zhaoxxqq zhaox...@163.com wrote:
and I write java server code like below:
import java.io.BufferedReader;
import java.io.BufferedWriter;
import java.io.IOException;
import java.io.InputStreamReader;
import java.io.OutputStreamWriter;
import java.io.PrintWriter;
import
jun yang yj13535428...@gmail.com wrote:
now, a conference 3002 with several users in it.
i want to record 3002 manually, but can't get the way.
i have try fs_cli use the command:
conference 3002 record /tmp/foo.wav
it response: conference 3002 not found
You need to specify the full
As a further note on this subject, temporarily downgrading to libtool 1.5.26
and rebuilding FreeSWITCH gave me a working mod_portaudio.so module.
Obviously this doesn't solve the problem, but it does prove that, as
suspected, the migration to libtool 2.2.6a was the cause.
Any suggestions on how
Matthew Fong mattdf...@gmail.com wrote:
What's the best way to put 2 bridged callers into a new conference? Must I
park both uuid's first, and then transfer both to an extension that will add
them to a new conference?
No, it's uuid_transfer with the -both option to transfer both legs to the
Dome Charoenyost d...@tel.co.th wrote:
May be need
action application=set data=hangup_after_bridge=false/
before first bridge
and also, reading this wiki page may help
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate
(see the discussion of multiple destinations)
I can report that this problem (the failure of mod_portaudio.so to be linked
properly) still persists as of revision 13970.
The operating system is Debian Testing, and the difficulty began after
upgrading from Libtool 1 to Libtool 2.2.6a.
If anyone else can reproduce this or suggest a means of
Chris Chen chris.chen2...@gmail.com wrote:
Brian, I would like to be one of the volunteers helping to report issues.
That's great. We need more volunteers.
For some FreeSWITCH users (of whom I am one), the user interface of Jira is an
obstacle to reporting bugs via that mechanism, for
Tamas jal...@gmail.com wrote:
Did you make bootstrap.sh and configure before compilation?
Yes. This was a clean export from svn, built by running the Debian debuild
tool, as in
svn export to a temporary directory, followed by debuild (after changing the
version number to make the package
Brian West br...@freeswitch.org wrote:
what are the error messages?
There aren't any. The build completes without error, but the module doesn't
load due to the undefined symbols.
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Sorry - I misread the question.
The error is:
freeswi...@default load mod_portaudio
-ERR [module load file routine returned an error]
2009-06-27 12:30:00.740316 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /opt/freeswitch/mod/mod_portaudio.so
**/opt/freeswitch/mod/mod_portaudio.so:
Mark Campbell-Smith mcampbellsm...@gmail.com wrote:
How can I get more information on this fault to file a bug report?
See the debugging FreeSWITCH page on the wiki, and set
param name=dump-cores value=yes/
in the FreeSWITCH core configuration (by default in switch.conf.xml), or use a
ulimit
Edmar Cruz darklio...@yahoo.com wrote:
Here is my dialplan on sip_profiles/external/myprofile.xml
extension name=dialmyprof
condition field=destination_number expression=^(\d+)$
action application=set data=gate_site_id=1/
action application=bridge
The above should
Saeed Ahmed saeedahmad1...@gmail.com wrote:
Can we also test dialplan using CLI, like dial in asterisk?
Have a look at the originate command.
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Nandy Dagondon g...@i.ph wrote:
hi,
i tested the latest SVN build (13884) using the sample configuration files
... no modifications whatsoever. but in sofia external profile, the IP
address is my internal address instead of my external IP address.
did i miss something here?
Try setting
Mathieu Rene mrene_li...@avgs.ca wrote:
action application=set
data=failure_causes=user_busy,recovery_on_timer_expire / and then
originate it.
Or if you're originating from a script, set that as a channel variable first.
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Bradley Brashier bjbrash...@gmail.com wrote:
I want to take a second and point out that while I may be complaining about
some difficulties I'm having, the process has actually been FAR easier and
faster than I had ever expected. This is a nice, solid product that works
amazingly well
Tim B timb0...@hotmail.com wrote:
Michael, I ran the debugging you asked. I also tried to post it to
pastebin.freeswitch.org but can't login. I used my login for the freeswitch
site, but that doesn't seem to work?? How do I gain acess?
