Lak,
Okay, it stood out right away: FS is trying u-law but Asterisk is trying
A-law. I'm not sure where the codec for openzap gets selected but you can
try modifying this line in vars.xml:
X-PRE-PROCESS cmd=set data=outbound_codec_prefs=PCMU,PCMA,GSM/
Try putting PCMA first. Also, I've never
Finally!!
Thank you Michael, I didn't know about status app. It satisfies all my
desires.
And again,
thanks for all the community for the strong support!
Artem
On Tue, Oct 13, 2009 at 10:48 PM, Michael Collins m...@freeswitch.orgwrote:
On Tue, Oct 13, 2009 at 8:32 AM, Artem Shiyanov
br...@freeswitch.org (Brian West) writes:
You shouldn't have to make clean usually ... doing so might break your
tree...
Why?
In any case on my mac (Leopard) neither make clean or distclean fully
cleans up afterwards. I would prefer to get a completely untouched
source tree after doing
I suppose he want to have a central dialplan and a dummy phone instead...
something as a MGCP phone behavior.
T.
On Tue, Oct 13, 2009 at 10:22 PM, Metik freeswitch-users-l...@metik.comwrote:
As evidenced by various DTMF interop issues (with RFC2833, inband,
etc) over the years, I would avoid
I fully agree that direct matching is much faster then pattern matching in
SQL.
One of my clients had same problem, he had around 12 million number prefixes
in a table and during each call an AGI script use to query that table to
find longest prefix match, but this use to take like 3-5 seconds
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 08:59 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
try sow start on h323 channel, there is a bug in faststart, i will fix it
later.
there are few things,
1. capability PCMU/PCMA needs to be inverted
2. when
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
may be needed later, i have no way
to test it on this time, i do it later.
Ok, will
Nice, I just converted this to Ruby/Sequel.
DB[:rates].first{{prefix=substring('number', 1, length(prefix))}}
Thanks for the help :).
On Wed, Oct 14, 2009 at 2:07 AM, TTNC - Adnan Barakat
techni...@ttnc.co.ukwrote:
Diego Viola wrote:
I'm using MySQL now but I will try PostgreSQL with the
DB[:rates].where(:prefix = substring('number', 1, length(prefix)).first
Rather.
On Wed, Oct 14, 2009 at 8:28 AM, Diego Viola diego.vi...@gmail.com wrote:
Nice, I just converted this to Ruby/Sequel.
DB[:rates].first{{prefix=substring('number', 1, length(prefix))}}
Thanks for the help :).
hi all:
i can get the member_id in the conference,but how can can i get the
number??
thx!
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
On Wed, Oct 14, 2009 at 10:16 AM, Tihomir Culjaga tculj...@gmail.comwrote:
2009/10/14 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-14 09:58 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
i need trace level 4 from mod_h323 and debug log of entire call, tcpdump
may be
hi simon, implementation is almost the same. here's my dialplan:
extension name=say_destination_info
condition field=destination_number expression=^0044(\d)$
action application=answer/
action application=playback data=misc/dialing.wav/
action application=say data=en name_spelled
Hello,
we have the following problem.
2 Fax machines are communicating via Freeswitch. One is externally
attached via a Telco who is able to handle T.38. The other one is
attached locally.
When 2 Fax machines start syncing each other, the Telco sends a SIP
UPDATE message with T.38 SDP, as it
Hi,
can any please let me know, how to add new extension(eg 1000.xml)
dynamically while running freeswitch?
while running freeswitch i have created new xml(eg: 1500.xml) file and i
have changed freeswitch.xml.fsxml still not working,
Thanks
Srinivasula Reddy K
lakshmanan ganapathy lakindi...@gmail.com said:
I was struggling to originate a call from the CLI.
originate openzap/1/1/9952248266 openzap/1/1/9952248266 says No route.
Aborting.
The syntax is incorrect.
originate openzap/1/1/9952248266 extension
For example,
openzap/1/1/9952248266 1234
I would still suggest using mod_lcr for this... If you have any real
volume, use postgresql with the prefix module.
It also supports IN lists, OR lists, optional quoting (since mysql is
retarded), and custom sql so you can interface with whatever stored
proc or deal with whatever database table
Hi,
I'm using event outboud socket(perl) in async mode.
