Hi,
I have a small question about running Freeswich in the background using -nc
option. Does freeswitch writes the log in any log file when running in
background mode?
If I have some print out statement in my custom mod, can I still see the
output of those logs somewhere?
Thanks,
Mark
I'm not sure if we understood each other correctly.
I meant calling from jabber to other sip phones. I'm not sure you can just
add a sip phone as
a buddy into e.g. gtalk or any other jabber service and call it.
So that why a bot (and only one but), which you would have as a buddy and
you would
On Wed, Dec 17, 2008 at 12:10:18PM -0500, Michael Jerris wrote:
On Dec 17, 2008, at 2:40 AM, Jason White wrote:
The code in bind6only_check in libs/sofia-sip/libsofia-sip-ua/tport/
tport.c looks
correct to me, but I can't find where the result is tested (it's in
mr_bindv6only). When
I bind mine independently without a problem on CentOS 5.2
/b
On Dec 18, 2008, at 2:53 AM, Jason White wrote:
I realized after posting that if the IPv4 port is bound by another
process,
then the attempt to bind to the IPv4 port in bind6only_check()
should return
-1, and hence the result
On Thu, Dec 18, 2008 at 03:01:12AM -0600, Brian West wrote:
I bind mine independently without a problem on CentOS 5.2
Thanks; friends of mine have access to Fedora boxes, so we'll compare
behaviour and try to sort it out.
___
Freeswitch-users mailing
Hello,
Im a newbie in FS. I get an error from the freeswitch cli:
2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL ERR
[database disk image is malformed]
I dont know what to do to remove this error! Can you help me?
Regards
Hello,
Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it was the same problem as
On Thursday 18 December 2008 22:20:24 fidibus83 wrote:
I’m a newbie in FS. I get an error from the freeswitch cli:
2008-12-18 10:09:14 [ERR] switch_core_db.c:100 switch_core_db_exec() SQL
ERR [database disk image is malformed]
I don’t know what to do to remove this error! Can you help me?
Brian,
That wasn't exactly what I meant :)
I have had Freeswitch connecting to GTalk directly as a client and that was
where I was getting the issues with sending anything to other domains.
I haven't actually tried the server profile with my own ejabberd server.
What I was planning to do was
Thanks. It's ok again!
-Ursprüngliche Nachricht-
Von: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] Im Auftrag von Hadley
Rich
Gesendet: Donnerstag, 18. Dezember 2008 10:38
An: freeswitch-users@lists.freeswitch.org
Betreff: Re:
Kriko,
Have a look at this, I used it to get my gtalk to fs working.
http://chesterton.id.au/blog/2008/01/02/freeswitch-google-talk-dingaling-jingle-all-the-way/
Cheers
Kirk
Date: Thu, 18 Dec 2008 09:37:38 +0100
From: kriko kristjan.ug...@gmail.com
Subject: Re: [Freeswitch-users] Call sip
I made a simple java example, following this guide
http://wiki.freeswitch.org/wiki/Java
so when someone calls it should print something in console.
I've also modified dialplan/public.xml, what I want is to intercept calls
from jabber and process them:
http://pastebin.com/m35de11d9
But there
I made a simple java example, following this guide
http://wiki.freeswitch.org/wiki/Java
so when someone calls it should print something in console.
I've also modified dialplan/public.xml, what I want is to intercept calls
from jabber and process them:
http://pastebin.com/m35de11d9
But
It is the right jar, I renamed it now to phoneTest.jar but still not
working.
Do I have to specify whole path to the program inside class or just the
class?
Remote debugging is working, but breakpoints never got triggered, so it is
not being
executed at all.
I'm trying to process a call
A note, logging is handled by mod_logfile, it has nothing to do if you
run in the background or not.