When I connect to pastebin.freeswitch.org I get a
this is an issue which I've been discussing with Brian West on IRC and in
e-mail correspondence, which I thought I should bring to the list so that
others can look at it as well.
The configuration
My external SIP profile has its ext-sip-ip and ext-rtp-ip set to
stun:stun.freeswitch.org. This is
selva kumar panse...@gmail.com wrote:
Hi,
I configured oubound in FS, it worked fine.
Then I configured inbound in FS,it also worked fine.But now the inbound
works fine and the outbound is not working.
What is the reason?
If you turn on debug-level logging, it might be possible
Jason White ja...@jasonjgw.net wrote:
The symptom is the following line in outgoing SIP messages while attempting to
establish a call to a gateway via the external profile:
o=FreeSWITCH 1245193059 1245193060 IN IP6 stun:stun.freeswitch.org
However, if I place an IPv6 call via the internal
Ken Rice kr...@suspicious.org wrote:
I doubt that header is exposed since its not a standard sip header. However
you could probably patch mod_sofia to expose it without too much trouble...
How difficult that would be is dependant on where in session that comes in
Using the info application
Raul Fragoso r...@etellicom.com wrote:
I can confirm the same issue, but it happens even with all the IPv6
stuff removed.
Thank you for the corroboration.
It only happens to me if I have the following in my external.xml profile:
param name=local-network-acl value=localnet.auto/
Note that I
Paul Mahler p...@ringcarrier.com wrote:
I have a large project coming up. I'm interested in using Freeswitch
instead of SER and Asterisk.
What is the current status of Freeswitch? Can I safely use it in a
large scale commercial environment? How active is the Freeswitch
developer
Durk de Beer durk.deb...@isp.solcon.nl wrote:
Hello I've minimized de xml files where possible to make a dialplan that is
as short as possible. Now do I've this funny effect to dial my extensions
who are running from 200 to 207. It seams that I'm able to dial an
extension in closed in a
Diego Viola diego.vi...@gmail.com wrote:
Since FreeSWITCH already uses PCRE, it would be cool if there is Parrot
support also.
Are you offering to write it?
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Brad Tuan brad.t...@gmail.com wrote:
If i don't want to use softphone function to transfer the call ,how to do
it??
uuid_transfer. Have a look on the wiki.
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Michael Jerris m...@jerris.com wrote:
Can you please re-test with current svn trunk. we added some new nat
busting code yesterday that may assist with this. You will need to
specify the new param name=local-network-acl value=localnet.auto/
param in the sofia profile (see
zhaoxxqq zhaox...@163.com wrote:
I have not use 'auth Cluecon' before sending api command.
I send other api have no problem.only play wav have problems
Try it from a telnet session. Start with auth ClueCon, then issue the API
command as shown in my example.
Unless you do something wrong, it
Bruce McAlister bruce.mcalis...@blueface.ie wrote:
I get past this initial error if I change my C compiler from
usr/bin/cc to /usr/bin/c99.
After changing the above, the compilation goes further, but I am now
faced with a different error:
Have you tried compiling with gcc? I would also
Brad Tuan brad.t...@gmail.com wrote:
I have tried
extension name=remoteFreeswitch
condition field=destination_number expression=^014(\d+)$
action application=bridge data=sofia/external/$1 at
192.168.141.187http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
:5080/
Also, if the other FS box is behind the same NAT you're on, you should be
using the internal profile:
sofia/internal/$...@192.168.xxx.xxx or whatever.
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Brad Tuan brad.t...@gmail.com wrote:
As title
Write a script that connects to the event socket and issues an api reloadxml
command.
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Brad Tuan brad.t...@gmail.com wrote:
I have tried
extension name=transfer_to_MeiLan
condition field=destination_number expression=^014(\d+)$
action application=bridge data=sofia/internal/$1%192.168.141.187/
Change the % to an @ in the above.
