Re: [Freeswitch-users] Callback to the user in ESL
Hi, Any help or suggestion regarding my previous post. Especially
I also noted that, if I don't receive any events, especially
SERVER_DISCONNECTED, then the connection is in established state, but once
I receive the SERVER_DISCONNECTED event, the connection is closed. Is it
correct??
Here is the program by which I confirmed the above!
require ESL;
use IO::Socket::INET;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort = '8447',
Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $con;
my $type = user/;
for(;;) {
# wait for any client to connect, a new client will get connected
when a new call comes in the dialplan.
my $new_sock = $sock-accept();
# Do fork and let the parent to wait for more clients.
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
# Extract the host of the client.
my $host = $new_sock-sockhost();
# file descriptor for the socket.
my $fd = fileno($new_sock);
print Host name is $host\n;
# Create object for the ESL connection package to access the ESL
functions.
$con = new ESL::ESLconnection($fd);
# Gets the info about this channel.
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid,
$info-getHeader(caller-caller-id-number),
$info-getHeader(caller-destination-number);
# Answer the channel.
$con-execute(answer);
# Set the event lock to tell the FS to execute the instructions in
the given order.
$con-setEventLock(true);
# Play a file Get the personal number from the user.
$con-execute(playback,/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav);
$con-execute(hangup);
while($con-connected())
{
my $e=$con-recvEvent();
my $ename=$e-getHeader(Event-Name);
print $e-serialize();
print $ename\n;
print Connection exists\n;
sleep(1);
}
print
Bye\n--\n;
close($new_sock);
}
I've not registered for any events.
In the above program I'm receiving the SERVER_DISCONNECTED event.
Output when receiving event:
Host name is 192.168.1.222
Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
Event-Name: SERVER_DISCONNECTED
SERVER_DISCONNECTED
Connection exists
Bye
When I comment the recvEvent line, I got the following output.
Host name is 192.168.1.222
Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000 to 9097
Connection exists
Connection exists
Connection exists
Connection exists
Connection exists
On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
I've tried the following program as per the suggestion that you've told.
But it seems, no success. Once the connection is closed, I created a new
connection and I send originate to originate a new call. But it is not
working.
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort = '8447',
Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $make_call;
my $con;
my $type = user/;
for(;;) {
my $new_sock = $sock-accept();
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
my $host = $new_sock-sockhost();
my $fd = fileno($new_sock);
$con = new ESL::ESLconnection($fd);
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid,
$info-getHeader(caller-caller-id-number),
$info-getHeader(caller-destination-number);
$con-filter(Unique-Id, $uuid);
$con-events(plain, all);
$con-execute(answer);
$con-setEventLock(true);
my $number=$con-execute(read,2 4
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav
accnt_number 5000 #);
while($con-connected())
{
my $e=$con-recvEvent();
my $ename=$e-getHeader(Event-Name);
my $app=$e-getHeader(Application);
if($ename eq CHANNEL_EXECUTE_COMPLETE and $app eq read)
{
my $num=$e-getHeader(variable_accnt_number);
print $num\n;
$con-execute(hangup);
}
}
if(!$con-connected())
{
print Connection not exists\n;
$con = new ESL::ESLconnection($fd);
$con-api(originate,user/1000 park());
print
Re: [Freeswitch-users] Callback to the user in ESL
Your using outbound socket and you hangup the call, so it tells you it
is done with the server disconnected message and drops the
connection. This is all as expected. I guess I don't understand what
you think is the problem. This code is doing exactly what I would
expect it to do.
Mike
On Nov 26, 2009, at 4:27 AM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
Hi, Any help or suggestion regarding my previous post. Especially
I also noted that, if I don't receive any events, especially
SERVER_DISCONNECTED, then the connection is in established state,
but once I receive the SERVER_DISCONNECTED event, the connection
is closed. Is it correct??
Here is the program by which I confirmed the above!
require ESL;
use IO::Socket::INET;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort =
'8447', Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $con;
my $type = user/;
for(;;) {
# wait for any client to connect, a new client will get
connected when a new call comes in the dialplan.
my $new_sock = $sock-accept();
# Do fork and let the parent to wait for more clients.
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
# Extract the host of the client.
my $host = $new_sock-sockhost();
# file descriptor for the socket.
my $fd = fileno($new_sock);
print Host name is $host\n;
# Create object for the ESL connection package to access the
ESL functions.
