I wouldn't call it donating per se... Its just giving it a place to
live with easy access for end users without having to do anything
extra go get it! ;)
/b
On Oct 12, 2009, at 11:27 PM, Tihomir Culjaga wrote:
> this will be perfect ... but it is up to Yuriy if he is willing to
> donate h
OK, then when i call to group number(911) how it will call to all the
registered members in group?
On Tue, Oct 13, 2009 at 11:55 AM, Brian West wrote:
> NO this is in the XML... not in the db table.
>
> /b
>
> On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:
>
> > in the same way there i
NO this is in the XML... not in the db table.
/b
On Oct 12, 2009, at 11:16 PM, srinivasula reddy wrote:
> in the same way there is any table for groups, how many groups are
> there? and information about groups(directory/default.xml this file
> having the group configuration).
__
HI Brain,
thank u very much for valuable reply,
you are right, sip_Registration table contains the registered endpoint
details, in the same way there is any table for groups, how many groups are
there? and information about groups(directory/default.xml this file having
the group configuration).
Thats called mod_fifo.
/b
On Oct 12, 2009, at 12:55 PM, William King wrote:
> I don't know if this was mentioned yet. It would be useful to have a
> way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark wo
Does anyone see a problem with hosting mod_h323 in our SVN? I would
like to centralize everything we can to reuse our issue tracking
resources and not fragment the community if possible.
/b
On Oct 12, 2009, at 2:43 PM, Tihomir Culjaga wrote:
> hi,
>
> finally i compiled it right ... had a s
Eli,
Well FreeSWITCH already keeps a list... but its the phone's job to
register to FreeSWITCH not the other way around. Their are various
ways to accomplish your goals but not sure how well each will work.
Check out the wiki for sip-force-contact.
/b
On Oct 12, 2009, at 4:37 AM,
Hi,
can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.
thanks
--
Srinivasula Reddy K
You need "async full"
/b
On Oct 12, 2009, at 10:59 PM, velusamy velu wrote:
> What is the problem? Please help me
___
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The sip_registration table contains the contacts for each registered
endpoint. Its not for the directory from a database per se... If you
wish to serve up your users and groups from a database check out the
XML Curl wiki page.
/b
On Oct 12, 2009, at 11:01 PM, srinivasula reddy wrote:
> Hi
Please DO NOT cross post.
/b
On Oct 12, 2009, at 11:02 PM, srinivasula reddy wrote:
Hi,
can any know where group information is exactly stored in sqllite
database, i have seen sip_registration here i can find the
registered users,
in the same way how i can i find the group information
Hi,
can any know where group information is exactly stored in sqllite database,
i have seen sip_registration here i can find the registered users,
in the same way how i can i find the group information, and which user
belongs to which user?
any help would be great.
thanks
--
Srinivasula Reddy K
Dear All,
I have set my socket mode as full. I have used ESL.pm to develop an
IVR. I encounter the situation that I need play some music file while
executing some external application. So, I used executeAsync method to play
the music file. But the executeAsync application didn't work.
Update to latest trunk and try again... then if it persists then
collect all the sip traces and debug logs as per the wiki on reporting
bugs.
Thanks,
Brian
On Oct 12, 2009, at 5:01 PM, Klaus Hochlehnert wrote:
P.S.: I’m using FS trunk from last week and newest Snom Firmware
_
"In many cases the issue with the docs isn't that they aren't complete but
rather that they are hard to find."
Agreed! The biggest problem with the wiki is that it is hard to find things.
How do books solve this problem they use an index. It is quite surprising that
the wiki software doesn't c
On Mon, Oct 12, 2009 at 5:26 PM, Gabriel Gunderson wrote:
> On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins
> wrote:
> > On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson
> wrote:
> >> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins
> >> wrote:
> >> > It seemed appropriate to do so, theref
exactly that is what i am trying to do now, if any help that would be great.
On Mon, Oct 12, 2009 at 5:07 PM, Eli Hayun wrote:
> Is it possible to keep a list of registered phone, and when FS will start
> it will register them all automatically?
>
>
>
> On Mon, 2009-10-12 at 13:19 +0200, Seven
It was a problem and has been fixed in the last trunk. Just update to the
latest code should be ok.
btw, the developers using jira to track bugs, so feel free to report one (as
you see http://jira.freeswitch.org/browse/FSCORE-463) if you think it's a
bug next time.
2009/10/12 Nagalenoj
>
> What
I won't try until I need that, but I believe it works. Thanks Brian.
2009/10/13 Brian West
> Fixed... svn up.