Brad Tuan brad.t...@gmail.com wrote:
This question can be separated into two part:
1.Pass a call to another FS
uuid_deflect or uuid_transfer, depending on whether the call has been answered
by the first FS instance or not. See the wiki.
2.Receive a call from another FS
Provide a dial plan
Brad Tuan brad.t...@gmail.com wrote:
When User1( User of FS1 ) call User2( User of FS2 ) ,
FS1 will pass the call to FS2 before answering,
You just need to write a dial plan extension that matches the call on FS1 and
bridges it to FS2.
For example:
include
extension name=remoteFreeswitch
zhaoxxqq zhaox...@163.com wrote:
I use event socket to send command to FS conference.
I send conference testconf play /root/test.wav in console. It worked ok.
I send api conference testconf play /record/test.wav by event socket. and
the response isDisconneted, Good bye.See you at
I should post the full session:
ja...@jdc:~$ telnet localhost 8021
Trying ::1...
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Content-Type: auth/request
auth ClueCon
Content-Type: command/reply
Reply-Text: +OK accepted
api conference 3300-192.168.0.2 play /tmp/msg.wav
Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
The problem is the null section
Yes, switch_simple_email is probably being called with a null first argument.
This shouldn't happen.
Which svn revision are you on? Does it still happen with the latest svn
revision?
Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
1.0.4pre8
It works for me with revision 13501. Mine is later than yours. Try upgrading.
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Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
I've upgrade to 13523 and I get the same result.
My only suggestion at this point is to debug it with gdb to find out why a
null argument is being passed to the function. There must be something in your
configuration or environment that differs
After ZRTP negotiation is complete (the ZRTP state machine has entered the
secure state), I get a number of lines in the log as follows (FreeSWITCH
rev. 13501):
2009-05-29 16:43:19 [DEBUG] switch_rtp.c:538 zrtp_logger() [zrtp protoco]:
ERROR! Decrypt failed. ID=14:DH s=SRTP authentication
Peter Olsson peter.ols...@visionutveckling.se wrote:
After using the latest trunk revisions I get no audio anymore. The last
working build I have is about 5 days ago. I havn't upgraded until today, so
I don't know exactly when this happened.
You could always check out some intermediate
Brian West br...@freeswitch.org wrote:
This is normal because the switch from clear to secure can happen
quickly on one end or the other and you'll have a few packets that get
thru before one end is ready... nothing to be worried about.
I thought that might be the scenario.
In a typical
Luis M. Zuccolo luismzucc...@yahoo.com.ar wrote:
I'm using postfix, that has a compatiblilty interface to sendmail.
I've used this with Sendmail successfully; it should work with Postfix too.
See the mailer-ap and mailer-app-args variables in
autoload_configs/switch.conf.xml and be sure they
Jason White ja...@jasonjgw.net wrote:
See the mailer-ap and mailer-app-args variables in
autoload_configs/switch.conf.xml and be sure they are set correctly for your
installation. Try running the Postfix sendmail program manually to be sure
that it is working correctly.
sendmail -t
Jim Burke j...@evolutiontel.net wrote:
extension name=On-Net_calls
condition field=destination_number expression=^103$
action application=set data=continue_on_fail=79/
action application=set data=continue_on_fail=true/
Why are you setting the same variable twice?
Jim Burke j...@evolutiontel.net wrote:
If I understand your comment correctly, I did not have both of the
above snippets in the dialplan at the same time. The dialplan was
modified continually to get the correct vars that worked for my
situation and then reloadxml to get them working.
Right,
Brad Tuan brad.t...@gmail.com wrote:
But,the response message change from 407 Proxy Authentication Required to
480 Temporarily Unavailable today.
Anybody can tell me what happen??
Your SIP trace might give you a clue as to what happened.
sofia profile external siptrace on
Brian West br...@freeswitch.org wrote:
Thank you... Now please tell 10 of your friends about FreeSWITCH ;)
Also, if you're a member of a Linux user's group or similar organization, now
might be a good opportunity to raise FreeSWITCH awareness on their mailing
list or at a meeting.