$con = new ESL::ESLconnection($fd);
# Gets the info about this channel.
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid, $info-
getHeader(caller-caller-id-number), $info-getHeader(caller-
destination-number);
# Answer the channel.
$con-execute(answer);
# Set the event lock to tell the FS to execute the
instructions in the given order.
$con-setEventLock(true);
# Play a file Get the personal number from the user.
$con-execute(playback,/usr/local/freeswitch/sounds/en/us/
callie/ivr/8000/ivr-welcome_to_freeswitch.wav);
$con-execute(hangup);
while($con-connected())
{
my $e=$con-recvEvent();
my $ename=$e-getHeader(Event-Name);
print $e-serialize();
print $ename\n;
print Connection exists\n;
sleep(1);
}
print Bye
\n--
\n;
close($new_sock);
}
I've not registered for any events.
In the above program I'm receiving the SERVER_DISCONNECTED event.
Output when receiving event:
Host name is 192.168.1.222
Connected call 022b79f8-d8c0-11de-8d50-596fac84e59e, from 1000
to 9097
Event-Name: SERVER_DISCONNECTED
SERVER_DISCONNECTED
Connection exists
Bye
When I comment the recvEvent line, I got the following output.
Host name is 192.168.1.222
Connected call 65b7f64a-d8c0-11de-8d50-596fac84e59e, from 1000
to 9097
Connection exists
Connection exists
Connection exists
Connection exists
Connection exists
On Tue, Nov 24, 2009 at 5:57 PM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
I've tried the following program as per the suggestion that you've
told. But it seems, no success. Once the connection is closed, I
created a new connection and I send originate to originate a new
call. But it is not working.
require ESL;
use IO::Socket::INET;
use Data::Dumper;
my $ip = 192.168.1.222;
my $sock = new IO::Socket::INET ( LocalHost = $ip, LocalPort =
'8447', Proto = 'tcp', Listen = 2, Reuse = 1 );
die Could not create socket: $!\n unless $sock;
my $make_call;
my $con;
my $type = user/;
for(;;) {
my $new_sock = $sock-accept();
my $pid = fork();
if ($pid) {
close($new_sock);
next;
}
my $host = $new_sock-sockhost();
my $fd = fileno($new_sock);
$con = new ESL::ESLconnection($fd);
my $info = $con-getInfo();
my $uuid = $info-getHeader(unique-id);
printf Connected call %s, from %s to %s\n, $uuid, $info-
getHeader(caller-caller-id-number), $info-getHeader(caller-
destination-number);
$con-filter(Unique-Id, $uuid);
$con-events(plain, all);
$con-execute(answer);
$con-setEventLock(true);
my $number=$con-execute(read,2 4 /usr/local/freeswitch/
sounds/en/us/callie/conference/8000/conf-pin.wav accnt_number 5000
#);
while($con-connected())
{
my $e=$con-recvEvent();
my $ename=$e-getHeader(Event-Name);
my $app=$e-getHeader(Application);
if($ename eq CHANNEL_EXECUTE_COMPLETE and $app eq
read)
Re: [Freeswitch-users] Requesting testing.
Hi Checked out svn checkout y'day. I am in the UK. Installed . Installed on Fedora 11 i386 box. : bootstrap.sh configue --without-libcurl make make install On startup only errors: PMP I'm not behind a NAT so OK Stacksize registered as too high and advised to use the -waste switch. Other than the stack thing all quiet on the new front, Sir regards Michael Jerris wrote: I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. Thanks Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Help Freeswitch with Voipuser Gateway
This is resolved. I could someone to call my VOIPUSER number and call
transferred to my designated extension. I could not get this to work
from my network ie calling from one of my extensions and setting that
the call be -rerouted to another extension.
All OK now.
Thanks folks
Otis wrote:
Has anyone got any suggestion how I can set up a gateway to receive
incoming call on extension 1001 please.
Any generic conf file will do. my username with my gateway is s=say
qwerty and password ytrewq
I have used the intruction from the link below without success.
Thanks.
Otis wrote:
Hello
Could anyone point out what I have missed please ?
