Developers using git HEAD. Did you tested the HEAD version on your box?
在 2010年7月1日 下午3:56，王洪永 写道：
>
> hi ,what is your freeswitch version? The revised test is normal I
>
> 王洪永
> 中企开源-IP通讯部
> 北京中企开源信息技术有限公司
> 电话：58022020-212
> MSN:linux...@msn.com
> E-mail:wanghongy...@ceopen.cn
> Tue, 29 Jun 201
agree with mrene that the google translation is horrible in this case.
2010/6/29 Anthony Minessale :
> turns out that change is horrible.
> i think his problem is he needs more connections because that change really
> messes things up
>
> 2010/6/28 Mathieu Rene
>>
>> I guess the source was easier
It should work with absolute path, I would suggest to try
@con.execute("record", "/opt/freeswitch/sounds/123.wav", .
and make sure the dir exists
2010/5/31 Bob Coleman
> I think you only have to specify it relative to the sounds directory, no
> need for full path.
>
> Test it by just s
I found that interestng and had tried that before, but didn't figure
out how to connect to FS though I have some other projects working
well with Erlang and FreeSWITCH. Will try more when I have time.
2010/5/27 Jan Berger :
> Hi Andrew,
>
> Good your in here :) - that looks promising from the doc.
Hi,
I'v seen lots improvements on trunk code, I just build a new server
with skype beta on FreeSWITCH Version 1.0.trunk (16848). Ubuntu 8.04
64bit.
It works well. however I seen some messages on debug level console.
Locking assertion failure. Backtrace:
#0 /usr/lib/libxcb-xlib.so.0 [0x7f1728a0
> FSComm is getting big so some organization is needed.
> Regards,
> João Mesquita
> FSComm Developer
Yeah, agree. I will let you know before I work on a new feature.
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Just an idea:
mark each gateway with a digit so people can chose gateway by prefix
with #1 #2 etc. And, better to make the digit setable so one may like
#gw1.
>>
>> Is the first provider assumed to be the default gateway?
>
> Right now, yes, but I am going to let you select tonight using the
>> Some bugs I found:
>>
>> 1) FreeSWITCH/FsComm user agent breaks xml parser, complains missing
>> ">", I guess it's a XML parser bug when it see "/" it expects an ">"
>> to be a "/>". Sorry I deleted the full log but I think it can be
>> replicated so let me know if I need to report to jira.
>>
>
the local copy
> that need to play nice with your modifications.
>
> Regards,
>
> João Mesquita
>
>
> On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote:
>>
>> Hi,
>>
>> FsComm is really cool. I played and make some improvements, see
>> rev16229 i
Thanks, see inline comments:
2010/1/11 João Mesquita :
>
> Forgot to reply line by line ...
>
> João Mesquita
>
>
> On Sun, Jan 10, 2010 at 2:12 PM, Seven Du wrote:
>>
>> Hi,
>>
>> FsComm is really cool. I played and make some improvements, see
>
Hi,
FsComm is really cool. I played and make some improvements, see
rev16229 in trunk(in my contrib dir). And note is stll has bugs when
sometimes the sound dev list doesn't sync well with FScomm on some
edge cases. Also ideally it can be automatically updated like XLite.
Because I cannot join th
Not related to this topic but http://www.bkw.org shows tons of dashes. :)
--
-
2009/10/29 Brian West
> Can you please test this patch http://www.bkw.org/celt.diff
>
> /b
>
> On Oct 29, 2009, at 9:09 AM,
We run on hardy, and even one server is gutsy.
2009/10/20 William King
> I'm running hardy as well in production. Are you using the nightlies
> packages, or building from source?
>
> Also, what is the problem or error message you get from trying to load
> mod_sofia?
>
> -William King
>
> Giovann
On Aug 1, 2009, at 1:00 PM, Anthony Minessale wrote:
> dont forget to propose what you are patching so we all approve of it
> if you plan to re-submit.
>
Thank you Anthony, I'd like to ask this question even no patch :)
Just like this one: http://jira.freeswitch.org/browse/MODENDP-236
And I'm
s. Also I have to re-diff
if I want to change change1 ...
7.
