Hi,
can any know how to implement fail over with freeswitch, please help me
Regards
--
Srinivasula Reddy K
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On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
can any know how to implement fail over with freeswitch, please help me
This issue has been debated many many times in the mailing lists.
(hint: no live call failover, HA with OpenSERet similia as
On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola diego.vi...@gmail.com wrote:
Very nice :)
On Sun, Oct 4, 2009 at 11:16 PM, Karl Vesterling k...@ken-ton.com wrote:
Folks;
Here's something that I did playing around w/ learning Apple Motion.
Me too: very nice!
-gmaruzz
--
Sincerely,
Giovanni
HI all,
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Hello?
On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
HI all,
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Hi all,
I've compiled the freeswitch with libpri support. But when I execute
oz libpri debug 1 all, I got the following error.
API CALL [oz(libpri debug 1 all )] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.
Here is my openzap configurations.
openzap.conf
[span zt PRI_1]
Sorry my mail client has some problem. I've send another mail with my
question. Kindly ignore this one.
On Mon, Oct 5, 2009 at 12:43 PM, Diego Viola diego.vi...@gmail.com wrote:
Hello?
On Mon, Oct 5, 2009 at 7:07 AM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
HI all,
On Mon, Oct 5, 2009 at 8:59 AM, Giovanni Maruzzelli gmar...@celliax.org wrote:
On Mon, Oct 5, 2009 at 8:24 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
can any know how to implement fail over with freeswitch, please help me
This issue has been debated many many times in the
hi,
FreeSWITCH as is have no live fail-over support, but this will change.
A live fail-over and redundancy mechanism is part of what SIGTRAN provides of
added values. I am working on this, but it will take time before this is on a
functional stage and available.
Also - SIGTRAN only
On Behalf Of lakshmanan ganapathy
...
I've compiled the freeswitch with libpri support. But when I execute
oz libpri debug 1 all, I got the following error.
API CALL [oz(libpri debug 1 all )] output:
src/ozmod/ozmod_libpri/ozmod_libpri.c: -ERR invalid span.
If you would start freeswitch
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost. This is an extract of my dialplan
extension name=Local
condition field=destination_number expression=^(10[01][0-9])$
it works,
thx!
T.
On Mon, Oct 5, 2009 at 12:31 AM, Michael Jerris m...@jerris.com wrote:
I updated the tiff lib to build better inline, try make tiff-reconf
Mike
On Oct 2, 2009, at 8:05 AM, Tihomir Culjaga wrote:
hello,
i just got the last trunk and tried to compile it on one of my
Thanks for pointing that.
I also tried that.
But in that case, I'm not able to make a call through openzap.
When I say
originate openzap/1/A/number number
It reported the following error
2009-10-05 17:45:47.733495 [ERR] ozmod_libpri.c:88 Can't destroy call 0!
API CALL
Hi,
When I record a call in FS, it only creates a 388-byte-long wav file. The
conversation is no written there, and FS deletes the file when the session
finishes.
What can cause this strange behavior?
Br/
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Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all users
register to the 'external' instance which acts as a gateway by 'internal'
instance (which in turn is controlled by my applicaiton with commands sent by
socket).
When user hangs up, the 'hanged up' event is propagated
Hello,
When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is
there any way to improve it? Is this a known problem?
Br/
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Fax tones are not played by the remote machine until after answer, the
tone_detect application starts a media bug that listens for the tone,
can you confirm the tone is happening at all. Maybe the issue here is
the timeout, try making that longer, or doing the tone_detect in
Hello.
I'm trying to configure stun in fs 1.0.4
vars.xml
X-PRE-PROCESS cmd=set data=bind_server_ip=auto/
X-PRE-PROCESS cmd=set data=external_sip_ip=stun:example.com/
X-PRE-PROCESS cmd=set data=external_rtp_ip=stun:example.com/
external.xml
param name=rtp-ip value=$${local_ip_v4}/
param
lakshmanan lakindi...@gmail.com said:
Thanks for pointing that.
I also tried that.
But in that case, I'm not able to make a call through openzap.