At the moment I configured a gateway voipuser as described here
http://www.onlinesolution.co.nz/viewtopic.php?p=119 :
Any suggestion as to what path I can take will be highly welcome
Thanks
.
Sam Abekah-Mensah wrote:
div class=moz-text-flowed style=font-family: -moz-fixedHi Michael
Thanks
I had set it to send incoming calls to extension 1001. This is in
the file abeka.xml in /usr/local/freeswitch/conf/dialplan/public
directory.
The contents are :
extension name=inbound-*userna...@sip.voipuser.org]
condition field=destination_number expression=08444846450
action application=transfer data=1001 XML default/
/condition
/extension
Is there
anything wrong with this please ?
Thanks
Michal Bielicki wrote:
Am 21.11.2009 um 23:15 schrieb Sam Abekah-Mensah:
I need help as I cannot receive calls through VOIPUSER. This is a
learning setup Attached are my conf files. What is wrong with them
? When I dial from a landline I get a continuous beep.
Attached are my gateway and the conf file to transfer. Sopfia
Status is my screen message. I can see a FAIL and cannot make head
or tail of all that message. Hopefully anyone using voipuser or in
fact any of you clever folks can make sense of this.
Thanks for your time.
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:2811 Activate
Buggy RFC2833 Mode!
2009-11-21 22:07:15.642652 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-11-21 22:07:15.650807 [DEBUG] sofia_glue.c:3071 Audio Codec
Compare [PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-11-21 22:07:15.672560 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/external/nob...@213.166.5.133 PCMA/8000 20 ms 160 samples
2009-11-21 22:07:15.676936 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf
payload to 101
2009-11-21 22:07:15.676936 [DEBUG] sofia.c:3455
(sofia/external/nob...@213.166.5.133) State Change CS_NEW - CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_INIT
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:83
sofia/external/nob...@213.166.5.133 SOFIA INIT
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:111
(sofia/external/nob...@213.166.5.133) State Change CS_INIT -
CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/nob...@213.166.5.133 [BREAK]
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:481
(sofia/external/nob...@213.166.5.133) State INIT going to sleep
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:398
(sofia/external/nob...@213.166.5.133) Running State Change CS_ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:484
(sofia/external/nob...@213.166.5.133) State ROUTING
2009-11-21 22:07:15.676936 [DEBUG] mod_sofia.c:130
sofia/external/nob...@213.166.5.133 SOFIA ROUTING
2009-11-21 22:07:15.676936 [DEBUG] switch_core_state_machine.c:78
sofia/external/nob...@213.166.5.133 Standard ROUTING
2009-11-21 22:07:15.696693 [INFO] mod_dialplan_xml.c:315
Processing anonymous-abeka in context public
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-unloop] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (PASS)
[unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-outside_call] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Absolute Condition
[outside_call]
Dialplan: sofia/external/nob...@213.166.5.133 Action
set(outside_call=true)
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-call_debug] continue=true
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/nob...@213.166.5.133 parsing
[public-public_extensions] continue=false
Dialplan: sofia/external/nob...@213.166.5.133 Regex (FAIL)
[public_extensions]
[Freeswitch-users] Problems with nat
I got a freeswitch that is behind nat and got three profiles.
External (all calls are going through a proxy):
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
Internal (phones on the same lan as FS)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=local-network-acl value=localnet.auto/
Wan (phones that are not in the same LAN, connecting from internet)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
The problem is that phones registered on the internal profile
gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in
all profiles.
Log from a call:
http://pastebin.freeswitch.org/11303
I'm running freeswitch with the -nonat option.
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[Freeswitch-users] Re-routing calls to PSTN
Hi folks Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Thanks. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] B Leg Account Code on Fail Over dialing
Hi Everyone,
How do I get the outbound sofia SIP route that the call took into a CDR
when using fail over dialing with the 'bridge' application?
...I have tried numerous approaches with no luck, this last attempt pasted
below did not work either:
dialString =
{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s
% (numberToDial, numberToDial)
session.execute(bridge,dialString)
I am using mod_python and this line is in the Python module called in the
dialplan.
Thanks!
