Woody Dickson wrote:
> I am getting a strange problem in my dialplan.
>
> After doing "SET", I want to use it in the next condition field. But then
> the value is not being set properly.
When parsing the dial plan, FreeSWITCH tests all of the conditions, then
builds a linked list of actions to
On 3-Jun-09, at 2:32 AM, Jim Burke wrote:
> Fernando,
>
> Try setting 'inbound-late-negotiation' in your SIP Profile. This will
> allow the call to hit the dialplan where you can set proxy_media.
> This also assumes you have bypass_media set to false in your dialplan.
>
> Alternatively I beleive
Fernando,
Try setting 'inbound-late-negotiation' in your SIP Profile. This will
allow the call to hit the dialplan where you can set proxy_media.
This also assumes you have bypass_media set to false in your dialplan.
Alternatively I beleive you can set "inbound-proxy-media" in the SIP
Profile an
Hi,
FreeSWITCH decides what to execute first, the set application runs
later (look a few lines later, you'll see lines beginning with
EXECUTE, this is when it runs).
If you need to use variables you've set in the DP, you need to use the
transfer application to make it go back into routing
Hello,
I am getting a strange problem in my dialplan.
After doing "SET", I want to use it in the next condition field. But then
the value is not being set properly.
Could someone please tell me what is wrong?
Thanks,
Woody
Here is the dialplan:
Here is the FS log
I only change freeSWITCH\conf\dialplan\default.xml
and add user xml from 9773~9773 in freeSWITCH\conf\directory\default
2009/6/3 Michael Collins
> okay, you will need to use pasteb
I had to upgrade again svn revision to use this switch, but it works.
Thank you.
On Wed, Jun 3, 2009 at 1:12 AM, Keith Laaks wrote:
> Hi,
>
>
>
> Try starting using the -nonat switch.
>
>
>
> Best Regards
>
>
>
> Keith
>
>
>
> *From:* Muhammad Shahzad [mailto:shaherya...@googlemail.com]
> *S
okay, you will need to use pastebin and post your configuration. anything
you changed from the default config, especially in the dialplan, but also
vars.xml, sip_profiles, etc. Turn on debug level logging (F8 or "console
loglevel 7") and also do the SIP trace. Make a few test calls and capture
all
..I've update my FS by SVN..
but the User-Agent became to FreeSWITCH-mod_sofia/1.0.trunk-"UNKNOWN"
Is that right??
And the displayname is still "97730002"...
What i confused is why "97730002" ??
( I have users from 9773~9773,but when I call them from 97710006 ,
the dis
Has anyone succeeded in sending fax on Windows with the following command line?
originate sofia/gateway// &txfax(/path_to_fax_file)
No matter how I specify that path (I even copied the file into the
installation folder, C:\Program Files\FreeSWITCH), I always got "[ERR]
mod_fax.c:518 process_fax()
I was updated my FS and rebuilt it.
It works
But when User2(FS2) accept the call from User1(FS1) ,
User2(FS2) display "call established",but User1(FS1) still display
"calling".
Why??
(I think maybe that I need to do some setting on FS2.)
2009/6/2 Brad Tuan
> FS1'
Hi,
Assume the following sinario:
A call B, B att_xfer to C
if no answer on C for a long time, B can cancel the att_xfer by
pressing a key and talk to A again.
Is that possible?
Thank you.
7.
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Thanks a lot ! This's what i want.
2009/6/2 Michael Jerris
> /usr/local/freeswitch/bin/fs_cli -x reloadxml
>
> On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote:
>
> > How to reload xml without using console command line??
>
> ___
> Freeswitch-users mailing
On Tue, Jun 2, 2009 at 3:14 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> You have a good point.
>
> On the other hand, it's just another random day in SVN trunk. =D
> Most projects don't offer SVN trunk you can play spin-the-bottle with and
> land on something production-ready. B
You now have -nonat and the hang on start up with the nat detection
code is fixed now.
/b
On Jun 2, 2009, at 6:37 PM, Nik Middleton wrote:
As Anthony comments later, using SVN for updates is usually a risky
business for most projects. We all have been blessed by fantastic
coding to date
As Anthony comments later, using SVN for updates is usually a risky
business for most projects. We all have been blessed by fantastic
coding to date with this project, that has lulled us into believing that
using the latest snapshot will be OK. This is the first time that I've
had problems.
You have a good point.
On the other hand, it's just another random day in SVN trunk. =D
Most projects don't offer SVN trunk you can play spin-the-bottle with and
land on something production-ready. But we are pretty close most of the
time.