Diego Viola diego.vi...@gmail.com wrote:
Hi, I have downloaded the latest freeswitch trunk, and when I do
reloadxml I get this.
Error [unterminated ${var}] in line
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
line 12
Any ideas? I haven't edited that file
Jens Vegeby j...@vegeby.nu wrote:
I think you can do that by creating another profile.
Then you can bind it to a specific IP address.
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Brad Tuan brad.t...@gmail.com wrote:
So , Anyone please tell me how can i do??
Turning off the ACL and setting auth-calls to false should be enough to do it.
To find out where the problem is in your configuration, set the log level to
debug if it isn't already, and read the logs carefully.
You
I've already discussed this with a few members of the community, but I would
like to raise it with a wider FreeSWITCH audience.
Since upgrading to libtool 2.2.6a (now the default in debian testing), I can't
successfully link mod_portaudio.so. The system is Debian Testing, x86_64
architecture.
With apologies for the incorrect address in the header, if you reply to this
follow-up instead of the original message we should be fine for the remainder
of the thread.
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FERNANDO VILLARROEL fvillarr...@yahoo.com wrote:
My install of wikipbx was succesfully, but i have problem for registering a
Softphone Xlite for testing; i look the next warning in FS_CLI:
freeswi...@internal 2009-04-28 06:13:58 [WARNING] sofia_reg.c:1701
sofia_reg_parse_auth() Can't find
Dale Trub dalet...@gmail.com wrote:
Does anyone have any recommendations about how to troubleshoot this?
sofia profile external siptrace on
and watch the SIP traces to see what happens.
Any known issues/patches in FS that could be biting us?
You didn't say which version you were running.
ravi hum ravi_...@yahoo.co.in wrote:
*call request is send from FreeSwitch to SIP Proxy server ( FreeSwitch ---
SIP Proxy Server)
please let me know how to solve this issue.
If there is a NAT device involved anywhere in your scenario, it's probably the
cause.
Lon Baker l...@kickasspixels.com wrote:
When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on
Ubuntu server 9.
I think there was a fix to fsctl to eliminate segfaults recently. If you
upgrade to trunk it might work now.
___
David Robinson pawzl...@gmail.com wrote:
Is this correct ? Am I missing something fundamental ?
My suspicion is that the RTP traffic isn't traversing the NAT properly. You
may have to configure the routers at both ends to forward the RTP packets to
the correct destinations. There is a good
adamF adam.falc...@gmail.com wrote:
Yes I am passing -nc when starting freeswitch and I can receive calls without
issue initially. If I wait 10-15 then try to place another incoming call
freeswitch will not pick it up. I haaven't found where anyone else has
reported this issue so I am at a
Andy a...@fabulous4.co.uk wrote:
The DTMF method was efault which I believe is info but I've now set it
explicitly to rfc2833 inband to see if that helps. Is there a way I can tell
from the logs that this is the case and that my config changes have worked.
This is in the logs, and (assuming
I now have another core file from FreeSWITCH, generated when it was
initializing the TLS during startup.
I know this should all be in a bug report - I tried Jira again yesterday, but
haven't been able to sort out the accessibility problems I was having with it.
If you would be interested in a
Saeed Ahmed saeedahmad1...@gmail.com wrote:
During Make Current I see no errors, at the end I get successful
installation message; I also tried to scroll up to see any possible errors
but I guess there was nothing.
Have you done a fresh checkout and tried to rebuild from the beginning?
When
I've narrowed this problem down.
When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833()
from switch_rtp.c is never called, as evidenced by freeswitch.log.
However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH
instance), do_2833() is called. It is
Jason White ja...@jasonjgw.net wrote:
It is also called if I use the voicemail
extension on my local FreeSWITCH.
Apologies for the nonsense - I meant that switch_rtp_dequeue_dtmf() is called
in that case, for DTMF detection.
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As a matter of interest, the other end (as reported in its SDP) is BroadWorks.