Scenario where I face problem:
When a call comes to an extension(1000), my program will play some message
to the user, and get some DTMF.
I'll get the DTMF event(as I'm in async mode) and store the digits in a
variable.
When he presses #,
try putting
ESL::eslSetLogLevel(7);
at the top so you can get a trace of the esl data on the stdout from your
script
also put some debug code to confirm you are getting the digit
On Wed, Oct 14, 2009 at 9:10 AM, Nagalenoj H. nagale...@gmail.com wrote:
Hi,
I'm using event outboud socket(perl)
the thing we want to make working nicer is the following:
we want the main/basic phonenumber (123456) to be reachable, so that
the telephone rings. but we also want it to be expandable with
ddi-digits.
example: dial the 123456 to reach the company, dial the 123456 1 to
reach the support.
in the
So with overlap you will have to keep refusing the call until the right
amount of digits are dialed.
This mode would send 1 then 12 then 123 then 1234 then 12345 then 123456 as
they were being dialed.
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to
the heartbeat you were seeing that needs a uuid was the session heartbeat
which is a per-call heartbeat.
indeed, the system heartbeat is not customizable.
On Wed, Oct 14, 2009 at 1:31 AM, Artem Shiyanov shiya...@gmail.com wrote:
Finally!!
Thank you Michael, I didn't know about status app. It
Never _EVER_ change the freeswitch.xml.fsxml file directly. If you need
something dynamic you would have to implement directory using mod_xml_curl,
otherwise, you change the files on the conf/ tree and do a reloadxml on the
CLI.
JM
On Wed, Oct 14, 2009 at 8:58 AM, srinivasula reddy
Hi,
Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work
TLS between freeswitch and a softphone (phonerlite), but I have problem
with Audiocodes during the TLS authentication. I've loaded the certification
but it still doesn't work. Can I debug the tls in freeswitch?
Hello,
Turning on debug I got the following messages:
tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): events NEGOTIATING
tls_connect(0x942e8f0): TLS setup failed
(error:0001:lib(0):func(0):reason(1))
tport_close(0x942e8f0): tls/82.77.201.227:64571/sips
Szasz Szabolcs
On
Michael,
I upgraded to v15152. There is no apparent change in the behavior. When I
call into a number handled by the lua script, I may get a fast busy signal,
or a recording saying the connection cannot be made.
But FreeSwitch is removed from memory as a result. I cannot see from the log
Look at eavesdrop on the wiki.
JM
2009/10/14 Nikita Belov nbe...@abisoft.spb.ru
HI all,
I want to configure FS to make special conference call between three users
(A, B, C). In this conference C will hear A and B, but A will hear only B.
Can I make it using FS API commands? Does
What method are you trying to use?
If you're using the event socket, it's in several of the parameters.
For instance, with an add-member event, you could look at
Caller-Username, Caller-Caller-ID-Name, or Caller-Caller-ID-Number
depending on your specific needs. There's also a uuid available if
I did svn up, ./configure, make and make install.
Do you want me to do make current before proceeding, or just try to make it
stop with the current build from the above commands?
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On
Hi, all,
I read the article Performance testing and configurations on the wiki.
Here is a Recommended from wiki:
libsofia only handles 1 thread per profile, so if that is your bottle
neck use more profiles
How can I use more profiles and listen on port 5060? Is it possible?
I used sipp to tested
On Wed, Oct 14, 2009 at 10:40 AM, Ryanny Lin ryan...@gmail.com wrote:
Hi, all,
I read the article Performance testing and configurations on the wiki.
Here is a Recommended from wiki:
libsofia only handles 1 thread per profile, so if that is your bottle neck
use more profiles
How can I
On Wed, Oct 14, 2009 at 4:58 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
Hi,
can any please let me know, how to add new extension(eg 1000.xml)
dynamically while running freeswitch?
while running freeswitch i have created new xml(eg: 1500.xml) file and i
have changed
Everyone,
Please go vote here:
http://www.freeswitch.org/node/208
Be honest - we need real feedback. Don't click the first choice unless you
really have the money. :)
Thanks everyone! We appreciate having such a great community.
-MC
___
On Wed, Oct 14, 2009 at 10:09 AM, Lars Zeb larc...@yahoo.com wrote:
I did svn up, ./configure, make and make install.
