Mark,
Thanks so much.. this now gives me an idea to start with something.
Kind regards,
Abilash
On Thu, Mar 27, 2008 at 12:24 AM, Mark Crane <[EMAIL PROTECTED]> wrote:
> "How can i input the telephone line (from my provider)
> into the server?"
>
> To do this you need an FXO. Most FXO's are car
> Are there any plans for documentation other than the wiki?
>
Yes! It has been slow going but we are trying to get some momentum
built up with the freeswitch-docs project. We're still in the very
early stages. Right now the wiki is pretty much everything, and you're
right in noting that it is
You can ignore everything in vars.xml and just set it. And
preprocessors directives can't be commented out.
/b
On Mar 28, 2008, at 4:32 PM, Tim Ferguson wrote:
> Thanks for the reply. My questions wasn't about those specific params
> but rather the ability to look up params overall I suppose.
Thanks for the reply. My questions wasn't about those specific params
but rather the ability to look up params overall I suppose.
Additionally, I found that if I set $domain is being set to 127.0.0.1
which I see in vars.xml domain=$${local_ip_v4}. I commented out this
line and manually set the dom
On Mar 28, 2008, at 3:53 PM, Tim Ferguson wrote:
> Hello,
>
> Is there more documentation, even at the code level I could use to
> reference what certain module variables are and can contain.
>
> For instance:
>
> default.xml:
>
>
> Where is the sip-ip param used, what are the possible values, c
Hello,
Is there more documentation, even at the code level I could use to
reference what certain module variables are and can contain.
For instance:
default.xml:
Where is the sip-ip param used, what are the possible values, can you
comma delimit multiple ip addresses, do you add additional sip
I've successfully build freeswitch on the following BSD machine:
FreeBSD bsd.quist.ca 6.2-RELEASE-p7 FreeBSD 6.2-RELEASE-p7 #2:
The only small caveat was that gmake was needed for the make install portion.
On Fri, Mar 28, 2008 at 8:24 AM, Mark Crane <[EMAIL PROTECTED]> wrote:
> I'm trying to ge
Ritesh,
Welcome to the FS community! Brian already mentioned a few things, like
the wiki and the IRC channel, so definitely get familiar with those.
One thing I'd like to recommend is that you check out the dialplan xml
files. (I hope you're comfortable with XML!) In the freeswitch conf
gs
> to learn.
>
> Best things to do are start here http://wiki.freeswitch.org and
> #freeswitch on irc.freenode.net
>
> Remember I'll be very glad to answer questions but the requirement is
> that you MUST put the info you learn on the wiki and pay it forward to
> others.
&g
and just to point it out, you can also do:
session1 = new Session();
session1.originate(session1, "{ignore_early_media=true}sofia/gateway/
asterlink.com/19184249378");
session1.execute("bridge", "sofia/gateway/asterlink.com/19184238080");
or even better
// this will transfer the channel into t
Great, guess I'll drop that from my config then. Sorry Nicolas must be
something else.
- Dale
On Fri, 2008-03-28 at 09:53 -0500, Brian West wrote:
> Dale,
> This was due to a rogue SRV record saying use TCP that has since been
> corrected after a month of talking to them about it.
>
> /
On Mar 28, 2008, at 8:07 AM, Ritesh Singh wrote:
Hi All,
I am very new to freeswitch. It will be great if some one can tell
me few things:
Welcome to the community.
1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y.
Then what changes i am supposed to do and at what plac
Dale,
This was due to a rogue SRV record saying use TCP that has since been
corrected after a month of talking to them about it.
/b
On Mar 28, 2008, at 8:50 AM, Dale Thatcher wrote:
> I had some slow connection problems with sipphone that were solved by:
>
>
>
> in the sip profi
I had some slow connection problems with sipphone that were solved by:
in the sip profile, might be worth a go. BTW this is a bug with
sipphone, not with Freeswitch.
- Dale
http://myhelpa.com - Sign up to be a Beta Helpa.
On Fri, 2008-
Hi All,
I am very new to freeswitch. It will be great if some one can tell me few
things:
1) If my ip is x.x.x.x and i want to call someone at ip y.y.y.y. Then what
changes i am supposed to do and at what place. I would like to use the
mod_portaudio for this purpose.
2) I started the windows fre
Thanks! It worked as advertised.
The only problem I have now, is my provider (I'm trying gafachi now),
I'm getting about one or two seconds delay on the audio, which is
pretty bad.
One other thing, it takes about 5 or 10 seconds to get a ring tone
after answering the first call, is there anyway t
Henk Oegema wrote:
>
>> Message: 2
>> Date: Thu, 27 Mar 2008 17:05:43 -0500
>> From: Brian West <[EMAIL PROTECTED]>
>> Subject: Re: [Freeswitch-users] How to reply ?
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <[EMAIL PROTECTED]>
>> Content-Type: text/plain; charset=US-ASCII; forma
On Fri, 2008-03-28 at 09:37 +0100, Henk Oegema wrote:
> > Message: 2
> > Date: Thu, 27 Mar 2008 17:05:43 -0500
> > From: Brian West <[EMAIL PROTECTED]>
> > Subject: Re: [Freeswitch-users] How to reply ?
> > To: freeswitch-users@lists.freeswitch.org
> > Message-ID: <[EMAIL PROTECTED]>
> > Content-T
I been putting FreeSwitch in production environment under FreeBSD 6.x/FreeBSD
7.x/FreeBSD 8.x/FreeBSD-stable/FreeBSD-current since March 2006 up to now and
it works
very well without the hassles that you encountered. Which makes me wonder with
regards
to your claim that you are creating a package
> Message: 2
> Date: Thu, 27 Mar 2008 17:05:43 -0500
> From: Brian West <[EMAIL PROTECTED]>
> Subject: Re: [Freeswitch-users] How to reply ?
> To: freeswitch-users@lists.freeswitch.org
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes
>
> Pres
I'm trying to get FreeSwitch installed on FreeBSD 7.
Right now trying to find all the dependencies.
* SVN:
Lots of choices here.
* GNUMAKE: The GNU version of make.
gmake-3.81_2 (required by autoconf-2.61_2)
* AUTOCONF: Version 2.50 or higher
ports/devel/autoconf261/
21 matches
Mail list logo