Mike
On Dec 18, 2008, at 3:29 AM, Jason White wrote:
On Thu, Dec 18, 2008 at 04:01:07PM +0800, mark morreny wrote:
I have a small question about running Freeswich in the background
using
On Dec 18, 2008, at 4:45 AM, Kirk Bateman wrote:
have had Freeswitch connecting to GTalk directly as a client and
that was where I was getting the issues with sending anything to
other domains.
I have seen this before specifically with gmail being unable to
federate presense... their
*Hi,
I am using JavaScript file to detect busy tone signals but I cant able to
detect the busy tone signals
*
*My JavaScript*
*
session1 = new Session();
session1.originate(session1, {ignore_early_media=true}sofia/default/
39841799...@172.20.191.228);
session1.execute(tone_detect, busy 400 r);
can you press f8 to set the FS console to DEBUG and take the same capture.
On Wed, Dec 17, 2008 at 8:45 PM, Scott Ellis scott.el...@novatex.com.auwrote:
After further checking, it does not seem like the authentication after the
challenge is being sent...
Are there any other settings I
did you turn up your console log level high enough to see it? The default
level is INFO
On Thu, Dec 18, 2008 at 6:09 AM, kriko kristjan.ug...@gmail.com wrote:
It is the right jar, I renamed it now to phoneTest.jar but still not
working.
Do I have to specify whole path to the program inside
What is the process for capturing and submitting a core dump?
I am messing around with the Cisco 79x1 phones and tcp and multiple reg.
I have a 7961 using tcp and a 7960 using udp both reg'd with the same
number and both showing up as registered. If I call out from the phone
using tcp, it
You've got ignore_early_media set to true but busy signals might be
sent during early media. Why are you ignoring early media?
Also, you might need to check your tone_detect syntax. You're set to
detect 400Hz but you haven't told the system what to do if it does
detect that tone. Please
I recently purged all freeswitch config and restarted configuring from
scratch. Using defaults, I modified public.xml dialplan config (added line
16 - 28):
http://pastebin.com/m5ece6e6f
and added a new config under jingle_profiles:
http://pastebin.com/d6e983b99
I register with phonelite or
Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch
-MC
Sent from my iPhone
On Dec 18, 2008, at 6:38 AM, pe...@networkoblivion.com
pe...@networkoblivion.com
wrote:
What is the process for capturing and submitting a core dump?
I am messing around with the Cisco 79x1 phones
If anybody wants to look at the core dump in gdb, here it is (the actual
core is 256Meg):
http://pastebin.freeswitch.org/6476
I know zip about debugging and gdb, but from looking through it, I see a
segmentation fault and it appears to be thread 15094. The last three
items in the bt full for
Is this a single occurrence or can you make it happen consistently?
-MC
Sent from my iPhone
On Dec 18, 2008, at 7:36 AM, pe...@networkoblivion.com
pe...@networkoblivion.com
wrote:
If anybody wants to look at the core dump in gdb, here it is (the
actual
core is 256Meg):
Peder,
Can you join us on IRC.
/b
On Dec 18, 2008, at 9:36 AM, pe...@networkoblivion.com wrote:
If anybody wants to look at the core dump in gdb, here it is (the
actual
core is 256Meg):
http://pastebin.freeswitch.org/6476
I know zip about debugging and gdb, but from looking
i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).
we use 3 freeswitch servers and the carrier passes calls to the three
fs servers randomly. if on fs server is not offline, the carrier sends
the call to the next fs.
this is generally
the deflect app.
/b
On Dec 18, 2008, at 10:36 AM, Dennis wrote:
i would like to know, what the best way is, to redirect an incoming
call from one fs (fs1) to another fs (fs2).
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
i had a look at the deflect app, but as far as i understand it, the
carrier has to support/understand it ans react on the signals.
is that right or does this have nothing to do with our carrier? or
does this work between fs servers in the same local network?
another similar question is: how to
What switch is your provider using?
