Set the codec negotiation to greedy
From: Helmut Kuper helmut.ku...@ewetel.de
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Mon, 09 Feb 2009 08:54:04 +0100
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Can I force the A-leg codec in FS?
Hello,
in my
Hello,
update, when I remove all ozmod_ from ozmod_libpri lines in Makefile, it
compiles without errors.
regards
helmut
On 09.02.2009 08:48, Helmut Kuper wrote:
Hello,
during some imporvements on q931toPcap as well as debugging my TDM PRI
problem with loosing state sync after some time I
That and not enclosing in single quotes, thanks
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 09 February 2009 00:56
To:
Recently I noticed that fs (1.0.3 RC1) is not processing the calls.
Previously about 50-60 calls were remaining active at any time. But then
suddenly, only 5-6 active calls were there. So I checked sip INVITE's in
ngrep, and I noticed that originator were sending CANCEL after INVITE. So I
I notice it offers 18 which is G729 but these listed below are 100%
invalid. There is no such thing as G.729a, G.729b or G.729ab that are
valid in the SDP. I suspect if you start FreeSWITCH and crank it up
to debug level (console loglevel debug) you'll clearly see why this
is taking
Hi Brian,
But issue here is that, FS is not processing any such calls and not sending
488 to the caller.
Also the sip trace I had provided was from the caller to fs. FS does not
even bridge the call to terminator in between the INVITE and CANCEL from the
caller.
It just gives so many errors in
Here is the correct codec sent to fs, but it times out again.
http://www.nabble.com/file/p21911561/correct_codec_with_cancel.txt
correct_codec_with_cancel.txt
Ankit Gandhi wrote:
Hi Brian,
But issue here is that, FS is not processing any such calls and not
sending 488 to the caller.
Also
In the absence of any directives on lua/mysql, is it possible to launch
a PHP script from lua? All I need to do is pass some data to update a
db record. I don't need to have any links to the call as I intend to
call is in the hang-up callback
Regards,
1) please do not report bugs on the mailing list.
2) please report the bug on jira http://jira.freeswitch.org according to the
rules: http://wiki.freeswitch.org/wiki/Reporting_Bugs
If you have an issue that you want us to correct you will have to try the
latest SVN trunk (not a snapshot) issue
Option 3 does not have slow performance, Java apps can be highly
scalable high performance when written right, this is a serious strong
typed language unlike lua and javascript.
I actually tested such solution against MySql cluster with 500 calls/m
load script, scaled just fine.
Contact me off
hint: when you have harder to program under cons: that's usually how you
find the right choice ;)
On Mon, Feb 9, 2009 at 8:01 AM, pauld pa...@versafon.com wrote:
Option 3 does not have slow performance, Java apps can be highly
scalable high performance when written right, this is a serious
On Mon, 2009-02-09 at 13:30 +, Nik Middleton wrote:
In the absence of any directives on lua/mysql, is it possible to
launch a PHP script from lua? All I need to do is pass some data to
update a db record. I don’t need to have any links to the call as I
intend to call is in the hang-up
Hello,
I'm looking for the best codec/scenario for the last-mile and checked
speex codec capabilities (http://www.speex.org/comparison/nb_codecs.png).
This will be FS-FS interconnect (where one side uses portaudio, aka a
simple softphone).
As I see, it would be worth to use Speex in VBR mode
hi,
i am having a small problem with the dtmf-sounds...
if i press a dtmf digit while i am bridged with another leg, the other
side will hear the dtmf sound.
this is very annoying and even worse in a conference, when multiple
people can press dtmf digits (for (un-)muting themselves or using
Hi. Well you need to install luasql, and work only with lua 5.0 or major.
You need a ODBC connection to MySQL.
And there is an lua script example:
#!/usr/local/bin/lua
require luasql.mysql
env = assert (luasql.mysql())
con = assert
Thanks so much Anthony but I have one more question:
I was checking the source file sofia_reg.c and it seems that the code had been
written iin such a way that FreeSWITCH can authenticate SIP agents based on
RFC2069 and RFC2617. Is that conclusion correct?
