Well, I'm just starting to use freeswitch, so my approach is probably for
from optimal. The point is I wanted that voicemail do not prompt for
passwords when the caller is a sip registered user, but I also wanted the
login requirement if the voicemail was called from some FXS port.
That lead me
Bear in mind that FS will accept both 2833 and INFO in any profile on an
inbound call. Param dtmf-type is valid only for outbound calls from the
profile.
Ognjen
On Thu, Dec 3, 2009 at 6:11 AM, Yehavi Bourvine
yehavi.bourv...@gmail.comwrote:
Hello,
I have Polycom phones which send only
Michael, Mark, Artem,
Thank you for your answers. I believe FS will suite our needs.
I've installed dedicated virtual machine (Centos) for FS and going to play
with it.
Thanks and regards,
Nikolay.
_
From: freeswitch-users-boun...@lists.freeswitch.org
Hello All,
I have a Digium TDM400P pci card with two FXO ports installed on my linux
box. I've connected an external telephone line to the first FXO port. But I
can't either make outgoing calls or receive incoming ones. Here are my
setups, please let me know where goes wrong.
*
/etc/zaptel.conf*
My freeswitch is using public IP. I setup a gateway registering to
voipstunt, and put it under internal profile. I tried to make call, and I
got no RTP back from the provider... Tried treating NAT issue by changing IP
address, internal IP, external IP. But no use, still getting no audio.
Finally,
I'm already using the latest wanpipe drivers (3.5.8), so yes.
François
On Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale wrote:
Did you
also update your wanpipe drivers and rebuild openzap again after you
upgraded it?
On Wed, Dec 2, 2009 at 2:12 AM, François Legal wrote:
So I
did
Thanks.
I did not succed to fincing the correct syntx with inline,
but the transfer application did work.
François
On Wed, 2 Dec 2009
12:21:54 -0600, Anthony Minessale wrote:
bind to the transfer app so
that it transfers the call to the vm extension that way the current
application is
Hello
In a thread back in March, I read that support for IAX in FreeSwitch is a
bit of kludge and since there's not much demand for it, chances are it won't
improve in the foreseeable future.
So I'd like some feedback from users who routinely connect to a FreeSwitch
server from various venues,
Hi,
Someone knows how to run periodically a JS script ?? The purpose is to write
to a db some global informations (Global Variables) about FS like every 5
minutes.
Thanks.
--
View this message in context:
http://old.nabble.com/How-to-run-a-JS-script-periodically-tp26625147p26625147.html
What about cron?
Create a cron entry like:
*/5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
But if you're just dumping global variables, you could easily retrieve them
directly from fs_cli without running an app and process the output however
you'd like:
Not sure about js, but in lua, you can use luarun to run a
long-running script like
loop
do sth.
sleep 5min
end
and also it can be set to start with freeswitch in lua.conf.xml
I guess you can also use jsrun to run js.
And, if you run every 5 min, why not use crontab?
fs_cli -x jsrun xx.js
First off, maybe this conversation is better suited to the dev list, and second
off, the current setup of where we do timers, where we poll, polling frequency
and architecture is the result of 4+ years of ongoing testing and optimization.
We have tried all different methods throughout.
You could also use the scheduler to run the jsrun command inside FreeSWITCH.
Mike
On Dec 3, 2009, at 8:31 AM, Rob Forman wrote:
What about cron?
Create a cron entry like:
*/5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
But if you're just dumping global variables,
http://wiki.freeswitch.org/wiki/Mod_commands#originate
Usage: originate call_url exten|application_name(app_args)
[dialplan] [context] [cid_name] [cid_num] [timeout_sec]
You can do this via shelling out to fs_cli like your example below or using esl
directly from php:
The behavior is probably expected, the unhelpful error is probably undesirable
but it would make a mess of the dial-plan to clean that up.
Mike
On Dec 2, 2009, at 9:19 PM, Lars Zeb wrote:
Is this reasonable given it was the only call in FreeSwitch at the time? How
can this situation be
Oh, it's not just one timer thread... Why, why is sql_thread keeps on
checking for messages every millisecond? Couldn't there be some signalling
implemented that will make the thread suspend on condition variable or a
socket/pipe in between?