What is in openzap.conf.xml? If you start fs_cli and enter oz list,
what does it show? Copy the ozmod lines from freeswitch.log to
Is it possible to treat a recorded voice as voice mail?
Assume that, I've recorded a conversation and I want this recorded file to
be treated like voicemail. So, I could check it like voicemail!!
--
View this message in context:
Dear friends,
I am trying with ESL outbound socket. I'm trying to make a call when I
receive ANSWER event. Now, I would want to do something like,
* Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
* If it is
That's is a somewhat vague position.
You did not mention which version of FreeSWITCH you are running, the phones
being used in your example, your configuration, the codecs in use etc.
BTW,
I think you should only ask one question at a time on this list. The list
is run by volunteers and it's
hi Mark,
This is an inbound call leg and media channel (so far) is open in reverse
direction only (application ringback). I'm afraid you have to answer the
call to be able to hear the fax tone.
T.
On Mon, Oct 5, 2009 at 2:32 PM, Michael Jerris m...@jerris.com wrote:
Fax tones are not
Yes, the stun thing was fixed after 1.4 I believe.
On Mon, Oct 5, 2009 at 6:43 AM, Vladimir Elizarov
xengelpubl...@gmail.com wrote:
How do I fix this? Upgrade fs to trunk?
--
-Rupa
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Hi,
Is is possible to override any of the setting specified in the conference
profile?
What I want to do is to have a default profile, and be able to modify
certain fields if necessary in the dialplan.
Alternatively, I would prefer to have a dynamic profile setting for the
conference to obtain
neat,
Here's some suggestions for your next ones. =p
Have them standing around the hologram trying to destroy the Death
Star(tm) that happens to look a lot like a giant 3d unix '*' character.
Then have one rebel say, wait!, why are we wasting our time... watch
this... and dial a number on his
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject
On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote:
Is it possible to treat a recorded voice as voice mail?
Assume that, I've recorded a conversation and I want this recorded
file to
be treated like voicemail. So, I could check it
Yes!
/b
On Oct 5, 2009, at 7:43 AM, Vladimir Elizarov wrote:
How do I fix this? Upgrade fs to trunk?
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On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
Is is possible to override any of the setting specified in the
conference profile?
Just the flags you can pass per user such as pin and mute
What I want to do is to have a default profile, and be able to
modify certain fields if
It seems many people are looking for ways to control gateways, resiliency of
termination, and limit on connections easily in FS. Here are some of the
thoughts I had, and I would like to hear what others think of this.
In tradition phone hardware you would define lines, put them into a group,
and
Espectacular!!.
FG
- Original Message -
From: Giovanni Maruzzelli gmar...@celliax.org
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, October 05, 2009 2:04 AM
Subject: Re: [Freeswitch-users] Youtube - FreeSWITCH Promo Video
On Mon, Oct 5, 2009 at 1:51 AM, Diego Viola
I've got a copy of Mike G's document that I'm reviewing. Like he said, it's
not 100% complete, however at first glance it looks like it would be a
perfect fit for a config example page. I'll get with Mike G shortly and
we'll have it up on the wiki in the next few days for everyone to review.
-MC
On Sun, Oct 4, 2009 at 4:09 PM, Nicolas Brenner nico...@medularis.comwrote:
Mike, how exactly should I format the file? I got the pcap file, how do I
convert it to text so that you can easily read it?
you can open it with wireshark, follow the TCP or UDP stream, then just copy
paste the text
Okay, I added the ; at the end of the sqlite3 select command and it just
returned to the sqlite prompt. No error was returned. Do you see
anything in my database (in the pastebin) that is incorrect? By the way,
the select command I put in the pastebin refers to the external config,
but the
Hi,
2009/10/5 Nagalenoj nagale...@gmail.com:
* Receive the events only for this uuid - I have done by registering
all events and filtering only for this uuid($uuid).
* If it is CHANNEL_ANSWER, originate a new call.
it's a filter in, not filter out :)
Now, How can I get the uuid of
Michael Collins m...@freeswitch.org said:
Confirmed: the Hicomm isn't sending anything at all in the SETUP message
except the usual stuff: dialed number, channel number, etc. Does the Hicomm
have any config parameters, like Caller ID presentation?