Michael
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Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: how about trying Fusionpbx.com ( GUI) -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On *Mon, 11/23/09, ram /talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users@lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis ab...@greatiam.com /mc/compose?to=ab...@greatiam.com wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' how about trying Fusionpbx.com ( GUI) Ram -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to run IVR application
Thank you very much MC . Its working :) . I started loving FS ;) On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins m...@freeswitch.org wrote: On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang lei.tl...@gmail.com wrote: you can do this in follow steps: 1.edit default.xml diaplan config file in your fs config directory(FS/conf/dialplan/default.xml), and section extension name=ivr_demo2 condition field=destination_number expression=^\*114$ action application=lua data=../ivr/test.lua/ /condition /extension 2. edit your ivr script, your can refer to http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in lua. 3. connect your sip phone to fs, and dial 114, this will launch your ivr application You can also do IVRs with static XML. I recommend you try out the demo IVR by dialing 5000. Now go look at the two main files that we used to build that IVR: conf/autoload_configs/ivr.conf.xml (menu structure) conf/lang/en/demo/demo-ivr.xml (phrase macros) it's overwhelming at first, however once you get the hang of it you'll appreciate how powerful it is. The wiki and the sample XML config files have lots of information so be sure to read as much as you can and try things. You can't break anything. :) -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] B Leg Account Code on Fail Over dialing
You need to use brackets [] not braces {} for per-leg variables.
On Thu, Nov 26, 2009 at 6:08 AM, Michael Toop
micha...@voxcore.voxtelecom.co.za wrote:
Hi Everyone,
How do I get the outbound sofia SIP route that the call took into a CDR
when using fail over dialing with the 'bridge' application?
...I have tried numerous approaches with no luck, this last attempt pasted
below did not work either:
dialString =
{provider=providerB}sofia/gateway/sipB/%s|{provider=providerC}sofia/gateway/sipC/%s
% (numberToDial, numberToDial)
session.execute(bridge,dialString)
I am using mod_python and this line is in the Python module called in the
dialplan.
Thanks!
Michael
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--
-Rupa
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Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX
Fedora and Centos installation instructions are very similar. You should be able to compile on Fedora without any problems that I'm aware of. Regards, Nik On Thu, Nov 26, 2009 at 06:24, Otis ab...@greatiam.com wrote: Thanks. I will try it . I am on Fedora 11 Mark Crane wrote: how about trying Fusionpbx.com ( GUI) -Ram I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was ready to release now but decided to do one more release candidate just to be sure. This should be the last release candidate before the release of version 1.0. The final release may be by the end of the week as long as no major issues are found. http://fusionpbx.com --- On *Mon, 11/23/09, ram /talk2...@gmail.com/* wrote: From: ram talk2...@gmail.com Subject: Re: [Freeswitch-users] GUI for Freeswitch -- wikiPBX To: freeswitch-users@lists.freeswitch.org Date: Monday, November 23, 2009, 10:54 PM On Mon, Nov 23, 2009 at 10:37 AM, Otis ab...@greatiam.com /mc/compose?to=ab...@greatiam.com wrote: Thanks. I have to get a centos box I guess. Much appreciated Samuel 'Otis' how about trying Fusionpbx.com ( GUI) Ram -Inline Attachment Follows- ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org /mc/compose?to=freeswitch-us...@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with nat
Are you doing this all on a linux box thats acting as your router
too? If not you don't need two profiles... you also don't need to set
the local-network-acl on ANY profile that isn't do anything with nat.
/b
On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote:
I got a freeswitch that is behind nat and got three profiles.
External (all calls are going through a proxy):
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
Internal (phones on the same lan as FS)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=local-network-acl value=localnet.auto/
Wan (phones that are not in the same LAN, connecting from internet)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
The problem is that phones registered on the internal profile gets
RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine
in all profiles.
Log from a call:
http://pastebin.freeswitch.org/11303
I'm running freeswitch with the -nonat option.
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[Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.
Is there any way to build a dial plan so that when an extension calls itself the call is automatically put to that users voice mail? Example, extension 1001 calling 1001 and is sent to voice mail (to receive messages). I know that there is a * code to get to voice mail, I cannot recall which one right now but my phones want to dial their extension to get to voice mail.I can modify the voice mail button but this works only for the first line registered at that phone. Any help is appreciated. Orien ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with nat
It's a windowsserver which is behind a router.
Which profile should local-network-acl be specified on?
