I was wondering if someone can tell me the command to turn on call
recording.
Ilan Perez
Webmaster
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Hi brothers,
Now that Freeswitch is there, the only problem remains is of choosing the
right language to use with it, as i hold the view, that you can get the best
from FS only by using the best and efficient call control language .
I am not sure about the following. Please clarify.
-For all
I have tried the following without success either:
'{origination_caller_id_number='+callee_number+',origination_caller_id_name=Klaus}
sofia/gateway/sip.gafachi.com/1'+caller_number, 15);
'{effective_caller_id_number='+caller_number+',effective_caller_id_name=Klaus}
yup
On Wed, 30 Jul 2008, David Knell wrote:
Hi Klaus,
You might want to try setting the variables before originating the call - the
'originate' will cause the call setup message to be sent.
--Dave
OK, Let me give you the full picture; here is what i do, but it still
doesn't work.
there is no space after the final } and the s in sofia
also please avoid the x = new Session(); x.originate() thing.
it's much easier to feed a dial string to the Constructor..
var dial_string =
{origination_caller_id_number=2121231234,origination_caller_id_name=fred}sofia/gateway/foo/2125551212
I have not benchmarked them but I will guess lua is the most efficient. I
really like perl though ;)
On Wed, Jul 30, 2008 at 5:46 AM, Ashutosh [EMAIL PROTECTED] wrote:
Hi brothers,
Now that Freeswitch is there, the only problem remains is of choosing the
right language to use with it, as
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
/b
On Jul 30, 2008, at 1:18 AM, Ilan Perez wrote:
I was wondering if someone can tell me the command to turn on call
recording.
Ilan Perez
Webmaster
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As long as you don't configure zt spans in openzap.conf that error is
really harmless and don't affect mod_openzap loading and the
configuration process for other types of spans (Pika or Wanpipe).
On Tue, Jul 29, 2008 at 8:38 AM, Hilda Farrad [EMAIL PROTECTED] wrote:
Hello,
We are trying to
Hi all,
A newbie question,
Is it possible to configure FreeSwitch to be using UDP only even fore the
signaling?
--
Tzury Bar Yochay
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mod_cdr is not a supported module and will not work with the current
api. You can take a look at mod_xml_cdr and mod_cdr_csv which are
supported and in tree,
Mike
On Jul 30, 2008, at 10:14 AM, Shehzad Pankhawala wrote:
Hi every body,
I have downloaded mod_cdr from svn as shown in
Doing so will cause your install to violate the RFC. But you asked so
here you go.
param name=bind-params value=transport=udp/ on the profile
should do what you want.
Remember the RFC says that TCP is NOT optional and is required to be
RFC compliant. ;)
/b
On Jul 30, 2008, at 8:53 AM,
On Jul 30, 2008, at 8:33 AM, Sangwoo Jin wrote:
Hi,
I I'm testing freeswitch with sipp.
My test configuration is the following:
Sipp(caller) - freeswitch - Sipp(callee)s
Testing loads are 5 CPS ~ 30 CPS and caller has hanged up a call as
soon as
receiving 200 OK.
In this testing
Hi,
thanks mike,
I want to get CDR into MySql.
organizing CSV and XML files for each user is some how combursome..
thanks
Shehzad
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You will probably need to solicit the time of the author of mod_cdr to get
the module up to date.
I am sure if you post a bounty, he will be able to get it operational again.
On Wed, Jul 30, 2008 at 9:29 AM, Shehzad Pankhawala
[EMAIL PROTECTED] wrote:
Hi,
thanks mike,
I want to get CDR into
Hi,
ok, Mike, i try to do that,
meanwhile,
dear Anthony, you please the same you told if you have contact with
author and then reply if any interesting come out.
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On Wed, Jul 30, 2008 at 8:29 PM, Shehzad Pankhawala
[EMAIL PROTECTED] wrote:
Hi,
thanks mike,
I want to get CDR into MySql.
organizing CSV and XML files for each user is some how combursome..
You can load the cdr from csv into mysql, LOAD DATA INFILE etc,
rehup freeswith to create a new
Is it possible to configure FS such that it will only handle SIP
Register messages internally and will pass all other messages via a SIP
trunk to an existing PBX. This would be very useful to gradually move
existing IP phones over to FS. The stage after that would be to define
that certain
Yes. That is possible. Its just a standard gateway on a sip profile.
/b
On Jul 30, 2008, at 11:04 AM, Simon Shaw wrote:
Is it possible to configure FS such that it will only handle SIP
Register messages internally and will pass all other messages via a
SIP trunk to an existing PBX. This
What if by violating an RFC one gets higher scalability and reliability -
would it harm anybody?
I only care about bottom line end of the day results. being lightweight
highly important in my case.
Thank you all for your prompt responsiveness -- freeSwitch community seems
to make a good feeling
Here's the agenda:
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_30
All are welcome.
-MC
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In comparison to audio handling, the load of the sip connections is
minimal and not worth considering. If reliability is one of your
metrics I would say TCP is more reliable than UDP.
Mike
On Jul 30, 2008, at 12:39 PM, Brian West wrote:
I honestly don't think it'll gain you much if any
Talking of trademarks.. Wandered over to the Gizmo Project website
earlier and noticed they've got a new business product compatible with
Freeswitch (sic). Now trademarks aside, it's FreeSWITCH isn't it? Kind
of like BlackBerry. I get frowned at by SWMBO when I don't spell it
right (or
Do I put this code.
extension name=ext-666
condition field=destination_number expression=^666$
action application=set data=RECORD_TITLE=Recording
${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}/
action application=set data=RECORD_COPYRIGHT=(c) 1980 Factory
You can put it in either place.. the extensions folder is next to the
last thing in line in the dialplan.
/b
On Jul 30, 2008, at 8:13 PM, Ilan Perez wrote:
In the dialplan default file or the actual extension file?
Also I want to apply to all extensions so I thought I would
Edit the first
I can't use JIRA because I'm waiting for confirmation of creating account.
Now, I tested again with the current svn trunk.
The lost memory of 1000 calls is 10 times more than 100 calls.
The following logs are my filtered valgrind's output.
I have removed the lost blocks log which may not have
Try running 500,000 calls at 100-200 cps with a duration of 1 second.
You'll always see something about lost memory on shut down and you
can't fully trust that output due to the usage of memory pools.
You'll see a running FreeSWITCH hit a high water mark and level off on
memory usage. I
As a suggested best practice I would recommend making as few changes
to the default config as possible. In the case of conf/dialplan/
extensions/ it is easy to put all of your custom extension definitions
in one or more distinct files. You get multiple benefits:
You don't have to sift through
I want to run freeswitch on low memory machine with 128M ram and no swap
partition.
On testing freeswitch on it, I have always seen the out of memory error.
Could I control the memory pools?
-Original Message-
From: [EMAIL PROTECTED] [mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of
mercutioviz in #freeswitch already solved it for me, a big thanks for you =D
The trick was to use ^9(\d{10})$. I need to get off and learn a bit of
regexp ;-)
Here is the log in case if someone wants to read it, maybe it will help some:
19:39 mercutioviz diegoviola: i've had a fair amount of
Hi everyone,
I have this in my default dialplan for dialing out:
extension name=outbound
condition field=destination_number expression=^9(\d+)$
action application=bridge data=sofia/gateway/teliax/$1/
/condition
/extension
It works fairly well... but when I try to dial other extensions
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