> /b
>
> On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
>
> http://jira.freeswitch.org/browse/MODCODEC-15
>
> Is it ok I assigned to you ?
>
> Thanks.
>
>
>
> _
Sure, I'm happy to put my little two cents to help the project :).
Diego
On Mon, Oct 12, 2009 at 10:38 PM, Michael Collins wrote:
>
> On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola wrote:
>
>> Hello,
>>
>> I have been doing some work recently on the FreeSWITCH wiki, to improve
>> things.
>>
>> You
This is what I have in my dialplan and the fax is detected
beautifully. Note that in my case, extension 1000 will ring for a
second or two before the fax is detected. So in your example, the fax
does not have time to be detected, the dialplan exists and the call is
hungup.
When the fax is detect
If anyone ran into the same problem, here is the fix that I got from [stkn] on
IRC:
make CFLAGS="-O2 -fPIC"
I hope it might help someone in the future.
Regards.
- Original Message
From: DJB
To: FREESWITCH-USERS MAILING LIST
Sent: Mon, October 12, 2009 4:45:27 PM
Subject: [Freeswi
On Mon, Oct 12, 2009 at 4:41 PM, Michael Collins wrote:
> On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson wrote:
>> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins
>> wrote:
>> > It seemed appropriate to do so, therefore I added a small snippet on the
>> > documentation guidelines:
>> >
>>
Hi,
I'm facing an issue and I don't know why this happens or what I can do to solve
this.
Here's the scenario:
- FS call timeout set to 30 sec
- Setting continue_on_fail=true and hangup_after_bridge=false
- 1 Snom phone and an external SIP/ISDN gateway (Lancom) connected to FS
established a cal
DJB wrote:
> Hello,
>
> I am trying to use luasql with freeswitch but having a hard time compile it
> with x86_64 Server running CentOS 5.3
>
> Linux 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64
> x86_64 GNU/Linux
>
> [r...@fsx1 luasql-2.1.1]# make
> export MACOSX_DEPLOYME
Hello,
I am trying to use luasql with freeswitch but having a hard time compile it
with x86_64 Server running CentOS 5.3
Linux 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64
GNU/Linux
[r...@fsx1 luasql-2.1.1]# make
export MACOSX_DEPLOYMENT_TARGET="10.3"; gcc -O2 -Wall
Opps. didn't realize it was closed. Here you go. Moving it to my seedbox
for permanent seeding.
-William King
Frank Carmickle wrote:
> Hello
> I would like to watch these files. Can someone seed please. Thank you.
>
> --FC
>
> ___
> FreeSWITCH-users
Hello
I would like to watch these files. Can someone seed please. Thank you.
--FC
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On Mon, Oct 12, 2009 at 3:33 PM, Gabriel Gunderson wrote:
> On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins
> wrote:
> > It seemed appropriate to do so, therefore I added a small snippet on the
> > documentation guidelines:
> >
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It
On Mon, Oct 12, 2009 at 3:11 PM, Diego Viola wrote:
> Hello,
>
> I have been doing some work recently on the FreeSWITCH wiki, to improve
> things.
>
> You can see some of my work here:
>
>
> http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
>
> I am try
On Mon, Oct 12, 2009 at 11:35 AM, Michael Collins wrote:
> It seemed appropriate to do so, therefore I added a small snippet on the
> documentation guidelines:
> http://wiki.freeswitch.org/wiki/Documentation_guidelines#Keep_It_Professional
Good guidelines...
Just in time to help the author of
if you are using ringback variable you now must also use
ignore_early_media=true either exported from A leg or in {} on the b leg
dial string to get the original behavior.
On Tue, Oct 6, 2009 at 10:41 AM, Lars Zeb wrote:
> http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14
Is this page still necessary:
http://wiki.freeswitch.org/wiki/Old_mod_python
I'd like to have the mod_python page only:
http://wiki.freeswitch.org/wiki/Mod_python
If there is something in the old one please let me know so we can move to
the new one and then get rid of the older one.
Thanks,
D
Hello,
I have been doing some work recently on the FreeSWITCH wiki, to improve
things.
You can see some of my work here:
http://wiki.freeswitch.org/index.php?title=Special:Contributions&limit=500&target=Diego.viola
I am trying to polish the wiki and give it a more professional and clean
look, t
On Mon, Oct 12, 2009 at 12:00 PM, Matthew Fong wrote:
> http://pastebin.freeswitch.org/10656
>
>
> Matthew,
Try continue_on_fail=true instead of hangup_after_bridge=false.
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
I think it will do what you want.