Thanks in advance,
Message: 2
How about PBlx I even have the domain name ;)
On Fri, Feb 6, 2009 at 3:16 PM, Michael Collins m...@freeswitch.org wrote:
P.S. - Yes, yes I know AstLinux isn't the best name for a distro
with FreeSWITCH. Depending on my success here I have some other
ideas...
How about KickAstLinux?
Just my 2c:
If you use curl with lighttpd and custom built fastcgi C responder (it
is really simple with fcgi libs - even I can do it :-) performance could
be not that bad.
Like I wrote in the past it can handle about 2000 reguest per second
(including SQL query wiht simple postprocessing).
1) don't use inband tones for dtmf.
2) post a bounty to have FS clip the audio for x milliseconds when a tone is
detected. (you will still hear faint clicks between the start of the tone
and when the clipping activates)
On Mon, Feb 9, 2009 at 8:59 AM, Dennis oderm...@googlemail.com wrote:
hi,
when an originate is unsuccessful the failure and the cause code is returned
as the reply to the originate request.
On Mon, Feb 9, 2009 at 9:16 AM, Dennis oderm...@googlemail.com wrote:
hi,
i am using socket outbound with fs.
if i do an originate over the console, for starting an outbound
1) set late-negotation=true in the sofia profile
2) set absolute_codec_string channel variable to the exact codec you want as
the first action in your dialplan.
On Mon, Feb 9, 2009 at 2:26 AM, Helmut Kuper helmut.ku...@ewetel.de wrote:
Hi Ken,
thx for the hint. It looks quite static, so I
Everything you can do in a static dialplan you can do via curl as
well. Multiple extensions, search/conditions are allowed. Don't sell
the curl short, it's very powerful and can get the ball rolling.
On Sun, Feb 8, 2009 at 21:21, Doug Blacksone dougblackst...@gmail.com wrote:
Hi,
Right now,
Can I assume that info/functions in lua are all available in the
embedded lua in FS?
Regards
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Alexandru Nedelcu
Sent: 09 February 2009 14:07
To:
It sounds like your automake got screwed up with some new changes. I
tried and was unable to reproduce this issue, can you test a fresh
checkout and see if you still see this issue?
Mike
On Feb 9, 2009, at 3:27 AM, Helmut Kuper wrote:
Hello,
update, when I remove all ozmod_ from
Hi Mike,
of course I can ... will do it tomorrow.
regards
helmut
On 09.02.2009 16:59, Michael Jerris wrote:
It sounds like your automake got screwed up with some new changes. I
tried and was unable to reproduce this issue, can you test a fresh
checkout and see if you still see this
pauld wrote:
The libs are there with correct symlinks, see below. I tested both
voices directly via swift command, works fine.
Any other ideas?
It's Cepstral 5.1, FS 1.0.2.
Unpredictable issues have been reported using cepstral 5 with
FreeSWITCH. I'd suggest using their 4.x release. If
hi,
i am using socket outbound with fs.
if i do an originate over the console, for starting an outbound call
without having an inbound call, and send the originate directly to the
socket, the socket is first started, if the call is in answer or
ringing state.
before this, i will not receive any
On Sun, Feb 8, 2009 at 3:14 PM, pauld pa...@versafon.com wrote:
The libs are there with correct symlinks, see below. I tested both
voices directly via swift command, works fine.
Any other ideas?
It's Cepstral 5.1, FS 1.0.2.
Well, first I recommend getting on latest trunk if that's at all
On Mon, Feb 9, 2009 at 7:47 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Can I assume that info/functions in lua are all available in the
embedded lua in FS?
Regards
Generally speaking that is a safe assumption.
-MC
___
I have two Asterisk boxes, A1 A2, each running a separate telephony
app.
We have an external SIP service with DID's N200 - N299.
