also was this 260 people in a single conference or multiple smaller
conferences?
/b
On Sep 17, 2009, at 1:58 PM, RobertT wrote:
Okay, I've performed some additional tests and this is what I've
found:
codecmax calls
speex (8kHz) 50
iLBC(8kHz) 50
PCMU(8kHz) 260(approx*)
GS
Are you using SIPP?
/b
On Sep 17, 2009, at 8:33 AM, email lists wrote:
> Now, after a few minutes of running calls at 20 calls per second with
> a 3 min. duration, I start seeing an influx of "480 Temporarily
> Unavailable" SIP error messages.
___
Fr
Just create an OpenBTS page on our wiki.
/b
On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote:
> Sorry, just realized that the sourceforge page is protected by
> password. I
> am happy to put the info in FreeSWITCH wiki, where does it make
> sense to
> add this project info?
___
Personally I would throw the phone in the trash. :P
In the default dialplan look at 5900 for park and 5901 for unpark.
/b
On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote:
> I am trying to create a simple call waiting dialplan as my phone does
> not have Recall button.
___
I think if you remove
it'll do what you want.
/b
On Sep 16, 2009, at 7:05 PM, Nandy Dagondon wrote:
> this makes sense. a workaround would be to provide an optional
> variable to delete recording file if it's less than N seconds.
> otherwise, it defaults to a preset duration.
>
> /nandy
Since we have no 's' extension or anything similar maybe if you tell
me what you're trying to do I can tell you how to do it.a
/b
On Sep 17, 2009, at 6:25 AM, Ahmed Munir wrote:
>
> Hi,
>
> How can I process s extension in FS? Is there other way around of
> doing it? Kindly advice me.
>
> --
NO you must not. The issue has been fixed in svn already please start
with a fresh tree.
/b
PS: end users should NEVER have to reswig.
On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote:
On Thu, Sep 17, Luis M. Zuccolo wrote:
Hi:
Since svn version 13523 to current I get this error:
make
You can do that in your dialplan. Not sure it'll fix all your options
unless you do both the gateway and the sip_invite_domain.
/b
On Sep 17, 2009, at 4:19 AM, Tzury Bar Yochay wrote:
> in which file under which section should I specify this
>
> data="{sip_invite_domain=${sip_from_host}}sof
Its a bug in 2.6.26 thru 2.6.28 kernels that impact the performance of
SQLite. He was specifically running SUSE.
/b
On Sep 16, 2009, at 11:07 PM, Jason White wrote:
> Please take this up with your Linux distribution as a bug report
> related to
> the kernel, and persist with it until it's s
These are not sip_dialogs as you think they are. These are used for
dialing-info dialogs for sip subscriptions.
/b
On Sep 17, 2009, at 2:51 AM, kokoska rokoska wrote:
>
> Hello,
>
> I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and
> based
> on logs FS connected to the dsns
Yah speex is a cpu hog (while you can tune it to use less)! Granted
it uses less bandwidth but on the server side it doesn't scale very
well.
/b
On Sep 16, 2009, at 5:11 PM, Jay Binks wrote:
> A few thing stuck out to me ...
> Mainly 50 calls and transcoding speex.
>
> Try it again with g711
I would be very interested in this also.
/b
On Sep 16, 2009, at 4:52 PM, João Mesquita wrote:
> I would be really interested to replay your test on Linux. Would you
> be willing to provide me all the details and relevant files so I can
> reproduce the test with a Linux box here?
>
> If yes,
Or you setup a gateway and set the from-domain
/b
On Sep 16, 2009, at 10:00 AM, João Mesquita wrote:
Is this what you are looking for?
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain
jmesquita
On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay > wrote:
Hi,
Currently, th
Might I ask what you are working on?Its interesting to hear what
people are doing with FreeSWITCH.
/b
On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote:
> perfect,
>
> thanks.
>
> T.
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Yes you're missing a switch_xml_free(xml); some place.
/b
On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote:
> hi,
>
> I've build a custom module for FS and everytihng work well except
> reloadxml command :P... m'I missing something in my module? ... i
> used mod_skeleton as a template when
Because you have 100rel disabled so PRACK will NOT show up in the
allow list.
/b
On Sep 15, 2009, at 3:38 PM, DJB wrote:
> I wonder whether anyone can tell me why the latest trunk has no
> PRACK comparing to the 1.0.4.