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Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for
that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
___
Rupa Schomaker r...@rupa.com wrote:
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
If someone could add it to Jira, I'll detail the issue here. The Jira
Remko Kloosterman r.klooster...@mtel.nl wrote:
Did you make a wireshark trace yet? You should be able to find out
exactly what's going on there, which protocol is used, etc. We've had
our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar
Anthony Minessale anthony.miness...@gmail.com wrote:
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
Thank you for the suggestion.
I tried both the Sonus and Cisco settings in the external profile (running
sofia profile
Diego Viola diego.vi...@gmail.com wrote:
Hi guys,
Nevermind with the ESL and EM thing.
I was wondering what the getBody() getHeader() and other ESL stuff
does behind the scenes, in raw socket, do you know?
Why not read the source code? This is free software and open-source, after
all.
A quick update: I can still reproduce the profile startup failure under
revision 13246, but I haven't hit the segfault again.
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Lars Zeb larc...@yahoo.com wrote:
I have installed from the 1.0.4pre7 tarball on a openSuse 11.1.
Why is it that after I launch freeswitch and type in either 'show' or
'status' at the console, it responds with 'Unknown command', but it does
accept 'shutdown'?
Maybe the mod_commands module
Just to add data to this:
PowerTOP 1.11 (C) 2007, 2008 Intel Corporation
Collecting data for 15 seconds
Detailed C-state information is not available.
P-states (frequencies)
2.34 Ghz 0.0%
2.00 Ghz 100.0%
Wakeups-from-idle per second : 405.4interval: 15.0s
no ACPI power
I know this isn't the place to report bugs; unfortunately, the Jira Web
interface isn't working for me due to accessibility issues. (If there is an
alternative way to submit reports that could be efficiently handled by the
developers, let me know).
A few weeks ago I reported problems with the
Paweł Pierścionek pa...@voiceworks.pl wrote:
boot Your kernel with divider=10 nohz=off options :)
Recent kernels are tickless which basically causes all freeswitch
timers/sleeps to fire at requested microsecond intervals.
With nohz kernels You get hundred times more system calls with
After upgrading to the 2.6.29 kernel (the Debian packaged version), FreeSWITCH
takes up more CPU time than usual, e.g., 7% as reported by top, and the load
average is high (e.g., 0.87) even when the machine is idle and there are no
calls in progress. When top is run, FreeSWITCH appears at the
paul.d...@gmail.com paul.d...@gmail.com wrote:
I am trying to remove default users from my FS installation, I removed
folder default with a bunch of users with numbers 1000 and up,
restarted FS, but it seems to be cached somewhere, I guess in internal
FS database. How do I purge it?
It
Peter P GMX prometheus...@gmx.net wrote:
has anybody tried successfully to setup a Nokia E71 (or similar symbian
S60 3rd phone) with Pjsip and TLS/SRTP?
TLS seems to work but what about the SRTP part?
Do you have log entries like this?
2009-04-24 11:05:19 [INFO] switch_rtp.c:782
Diego Viola diego.vi...@gmail.com wrote:
I got it, thanks people :D
Could you now add it to the documentation?
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Diego Viola diego.vi...@gmail.com wrote:
Let me know if you have some nice ideas for my presentation, I already got
some by myself, but more are always welcome :).
You could demonstrate the flexibility of the dial plans, in particular the use
of regular expressions and the dial plan syntax to
Pete Mueller p...@privateconnect.com wrote:
Is there a way to detect DTMF during bridged conversation?
You can use bind_meta_app in your dial plan; see the wiki for details.
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Nicolas Brenner nico...@medularis.com wrote:
Hi, I am not very familiar with FS internals, but I recently found this
new db engine called couchDB. Looks pretty interesting, and its main focus
is scalability.
Has anybody played with couchDB? does it make sense to replace sqlite with
couchDB in
David Robinson pawzl...@gmail.com wrote:
Is this how I should be doing this ? I want to specify a different
gateway for a different rexep. Please give me some idea what path I
should take.