#0 do_sleep (t=1000) at src/switch_time.c:109
#1
Michael Collins wrote:
On Wed, Dec 2, 2009 at 9:58 AM, Frank Carmickle fr...@carmickle.com
mailto:fr...@carmickle.com wrote:
On Wed, Dec 02, Otis wrote:
Snip...
Thanks.
I would like all extensions on say server A to be contactable
by those
on server
http://www.freeswitch.org/
On the right side. Join IRC
Just fill in a nickname and click JOIN IRC
-- W
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Btw, I have these popping up in my logs from time to time:
2009-12-03 09:42:06.035294 [DEBUG] switch_core_state_machine.c:314
(sofia/external/xx...@4.68.250.148) Running State Change CS_HANGUP
2009-12-03 09:42:06.035294 [CRIT] switch_time.c:473 Virtual Migration
Detected! Syncing Clock
In this
If doing this, I'd suggest checking for a global var to see if the script
should terminate itself. Otherwise, you'll have to bring down the whole
freeswitch to stop this script.
On Thu, Dec 3, 2009 at 7:28 AM, Seven Du dujinf...@gmail.com wrote:
Not sure about js, but in lua, you can use
Hello,
It was all ok until yesterday when i updated to svn 15761(last update before
that was about 4 days ago), Now I have this issue:
someone from the pstn (555) calls through my FXO gw (10.1.1.90) to ext
200
200 picks up, then 200 transfers the call to 205
call gets lost (it used to
Try trunk again
On Wed, Dec 2, 2009 at 5:33 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
I am not sure what you are sending over the socket but you have a queued
hangup being processed on line 640 of your pastebin
are you executing any commands with a ! character in it by any
The easiest place to do this is at the point you send the calls to FreeSWITCH.
How are the calls coming in?
Mike
On Dec 2, 2009, at 7:49 PM, Tim Uckun wrote:
I have read some of the archived emails about HA, loadbalancing,
failover etc and I am still a bit confused about how I could set up
You can turn up the full freeswitch debug or enable the siptrace on the sip
profile to get more information about this. This looks like a nat related
issue getting no response from the provider. A sip trace is probably the best
tool to figure this one out.
sofia profile internal siptrace
You may want to try this again with latest svn trunk. We have done quite a lot
of work to make nat support much better sense 1.0.4
Mike
p.s. I can't comment about version 1.4 due to broken flux capacitor.
On Dec 3, 2009, at 4:36 AM, Henry Huang wrote:
My freeswitch is using public IP. I
with the right clients, it nearly always works well. with a client that does
not support stun or at least rfc 3581 the results are much more sketchy and
require more hacks on the server side, but with enough effort can almost always
be made to work.
Mike
On Dec 3, 2009, at 7:17 AM, Fred-145
If you see that message then your machine/os/combo is having some problems
keeping up.
It's not the timer missing anything its the monotonic clock detecting a 1
second or more differential from what its next prediction for the time
should be. The best way to trigger this would be to suspend FS
what revision were you at prior to upgrade or can you narrow the range of
versions that broke this any more (or even better the exact version that broke
this). Please post this bug to http://jira.freeswitch.org.
Mike
On Dec 3, 2009, at 10:30 AM, Milena wrote:
Hello,
It was all ok until
This got fixed in version 15773, thank you very much
2009/12/3 Michael Jerris m...@jerris.com
what revision were you at prior to upgrade or can you narrow the range of
versions that broke this any more (or even better the exact version that
broke this). Please post this bug to
to late it's fixed now.
On Thu, Dec 3, 2009 at 10:21 AM, Michael Jerris m...@jerris.com wrote:
what revision were you at prior to upgrade or can you narrow the range of
versions that broke this any more (or even better the exact version that
broke this). Please post this bug to
you could check if the uuid is blank with an expression and playback an
audio warning that it's an invalid call.
On Thu, Dec 3, 2009 at 8:08 AM, Michael Jerris m...@jerris.com wrote:
The behavior is probably expected, the unhelpful error is probably
undesirable but it would make a mess of the
Have you checked out Redfone? While I haven't attempted to implement it yet,
my Redfone foneBridge2 claims to be able to handle load balancing and
failover between two Asterisk/Freeswitch servers.
