Hello,
I'm attempting to get Mix Meeting work with freeswitch they use IP
authorization only, however username / password are required in the config
file.
If I set them to an arbitray value, Mix returns "forbidden".
If I leave them out, then Freeswitch complains that the gateway is invalid.
un 12, 2008 at 1:32 PM, Brian West <[EMAIL PROTECTED]> wrote:
> I know change the {} to [] this has to be a bug introduced in a recent
> patch in trunk.
> /b
>
> On Jun 12, 2008, at 12:12 PM, Nick Temple wrote:
>
> Show me the INVITE:
>
>INVITE sip:[EMAIL PROTECT
io 28950 RTP/AVP 0 8 3 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20
On Thu, Jun 12, 2008 at 1:05 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Show me the INVITE.
e can help them put a listing on that
> page
> as soon as their interop works ?
>
>
>
> Matt
>
>
> On Wed, 11 Jun 2008, Nick Temple wrote:
>
> > Hello,
> >
> > I am attempting to get Gafachi setup on our system.
> > They are telling me that
ngly tiny)
hurdle.
I'm at work right now, I'll try to get on IRC this evening or tomorrow.
Nick
On Thu, Jun 12, 2008 at 9:20 AM, Brian West <[EMAIL PROTECTED]> wrote:
> I don't get what you're doing here.. effective_caller_id_number and
> origination_caller_id_n
West <[EMAIL PROTECTED]> wrote:
>
> On Jun 11, 2008, at 8:18 PM, Nick Temple wrote:
>
> call = "{ignore_early_media=true,originate_timeout=30}sofia/gateway/
> sip.gafachi.com/1XX";
>
>
>
> Try to export instead of set. The variable isn't on the b-
Hello,
I am attempting to get Gafachi setup on our system.
They are telling me that the caller id is not being sent with the request,
and are thus rejecting the calls.
Is there a short FAQ on how to send the caller id?
in vars.xml, I have:
call.js:
call = "{ignore_early_media=true,origi
oxy values can be left out. We will
> just use that.
> /b
>
> On Jun 4, 2008, at 6:14 PM, Nick Temple wrote:
>
>
>
>
>
>
>
>
>
>
>
>
>
> ___
> Fr
Here's my setup:
conf/sip_profiles/outbound/lesnet.xml
Then:
sessionx.originate(session, "sofia/gateway/lesnet/1502XXX", 30);
HTH,
Nick
On Wed, Jun 4, 2008 at 6:30 PM, Klaus Teller <[EMAIL PROTECTED]> wrote:
> Hi Brian,
>
> I created a LES profile loca
Is there an API or module that implements a routine similar to Asterisk's
"WaitForSilence"?
I'm looking for very basic "wait until someone is done speaking before
playing audio" functionality.
Or maybe an example app that does something similar?
I've searched the site, but don't see anything imm
Still testing, but it seems to be working now! -- thx.
On Fri, May 2, 2008 at 2:02 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Yes please update.
> /b
>
> On May 2, 2008, at 12:27 PM, Nick Temple wrote:
>
> rc3 tarball - I can upgrade to latest svn if that could help (
rc3 tarball - I can upgrade to latest svn if that could help (looking
forward to rc4).
On Fri, May 2, 2008 at 12:57 PM, Brian West <[EMAIL PROTECTED]> wrote:
> Nick, Before we go any further what svn rev are you running?
>
> /b
>
> On May 2, 2008, at 11:23 AM, Nick Tem
to be this way but your first script has /
> tmp/test.wav and your second script has /mnt/test.wav - was that
> intentional?
>
> -MC
>
> Sent from my iPhone
>
> On May 1, 2008, at 9:28 AM, "Nick Temple" <[EMAIL PROTECTED]> wrote:
>
> > Hello,
>
Hello,
I am extremely new to VOIP in general and have been using freeswitch only a
couple of days,
so if I have missed something in the docs please point me in the right
direction.
I am attempting to create an outbound call system -- mostly research and
playing at
this point.
The initial goal is
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