Here's my point of view:
That particular addition was a
We are working to correct it. So hold on ;)
/b
On Jun 2, 2009, at 4:53 PM, Lars Zeb wrote:
Brian,
I’m probably not the only one here, but much of what I have to do to
get Freeswitch going is new to me. Never installed or really worked
with Linux and scripting; just a little xml. It is ch
Brian,
I'm probably not the only one here, but much of what I have to do to get
Freeswitch going is new to me. Never installed or really worked with Linux
and scripting; just a little xml. It is challenging. Freeswitch is
interesting, appealing and challenging. The work your group has done is
a
Dear,
I can't solve my problem, i was try with:
and:
in freeswitch.xml
But receive the same log:
http://pastebin.freeswitch.org/9204
Anyone help me.
Fernando
--- On Mon, 6/1/09, FERNANDO VILLARROEL wrote:
> From: FERNANDO VILLARROEL
> Subject: Re: [Freeswitch-users] Passthru mode
>
You'll need to set the variable default_language
/b
On Jun 2, 2009, at 2:42 PM, Dome Charoenyost wrote:
Dear sir,
i create mod_say_th for Thai language. i found some problem
about sound-path.
I have config th.xml in conf/lang/th/
...
when i try
Freeswitch still looking sounf file i
Dear sir,
i create mod_say_th for Thai language. i found some problem
about sound-path.
I have config th.xml in conf/lang/th/
...
when i try
Freeswitch still looking sounf file in /sounds/en/us/callie (en sound-path)
Someone help me please
Dome C.
On Tue, Jun 2, 2009 at 11:37 AM, Rudolf Denert wrote:
> Hello,
>
> I'm not sure which one is it. But I think I send the digits in RFC 2833.
> All devicves are supporting RFC 2833.
>
Actually, that's very interesting. The start_dtmf app looks for in-band
DTMFs, so if you have to have that in orde
Hi,
Try starting using the -nonat switch.
Best Regards
Keith
From: Muhammad Shahzad [mailto:shaherya...@googlemail.com]
Sent: 02 June 2009 14:39
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch taking too long to start up
Yes, this
Hello,
I'm not sure which one is it. But I think I send the digits in RFC 2833. All
devicves are supporting RFC 2833.
The equipment:
(VoATM)
Allied Data Copperjet 1614 (ISDN)
Siemens Euroset 5020 phone
(MGCP)
Thomson SpeedTouch 780WL
Siemens Euroset 5020 phone
(SIP)
AVM Fritz!Box 7170
Siemens
On Tue, Jun 2, 2009 at 9:50 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> effective_* is *NOT EVER* valid in the dial string. they are settings of
> an existing session to control what caller id they pass.
>
>
FYI, I've updated the wiki to reflect this fact and to make it complete
On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert wrote:
> Hello,
>
> I have problems with sending DTMF. The freeswitch server recieves the
> digits from my telefon (SIP account) if I activate DTMF with:
>
> in my dialplan.
>
> The line
> <--param name="dtmf-type" value="rfc2833"/-->
> in sofia.conf
On Tue, Jun 2, 2009 at 6:41 AM, Brad Tuan wrote:
> How to update FreeSWITCH-mod_sofia/1.0.3-12163??
Your best bet is to use SVN trunk. It is the most stable version available,
even more stable than the latest 1.0.4pre8 release candidate. Back up your
entire freeswitch folder in case there's an
Hi Nik,
Yes and no, respectively.
Cheers --
Dave
- Original Message -
From: Nik Middleton
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, June 02, 2009 6:04 PM
Subject: [Freeswitch-users] Outbound socket question
Hi Guys,
I'm going some work with outbou
yes the socket remains open the duration of your connection.
and the uuid becomes optional at that point for sendmsg but may still come
into play for some FSAPI based commands
like uuid_getvar
On Tue, Jun 2, 2009 at 12:04 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> Hi Guys
Hi Guys,
I'm going some work with outbound socket, and have a few questions.
When each call is answered, I get a connection to my server socket.
Is it right to assume that this connection will remain for the duration
of the call?
If so, do I still need to pass the UUID when I call a
effective_* is *NOT EVER* valid in the dial string. they are settings of an
existing session to control what caller id they pass.
On Tue, Jun 2, 2009 at 11:40 AM, Rupa Schomaker wrote:
> I've fixed mod_lcr now. It should have been setting
> origination_caller_id_number not effective_caller_i
I've fixed mod_lcr now. It should have been setting
origination_caller_id_number not effective_caller_id_number.