Do you want me to do make current before proceeding, or just try to make it
stop with the current build from the above commands?
Lars,
The make current method was
m...@freeswitch.org (Michael Collins) writes:
Please go vote here:
http://www.freeswitch.org/node/208
Be honest - we need real feedback. Don't click the first choice unless you
really have the money. :)
There's a missing option IMO at least for SOHO people like
myself. That's better
On Wed, Oct 14, 2009 at 12:46 PM, Simon J Mudd sjm...@pobox.com wrote:
m...@freeswitch.org (Michael Collins) writes:
Please go vote here:
http://www.freeswitch.org/node/208
Be honest - we need real feedback. Don't click the first choice unless
you
really have the money. :)
There's
That's what asynchronous means do not block
try $con-setEventLock(true); with the socket in async mode, then your
requests will stack in order.
On Wed, Oct 14, 2009 at 12:50 AM, velusamy velu velu.techni...@gmail.comwrote:
Dear All,
I am implementing an IVR framework using Perl event
On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd sjm...@pobox.com wrote:
Hi again,
I'm trying to build up a soho dialplan to replicate an existing Asterisk
setup I want to replace.
I'm editing
/usr/local/freeswitch/conf/dialplan/default/00_mydomain.com.xml to add
some dialplan rules for
I am trying to make a test call to the FreeSwitch conference address,
sip:8...@conference.freeswitch.org on a Bria softphone. However, it fails
with a 404 Not found message.
I have struggled with this for a while. Is there something special I must do
on the Bria to make this happen correctly?
Hello, a few weeks ago I asked about a setup, and about using the
freeswitch package on pfSense. Since then I have received most of my
hardware and started setting up freeswitch.
First my setup
WWW -- 1st pfsense box -- private (192.168.2.xxx) -- pfsense w
freeswtich -- Test (192.168.3.xxx) --
Hi,
I just upgraded FS from 15094 to 15161 and after the upgrade I have the
following problem:
I use t38modem together with a SIP/ISDN gateway.
With the old release everything fax related works fine. I call the fax - I
hear a fax tone.
With the new release I call the fax - I don't hear a fax
On Wed, Oct 14, 2009 at 4:48 PM, Klaus Hochlehnert maili...@kh-dev.dewrote:
Hi,
I just upgraded FS from 15094 to 15161 and after the upgrade I have the
following problem:
I use t38modem together with a SIP/ISDN gateway.
With the old release everything fax related works fine. I call
Hey Orien,
All of these guys are more experienced than I, but I'll try to throw you a
line.
I'll have to check a thing or two out at work, but here's an observation or
two (I'll post again tomorrow morning after I get in).
The mult-homed example is set to work in a very specifc way: A
Mike,
Thank you for your quick response, this has been driving me nuts.
in my third attempt I tried to follow the mulit home example. I did set
the phones to register to the local lan IP. actually had them
registering, but it did help it in any way still the same problem.
After my post I
Sounds like bria is sending the invite via the proxy which is your
local FS box and NOT direct.
/b
On Oct 14, 2009, at 4:26 PM, Lars Zeb wrote:
I am trying to make a test call to the FreeSwitch conference
address, sip:8...@conference.freeswitch.org on a Bria softphone.
However, it fails
Why exactly are you doing proxy media?
/b
On Oct 14, 2009, at 4:56 PM, Klaus Hochlehnert wrote:
Does anyone have a solution for this?
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
Make current will NOT touch your configs and is designed to prevent
build skew.
/b
On Oct 14, 2009, at 5:14 PM, Klaus Hochlehnert wrote:
The conf and the scripts dir were untouched after that (which I
wanted because I didn’t want to reconfigure everything).
I'll get to you in the morning (where do you live?) and we'll see what's
what.
On Wed, Oct 14, 2009 at 10:30 PM, Orien Love or...@tx.rr.com wrote:
Mike,
Thank you for your quick response, this has been driving me nuts.
in my third attempt I tried to follow the mulit home example. I did
Everyone repeat after me:
make current is my friend
make current is my friend
make current is my friend
ALWAYS use make current unless you know more about FreeSWITCH than the
project's authors do.
-MC
On Oct 14, 2009, at 8:56 PM, Brian West br...@freeswitch.org wrote:
Make current will NOT
48 matches
Mail list logo