On Aug 1, 2009, at 4:52 AM, Brian West wrote:
> svn diff path.tofile.c > patch1.diff
> svn diff path.tootherfile.c > patch2.diff
>
> /b
>
> On Jul 31, 2009, at 3:42 PM, Seven Du wrote:
>
>> Hi,
>>
Hi,
Let's say I made 2 changes to one file, and svn diff can generate one
diff file, but I want two changes in different patch files so
developers may apply both or only one to trunk.
The simple way would be do only one change one time, but that one
might be take time or never to be applie
HI,
I think it should be Log-Line, too simple to make a patch, but should
I report on jira?
version 14048.
Dialplan: portaudio/1 Regex (FAIL) [group_dial_sales]
destination_number(1) =~ /^2000$/ break=on-false
Content-Type: log/data
Content-Length: 75
Log-Level: 7
Text-Channel: 1
Log-File:
Great! Then I guess the Mutilang extension will be available soon.
On Apr 25, 2009, at 7:02 AM, Michael Collins wrote:
> FYI,
>
> The FreeSWITCH wiki site has been upgraded to MediWiki version 1.14.
> Also, we've migrated the web host and back-end database to our new
> infrastructure. All us
And, actually the value to be normalized is 32bit and which generally
is a sum of two 16bit values( read and write ), so it's probably easy
to exceed 32767 if both end speaking at the same time.
On Apr 7, 2009, at 12:15 PM, Nic Bellamy wrote:
> Anthony Minessale wrote:
>> 2) the macro is desi
the data that is real rlen is how many bytes on the read channel and
wlen the write, we normalize it so if one of the other is a
difference size we only change samples that are in the range where
actual audio occured.
On Mon, Apr 6, 2009 at 10:44 AM, seven du wrote:
Well, Thank you Anthony
stion is.
On Mon, Apr 6, 2009 at 9:35 AM, seven du wrote:
Hi, can someone explain this to me?
In switch_core_media_bug.c, around line 173:
for (x = 0; x < blen; x++) {
int32_t z = 0;
if (x < rlen) {
Hi, can someone explain this to me?
In switch_core_media_bug.c, around line 173:
for (x = 0; x < blen; x++) {
int32_t z = 0;
if (x < rlen) {
z += (int32_t) *(fp + x); //what's
difference here
Hi developers,
I'm using rev 11066 and mod_skypiax, something happened and left dead
channels. I know the version is very old and I bet it has been fixed
in trunk. So I didn't put on jira. I will arrange an upgrade and test
sometimes later. Just a FYI.
The problem is when I use uuid_kill, i
around half time before deadline */
>
> /b
>
>
>
> On Feb 28, 2009, at 12:32 PM, seven du wrote:
>
>> Thanks Brian. I didn't know that before. But doesn't set the timer to
>> 0 disable it? Or can I set to a very large value say 999?
>>
>
ry 60 seconds.
>
> /b
>
>
> On Feb 28, 2009, at 1:24 AM, seven du wrote:
>
>> Hi Math,
>>
>> in 200 OK session-timeout, do you mean this?
>>
>> Session-Expires: 120;refresher=uas
>>
>> I added the session-timeout to my sip profile and
Hi Math,
in 200 OK session-timeout, do you mean this?
Session-Expires: 120;refresher=uas
I added the session-timeout to my sip profile and I can see
SESSION_TO 0
in console when I typed in " sofia status profile default " , however
the Session-Expires will be always 120 in the 200O
easy to
implement something in FS like "record-sip-session" and record all the
SIP messages related to a session. I'm not sure whether that works.
On Feb 25, 2009, at 11:32 PM, Anthony Minessale wrote:
try tcpdump or tshark
On Wed, Feb 25, 2009 at 9:06 AM, seven du wrote:
Hi,
Hi,
I want to get all(or certain) SIP messages in FS(perhaps I want to
write a module), What's the easiest or best way to get that? or where
I can find it in code?
I know we can see SIP messages with TPORT_LOG enabled, It's in
mod_sofia or sofia-lib? Why it is only possible to set that on
es. You could either record one
> RTP stream per pcap or stuff both streams into the pcap, and possibly
> even the sip signaling. I've used sipp to play back pcap files in the
> past to reproduce polycom bugs. You must give it a pcap file
> containing a single rtp stream.