I believe it does, but I don't have access to
I can see how BLFs and Presence are managed, however I haven't found much
documentation on SLAs and BLAs. What is the RFC(s) that Freeswitch used to
implement SLAs and BLAs? Do they differ from BLFs?
Best Regards,
Jerry
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Because TRUNK is stable... its only fixes going in usually and if
things do break they don't stay broken for long.
Ask anyone our trunk is more table then most commercial products.
/b
On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote:
Ok. Brian, why fs no two branches of the stable and
On Mon, Oct 5, 2009 at 10:50 AM, russell.mosem...@cune.org wrote:
Michael Collins m...@freeswitch.org said:
Confirmed: the Hicomm isn't sending anything at all in the SETUP message
except the usual stuff: dialed number, channel number, etc. Does the
Hicomm
have any config parameters,
Nice script Anthony, that would be amazing to have on video ;)
On Mon, Oct 5, 2009 at 3:02 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
neat,
Here's some suggestions for your next ones. =p
Have them standing around the hologram trying to destroy the Death
Star(tm) that happens
Ladies and Gentlemen,
Thank you for calling in to the weekly FreeSWITCH conference call. Last
week's agenda was rather light, so if you have things that you would like to
have discussed please be sure to add them here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_09
Here are a few updates
Michael Collins m...@freeswitch.org said:
If you need proverbial ammo let me know. If he speaks Q931 then your
pastebin is the ultimate proof for him that the hicom is not sending any
caller ID info. In any case, I'm here if you need assistance.
Heh, I have the exact opposite problem. I don't
On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
http://wiki.freeswitch.org/wiki/Mod_limit
Of course I've read that. But it only affects the number of outgoing
calls (at least for gateways - chapter Using mod_limit with an
outbound gateway). But I would like to limit the
Direction doesn't matter, it uses realm's and a few other vars. Use
the same vars for both directions.
On Mon, Oct 5, 2009 at 14:57, Dmitry Bely dmitry.b...@gmail.com wrote:
On Mon, Oct 5, 2009 at 2:30 AM, Michael Jerris m...@jerris.com wrote:
http://wiki.freeswitch.org/wiki/Mod_limit
Of
Incoming Call-Limit http://www.howtonix.com/?p=89
Outbound Call-Limit http://www.howtonix.com/?p=86
On Tue, Oct 6, 2009 at 2:19 AM, SP spr...@gmail.com wrote:
Direction doesn't matter, it uses realm's and a few other vars. Use
the same vars for both directions.
On Mon, Oct 5, 2009 at
On Tue, Oct 6, 2009 at 12:19 AM, SP spr...@gmail.com wrote:
Direction doesn't matter, it uses realm's and a few other vars. Use
the same vars for both directions.
Unfortunately it does.
action application=limit data=$${domain} my_provider 1/
generates limit_exceeded for the second
On Tue, Oct 6, 2009 at 12:32 AM, M.Emran monem...@gmail.com wrote:
Incoming Call-Limit http://www.howtonix.com/?p=89
Outbound Call-Limit http://www.howtonix.com/?p=86
But what if I need to limit the total number of calls (in my case == 1)?
On Tue, Oct 6, 2009 at 2:19 AM, SP spr...@gmail.com
did you use the application limit on the inbound call? You'll need to
in order to account for it.
On Mon, Oct 5, 2009 at 15:39, Dmitry Bely dmitry.b...@gmail.com wrote:
On Tue, Oct 6, 2009 at 12:19 AM, SP spr...@gmail.com wrote:
Direction doesn't matter, it uses realm's and a few other vars.
Ok using windows xp x64 here. I download from the trunk as expected.
I fire up VS 2005 and I open the VS 2005 solution. Yes it does say
unsupported.
But I get two missing header files:
freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error
C1083: Cannot open include file:
Trying to achieve dynamic binding of user directory. In short: It's not
doing the authorization properly. I can use curl in the command line and
it works perfectly, specifying BASIC auth.. however with the freeswitch
module it returns HTTP 401.
