Hi,
I'm using "record_session" to record a one-legged call, that is, a call
which is established between a FS extension that plays audio files and a
person. The call is recorded OK, but the volume of the person talking is
several times lower compared to that of FS playback.
Is there a way
Just issue "make current" and you'll be all set.
/b
On Sep 30, 2008, at 5:08 PM, jflowers wrote:
> Once I have done whatever I need to do to get to the svn version,
> can I just
> do svn update, ./configure make && make install? Can I use make
> speedy-current? Reason I ask is that this is a
I have a several weeks old freeswitch-snapshot installed on FreeBSD 6.2 that
I would like to update and keep current with svn but I am unsure of the
correct procedure.
I checked out freeswitch trunk, ran bootstrap.sh, ./configure and make but
am having second thoughts about running make install o
The main reason for that is because we are in the beta stage for 1.0.2
so every commit is one step closer to the next release so we are encouraging
as many ppl as we can
to try it so make sure all the problems they may encounter have been
addressed.
once we get the 1.0 branch where we want it we w
HelloFreeSWITCHers,
I am just curious as to which version of FS you all run on your in
use servers?
It seems like the SVN of trunk is being recommended to fix a lot of
issues, but are people actually using trunk on customers servers? I
am running 1.01 and haven't had issues with it, but I
if you are using event_socket you can subscribe to the channel_state event
and it will tell you
every time the channel changes states.
On Tue, Sep 30, 2008 at 1:29 AM, Baskar <[EMAIL PROTECTED]> wrote:
> Hi,
>
> I don't understand.could you please tell me in brief? . I want to check my
> channel
Hi,
I don't understand.could you please tell me in brief? . I want to check my
channel status whether it is ringing or answer or hangup.
On Mon, Sep 29, 2008 at 11:38 PM, B Karthik <[EMAIL PROTECTED]> wrote:
> Set hangup_after_bridge variable to false and run the info command after
> bridge, you
Jeremy,
Short answer:
The commercial version of zoiper http://www.zoiper.com/ (30 euros) has
the features you request listed but I only ever used their 'lite' version.
Long answer:
I found the 'lite' version on Leopard useful but it has is weirdness -
some I would attribute to be a stripped d
hopefully on 2 different servers?
please send a private email with the box login credentials to
[EMAIL PROTECTED]
and we will contact you.
On Tue, Sep 30, 2008 at 4:06 AM, Jon Bruel <[EMAIL PROTECTED]> wrote:
> OK, you can access it directly now, and yes it would be fine if we can
> find out w
There is a separate apply-inbound-acl and apply-register-acl
And yest the point of apply-inbound-acl is that all matches from that ip
will be allowed in blindly.
it's the opposite of doing Digest auth.
There is still a way to associate an a ip range with a user so when you are
let in over
acl it
OK, you can access it directly now, and yes it would be fine if we can
find out what's wrong once for all. Phone is: +45 45 16 1001. I'm on
CET.
Regarding the calling application: It is Asterisk, setting up one leg to
MOH and the other to FS, so audio is certainly going through.
Regarding the resp
Hi,
Without the info itself I am able to get the Answer state and lots of
varialbles, is it possible that I can get some variables via api?
On Mon, Sep 29, 2008 at 11:38 PM, B Karthik <[EMAIL PROTECTED]> wrote:
> Set hangup_after_bridge variable to false and run the info command after
> bridge
The software in question is a custom dialplan replacement module for
freeswitch that is available for licensing...
As far as the setup goes you set up the sip profiles as normal, ratedecks
and other items are loaded into a DB and routing happens automagically...
> From: Noah Silverman <[EMAIL
That makes sense.
However it might make sense for me to add something to the wiki about
this. It isn't documented anywhere that an "allow" in the acl will
bypass the directory and registration.
On a separate topic, I was just reading a post of yours from February
where you describe your LC
The ACL is a way to specify a group of trusted machines and the system will
bypass auth on those calls...
If you need something from the directory don't use the ACL...
If you don't want FS to respond to SIP from unknown IP Addresses that's a
more appropriate job for your firewall software (iptabl
Hi,
As some of you are probably aware, I've had a really hard time getting
asterisk to work with FS.
The effective_caller_id_number and the accountcode were not getting
logged or passed through on outgoing calls.
I finally solved the problem, but attribute it to some unexpected
behavior of
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