-AF
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
I'm sorry if I sounded that way. Did mean to. :)
Yes, it's an embedded platform. It's an ALIX board with AMD Geode LX800 chip
and 256MB of RAM. http://www.pcengines.ch/alix2d3.htm
Line offset difference is due to some minor logging changes I made to see
who's allocating timers and how often.
Tony,
The call no longer hangs up but we still only get hold music in one
direction - if the callee places the caller on hold there is no music.
PB here:
http://pastebin.freeswitch.org/11378
This was on rev 15773.
Thanks again Tony!
On Thu, Dec 3, 2009 at 10:56 AM, Anthony Minessale
I don't think it's the board itself...
We have extensively tested FreeSwitch (no modifications) on that exact
board with AstLinux and have it running at multiple customer
locations.
No timing errors, no warnings or errors of any kind. Pretty standard
really just don't expect too much from the
I have copied 1001.xml in directory/default to a test user 2319.xm
changing or instances of 1001 in the file to 2319. I then went into
default.xml in directory folder and in one of the groups just mimicked
1001 details by changing 1001 to 2319.
Connecting to FS gives Forbidden message.
I know people with hardware out there in production based on arm11 and those
are pretty small processors, not sure how they compare to this. In regards to
the DISABLE_1MS_COND, try getting rid of that, it did increase performance on
the high end but may be better for you on the low end with
On Thu, Dec 3, 2009 at 9:46 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
I have copied 1001.xml in directory/default to a test user 2319.xm
changing or instances of 1001 in the file to 2319. I then went into
default.xml in directory folder and in one of the groups just mimicked
1001
What about the things I spent time suggesting in my last email?
Did you try them because I was actually curious if they made any impact.
On Thu, Dec 3, 2009 at 11:29 AM, eaf erandr-j...@usa.net wrote:
I'm sorry if I sounded that way. Did mean to. :)
Yes, it's an embedded platform. It's an
Yehavi,
There are a few variations of transmitting this information... If you
have already enabled a supplemental isdn service profile, try adding the
following to the PRI you are using:
(config-if)#isdn outgoing ie facility
(config-if)#iisdn outgoing ie extended-facility
(config-if)#isdn
I've sent deep-breath message to the dev list.
Just-in-case, here is a cross-post:
Hi there!
This message is a forward from user-mail-list.
I'm trying to fix such a problem:
FreSwithch compiled from SVN-trunk, date = 11/02/2009.
What is need: connect two users, initially one is on the
Hi guys,
i have a weird problem with my dialplans. For the moment, i have only 2
«usable» extensions. They were working #1 yesterday, but this morning i
realize i forgot to compile mod_python, so i go back into my source
folder and modify the modules.conf to uncomment mod_python, did a make
and
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and then put the user in that domain. Isn't the
default domain that of the server FS is running on ?
2319.xml is in
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and then put the user in that domain. Isn't the
default domain that of the server FS is running on ?
2319.xml is in
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and then put the user in that domain. Isn't the
default domain that of the server FS is running on ?
2319.xml is in
You mean, upgrading to the trunk and disabling RTP timers? Yes, I did. I
thought I responded back. Perhaps it didn't make through though, as I just
emailed back to the list instead of using nabble.com...
Anyway, upgrading to the trunk didn't change much, forcing SPA to 30ms went
w/o any effect
other than configuration/syntax problem it could be a simple character/file
encoding problem or may be improper file
permissions!
On Thu, Dec 3, 2009 at 11:29 PM, David Laperle dlape...@rsslex.com wrote:
Hi guys,
i have a weird problem with my dialplans. For the moment, i have only 2
From de console:
sofia profile external siptrace on
or
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Unfortunately this didn't help... Incoming calls from ISDN to SIP sends back
to ISDN the name of the destination, but not the other way around...