>
> Se
not the whole branch,
so I think the best way to make a patch is run svn diff from the mod's
root directory, is that right?
On Jan 23, 2009, at 11:42 PM, Raymond Chandler wrote:
> seven du wrote:
>> hi Giovanni,
>>
>> I just made some improve on skypiax, s
hi Giovanni,
I just made some improve on skypiax, so you can call skypiax/ANY/
another_skypename, it will automatically chose an available channel.
not good, but it works.
put the following code directly before the following line:
for (i = 0; i < SKYPIAX_MAX_INTERFACES; i++) {
And
Hi FreeSWITCHers,
1) FreeSWITCH support G729 codec in passthrough mode, it is normally
engough if both call-legs suport G729. But there is no way to do
recording if you can't decode it. I wrote a small module called
mod_recpld. The idea is to record the raw payload in rtp packets to
files
he one put the bounty :)
> Do you mind to add some $$$ to the fund? That could help the thing
> make
> happen.
>
> Regards,
>Tamas
>
> ps: Right waiting for response from consulting@ for the request.
>
> seven du írta:
>> Hi developers,
>>
>> No
Hi,
For production use, we want the service run consistently and long as
possible. As we have command tools to reload xml configurations and
restart gateways, I have a problem to make my configuration take
effect before restarting the whole FreeSWITCH server.
Here are two questions:
1) Wh
Hi developers,
Now and then we need to know some statistics message and sure we need
some way to measure it. My ideas just like this:
http://wiki.freeswitch.org/wiki/Bounty#RFC_3611_-_RTP_Control_Protocol_Extended_Reports_.28RTCP_XR.29_support
Is there some plans to build this kind of function
Giovanni,
Thank you for taking mu suggestions take into account.
Another idea: Make the skype in context and dialplan
configurable(other than 5000?)
maybe useful. But not sure, due to the limits of skype in, perhaps the
only use of
skype in is transfer into an ivr.
And, Just found a easy wa
Hi Giovanni,
In addition to the command line tools, I suggest taking following
things in to account:
1) originate skypiax/wrong_skype_name won't cause core dump
2) currently we can only call skypiax/skypiax1/other_skype_name, can
we implement
something like openzap, so skype group can
is there some commands like sofia status?
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Hi Giovanni,
Sorry for the previous mail. I noticed it should be originate skypiax/
my_skypy_account_name/another_skypy_name, it works, great!
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hi Giovanni,
I successfully loaded skypiax on another ubuntu gutsy machine, the
privious problem maybe because it is in a Xen VM.
But, when I call out with originate skype/echo123 &echo, it shows
chan_not_implemented, however, I think it should be skypiax(2009-01-16
16:49:06 [NOTICE] switch
Hi,
I tested skypiax on Linux db1.veecue.com 2.6.24-19-xen (Ubuntu hardy
in Xen), after I load the mod_skypiax, the console will stuck.
I'm not sure, anything is following the wiki. However, On the Desktop
computer configuring skype, there are multiple snd drivers,
default
HW dummy
other
I
Brian,
I tested using originate
{ignore_early_media=true,absolute_string=g729,ringback=/sounds/
somefile}sofia/default/1000 &bridge({ignore_early_media=true,
ringback=/sounds/somefile,absolute_codec_string=g729}sofia/gateways/
/0)
To listen the ring back, you need set ignore_early_m
>> sorry, mod_native_file only works on non-PASSTHROUGH codecs.
>> previously I modified mod_g729, so FreeSWITCH takes G729 codec as
>> transcoding codecs, mod_native_file will open sound.G729 for
>> ringback( and playback). However I need to make sure both legs
>> using g729 codec.
_
> We are using the mod_native_file. It's case sensitive from my
> experience. I tested PCMU, PCMA and G729, work fine.
> And it also works on ringback tone, just set the channel variable to
> ringback=/sounds/somefile will work.
> Just one thing confused me. if I set a ringback tone, the G729 c
Hi list-
Some custom variable not get send out by event_socket.
Like
2008-12-23 18:12:51 [DEBUG] switch_ivr.c:451 switch_ivr_parse_event()
sofia/outbound/015810303546 Command Execute set(called_trainers=55)
we set some variables(including some custom variables)
on a channel A-leg before bri
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