/b
On Dec 18, 2008, at 10:52 AM, Dennis wrote:
i had a look at the deflect app, but as far as i understand it, the
carrier has to support/understand it ans react on the signals.
is that right or does this have nothing to do with our carrier? or
does this
sorry, i do not know that. i could ask tomorrow.
is deflect, what i understand? the provider has to support it? if yes,
what could i tell and ask the provider, to find a solution to this
problem? the provider is quite open for new ideas, although we do not
want to be to dependant on the provider
Well its a standard SIP Refer, They may not support it for good reason.
/b
On Dec 18, 2008, at 11:07 AM, Dennis wrote:
is deflect, what i understand? the provider has to support it? if yes,
what could i tell and ask the provider, to find a solution to this
problem? the provider is quite open
It seems to be related to 20ms vs 30ms ptime.
What are the 2 devices and what rev of FS are you on?
There was more code added in the last few weeks to smooth out this
occurrence.
you can also opt to declare your codec prefs at 30ms
p...@30i instead of just PCMU in the conf.
On Thu, Dec 18,
so if they do not suport it (which has to be seen), is there another
way to redirect a call from one fs to another without the provider?
like redirect from one fs to the other over the local lan?
2008/12/18 Brian West br...@freeswitch.org:
Well its a standard SIP Refer, They may not support it
do they follow a 302 redirect? Because if the call isn't answered yet
then you can do a redirect
/b
On Dec 18, 2008, at 11:27 AM, Dennis wrote:
so if they do not suport it (which has to be seen), is there another
way to redirect a call from one fs to another without the provider?
like
Hello
I've checked out the latest trunk, the problem is still left.
Im using a Linksys SPA8000 analogue telephone adapter as one device. The
other call comes through the sip gateway (from PSTN).
I'll try to specify 30ms.
Regards,
Jonas
On Thu, Dec 18, 2008 at 6:20 PM, Anthony Minessale
sorry, this is to difficult for me. what does that mean?
they pass a call to one of our fs. then we see, that the call should
be on another fs. we know, that the call is on the wrong fs, before we
send an answer. so we could react accordingly.
2008/12/18 Brian West br...@freeswitch.org:
do
I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to redirect to.
Gabe
Dennis
When using bypass_media (aka. no_media) mode between an X-lite softphone and
Prophacy ASR, I get intermittent crackiling background noise with the audio
that I'm hearing.
How do I get rid of this?
___
Freeswitch-users mailing list
Gabriel Kuri wrote:
I've tried to do the same and in my own experience, most carriers don't
accept 302 redirects. What I've seen is they take the 302 as a failure
and move on to the next switch, so worse case with 3 switches, it will
take 2 retries before hitting the switch you want them to
so at least they should react on a 302? this could help, although i do
not really understand, what happens on a 302.
if they support it, they would receive the target fs server ip where
they should try next with deflect?
if everything does not help and is not possible: what could i do else?
it
I agree with Ray ... using a 3XX series message is a bad idea ... or you
could put OpenSer in front using the LCR module ... 503 to OpenSer and it
would route to the next gateway in the gateway group.
I have yet to work with any carrier that handles 3XX series correctly except
for some of the TCAP
I'm no expert, but I believe in media bypass mode freeswitch isn't handling
media so it's not a fs fix, it would be the quality of connection for each of
the originator/terminator, fs just directs each endpoint to set's up a point to
point connection for RTP.
Is this right?
mszla...@aol.com
We do not support registration fetching.
Mike
On Dec 18, 2008, at 9:56 AM, kriko wrote:
I recently purged all freeswitch config and restarted configuring from
scratch. Using defaults, I modified public.xml dialplan config
(added line
16 - 28):
http://pastebin.com/m5ece6e6f
and added a
If you need to do load balancing, you could set up a conference_a domain on one
switch, conference_b on the second, conference_c on the third, then use
xml_curl to dialplan and bridge the call to the right domain...
But again, I am no expert... Just a noob trying to be creative. :P
Chris
Today I also tried playing a wav file with the play application and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The point is that FS plays the file for several seconds, but I don't
hear any sound.
