into the dialplan and end the script
// thus reducing overhead of leaving JS open.
session1.execute(transfer, 19184238080);
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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sip:[EMAIL
oops i meant \$
On Thu, Apr 10, 2008 at 4:24 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
try escaping the $ with /$
On Thu, Apr 10, 2008 at 2:21 PM, Tamas Cseke [EMAIL PROTECTED]
wrote:
Hello,
I would like to mark the uuid of the called channel for further process.
(Spying
this message to freeswitch-users' list.
On Thu, Apr 10, 2008 at 8:35 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
sure let us know what you need.
Despite FreeSWITCH didn't reach 1.0 yet, would be nice to list some
cases of success. I've heard We've never heard of FreeSWITCH, is
anyone using
at 4:24 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
try escaping the $ with /$
On Thu, Apr 10, 2008 at 2:21 PM, Tamas Cseke [EMAIL PROTECTED]
wrote:
Hello,
I would like to mark the uuid of the called channel for further
process.
(Spying this channel with eavesdrop
point and do not riddle my
email with comments and I will be happy.
We look forward to your success with FreeSWITCH and welcome to the
community.
On Fri, Apr 11, 2008 at 10:23 AM, kokoska rokoska [EMAIL PROTECTED]
wrote:
Anthony Minessale napsal(a):
The main reason we write the port
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:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent:* Monday, April 14, 2008 4:33 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Dingaling In Component Mode How To
CallJingleTo SIP
The easiest way to do it in component
/freeswitch-users
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really need).
Thanks!
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*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent:* Tuesday, April 15, 2008 3:31 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] event BACKGROUND too late
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sip
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of
bypass_media and examine all of your sip profiles for the
inbound-bypass-media param.
On Mon, Apr 21, 2008 at 8:52 AM, kokoska rokoska [EMAIL PROTECTED]
wrote:
Thank you very much, Anthony, for your answer!
Anthony Minessale napsal(a):
do you have bypass_media=true set or the inbound-bypass-media
, Apr 21, 2008 at 10:43 AM, kokoska rokoska [EMAIL PROTECTED]
wrote:
Anthony Minessale napsal(a):
please set the environment variable TPORT_LOG=1 to log the sip traffic
inline.
then capture all of the console output reproduce your issue
before you try the call, please enter the command
the final config
will contain the actual ip.
On Mon, Apr 21, 2008 at 1:47 PM, kokoska rokoska [EMAIL PROTECTED]
wrote:
Anthony Minessale napsal(a):
2008-04-21 18:47:48 [DEBUG] sofia_glue.c:1523 sofia_glue_activate_rtp()
AUDIO RTP [sofia/0002*00/800123456] 0.0.0.0:21322-212.20.119.45
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sip:[EMAIL PROTECTED] [EMAIL PROTECTED
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sip:[EMAIL
at
the beginning, which means the delay increase.
Thomas
PS: i attach a patch for better understanding
--
*Von:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *Im Auftrag von *Anthony
Minessale
*Gesendet:* Dienstag, 6. Mai 2008 00:38
*An:* freeswitch-users
(with codec-ms = 60ms), and with
15-20ms jitter.
Anthony Minessale wrote:
Have you tried setting the codec-ms in the portaudio.conf.xml to 60 or 120
ms?
Maybe the soundcard is not able to do 20ms intervals and portaudio is
doing the least common multiple and chopping it up for us.
I think what's
RTP payloads are a 7 bit number 0-128 divided in 2.
0-96 are reserved by the iana
97-128 are considered dynamic.
When matching codecs between 0 and 96:
1) The payload MUST be matched by the _number_.
When matching codecs between 97 and 128: (dynamic range)
1) The payload MUST be matched by the
]
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sip:[EMAIL PROTECTED] [EMAIL
it's invalid because you hungup
On Thu, May 15, 2008 at 3:49 PM, Zelei Csaba [EMAIL PROTECTED] wrote:
Tried it. It's ok with the threeway now, but still get the invalid uuid
error if I hang up
2008/5/15 Anthony Minessale [EMAIL PROTECTED]:
There was a bug regarding that, try revision 8412
I tell
it to hang up me and bridge them together?
2008/5/15 Anthony Minessale [EMAIL PROTECTED]:
it's invalid because you hungup
On Thu, May 15, 2008 at 3:49 PM, Zelei Csaba [EMAIL PROTECTED]
wrote:
Tried it. It's ok with the threeway now, but still get the invalid uuid
error if I hang
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in a proper way that it isnt need to
hang up the other channel
Anthony Minessale wrote:
try latest trunk i think i made what you want work too.