-MC
__
hi,
finally i compiled it right ... had a stupid issue with ekiga and wrong
ptlib in place...
anyhow, i loaded the module and will continue the tests tomorrow ...first
thing i arrive in my office :P
freeswi...@subzero>
freeswi...@subzero>
API CALL [console(loglevel 7)] output:
+OK console log
Thanks Michael. I'll go through the resources you mentioned.
Thanks,
Vinuth.
On Tue, Oct 13, 2009 at 2:15 AM, Michael Collins wrote:
>
>
> On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur > wrote:
>
>> Hi,
>> Does Freeswitch detect all of these hangup cases mentioned here [
>> http://wiki.free
On Mon, Oct 12, 2009 at 12:58 PM, Vinuth Madinur
wrote:
> Hi,
> Does Freeswitch detect all of these hangup cases mentioned here [
> http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
> Trunk provider?
>
> If not, should I put in tone_detect application in the dialplan for
>
2009/10/12 João Mesquita
> I would say that the parking meter is a good idea and it is the default
> behavior of parking on legacy PBXs. Since we always do _more_, what do you
> think about having the option to transfer to any extension instead of just
> the one that transfered the call?
>
>
Gent
On Mon, Oct 12, 2009 at 12:55 PM, William King wrote:
> I don't know if this was mentioned yet. It would be useful to have a way
> to have the parking lot automatically find the next available spot and
> tts it to the person parking the call.
>
> Then the auto unpark would pop off the lowest numb
Hi,
Does Freeswitch detect all of these hangup cases mentioned here [
http://wiki.freeswitch.org/wiki/Hangup_causes] when using it through a SIP
Trunk provider?
If not, should I put in tone_detect application in the dialplan for
detecting the SITs?
Won't freeswitch have to depend on the SIP statu
I don't know if this was mentioned yet. It would be useful to have a way
to have the parking lot automatically find the next available spot and
tts it to the person parking the call.
Then the auto unpark would pop off the lowest numbered lot, or return
fail if there is nobody in the parking lots e
I would say that the parking meter is a good idea and it is the default
behavior of parking on legacy PBXs. Since we always do _more_, what do you
think about having the option to transfer to any extension instead of just
the one that transfered the call?
Regards,
jm
On Mon, Oct 12, 2009 at 4:20
Michael Collins said:
> On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi
wrote:
>
> >
> >
> > On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins
wrote:
> >
> >> FYI,
> >>
> >> The FreeSWITCH devs have added valet parking! Check it out:
> >> http://www.freeswitch.org/node/207
> >>
> >> Let us know wh
On Sat, Oct 10, 2009 at 8:18 PM, mayamatakeshi wrote:
>
>
> On Fri, Oct 9, 2009 at 10:42 AM, Michael Collins wrote:
>
>> FYI,
>>
>> The FreeSWITCH devs have added valet parking! Check it out:
>> http://www.freeswitch.org/node/207
>>
>> Let us know what you think.
>>
>
> Very nice.
>
> But I think
http://pastebin.freeswitch.org/10656
On Tue, Oct 13, 2009 at 1:34 AM, Michael Collins wrote:
> Turn on debug, make another test call, and pastebin the output.
> -MC
>
>
> On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins wrote:
>
>>
>>
>> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong wrote:
>>
Lak,
Okay I will need a little bit of time to dig into the IE's and what they
contain. In the meantime can you tell me who the carrier is? I'd like to
find out if they have some specific requirements. The fact that it doesn't
work with libpri surprises me because that would mean that Asterisk syst
On Tue, Oct 6, 2009 at 8:41 AM, Lars Zeb wrote:
> http://pastebin.freeswitch.org/10612
>
>
>
> I having been running v14996 OK for a while. I have upgraded a couple of
> times after, but every time, an inbound call is hung up on. The only thing
> that has changed is the upgrade. This morning I u
Turn on debug, make another test call, and pastebin the output.
-MC
On Mon, Oct 12, 2009 at 11:11 AM, Michael Collins wrote:
>
>
> On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong wrote:
>
>> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
>> bridge...
>> when an ALLOTTED_TIMEOUT
On Sun, Oct 11, 2009 at 10:48 AM, EdPimentl wrote:
> http://www.assistivetech.net/search/productDisplay.php?product_id=18854
> -E
>
> Let's keep an eye on this but I doubt that an assistive technology
application is going to create any real confusion.