We want to direct the incoming SIP calls so that the DID's N200 -
N219 go to Asterisk server A1 and N220 - N299 to Asterisk
server A2.
kokoska rokoska wrote:
Just my 2c:
If you use curl with lighttpd and custom built fastcgi C responder (it
is really simple with fcgi libs - even I can do it :-) performance could
be not that bad.
hmmm, mod_xml_curl using C, interesting thought.. all of the
complexities of writing your own
Hello Anthony,
:D yes that's what I'm doing ... beneath some code changes in openzap
... So I found a real timestamp in pcap is quite usefull if you have
more than one call at a time ... I added that function today. It uses
libapr-1 functions. Unfortunately I introduced a dependency to
libs/apr
We can not add apr dependency in openzap, we should use the native
openzap calls instead. If there is anything you NEED that you don't
have, please let me know and we will try to add replacement functions.
Mike
On Feb 9, 2009, at 1:17 PM, Helmut Kuper wrote:
Hello Anthony,
:D yes
Hi Mike,
I would like to have a function which gives current time in sec, usec
since unix epoch. It's only for pcap timestamp. I found a zap_time_now()
somewhere in openzap maybe it helps ...
regards
helmut
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Freeswitch-users mailing list
That's why I chose mod_xml_curl as a demo for the xml_hook api. It's not
only a demo, it's rather functional =D
You have 2 choices other than using the stuff we already have in tree.
1) write a custom dialplan module, this module gets a single callback
function a dialplan_hunt function that has
should be fixed in latest trunk
On Sat, Feb 7, 2009 at 7:56 PM, Brian West br...@freeswitch.org wrote:
You have a \ somewhere in your path... which doesn't make sense... you're
on windows.
Can you open a jira... I think this was the cause
Hi Guys,
I have an IVR that's working fine on internal extensions, but when a
call is via my sip GW, they're not being trapped.
I have tried the following in the gw profile
param name=dtmf-type value=rfc2833/
param name=rfc2833-pt value=101/
param name=pass-rfc2833 value=false/
I
Further to this message, DTMF works with PMCU but not with PMCA which is
the native format for this sip provider.
Regards
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Nik
this was the wrong thread, i have no idea if this is fixed or is even a real
issue.
On Mon, Feb 9, 2009 at 2:03 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
should be fixed in latest trunk
On Sat, Feb 7, 2009 at 7:56 PM, Brian West br...@freeswitch.org wrote:
You have a \
On Mon, Feb 9, 2009 at 12:21 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Further to this message, DTMF works with PMCU but not with PMCA which is the
native format for this sip provider.
Any chance you could get some debug information? I'm wondering what is
actually being sent
Forgive me, I'm not sure how I get that info with FS, can you enlighten
me?
DTMF also works with GSM and others, but not Alaw
Regards,
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael
On Mon, Feb 9, 2009 at 1:34 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
Forgive me, I'm not sure how I get that info with FS, can you enlighten
me?
I was thinking of something like Wireshark. You can also check out this:
Whenever I try to record and IVR or Voicemail Greeting, it will record
and playback, but playback does something like this.
Please leave a message ... Message
It plays the end of the sound file AGAIN after playing the sound file.
I've tried leaving extra time before and after speaking, but it
Can you tell me what SVN rev you're on?
/b
On Feb 9, 2009, at 12:04 PM, Blake France wrote:
Whenever I try to record and IVR or Voicemail Greeting, it will record
and playback, but playback does something like this.
Please leave a message ... Message
It plays the end of the sound file
Yes search the mailing list people have interoped with BT in record
time. On another note you hijacked the DTMF not being recognized by
clicking reply, deleting the text and changing the subject. Please
try not to do that in the future, click new message input
John,
Here is the post
http://lists.freeswitch.org/pipermail/freeswitch-users/2007-December/001825.html
Shannon,
I want to make sure everyone knows that list etiquette is critical to
keep the SNR low. ;)
Anyway welcome to FreeSWITCH, sit back, relax and enjoy the ride... ;)
What is funny the post about this from David Knell was also a thread
hijack :P
/b
On Feb 9, 2009, at 6:04 PM, Shannon wrote:
Test A - proper user list manners -FAIL :)
On 2/9/09, Brian West br...@freeswitch.org wrote:
Yes search the mailing list people have interoped with BT in record
Oops - I did it again ;-)
--Dave
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There arent known issues cause I don't think anyone else has tried it hah
From: Kristian Kielhofner kristian.kielhof...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 10 Feb 2009 02:43:49 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users]
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