Make sure that FreeSWITCH actually reaches your extensions while searching the
dial plan. Order is
Matthew Fong mattdf...@gmail.com wrote:
My question is, is there a way to use mod_vmd to detect if an answering
machine or human has picked up within the first 1-2 seconds after being
answered?
Probably not. If you have an algorithm in mind that would achieve this with a
high degree of
mszla...@aol.com mszla...@aol.com wrote:
Is there a way to change $${local_ip_v4} in one place.
Of course. That's why it's a variable.
X-PREPROCESS cmd=set data=local_ip_v4=10.10.1.2/
this goes in vars.xml, substituting the desired address.
___
xbipin bi...@xbipin.com wrote:
any1 have any idea how what to sue in dialplan such that calls from a single
id go to a specific gateway only with blind registration enabled, this is
the only major issue im having.
Perhaps you could match the source address in the dial plan and then bridge or
dujinfang dujinf...@gmail.com wrote:
However, the caller do need to hear the early media to figure out
what's going on. If I set ignore_early_media=false, only the first one
tried.
Could you use ring_ready? that way, the calling SIP phone should generate the
ringback.
seven dujinf...@gmail.com wrote:
Agree, I think the author better to document the code first.
Well, actually... it's already done. It's called API documentation, and
consists of specially written comments in the code.
This is not user-level documentation, however; it exists to help programmers
Alfonso Pinto elhod...@gmail.com wrote:
One question more, maybe a stupid one: How can I search the archives?
http://www.gmane.org/
The searching tool they use, Xapian, tends to give good relevance ranking, at
least in my experience.
___
Brian West br...@freeswitch.org wrote:
This isn't a buffet where you pull up and demand things be one way or the
other... this is a community where you start helping. I would love to
see more helping and less demanding!
So would I.
I regularly scan the mailing list looking for questions to
Michael Collins m...@freeswitch.org wrote:
I like it... The Wiki Tax
It's an excellent suggestion.
As an aside, would it be possible for the wiki administrator to modify the
configuration so that there is a means of subscribing without having to deal
with a captcha?
For people who can't see
badeguruji badegur...@yahoo.com wrote:
Q. what voip related software will i need to run above setup?
FreeSWITCH
Q. what hardware (like phone card etc.) do i need in my PC?
A digium or similar card, and a SIP phone of course.
Q. will i be able to use my existing phone number? (i do not have
While I was trying to obtain more detailed logs of my portaudio problems,
FreeSWITCH crashed, leaving a core file.
The backtraces are here:
http://pastebin.freeswitch.org/7998
As far as I can remember, at the time of the segfault, one channel was trying
to connect and not succeeding; I had just
James H Thompson j...@lj.net wrote:
I've been considering mirroring some of the major voip mailing lists on
voip-info.org
into a forum of somekind.
Have a look at http://www.gmane.org/
and note that you can post via NNTP or via the WEb.
This mailing list is subscribed to gmane.
Addison Martin freeswi...@servercorps.com wrote:
Also, moving the list to Google Groups would allow email OR threaded
views, and personally I like them better than nabble.
http://dir.gmane.org/gmane.comp.telephony.freeswitch.user
Would any of those views suffice?
I've read the ipv6(7) manual page now.
Unfortunately
echo 1 /proc/sys/net/ipv6/bindv6only
doesn't solve the problem as the manual page suggests it should:
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This has occurred with a number of recent revisions. If anyone can reproduce
it or suggest debugging steps, I'll gladly supply more information. Jira isn't
convenient for me due to X issues at the moment, unless there's a way to
interact with it other than via a Javascript-capable Web browser.
Jason White ja...@jasonjgw.net wrote:
It looks like an operating system issue to me.
Furthermore, the following message on linux-kernel appears relevant.
http://linux.derkeiler.com/Mailing-Lists/Kernel/2005-03/3988.htm
From what I have been able to ascertain, Red Hat/Fedora kernels don't seem
I have TLS enabled in my internal and internal-ipv6 profiles as per the stock
configuration.
When FreeSWITCH is started, sometimes either of the profiles fails to
initialize, with an Unable to create SIP UA for profile error in the log. If
I then start the profile manually
sofia profile
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