-MC
>
>
> __
still no luck...
freeswi...@matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 18:25:44.345480 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/sip_1 [3fc6efb2-e4fa-454a-abb7-ebe39da748f5]
2009-10-12 18:25:44.489
On Mon, Oct 12, 2009 at 10:42 AM, Matthew Fong wrote:
> when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed
> bridge...
> when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
> recognized (I think). Is there anyway to get an alloted_timeout to continue
> after
On Mon, Oct 12, 2009 at 4:01 AM, homqua wrote:
>
> Hi,
> I have implemented the solution for tone detection in wiki, and also answer
> the channel before detecting the tone:
>
>
>
>
>
>
>
>
>
>
> data="/usr/local/freeswitch/storage/fax/${caller_id_number}-$
when I add a leg_timeout, I get an ALLOTTED_TIMEOUT from my failed bridge...
when an ALLOTTED_TIMEOUT is received, the hangup_after_bridge=false is not
recognized (I think). Is there anyway to get an alloted_timeout to continue
after bridge (failure)?
revised dialplan & cmd output
On Sun, Oct 11, 2009 at 4:49 PM, Brian West wrote:
> I have tried to police the wiki when things like this appear.. its one
> thing to crack a joke in fun from time to time... but to put stuff
> like that on the wiki isn't acceptable.
>
> /b
>
>
It seemed appropriate to do so, therefore I added a
On 2009-10-12 18:03 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
i already write this, ptlib 2.6.5, tou can find link to it on oplalvoip.org,
h32plus latest CVS version,
you can find it on www.h323plus.org.
TC>hi,
TC>
TC>can't make it...
TC>
TC>subZero:~/freeswitch-trunk$ make mod
This is because FS is a B2BUA ... not a proxy. You should consider
OpenSER/SIPS/Kaemillio for this type of application.
SDR
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On Sun, Oct 11, 2009 at 2:40 PM, Diego Viola wrote:
> I'd like to add this for the next weekly conference.
>
> I have added a few events to the event list, as you can see here:
>
> http://wiki.freeswitch.org/wiki/Event_list
>
> But I need more help from the community to complete that and add cont
Hi, all
I used sipp to test to forward an un-implemented SIP method, but I got 501
Not Implemented.
Can freeswitch be a SIP proxy server?
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 10.10.59.161:5061
From:
To: ;tag=gBpay50SD2SUD
Call-ID: 10-27...@10.10.59.161
CSeq: 3 TESTA
User-Agent: FreeSWITC
Fixed... svn up.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
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Nope, I wanted to make sure that this is indeed a bug. I opened an issue in
JIRA before regarding some other matter and it turned out to be my mistake,
so I decided to try mailing list first this time.
MA
Brian West wrote:
>
> Did you open a jira and attach all the info?
>
> /b
>
> On Oct 12
hi,
can't make it...
subZero:~/freeswitch-trunk$ make mod_h323
making all mod_h323
Compiling mod_h323.cpp...
quiet_libtool: compile: g++ -g -ggdb -I/usr/local/include/ptlib
-I/usr/local/include/openh323 -I. -DPTRACING=1 -D_REENTRANT -fno-exceptions
-I/home/tculjaga/freeswitch-trunk/src/include
doh! thanks!
On Mon, Oct 12, 2009 at 10:33 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> because the regex is on 1997 not 1999
>
>
>
> On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong wrote:
>
>>
>>
>> > data="hh/hh-unable_to_connect_contact.wav"/>
>>
>>
You are asking how you can do asynchronous actions park then play a sound
before park exits without enabling async mode. Think about that.
you could use new valet_parking that parks with music i guess
or tell the channel execute playback instead of park since playback an park
are almost the same t
because the regex is on 1997 not 1999
On Mon, Oct 12, 2009 at 10:25 AM, Matthew Fong wrote:
>
>
> data="hh/hh-unable_to_connect_contact.wav"/>
>
>
>
>
> my extn 1999... since it looks from the output like it's transferring, just
> don't know why it's di
2009-10-12 15:22:47.015952 [NOTICE] switch_core_state_machine.c:179 Hangup
sofia/internal/sip_1 [CS_EXECUTE] [NORMAL_CLEARING]
might be the line..or the entire output is below
freeswi...@matthew-laptop> originate sofia/internal/sip_1%192.168.1.10 1920
2009-10-12 15:21:44.029517 [NOTICE] switc
my extn 1999... since it looks from the output like it's transferring, just
don't know why it's disconnecting the call instead of playing the .wav and
parking.
On Mon, Oct 12, 2009 at 10:23 PM, Matthew Fong wrote:
> 2009-10-12 15:22:47.015952 [NOTICE] s
which line is hanging up your A (inbound) leg?
look for a blue line that says "Hangup xyz" that matches it so i can
see.