On Tue, Jun 2, 2009 at 6:04 AM, Yuriy Ivzhenko wrote:
> Hello all.
>
> I have test the lcr overriding the Caller ID functionality.
>
> It return dialstring, that contains 'effective
I thought it is a problem, made a jira:
http://jira.freeswitch.org/browse/XML-2
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Hello,
I have problems with sending DTMF. The freeswitch server recieves the digits
from my telefon (SIP account) if I activate DTMF with:
in my dialplan.
The line
<--param name="dtmf-type" value="rfc2833"/-->
in sofia.conf.xml is active, too.
But than I have the problem that the other phone
On Jun 2, 2009, at 11:11 AM, Rex_Alex wrote:
Hello,
My public.xml configration is:
$1 will not exist in this case because your regular expression doesn't
capture anything. So replace $1 with your target number or use
^(123456)$
My default.xml configration is:
Hello,
My public.xml configration is:
My default.xml configration is:
When I am trying to call 123456 from my mobile no. Not able to see any
logging in FS console. Please assist where I am going wrong? Or do I requir
/usr/local/freeswitch/bin/fs_cli -x reloadxml
On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote:
> How to reload xml without using console command line??
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can you try in the square brackets using http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number
instead? I think effective will work if you set it but not in the
square brackets.
Mike
On Jun 2, 2009, at 7:04 AM, Yuriy Ivzhenko wrote:
Hello all.
I have test the lcr
UPDATE: I just looked at my firewall rules and looks like I scrapped all the
logic I was attempting and now I just port forward ANYTHING coming from the
IP of my provider gateway to my freeswitch box. Seems to be working fine.
On Tue, Jun 2, 2009 at 7:35 AM, Darin Weeks wrote:
> Have you setup
I would route the DID to the host and port 5080 if you are using the
default config, and make an extension in dialplan/public.xml to catch
the DID. Press F8 to see the debug information if not sure what DID
string should be matched.
On Jun 2, 2009, at 10:14 PM, Rex_Alex wrote:
Hello, I h
Have you setup an inbound gateway similar to this?
http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing
You also need to setup your dialplans for the inbound this page among
others has more info:
http://wiki.freeswitch.org/wiki/Quick_Start
Finally, your FIREWALL can be the most crit
sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M)
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Hi,
I always got 0 messages when using web. Finally I added some debug
information in the code and get this:
2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883
voicemail_api_function() port:[8080]
2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884
voicemail_api_function() uri:[/domains/192.168.1.
Hello,
I have installed FS, and tested outbound successfully. Now I am just trying
to do the inbound testing.
I got the Inbound DID. Please suggest me what changes should I make and
where?
Thanks,
Rex
--
View this message in context:
http://n2.nabble.com/Inbound-using-FS-tp3012286p3012286.htm
So I need to new a User(97719006) in directory\default ??
2009/6/2 Brian West
> I would update if I were you! :) Anyway something had to have changed it
> it won't magically do it.
> /b
>
> On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
>
>
>
How to update FreeSWITCH-mod_sofia/1.0.3-12163??
2009/6/2 Brian West
> I would update if I were you! :) Anyway something had to have changed it
> it won't magically do it.
> /b
>
> On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
>
> User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
>
>
>Brian
I would update if I were you! :) Anyway something had to have
changed it it won't magically do it.
/b
On Jun 2, 2009, at 8:28 AM, Brad Tuan wrote:
User-Agent: FreeSWITCH-mod_sofia/1.0.3-12163
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_
When send 100 Trying:
From: 97719006 ;tag=124388224932run00
But when send INVITE:
From: "Extension 97730002"
>;tag=U3QF8QUp1F3tQ
What happened between sending Trying and sending INVITE ??
send 556 b
Then it had to be passed in from the proxy.
/b
On Jun 2, 2009, at 8:11 AM, Brad Tuan wrote:
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
_
I don't have a 97719006 User in my FS.
It was passed from another sip proxy.
2009/6/2 Brian West
> Chances are that is what you set it to on the user. Verify the users
> settings in the directory.
> /b
>
> On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
>
> Why the Caller-Username is "97719006"
Chances are that is what you set it to on the user. Verify the users
settings in the directory.
/b
On Jun 2, 2009, at 6:28 AM, Brad Tuan wrote:
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name
is "Extension 97730002"??
Brian West
br...@freeswitch.org
-- Meet us at Clue
Its coming soon!