Thanks! __Yehavi:
2009/12/3 Metik freeswitch-users-l...@metik.com
Yehavi,
There are a few variations of transmitting this
Hi, Anthony and Mike,
With the latest version from SVN, I was able to remove the warning sample
rate not matching. But the remote RTP port was still changed after after
playing the vm greeting. See below,
2009-12-03 13:44:46.901216 [INFO] switch_rtp.c:1975 Auto Changing port from
On Thu, Dec 3, 2009 at 10:34 AM, Samuel Abekah-Mensah ab...@greatiam.comwrote:
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and then put the user in that domain. Isn't the
On Thu, Dec 3, 2009 at 10:29 AM, David Laperle dlape...@rsslex.com wrote:
Hi guys,
i have a weird problem with my dialplans. For the moment, i have only 2
«usable» extensions. They were working #1 yesterday, but this morning i
realize i forgot to compile mod_python, so i go back into my
I am trying to rewrite all my javascript scripts into Lua scripts. I have run
into the problem of core_db access. This can be achieved with Spidermonkey, but
apparently not with Lua. I have tried to get the binary for Lua (using apt-get)
but I get an error when I require the sqlite.so:
In latest trunk you can run the core db in your same mysql db.
other than that we would need to create an object from our lua module
similar to how it was done in js.
On Thu, Dec 3, 2009 at 2:05 PM, Jon Bruel j...@consiglia.dk wrote:
I am trying to rewrite all my javascript scripts into Lua
On Fri, Dec 4, 2009 at 4:59 AM, Michael Jerris m...@jerris.com wrote:
The easiest place to do this is at the point you send the calls to
FreeSWITCH. How are the calls coming in?
From an as of now unkown SIP trunk provider (we are still in
negotiations with a couple of companies).
Hi, I solved this issue. the reason is because of the different port number
between the the one in SDP and the one in real RTP stream. This is very nice
feature.
e
On 12/2/09, Erwin Davis davis.er...@gmail.com wrote:
Hi, I got a weird issue when I dialed an extension and listen to a recorded
On Fri, Dec 4, 2009 at 5:56 AM, Adam Ford li...@redbonez.net wrote:
Have you checked out Redfone? While I haven't attempted to implement it yet,
my Redfone foneBridge2 claims to be able to handle load balancing and
failover between two Asterisk/Freeswitch servers.
That would be my choice for
no,
I mean the one after that that you must have completely skipped with a
command line option to try and a param to set in the config. It somewhat
annoys me for taking the time to compose it now. I wrote all of the code
you are talking about myself and I was trying to give you some
Oh, you mean giving FS higher priority? Yeah, as a last resort I'll do that.
At the moment, I hope it won't be necessary as I can make those hyper
threads behave, and will see how that goes first. I see where your
implementation could be coming from. There is a queue of SQL queries in
sofia.c
Hi All,
I managed to borrow a SPA3102 with the latest firmware and have got it
to register using TLS, but I am still struggling with SRTP. Has
anyone managed to get SRTP working with the Linksys devices and if so,
can they direct me on how to do this.
I have generated a mini-certificates and
AFAIK, the Cisco/Linksys SPA series ATAs do not support SDES key
exchange to appropriately support SRTP and FreeSWITCH. They do their
proprietary Sipura key exchange only, not sure if Cisco plans on
upgrading the firmware to ever support SDES on the ATAs. They added
support for SDES to their IP
you can try xlite too.
On Thu, Dec 3, 2009 at 8:05 PM, Mark Campbell-Smith
mcampbellsm...@gmail.com wrote:
Hi All,
I managed to borrow a SPA3102 with the latest firmware and have got it
to register using TLS, but I am still struggling with SRTP. Has
anyone managed to get SRTP working with
Quote: Cisco/Linksys SPA series ATAs do not support SDES key exchange
to appropriately support SRTP and FreeSWITCH
I'll check with Cisco regarding their implementation then and try to
find out when/if they will support standard SRTP encryption.
So, back to my origianal question then. Are there
is it possible to run a javascript at the end of dialplan to generate cdrs?
because (mod_cdr_csv) is giving me hard time as it rotates Master file on
machine reboots or shutdown signals.
javascript or LUA for preferences?
thank you
___
FreeSWITCH-users
The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the
Grandstream and Mediatrix devices (although I've never tried either
one with FreeSWITCH).
I've personally never had any good experience with the Grandstream
ATAs. The Mediatrix ATAs are OK devices, but I've never personally
fs_cli looks like a good idea. I will try that. Many thanks Rob
Rob Forman wrote:
What about cron?
Create a cron entry like:
*/5 * * * * /usr/local/freeswitch/bin/fs_cli -x jsrun yourscript app()
But if you're just dumping global variables, you could easily retrieve
them
directly
Sigh,
You just took it up a notch in terms of disdain and sarcasm.