>
> Here is the sip message:
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.tru
The friday meetings are where we all collaborate on these group
efforts and discuss project direction, goals and areas where people
can help out more.
Did we ever find someone to officially take over the Debian packages?
/b
On Sep 15, 2009, at 9:56 AM, Michael Gende wrote:
Answered my o
Do you have Late Negotiation on? Also is this the only FreeSWITCH log
output you have in this transfer?
/b
On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote:
> Hello Jason,
>
> Sorry for the delay in answering - I saw your reply only now as it
> got burried with some other stuff...
>
>
making sure you know that now.
Thanks,
Brian West
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Run system(); from the dialplan...
/b
On Sep 14, 2009, at 8:58 AM, Tihomir Culjaga wrote:
Hi,
i just have a maybe dummy question but it is still a question :P
in my case ${service_instance} is something dynamic and has to be
created on the fly.
Is there any way FS can create a di
HAHA I couldn't have said this better!
/b
On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote:
> The first hint was when the firmware rev began with the letters POS
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On Sep 13, 2009, at 9:27 AM, Morten Henckel wrote:
However when i analyse the rdecording the Digits are being cut off
down to 10 msec "bursts" - I trust its FS that cust the DTMF in
order to avoid further propogation inband to second leg of the call.
Nope if its rfc2833 its not us cutting
This means the far end is sending you a challenge and we do not know
how to answer it... please review how to setup a gateway on the Wiki
so you can authenticate.
/b
On Sep 13, 2009, at 6:47 PM, paul.d...@gmail.com wrote:
> Only abnormal things I can see in FS logs are:
> 2009-09-13 19:17:31
You have a merge conflict please svn revert sofia.c
/b
On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote:
> Hi Folks,
>
> I've got a compilation error with the latest codes (r14842)
>
> Making all in packages
> Creating mod_sofia_la-mod_sofia.lo
> Compiling mod_sofia.c ...
> Creating mod_sofia_la-
I haven't seen this issue in 8.12 either... Maybe thats why 8.11
isn't on the website last I checked?
/b
On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote:
RESOLVED!!!
Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00
I forgot that (a long long time ago) I had dropped
that means the media isn't actually making it to the FS server I
suspect.
/b
On Sep 13, 2009, at 7:22 AM, Woody Dickson wrote:
> Hi,
>
> While trying to record some sounds with the voicemail app, I keep
> getting message saying my record is below the minimal length even I
> was actually st
Also I'm going to suspect you have removed the domain aliases from the
profile. If you have then you can't just do sofia_contact
u...@domain... You must do sofia_contact profile/u...@domain since
your hint for the domain is no longer on the profile.
/b
On Sep 12, 2009, at 10:25 AM, Anthony
Sounds like you have Force-RPORT on which you can't do with a 7960.
/b
On Sep 12, 2009, at 6:24 PM, Karl Vesterling wrote:
>
> Seems normal, right??? Keep scrolling, or search for "JUST WRONG!" and
> you'll see it below...
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I'll dig it up this weekend and get you a copy of it.. its a perl
script that writes out some js that I run via jsrun
/b
On Sep 11, 2009, at 3:42 PM, Muhammad Shahzad wrote:
> great, can you share it with me?
>
> Thank you.
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Kewl I have a fuzz test I do also thats automated that throws all
kinds of crazy stuff at all the api's.
/b
On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote:
> sure, i have a full QA department who will take case of all possible
> cases. Then it can be tested by our community.
>
> Thank y
make samples
/b
On Sep 11, 2009, at 1:03 PM, Mark Sobkow wrote:
> When I try to d a load mod_sofia, I get an error message indicating
> that
> Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the
> autoload directory, which I _thought_ was the main sofia configuration
> file.
Also for tests make sure you fuzz test it also .. giving it invalid
data shouldn't crash ... so try that when you're done too.
/b
On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote:
> actually, mod_dingaling is not reading configuration from xml_curl
> unless we reload mod_dingaling, which
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11
Here is the agenda please review and add to it anything you think we
should cover.
Thanks,
Brian
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Thats normal too.
/b
On Sep 11, 2009, at 2:26 AM, Anatoliy Kounitskiy wrote:
> It's normal to have to two records for a call - Start and Stop
> message.
>
> From what i see - you have one start and stop for each leg of the
> call.