So I've taken a close look at the network
Ok I found spandsp.h. It is a case of the project file being out of
date. No surprise. ptlib.h is still not found.
Ok using windows xp
Are you using something other than apache?
/b
On Oct 5, 2009, at 1:25 PM, Christian Damianidis wrote:
The inconsistent POST request sent by the module causes freeswitch
to hang for 1-2 minutes during start-up.
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Have you updated today?
/b
On Oct 5, 2009, at 3:49 PM, David Clark wrote:
Any idea what is up?
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voip codecs is fixed, ptlib I can't recall if we ever did full build
integration or if you needed to manually download the libraries, can
someone who has done mod_opal build on windows comment?
Mike
On Oct 5, 2009, at 5:14 PM, David Clark wrote:
Ok I found spandsp.h. It is a case of the
Haha classic !!!
Can't wait for the next installment in the series !!
J
On 06/10/2009, at 1:02, Anthony Minessale
anthony.miness...@gmail.com wrote:
neat,
Here's some suggestions for your next ones. =p
Have them standing around the hologram trying to destroy the Death
Star(tm) that
We are building our own in-house developed Teo phones. I also have
CounterPath's Bria Professional phone. For test purposes, I have one snom
phone and a couple Polycomm phones.
Jerry
-Original Message-
From: Brian West [mailto:br...@freeswitch.org]
Sent: Monday, October 05, 2009
The Revenge of the Sip
On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks jaybi...@gmail.com wrote:
Haha classic !!!
Can't wait for the next installment in the series !!
J
On 06/10/2009, at 1:02, Anthony Minessale anthony.miness...@gmail.com
wrote:
neat,
Here's some suggestions for your next
Sadly the budget of time and props can't afford such extravagance...
But OMG I'm still laughing...
I was e-mailing earlier off the list, and came up with some nice
names that could be put in the credits...
Like:
Anthony Minnessale -as- Obi-Code-Kenobi (But I do like Master Coda
from
NOW THAT might be worth doing!
Best Regards,
Karl J. Vesterling
k...@ken-ton.com
202-461-3231 x0
On Oct 5, 2009, at 6:35 PM, Giovanni Maruzzelli wrote:
The Revenge of the Sip
On Tue, Oct 6, 2009 at 12:16 AM, Jay Binks jaybi...@gmail.com wrote:
Haha classic !!!
Can't wait for the next
Openzap.conf.xml
configuration name=openzap.conf description=OpenZAP Configuration
settings
/settings
libpri_spans
Openzap.conf.xml
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=7/
/settings
libpri_spans
span name=PRI_1
param name=node value=cpe/
param
Thanks for your help Mike and Tihomir.
A little more playing around and I found that having action
application=fax_detect/ as well as action
application=tone_detect data=fax 1100 r +5000 transfer fax XML
features / do not work together.
Simply by removing fax_detect, the fax is detected
There was no sane way of doing that, so I ended up logging the trace from
the cli.
Here's the bad registration:
- http://pastebin.freeswitch.org/10605
Here's the good one:
- http://pastebin.freeswitch.org/10606
I am not sure if the second one is complete because for some reason the
first few
Michael,
Thanks for wiki-fying my text-only attempt at some user doc. I should have
done that for you. I actually have an updated version with many corrections
and the end tabs filled in. Can you point me to info on how I can amend and
append what you have kindly put up?
Mike G.
On Mon, Oct 5,
Ignore my previous email, the traces were incomplete, got much better (and
complete) traces with ngrep (found a suggestion from Brian in the list
archive, thanks!)
The gateway that registers:
- http://pastebin.freeswitch.org/10607
The one that doesn't:
- http://pastebin.freeswitch.org/10608
Further playing around and everything is working fine (even the
emailing). I'm not sure what I changed though to document it.
cheers
/M
On Mon, Oct 5, 2009 at 12:03 AM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi
I was hoping someone could help me to setup the fax detection /
try using the hanup hook
action application=set data=api_hangup_hook=system ${syscmd}/
On Mon, Oct 5, 2009 at 23:15, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Thanks for your help Mike and Tihomir.
A little more playing around and I found that having action
application=fax_detect/
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