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0xb7ef0b36 in dummy_worker (opaque=0x81c58a0) at
threadproc/unix/thread.c:138
#10 0xb7e0b31b in start_thread () from /lib/tls/i686/cmov/libpthread.so.0
#11 0xb7d5157e in clone () from /lib/tls/i686/cmov/libc.so.6
Any ideas?
regards
helmut
Anthony Minessale schrieb:
| I thought I already
speaking slots we can open up if anyone wants to
speak.
I look forward to seeing everyone there!
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and organization (if applicable) and a description
about how you use FreeSWITCH.
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to fork call back to the dialplan, not to the endpoints - ie
parallelize walking through the dialplan from some point. (Nearly same
thing Asterisks Local channel does.)
Any sugegstions are welcome :-)
Best regards,
kokoska.rokoska
Anthony Minessale napsal(a):
the originate string can
with 120ms interval, I still hit around 170 channels limits
in
one room.
Let me play around more with multiple room setup.
Rgds,
Johny K.
On Thu, May 29, 2008 at 10:56 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
We don't do much testing on the conference but i can give
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,
Johny K.
On Thu, May 29, 2008 at 10:56 AM, Anthony Minessale
[EMAIL PROTECTED] wrote:
We don't do much testing on the conference but i can give you a
few
pointers.
is it a 32 or 64 bit box?
if it's 32 you can try this as root before you start up.
ulimit
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GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED]
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FreeSWITCH Developer Conference
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/options/freeswitch-users
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yes update and try trunk.
On Mon, Jun 2, 2008 at 9:23 AM, Sune Kristensen [EMAIL PROTECTED] wrote:
I am using the 1.0.0 release. I can try the svn code if that is what you
mean by latest J
/Sune
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent:* Tuesday, June 03, 2008 2:01 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Shutting down RFC
Can you contact company who makes this phone and ask them to explain what
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…
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*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent:* Monday, June 02, 2008 2:00 PM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] OZ question: disconnected number
handling
they are sending inband
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sip
. We haven't tried on anything else.
I assume you tested it on a linux machine.
I guess if it's not replicating there, it must be a windows specific issue.
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*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent
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Anthony Minessale II
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Anthony Minessale II
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Anthony Minessale II
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I'll be going on vacation till June 21st so our production may lag just a
teenie bit.
Don't worry, we should be releasing 1.0.1 while I'm gone and with the slow
down it gives our community time to prepare for the upcomming ClueCon 2008
http://www.cluecon.com
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Anthony Minessale II
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Anthony Minessale II
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?
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*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Anthony
Minessale
*Sent:* Tuesday, June 24, 2008 7:14 AM
*To:* freeswitch-users@lists.freeswitch.org
*Subject:* Re: [Freeswitch-users] Recent VAD enhancement?
check the packet flow in wireshark also
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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=%(2000,4000,440,480));
The main problem i have is that there is no tone at all. No dial tone, and
no ring tone. Any wrong with this code?
Thanks,
Klaus L'Imbecile.
Original-Nachricht
Datum: Mon, 23 Jun 2008 08:45:56 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users
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try again on trunk
On Thu, Jun 26, 2008 at 3:32 PM, Anthony Minessale
[EMAIL PROTECTED] wrote:
or you could get the record thing working instead of overlooking it.
it might be a bug so we should figure it out.
On Thu, Jun 26, 2008 at 3:13 PM, e schmidbauer [EMAIL PROTECTED]
wrote:
i've
[6410867975] cause:
[CHAN_NOT_IMPLEMENTED]
Thanks,
Klaus.
Original-Nachricht
Datum: Thu, 26 Jun 2008 09:13:41 -0500
Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Java Support
If you want
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Von: Anthony Minessale [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Java Support
put
session.setVariable(ringback, file path to some audio file);
before you bridge.
There are more complicated ways to do it if you really want but only
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on with headphones all the times :) They have other work to do!
Anthony Minessale wrote:
if you add these 2 extensions: 7010 will be for agents who will hear
music till someone calls
and 7011 will be the customer who will hear hold music until an agent is
free.
extension name=test
condition
Right you are,
sorry forgot that bit.
update and try again.
On Tue, Jul 8, 2008 at 2:12 PM, Andy Spitzer [EMAIL PROTECTED] wrote:
Woof!
On Tue, 08 Jul 2008 13:27:40 -0400, Anthony Minessale
[EMAIL PROTECTED] wrote:
Try latest trunk, I added support for what you want.
Thanks. I just
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