>
> Regards,
> AK
_
No you should never be doing your billing inline like this. You
should be doing this externally of your application not inside your
dialplan.
/b
On Sep 10, 2009, at 11:40 PM, Ahmed Munir wrote:
> Thanks for reply, well actually I'm doing billing after call hangup.
> If h extension is inte
You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if
the IP changes sofia will bounce the profile and update the IP.
/b
On Sep 11, 2009, at 7:55 AM, jun yang wrote:
i also found that:
2009/7/17 Raul Fragoso :
> You can not do that with a single profile. Each profile is bound
You need to telnet to the socket or use fs_cli... example...
telnet 0 8021
auth ClueCon
events all plain (or what ever commands you wish to run)
/b
On Sep 11, 2009, at 3:48 AM, jun yang wrote:
> how can i subscribe to custom event in cli.
> cli: load mod_event_socket
> say Module mod_event_so
Next Bug? Huh? :P
/b
On Sep 11, 2009, at 2:32 AM, Benedikt Fraunhofer wrote:
>
> Thx again and looking forward to the next bug :)
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You'll get this for transfers, and each leg... Can you elaborate on
the call scenario more?
/b
On Sep 10, 2009, at 3:47 PM, email lists wrote:
Hello,
Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
RADIUS messages being generated for individual calls (sample
message
There will NEVER be an easy way to do billing inline and we never
support such crazy because its WRONG. That said... You're not the
first to approach FreeSWITCH with an Asterisk mentality.
Billing inline = NO
Billing direct to DB = NO
Post Processing = YES, do this.
Nibble/Heartbeat billing v
What puzzles me is why you can't just update to SVN trunk to get
uuid_exists in the first place? Its going to be 1.0.5 the more people
we have testing the faster that happens.
/b
On Sep 10, 2009, at 4:24 PM, Phillip Jones wrote:
> Strangely - the uuid_getvar uuid
> workaround does not wo
This does now...
/b
On Sep 10, 2009, at 4:52 AM, Jason White wrote:
>
> Yes, but by default the internal profile doesn't handle nat, which
> is why (if
> I recall correctly) it has been recommended that the external
> profile be used
> to register clients that are not on the local network wh
Those configs will still work.
/b
On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
Hi there,
the internal.xml and external.xml examples are for situations where
FS is running inside a company's private network, behind a NAT
router. So internal.xml connects the clients to FS without crossi
So if you have an extension name that is "testing" and the
destination number is "testing" then if testing is at the bottom of
the dialplan auto_hunt will make it warp right to it.
/b
On Sep 9, 2009, at 8:29 AM, Max Ivanov wrote:
> Hi all!
> Is there any difference between auto_hunt=True a
This looks and sounds like a case where pjsip isn't listening to our
SDP. If we 200 OK with speex on 102 and the far end starts sending it
on 98 then I suspect the client is broken if I'm not mistaken.
/b
On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote:
> Hi,
>
> Owe to the network bandw
Its your gateway provider not squelching the DTMF I suspect.
/b
On Sep 8, 2009, at 12:26 PM, Bradley Brashier wrote:
> The issue is that when a command is pressed on one phone in the
> conference, all users hear the tones of the first key pressed. My
> expectation is that no other users should h
00 AM, Humberto Quintana wrote:
Hi Brian,
Yes , the Call-Id is the same for the 2nd and 3rd transaction but
the branch parameter in the Via header is different. Please check
the capture below.
Thanks,
Humberto
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Not automatically. But you could use the event socket to get the
message and forward it via ESL.
/b
On Sep 5, 2009, at 1:26 PM, Juan Backson wrote:
>
> If so, how can it be done?
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You're missing some required args there.
chat []
/b
On Sep 5, 2009, at 11:19 AM, Juan Backson wrote:
freeswi...@localhost.localdomain> chat sip|180...@192.168.1.102|
testing
API CALL [chat(sip|180...@192.168.1.102|testing)] output:
___
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You need to install db4-devel.
/b
On Sep 5, 2009, at 7:11 AM, Tina Martinez wrote:
> When I attempt to execute the make perlmod, I currently get the
> following error:
>
>/usr/bin/ld: cannot find -ldb
>collect2: ld returned 1 exit status
>make[1]: *** [ESL.so] Error 1
___
No you should have installed gdbm-devel.
