Ciao Ed,
for Skype, you can have a look at: http://wiki.freeswitch.org/wiki/Skypiax
Feel free to ask for more info if the wiki page is not complete/clear.
Ciao for now,
Giovanni
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9,
Hi Abdul,
Brazil does much more than that but we still on latim america and according
with 1st world countries, we are
considered non developed country.
The most funny thing about that: some people still thinks that we (brazillians)
lives in the forest with monkeys :-)
Regards,
Rodrigo Telles
Hi,
After some testing I came to the following conclusions :
1) The problem (timeouts and retries) I describe below only happens when
there is no radius server responding on the other side.
2) It only happens when using the latest cvs version of radiusclient. If
you use version 1.1.6 it
Hi Steve,
My point is more towards the high import taxes here. A foreign company
won't pay as much taxes to export to Brazil as brazilians would pay
import taxes for such goods.
I will give you a simple example: a Snom 320 phone would cost roughly US
$180.00 if bought in the US. Here it would
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
again an update to my little project:
I enhanced mod_openzap and ozmod_isdn so that I'm able to start and stop
q931ToPcap generation in ozmod_isdn from FS's console. q931ToPcap itself
isn't implemented yet.
To give you an idea about the
Please see in line comments.
Anthony Minessale wrote:
Sounds like having cake and eating it too.
The risk is obvious when using radius or some other additional
protocol for AAA that you will have trouble if the server is down.
Radius was designed to be fast and redundant, so typically you
Can you describe what you're trying to do and how you're doing it?
Maybe an example?
/b
On Jan 27, 2009, at 12:55 AM, Stephen Crosby wrote:
I'm running rev 11131, and I cannot make these hangup events work in
javascript... does anyone have a working example? I tried this script
from the
Helmut,
Nice work! Thanks for doing this.
-MC
On Tue, Jan 27, 2009 at 7:15 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Looks good, you might want to consider making the file name that is
generated be an optional argument but it's not necessary just a suggestion.
On Tue, Jan
Would you mind giving us some more information? Please use
pastebin.freeswitch.org to post your configuration files. Also, if you
could review the information here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs
it will help you gather what you need. See the section on openzap to
know what
Nope.
Tried all three ways, im stumped.
I had this issue with another provider and we never figured it out and then
I never used them anyways, but with new other provider I have to get it to
work.
Talked with the carrier, they say we are sending user as the CID itself.
Super weird, not sure
Jason Garland wrote:
If you want Speex support you need to target the chipset manufacturers:
Here is the Texas Insturments chipset that Polycom uses in the IP650
CPU is TNETV1050/C55x, rev 2 running at 162MHz with memory at 125MHz.
And here are the codecs that Chip supports from TI's
Out of curiosity which SIP messages have you been watching for on the
event socket? Also, how are you connected to the event socket? Are you
subscribing to all events and sifting through them to confirm that no
events are being fired when SIP messages are being sent?
-MC
Sent from my iPhone
Yeah that fixed it!
I have never even seen this option in the docs before, but that sure did the
trick.
Thanks guys, this was driving me nuts!
On Tue, Jan 27, 2009 at 7:26 PM, Raymond Chandler
intralan...@freeswitch.org wrote:
Ron McCarthy wrote:
Nope.
Tried all three ways, im stumped.
Great thanks to Jason for sharing Cherebrum's great discovery, this works
like a charm on my Ploycom IP 320 with G722 codec.
Chris
On Tue, Jan 27, 2009 at 4:48 PM, Jason Garland jgarl...@jasongarland.comwrote:
Something like this might do it... ;)
?xml version=1.0 standalone=yes?
!-- SIP
Hi,
My settings does not allow me to test the following right now. So I'm wondering
if somebody knowledgeable could help me answer the following question.
I do know that if i call Freeswitch, i can use Javascript to read DTMF even
without answering the call. My question is can i do this even
If the dtmf is in the media stream ie 2833 and you can't establish
media then no you wouldn't. Have you tried to do a pre_answer
instead of an answer to establish early media?
/b
On Jan 27, 2009, at 11:04 PM, Klaus Teller wrote:
Hi,
My settings does not allow me to test the following
On Wednesday 28 January 2009 08:38:17 Ron McCarthy wrote:
Yeah that fixed it!
I have never even seen this option in the docs before, but that sure did
the trick.
Please, add this to wiki.
--
С уважением, Кривушин Михаил
Ведущий специалист отдела телекоммуникаций,
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