I think what is happening is you are getting early media so the bridge is
actually working then when nobody answers it dies but technically the bridge
worked.
On Mon, Oct
On 2009-10-12 09:43 -0500, Brian West wrote freeswitch-us...@lists.freeswit...:
BW>We can host this in our SVN if you wish?
If in fs svn i think yes. But i think may be little time later?
i don't known is it builds on trunk because i develop it on 1.0.4.
BW>/b
BW>
BW>On Oct 12, 2009, at 8:31 AM
I think think this might be a bug, but wanted to post here instead of Jira
in-case I'm overlooking a configuration variable
Dialplan
API Command
originate sofia/internal/sip_1%192.168.1.10 1920
When the bridge to 14159927717 fails (NO_ANSWER) bo
We can host this in our SVN if you wish?
/b
On Oct 12, 2009, at 8:31 AM, Georgiewskiy Yuriy wrote:
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but
seems it work, but should be buggy,
to build need libpt 2.6.5 and h323plus cvs version, i test it now on
fs 1.0.4.
___
Hi,
I have implemented the solution for tone detection in wiki, and also answer
the channel before detecting the tone:
But FS cannot recognize the tone, and therefore cannot move to fax
extension. Below are the error in F
2009/10/12 Georgiewskiy Yuriy
> On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote
> freeswitch-us...@lists.fre...:
>
> TC>Hi Yuriy,
> TC>
> TC>can you share what you have so far, I'm sure we can help with RTP
> part...
>
> ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems
>
it works in either case with or without media
the syntax for setting the frequency was answered above.
On Mon, Oct 12, 2009 at 12:38 AM, Artem Shiyanov wrote:
> Michael, Diego,
> thanks for the rapid answers!
>
> As far as I understand, "enable_heartbeat" app is launching
> SESSION_HEARTBEAT ev
On 2009-10-08 20:32 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC>Hi Yuriy,
TC>
TC>can you share what you have so far, I'm sure we can help with RTP part...
ftp://srv.icf.org.ru/pub/soft/f/freeswitch/mod_h323/ alfa code, but seems it
work, but should be buggy,
to build need lib
Did you open a jira and attach all the info?
/b
On Oct 12, 2009, at 3:47 AM, Maciej Aniserowicz wrote:
Yes, I confirmed that with Wireshark (filter "rtp and ip.src ==
). RTP packets are sent every 20ms.
MAniserowicz
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Perfect! Thanks.
/b
On Oct 12, 2009, at 1:15 AM, Seven Du wrote:
http://jira.freeswitch.org/browse/MODCODEC-15
Is it ok I assigned to you ?
Thanks.
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Ya ok. Here is the required stuff
Configuration: (PRI span)
Log while starting freeswitch:
http://pastebin.freeswitch.org/10646
Log when making a call:
http://pastebin.freeswitch.org/10647
Configuration: (LIBPRI SPAN)
Log while starting freeswitch:
http://pastebin.f
Is it possible to keep a list of registered phone, and when FS will
start it will register them all automatically?
On Mon, 2009-10-12 at 13:19 +0200, Seven Du wrote:
> try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open
> them. not sure how to do that on windows, but on linux:
try open YOUR_FreeSWITCH_INSTALL_DIR/db/*.db, you need sqlite3 to open them.
not sure how to do that on windows, but on linux:
# sqlite3 xx.db
sqlite> select * from sip_registration;
2009/10/12 srinivasula reddy
> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in
Yes, I confirmed that with Wireshark (filter "rtp and ip.src == ).
RTP packets are sent every 20ms.
MAniserowicz
- Original Message -
From: Michael Jerris (via Nabble)
To: Maciej Aniserowicz
Sent: Monday, October 12, 2009 12:21 AM
Subject: Re: [Freeswitch-users] Bad sound q
Any one please help me to solve the mentioned problem
On Sat, Oct 10, 2009 at 3:07 PM, velusamy velu wrote:
> Dear All,
> I am using ESL.pm module to control the dial plan application. I want
> to play some music while executing the some external scripts. I executed
> park after th
FreeSWITCH 1.0.2?
That's more than a year old I think, you should really update to 1.0.4 or
latest SVN trunk.
Diego
On Mon, Oct 12, 2009 at 4:14 AM, srinivasula reddy <
srinivas.ksvre...@gmail.com> wrote:
> Hi Mike,
>
> Thanks for your valuable reply,
> when i install freeswitch1.0.2 in my mach
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