/b
On Sep 5, 2009, at 7:11 AM, Tina Martinez wrote:
> Prior to this, I encountered a similar error (cannot find -lgdbm)
> where I had to
> create a link to fix:
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I'm going to gess the call-id is the same for the second
transaction... can you provide a more detailed trace?
/b
On Sep 4, 2009, at 11:06 AM, Humberto Quintana wrote:
Hello,
I'm a new Freeswitch user. After some reading I put Freeswitch
(Version 1.0.4) to work as Session Border Controlle
make sure your firewall is not up
/b
On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote:
Hi,
I just installed freeswitch as a replacement for our Asterisk
Server. I want to untimately do Conferencing with it as I have heard
is it pretty good at it.
I have it compiled and up and
All the variables are there in XML_CDR too.
/b
On Sep 4, 2009, at 6:28 PM, Rogelio Perez wrote:
> From the mod_nibblebill documentation:
>
> At the end of a call, the module sets a variable named
> nibble_total_billed. You can use mod_cdr to record this variable to
> your CDR log.
>
> Is it poss
Can you send it to me with the data filled out off list please.
/b
On Sep 4, 2009, at 4:33 PM, Dmitry Bely wrote:
> It's fairly standard:
>
>
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show me your XML for the gateway please.
/b
On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote:
I'm started to suspect another thing.. Successful register (SIP
phone) contains
REGISTER sip:Domain SIP/2.0
while unsuccessful one is
REGISTER sip:1.2.3.4 SIP/2.0
What parameter is responsible for
Try filling out contact-host too. But if the far end gets pissed
about your contact they are broken.
/b
On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote:
Well, you are right. Looks like the problem is not with authorization
but in the line
Contact:
that the gateway would like to see as
C
Hi all
anyone know where I can find UK English recordings for the FS prompts
(assuming there are any)? (I've googled to no avail). Alternatively is
there a list of the text used so we can record our own?
Regards
Brian
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I cannot change the way SIP Authentication works. The first register
is always sent without an authorization header then is challenged. If
you're getting an instant 403 then you have something wrong in your
config and the remote system doesn't like it. Please contact your
provider and as
FreeSWITCH is not a proxy so you'll have to look at all the variables
to see if you need to send the invite out use the info app.
/b
On Sep 3, 2009, at 8:27 PM, Lars Zeb wrote:
I tried to dial sip:8...@conference.freeswitch.org via a Bria
softphone. Why does the parsed regex look like:
There will not be an authorization header on the first register
attempt... it only happens once we are 401/407'ed and the phone comes
back and registers again.
/b
On Sep 3, 2009, at 3:26 PM, Dmitry Bely wrote:
Unfortunately even after that there is no "Authorization:" header in
the REGISTE
You're not on the latest SVN that was recently added.
/b
On Sep 3, 2009, at 2:55 PM, Phillip Jones wrote:
Hi there:
The wiki states:
uuid_exists
Checks whether a given UUID exists.
Usage: uuid_exists
However when I call via an API call I get:
INVALID COMMAND!
I also don't see it in
Please join IRC if you experience the issue again #freeswitch on
irc.freenode.net
/b
On Sep 3, 2009, at 11:43 AM, Christian Löschenkohl wrote:
> sorry i can not follow you
> i build everthing from scratch (download source, unpack and build)
>
> what i mean with the build-essential package is a
Can you try NOT using a package? I have a theory that the package has
a few optimization flags in it that breaks things.
/b
On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote:
> Since you are using the debian package, the files will be in
> /opt/freeswitch not /usr/local/freeswitch.
Sounds like you have some build skew... can you tell us how you built
FreeSWITCH?
/b
On Sep 3, 2009, at 2:29 AM, Christian Löschenkohl wrote:
> hello
>
> we have regular (every 4-6 days) stability problems with freeswitch
> when the problme occurs
>
> - no registers are done bythe server (olny
well your clock shouldn't be going back in time... that is unless you
have figured out time travel or passed thru some star trekish temporal
wake.
For the most part its a harmless warning unless its happening every
second or so.
/b
On Sep 2, 2009, at 2:43 PM, Phillip Jones wrote:
> Hi th
Her real name is Katherine
/b
On Sep 2, 2009, at 2:27 PM, Carlos S. Antunes wrote:
> http://www.gmvoices.com
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UNSUBSC
Using esl + perl you could do it.