/b
On Jun 2, 2009, at 6:23 AM, David Knell wrote:
At the risk of evisceration (but with the intention of helping avoid
future brain dead build vs. idiot admin debates), I'd suggest that,
when significant new bits are added to the switch core, they should
default to being
Yes, this resolves the problem.
Thank you.
On Tue, Jun 2, 2009 at 5:27 PM, dujinfang wrote:
> Actually Brain mentioned that you can comment out switch_nat_init(); in
> switch_core.c
> On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote:
>
> As I understand it, a new ‘feature’ was added over the we
Hey Gents,
What is the Jira for this issue?
Dale,
Did you get any SIP traces. I am interested to have a look. You can
use NGREP if your system is Linux.
Regards,
Jim
On Thu, May 28, 2009 at 11:08 PM, Anthony Minessale
wrote:
> btw,
>
> 3 and 4 are not useful without 1
> we only debug issu
Hi,
I read the fifo section of the wiki and what is not clear are:
What is the meaning of fifo_orbit_announce?
What is the meaning of fifo_override_announce?
Is it possible to create a scenario where the caller can hear "Agent #123 is
going to attend to your call"?
Any help will be greatly appre
Steve Underwood wrote:
> Nobody has yet adapted Freeswitch for the Blackfin, and they probably
> won't. The Blackfin lacks an MMU and cannot run Linux - it runs uCLinux,
> which is a cut down Linux for machines of this type. It is quite
> troublesome to get memory management to behave sanely o
Actually Brain mentioned that you can comment out switch_nat_init();
in switch_core.c
On Jun 2, 2009, at 7:08 PM, Nik Middleton wrote:
As I understand it, a new ‘feature’ was added over the weekend to
resolve NAT. If you’re firewall is not allowing ICMP then FS waits
until it times out.
Why the Caller-Username is "97719006" but the Caller-Caller-ID-Name is
"Extension 97730002"??
EVENT DUMP:
Channel-State: [CS_ROUTING]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/sip:97730...@210.68.184.192:62101
;rinstance=16b8076934af7da9]
Unique-ID: [342618e3-84cd-494b-b745-760b60639
At the risk of evisceration (but with the intention of helping avoid future
brain dead build vs. idiot admin debates), I'd suggest that, when significant
new bits are added to the switch core, they should default to being off and
require a configuration option to turn them on. Such config optio
As I understand it, a new 'feature' was added over the weekend to
resolve NAT. If you're firewall is not allowing ICMP then FS waits
until it times out. At this time there is no option to disable it.
Regards
From: freeswitch-users-boun...@lists.freeswi
Hello all.
I have test the lcr overriding the Caller ID functionality.
It return dialstring, that contains 'effective_caller_id_number' variable.
But that variable has no effect.
I try test configuration
There is no result. (caller id number not changed)
But If I
Firstly, thanks for continuing to provide a superior piece of VOIP software.
I have a couple of small and unrequested suggestions:
1: have "make current" after the svn update do ` (./configure && make) || (./boostrap.sh && configure && make)` instead of what it does now which is presumably the
Hi,
I have just upgraded Freeswitch from svn revision 12432 to 13544. I am using
32bit CentOS 5.3, "make current" command completes successfully without any
errors but when i start freeswitch it take considerable time (roughly 90 -
120 seconds) to start up. During this time no message is display o
FS1's IP is 192.168.141.182 and FS2's IP is 192.168.141.187
These two FS are in the same LAN.
I just try to pass one sip call from one FS to another.
If it works, next is FS1( PublicIP ) to FS2( PublicIP ).
2009/6/2 Ken Rice
> Dumb question... Is 187 the local fs machine? You should have the
Dumb question... Is 187 the local fs machine? You should have the IP address
of the remote FS machine
From: Brad Tuan
Reply-To:
Date: Tue, 2 Jun 2009 17:00:29 +0800
To:
Subject: Re: [Freeswitch-users] How to pass a call from one FS to another FS
??
the same message
2009-06-02 16:49
the same message
2009-06-02 16:49:01 [INFO] mod_dialplan_xml.c:233 dialplan_hunt() Processing
100
1->97710001 in context default
2009-06-02 16:49:01 [WARNING] mod_sofia.c:2546 sofia_outgoing_channel()
Cannot l
ocate registered user 97710001 at 192.168.141@internal
2009-06-02 16:49:01 [
Brad Tuan wrote:
> I have tried
>
>
>
>
Change the % to an @ in the above.
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Brad Tuan wrote:
> As title
Write a script that connects to the event socket and issues an api reloadxml
command.
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UNS
As title
How to reload xml without using console command line??
-Brad
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