Why do people always only apologize sarcastically?
I asked you to try the -hp and turn off the monotonic clock just to gather
the results to help you. You completely missed it and just went on about
the threads. Please save
why not try mod_xml_cdr?
2009/12/4 Mouncif Benniane mounci...@gmail.com:
is it possible to run a javascript at the end of dialplan to generate cdrs?
because (mod_cdr_csv) is giving me hard time as it rotates Master file on
machine reboots or shutdown signals.
javascript or LUA for
You didn't say the exact error was. was 10.15.0.91 == aaa.bbb.ccc.ddd ?
2009/12/4 Samuel Abekah-Mensah ab...@greatiam.com:
Hi
Sorry .xm is a typo. I actually shut down the server and restarted. The
log says I need to create a domain of aaa.bbb.ccc.ddd (which is the
server IP address ) and
Do you sometimes/often get issues where SIP (UDP5060) or RTP (UDPwhatever)
ports fail being opened dynamically to work properly, or does SIP today
really work well over NAT firewalls?
Yes I get issues quite a bit with the server being behind a firewall.
IAX is much nicer in this
Tim Uckun timuc...@gmail.com wrote:
Yes I get issues quite a bit with the server being behind a firewall.
IAX is much nicer in this circumstance.
I just set up an IPv6 over IPv4 tunnel and nat goes away.
I have native IPv6 over ADSL now, as part of a trial that my ISP is
conducting. As a
so your registering to the provider to get the calls? If so, this gets tricky,
the provider likely does not support multiple registrations, even if they did
they probably send the call to both registered endpoints. With this big
unknown its not very easy to suggest a good solution. If I were
Hi there,
It it possible do something like:
extension name=rtp
condition field=destination_number expression=^2127776252$
action application=answer/
action application=playback data=rtp://192.563.41.246:27378/
/condition
/extension
Basically I have need to connect to incoming calls
Cheers Gabriel.. thanks for the information.
I'll look at the Mediatrix ATA's as an alternative - has anyone had
experience with those and TLS/SRTP?
On Fri, Dec 4, 2009 at 10:25 AM, Gabriel Kuri gk...@ieee.org wrote:
The ATAs I'm aware that claim support for TLS and SRTP w/ SDES are the
Hello,
I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they
should support TLS also (will try it next week; up to now I preffered to not
use TLS so I can sniff the traffic and debug things).
Regards, __Yehavi:
2009/12/4 Mark Campbell-Smith
I am taking my words back... The Cisco sends back what I want.
I got confused because the Nortel sends the name only for the connected PBX
and not for the othes ones (although it gets this infomation from them).
Thanks, __Yehavi:
2009/12/3 Yehavi Bourvine
Thanks Yehavi,
I would be very interested to find out how your test goes... can you
report back after you have tested it?
Thanks!
On Fri, Dec 4, 2009 at 3:38 PM, Yehavi Bourvine
yehavi.bourv...@gmail.com wrote:
Hello,
I have AudioCodes MP and Vega ATA adapters. They both support SRTP; they
Hi All,
This is a great list, thanks for all of the support!
For my IVR app running on FS, we we accept potentially long audio
recordings. Is it possible (in lua) to save recorded as mp3?
Thanks,
Neil
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Hi Neil,
If you have mod_shout loaded and use a .mp3 file as you recording
filename, it'll automagically encode it.
Cheers,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 4-Dec-09, at 1:28 AM, Neil Patel wrote:
Hi
Thanks all for your help. I got around this by running ./Setup and
installing wanpipe in TDM API mode (it says it's the default for FS). I then
uncommented the mod_openzap line in modules.conf when installing FS. Finally
I ran wancfg_fs which creates appropriate config files for you for your FS
Anthony, you advised me to use MySQL as the core database in order to access it
from Lua. I'm testing that as a work-around.
Still, I guess that your choice of SQLite as the default core database have
been taken from efficiency or stability considerations. Using MySQL through an
ODBC-connector
ODBC isnt as bad as its used to be. We use it with postgresql every
day and are very happy with it.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 4-Dec-09, at 1:40 AM, Jon Bruel wrote:
Anthony, you advised me to use
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