/b
On Sep 2, 2009, at 9:27 AM, Mathieu Parent wrote:
> Hi,
>
> I wanted to run an "originate" command when a MESSAGE_WAITING event
> is fired.
>
> Is there a simpler way than creating a daemon listening to the event
> socket ?
>
> Thanks
>
> Mathieu Parent
I know people that have deployed on windows... not a huge problem just
hasn't been load tested like linux... we don't have the resources or
time to load test every single platform, tune and tweak it. The
community can help out with this area a lot.
/b
On Sep 2, 2009, at 8:01 AM, Diego Tor
uuid_getvar
/b
On Sep 2, 2009, at 8:16 AM, Tristan Mahé wrote:
> Hi,
>
> just a fast 2cent:
>
> get var via channel status ? ( variable_res )
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What are you doing in these lua scripts? Because there are a few
things you can do in the lua script itself that will cause you to leak
like crazy due to improper use.
/b
On Sep 2, 2009, at 7:59 AM, Benedikt Fraunhofer wrote:
Hello *,
2009/8/31 Rupa Schomaker :
Isn't there a known issue
You don't have to do that anymore... the default profile on 5060 will
work when the users are on the public internet also. No need to have
two profiles anymore.
/b
On Sep 2, 2009, at 7:50 AM, Juan Backson wrote:
> Hi,
>
> Things are working find before I tried using public IP ( behind
> N
Try this one.
Outbound
Inbound
/b
On Sep 1, 2009, at 4:40 AM, NOx-WHV wrote:
>
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gateways do not go into the table... ONLY inbound registrations to the
profile do.
/b
On Sep 1, 2009, at 5:16 AM, Peter P GMX wrote:
> Hello Brian,
>
> I've done this. FS creates the tables sccessfully, but then doesn't
> fill
> them.
&
Use valgrind.
/b
On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote:
>
> The other two freeswitch-machines in this scenario play well, they're
> dumb load-sinks. what makes this machine different is on one hand the
> lua-scripts that are run on it and on the other hand it's the only
> machin
On the profile.
/b
On Aug 30, 2009, at 5:25 PM, Peter P GMX wrote:
> Hello,
>
> is there a chance to have sofia_reg_external in odbc/mysql instead of
> sqlite?
> In a B2BUA environment we have thousand of external registrations
> during
> a migration phase, and it would be good to have easy
Make sure you rm -rf /opt/swift then reinstall
/b
On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote:
> Nop, the 64bit cepstral doesn't work either:
>
> Creating mod_cepstral.so...
> /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when
> searching for -lswift
> /usr/bin/ld: cann
http://downloads.cepstral.com/cepstral/x86-64-linux/Cepstral_Allison_x86-64-linux_5.1.0.tar.gz
Make sure you download the 64bit version.
/b
On Aug 30, 2009, at 2:33 PM, Pete Mueller wrote:
> I don't think Cepstral supports 64-bit, as the SDK is 32-bit.
_
make sure you run ldconfig after you add the line then try to compile
it again.
/b
On Aug 30, 2009, at 1:59 PM, Max Bridgewater wrote:
> Please bear with me. Here is the content of my /etc/ld.so.conf:
>
> include ld.so.conf.d/*.conf
> /opt/swift/lib
Yes please add /opt/swift/lib to /etc/ld.so.conf as per the FAQ.
/b
On Aug 30, 2009, at 1:33 PM, Max Bridgewater wrote:
> /usr/bin/ld: cannot find -lswift
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difference. Will let ya know.
On Sun, Aug 30, 2009 at 1:58 PM, Brian West
wrote:
You still have to have the voices installed... the SDK comes with
each voice for Linux ... for windows you have to have the SDK...
/b
On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
Hi,
I was trying
:
http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
Unfortunately, the build fails as show in the log message
hereunder. In a previous thread, Brian says that the SDK is needed.
Yet on the above wiki page, it is said that the SDK is not needed.
Any suggestion
Unless you have installed the Cepstral SDK you can't compile
mod_cepstral. Please visit www.cepstral.com to purchase voices and
the SDK is included.
/b
On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote:
> mkdir .libs
> Compiling mod_cepstral.c ...
> mod_cepstral.c:41:19: error: swift.h: No suc
Group call works for me on mine... I suspect your config is not
correct... tar up your directory and mail it to me off list if you can
please.
freeswi...@default> version
FreeSWITCH Version 1.0.trunk (14665M)
freeswi...@default> group_call default
[presence_id=1...@example.com]sofia/internal/
The sip stack needs to be modified to spin that data up into the state
machine so that it can take over calls once the fail over takes
place... its not an easy task.