I also looked at the libraries
Yes Chris you are right. FreeSWITCH isn't involved in the media at all.
/b
On Dec 18, 2008, at 12:14 PM, Chris wrote:
I'm no expert, but I believe in media bypass mode freeswitch isn't
handling media so it's not a fs fix, it would be the quality of
connection for each of the
thanks for all your help!
this sounds interesting. it seems, that these codes should be
available by default with sip!? is this right?
i will talk to the carrier tomorrow and ask, what is possible.
as far as i can see, i am always dependant on the carrier? there is no
way to pass a call from
shout does not play wav files it plays mp3 files.
Mike
On Dec 18, 2008, at 1:29 PM, Peter P GMX wrote:
Today I also tried playing a wav file with the play application
and it
worked. However accessing the same file through shout:// didn't work
with freeswitch (with Totem it worked).
The
Man, I can't win with this one.
I can bypass media between two endpoints with some static but what I really
want FS to do is process the audio before it's passed on.
However, getting FS involved is something I haven't had any success in with
these two endpoints ... so far.
Thanks for
Can you answer the questions and possibly go online on IRC so we can debug
your issue?
On Thu, Dec 18, 2008 at 9:36 AM, pe...@networkoblivion.com
pe...@networkoblivion.com wrote:
If anybody wants to look at the core dump in gdb, here it is (the actual
core is 256Meg):
Carole O. wrote:
Hello,
Thanks for your answers!
Concerning the creation of a new variable for the conference the problem is
that I do not create channels from the conference. I call separately a new
member on a new channel and add it on the conference only if he agrees to
enter it. So it
On Thu, Dec 18, 2008 at 1:04 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
I agree with Ray ... using a 3XX series message is a bad idea ... or you
could put OpenSer in front using the LCR module ... 503 to OpenSer and it
would route to the next gateway in the gateway group.
I have yet to work
Excellent advice. So just letting L3 know the IP's and you'll be fine.
/b
On Dec 18, 2008, at 1:21 PM, Kristian Kielhofner wrote:
Level(3) readily supports 302s if the destination IP of the new
contact has been made known to Level(3) beforehand. You can't 302
just anywhere but you can
I can make it happen on demand. All I have to do is call the shared
number and it crashes. I'll hop on IRC in a bit.
Michael S Collins wrote:
Is this a single occurrence or can you make it happen consistently?
-MC
Sent from my iPhone
On Dec 18, 2008, at 7:36 AM,
Anthony,
Hopefully I have everything that you need this time. (Thanks for the
help!)
Also, running under Ubuntu 8.10, as a VMWare workstation vm, on Vista.
Phones are registering ok with FreeSwitch and I can make calls ok phone
to phone etc.
Scott
FreeSWITCH Version 1.0.trunk (10760)
On Thu, Dec 18, 2008 at 07:02:18AM -0800, Michael S Collins wrote:
Check out this page:
wiki.freeswitch.org/wiki/Debugging_Freeswitch
In the long term (i.e., when more important matters aren't at issue), it might
be a good idea to modify the build process so that the debug symbols are
written
Thanks for the response Brian. I don't understand what bind_meta does, or how
it can help me. Is it something I can use from my Python script? I searched
for a description of it, but I was unable to find one. Could you please point
me towards some documentation, or maybe quickly explain it?
This is a function of the packaging system, not the build system and
at least the debs do have this already. On a related note I fixed the
Sofia build to include proper symbols now all the time in the debug
build.
Mike
On Dec 18, 2008, at 11:35 PM, Jason White ja...@jasonjgw.net wrote:
On Thu, Dec 18, 2008 at 11:41 PM, Jason White ja...@jasonjgw.net wrote:
On Fri, Dec 19, 2008 at 12:24:24AM -0500, Michael Jerris wrote:
This is a function of the packaging system, not the build system and
at least the debs do have this already. On a related note I fixed the
Sofia build to
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