCH environment, calls are originated out to customers who are
placed into a fifo upon answer. There are members (x-lite endpoints) in this
fifo who handle those customer calls. I am writing a monitoring application
that uses event_socket to watch the channels involved in this
fifo list issue this API and get the fifo XML and get the caller's
uuid out of the list.
/b
On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote:
The short version of my question is this: how do I programmatically
determine which channel uuid the consumer channel in a fifo is
connected to?
Hello,
in our dialplan we have enabled multiple-registrations, so 2 phones can
register on a single directory entry.
param name=multiple-registrations value=true/
Both phones are registered, both phones can be called and each phone can
call the other phone.
However in an attended_transfer
Regarding Mac OSX 10.5/6 can you point me where the latest FS binary file
is?
Thanks in advance,
-E
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
On Tue, Dec 8, 2009 at 3:59 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Thought I’d send this little hurrah! As there seems to have been a lot
of negativity on this list lately.
I hereby multiply all the negative comments by -1. :P
-MC
That is more dependent on the endpoint than on the switch itself. I guess
you can always use mod_limit to come up with some crazy key to identify one
endpoint or the other but still it seems overly complicated for something
that is not supposed to be working this way.
You can also park the call
Nik Middleton wrote:
I cannot imagine doing what I'm using FS for, with any other product. Yes
it's frustrating at times, but this is largely down to a lack
documentation/samples.
Speaking of which... would this layout be good for a book on Freeswitch?
Preface
1. VoIP, Freeswitch, FS vs.
Looks good, but you've missed out billing and the key one, the event
socket which could be a chapter in it's self.
Do you have a publisher for it yet?
Regards
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On
Yes, I have two extensions.
I can even make them join a group, and if I call the group, the two extensions
will ring.
08-12-2009 kl. 23:45 skrev Nandy Dagondon nandy1...@gmail.com i
meddelelsen 7d0bfd8c0912081445v124dd6cs9174a201eb109...@mail.gmail.com:
have you created Extension 1002?
This is a new install, but it's grabbed from a pfSense repository.
09-12-2009 kl. 10:28 skrev Mark Crane mc...@yahoo.com i meddelelsen
187489.95329...@web56408.mail.re3.yahoo.com:
Is this a new install of the FreeSWITCH package or is it an upgrade from and
earlier package?
Mark J Crane
Hi Michael
Thankyou for the excellent wiki article, yes, I did follow your guide there,
all the way except to 'dialplan' and it seems that's the problem at the moment.
I would very much like to create the dialplan in the webinterface, and not in
the public.xml file.
But at the moment it only
Preface
1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc.
2. Choosing hardware options (server, phones, gateways)
3. Setting up FS
4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS
gateways, etc.)
5. Administering FS (CLI and GUI)
6. Customizing dialplan
Visit the friday meetings and we can help if you document it. ;)
/b
On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote:
I found the rosetta stone useful though woefully lacking in volume.
I guess that's true overall with the project.
___
I found the rosetta stone useful though woefully lacking in volume.
I guess that's true overall with the project.
Documentation is neither easy nor glamorous. The woefully lacking
documentation has been provided by a little group of people who've done a
big bit of documenting and a big group
Hey All. I am trying to get freeswitch to route to my socket handler
and am having a problem.
I am running freeswitch inside a virtualbox VM for testing purposes.
The vitualbox communicates with my host via the host only adapter.
The VM IP address is 192.168.56.3 and the laptop has the iP
That is what is nice about our community I'm more than willing to
answer the questions if you document them... as are many others in the
core team...we just have a lot to do and I think the best repayment is
documentation! ;)
/b
On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote:
On Thu, Dec
Dear FreeSWITCHers,
As of Friday Dec. 11th we will NOT accept any more bug reports on
1.0.4. You need to be on a 1.0.5pre or SVN trunk. 1.0.4 is over 6
months old and I really suspect your issues in 1.0.4 are already
fixed. We will release a new pre every monday morning till
Load sharing feature is coming off our Lucent Telica switch.
From: Tim Uckun timuc...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, December 9, 2009 2:26:41 AM
Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high
do you have something listening on 8084 ?
On Wed, Dec 9, 2009 at 4:35 PM, Tim Uckun timuc...@gmail.com wrote:
Hey All. I am trying to get freeswitch to route to my socket handler
and am having a problem.
I am running freeswitch inside a virtualbox VM for testing purposes.
The vitualbox
On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
do you have something listening on 8084 ?
Yes.
I figured out the problem. There was already an extension called 8084
and it overwrote the extension I defined.
Which brings me back to a question I had
the dialplan is dynamic there is no such thing
you have to look in your dialplan xml files because it's served up live.
FS has a different paradigm than asterisk.
On Wed, Dec 9, 2009 at 8:00 PM, Tim Uckun timuc...@gmail.com wrote:
On Thu, Dec 10, 2009 at 2:46 PM, Anthony Minessale
Hello,
I asked this question on my local linux user group mailing list, and got the
recommendation to ask here.
Anyway, at the moment I am running Asterisk on an IP04 embedded system.
http://www.rowetel.com/ucasterisk/ip04.html
It works well most of the time, however there are some bugs that
Brian,
I have been making efforts to fully support FreeSWITCH in AstLinux.
Our primary targets are low powered x86 boards like the Soekris and
Alix. x86, powerful enough, cheap enough (as low as $100), and about
12 watts. Not bad.
The Soekris net5501 and standard case will (I believe) take
On Thu, Dec 10, Brian May wrote:
Hello,
I asked this question on my local linux user group mailing list, and got the
recommendation to ask here.
Anyway, at the moment I am running Asterisk on an IP04 embedded system.
http://www.rowetel.com/ucasterisk/ip04.html
It works well most of the
Brian May br...@microcomaustralia.com.au wrote:
I do have a spare TDM400p card, although as it is full height, suspect this
isn't going to help.
Have a look at http://www.yawarra.com.au/
Some of their hardware (notably the Soekris Engineering boards:
http://www.soekris.com/) has a PCI slot.
how big does need to get before it rotates, what's the size exactly?
also how do I do it through dialplan via javascript?
On Fri, Dec 4, 2009 at 6:48 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
set rotate-on-hup to false in the cdr_csv config file
then it will only rotate when
Kristian Kielhofner wrote:
The Soekris net5501 and standard case will (I believe) take a full
height card. Then again you could use any board and get an external
SIP gateway (ATA). We don't currently support OpenZAP with FS in
AstLinux but I'd love to add support for it eventually.
Jason White wrote:
Have a look at http://www.yawarra.com.au/
Ok, found the net5501:
http://www.yawarra.com.au/hw-net5501.php
And here it is assembled for you:
http://www.yawarra.com.au/product.php?productCode=HW-NT55
I am not quite sure on one aspect, for extensions to work the TDM400P
I think I fixed the spandsp cross compile issues tonight, but I suspect there
is a good chance that I broke other builds in the process. I also did a bunch
of work to make the OS X Snow Leopard build cleaner today. Testing would be
much appreciated on both.
Mike
On Dec 9, 2009, at 10:47 PM,
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