/b
On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
Is itnpossible to have a db cluster know the state of each and every
c
No you shouldn't have to touch ANYTHING in the default configs at all
to get 1000 with password 1234 to register... I recommend you:
cd /usr/local/freeswitch
mv conf conf.old
then do make samples again.
/b
On Aug 29, 2009, at 10:40 AM, tom wrote:
>
> ok, what u sent is fine, i have that. do i
No, enable sip trace I suspect your config is wrong in the end point.
/b
On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote:
> I am using a fresh install of freeswitch trunk and I am unable to
> register a phone on port 5080. I set the the registrar of the phone to
> server.host.com:5080 but when
this?
//yes im using the 1.0.4 tar
On Sat, Aug 29, 2009 at 11:22 AM, Brian West
wrote:
Are you on the latest configs? This message is unlikely in the
default configs. Please double check that you have the internal.xml
from the default install it forces the register domain to match the IP
s
Are you on the latest configs? This message is unlikely in the
default configs. Please double check that you have the internal.xml
from the default install it forces the register domain to match the IP
so you won't get this error.
/b
On Aug 29, 2009, at 10:19 AM, tom wrote:
> yes u are r
follow the howto on the wiki.
/b
On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote:
> You still have hardware failures and fail-over is also useful for
> hit-less maintenance on boxes.
>
> I'd be interested to know how Brian West was approaching his live
they don't go in sip_profiles... its in conf/directory/default/ and
they are all there already. Maybe your configs were not properly
installed?
/b
On Aug 29, 2009, at 8:46 AM, tom wrote:
>
> what do i have to do now to get it
> a) internally running
> b) talk to an external cllient?
_
You can also uuid_dump the uuid to see the info also.
/b
On Aug 29, 2009, at 3:17 AM, Jay Binks wrote:
Check out INFO
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
Throw that in your dialplan the look at your logs... You should find
what your after..
Jay
__
Can you get a sip trace and console trace because this is highly
unlikely.
/b
On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote:
However, I did run into one issue, not sure if its somehow related
to the phones I'm using (softphone of unknown quality (weephone) on
iphones)... I'm using th
report the issue on jira If its dumping a core something is wrong
and we need to fix it.
/b
On Aug 29, 2009, at 9:07 AM, Max Bridgewater wrote:
> Hi,
>
> Whenever i stop freeswich, it creates a core dump. How can i disable
> that?
>
> thanks,
> Max.
>
_
Don't confuse this with the actual group/ endpoint.
/b
On Aug 25, 2009, at 11:41 AM, João Mesquita wrote:
> According to wiki:
> group,[insert|delete|call]::,group [insert|delete|
> call
>
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Now you have to actually catch this in your dialplan on FreeSWITCH and
execute the bridge application to the dialstring provided
/b
On Aug 28, 2009, at 11:42 AM, Hristo Benev wrote:
> variable_sip_redirect_dialstring_0: [sofia/internal/
> sip:fra...@peer_01]
> variable_sip_redirect_dialstring
Yes you own the recordings. Its work for hire. Do you want to
include the recordings in our sound repo?
/b
On Aug 27, 2009, at 3:13 PM, Carlos S. Antunes wrote:
> Mike,
>
> Sure! I am planning on doing a session "soon", maybe in a couple of
> weeks or so. My only question is whether GM Voi
Thats still debatable The RFC's on these things read like chinese
stereo instructions.
/b
On Aug 28, 2009, at 12:13 AM, Carlos S. Antunes wrote:
> Brian,
>
> You've been vindicated. Callcentric is now advertising zero weighted
> SRV records! :)
>
> I'
/me points to the first question on the FAQ ;)
/b
On Aug 27, 2009, at 1:18 PM, Anatoliy Kounitskiy wrote:
> Thank you all for the great job :)
> Now it works as I wanted!!
>
> Tomorrow I'll will try to update the wiki :)
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Are the phones behind the same nat as FS?
/b
On Aug 27, 2009, at 10:39 AM, Dennis wrote:
> we played with using a stun-server and had a small success. if we are
> connected with a stund-server, the hangup will not come after 120
> seconds. we can also call the 5900 and listen to the moh. this wo
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