When I bridge calls to the outside world, should I use
sofia/internal/phoneNumber@gateway or
sofia/external/phoneNumber@gateway?
On Thu, Nov 26, 2009 at 4:42 PM, Brian West br...@freeswitch.org wrote:
Are you doing this all on a linux box thats acting as your router too? If
not you don't need two profiles... you also don't need to set the
local-network-acl on ANY profile that isn't do anything with nat.
/b
On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote:
I got a freeswitch that is behind nat and got three profiles.
External (all calls are going through a proxy):
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
Internal (phones on the same lan as FS)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=local-network-acl value=localnet.auto/
Wan (phones that are not in the same LAN, connecting from internet)
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip value=$${external_sip_ip}/
param name=local-network-acl value=localnet.auto/
The problem is that phones registered on the internal profile
gets RECOVERY_ON_TIMER_EXPIRE error after 40-60 seconds. Audio works fine in
all profiles.
Log from a call:
http://pastebin.freeswitch.org/11303
I'm running freeswitch with the -nonat option.
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Re: [Freeswitch-users] XML config file parsing
On Tue, Nov 24, 2009 at 5:48 AM, Eliot Gable egable+freeswi...@gmail.com wrote: Or, you can use something like Smarty to cache your generated XML on your web server and only invalidate those cached results when you change something that will impact them. That sounds like a pretty sane way to go bout it. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with nat
In this case you should not need 2 profiles either. On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: It's a windowsserver which is behind a router. Which profile should local-network-acl be specified on? When I bridge calls to the outside world, should I use sofia/internal/phoneNumber@gateway or sofia/external/phoneNumber@gateway? On Thu, Nov 26, 2009 at 4:42 PM, Brian West br...@freeswitch.org wrote: Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.
Of course. Please read through the default configs and the getting started guide and xml dialplan information on the wiki. Mike On Nov 26, 2009, at 12:38 PM, Orien Love wrote: Is there any way to build a dial plan so that when an extension calls itself the call is automatically put to that users voice mail? Example, extension 1001 calling 1001 and is sent to voice mail (to receive messages). I know that there is a * code to get to voice mail, I cannot recall which one right now but my phones want to dial their extension to get to voice mail.I can modify the voice mail button but this works only for the first line registered at that phone. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445) On Nov 26, 2009, at 2:22 PM, Frank @ Impact wrote: “GREAT SCOTT!!! Cannot execute batched statements! If you are using mysql, make sure you are using MYODBC 3.51.18 or higher and enable FLAG_MULTI_STATEMENTS” I realize a bit off of list topic… But I do have mysql 3.51.18 and higher but for the life of me , I cannot seem to get the DSN config setup so that the odbc connector seems to tell FS that it can do multi statements. Anyone have any insight on how and where to set this flag? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.
Of course. Please read through the default configs and the getting started
guide and xml dialplan information on the wiki.
Mike
This is of interest to me as well, would that be something like this:
extension name=ext_100_vm
condition field=caller_id_number expression=^100$/
condition field=destination_number expression=^100$
action application=answer/
action application=voicemail data=check $${voicemail_profile}
$${domain} 100/
/condition
/extension
Could anyone versed in xml and variables comment on this so it generically
checked
if the extension dialed was of your extension length, like ^(\d{3})$ then if it
matched
your caller_id_number go into the action so you could leave it as $1, not 100
in my case?
That way you could only have one of these plans work for all extensions.
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Re: [Freeswitch-users] odbc FLAG_MULTI_STATMENTS
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote: http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html MySQL Connector/ODBC now supports batched statements. In order to enable cached statement support you must switch enable the batched statement option (FLAG_MULTI_STATEMENTS, 67108864, or Allow multiple statements within a GUI configuration). Be aware that batched statements create an increased chance of SQL injection attacks and you must ensure that your application protects against this scenario. (Bug#7445 http://bugs.mysql.com/7445) so, is this the right patch ? http://bugs.mysql.com/file.php?id=6994 T. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.
condition field=destination_number expression=^${caller_id_number}$
On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Of course. Please read through the default configs and the getting
started guide and xml dialplan information on the wiki.
Mike
This is of interest to me as well, would that be something like this:
extension name=ext_100_vm
condition field=caller_id_number expression=^100$/
condition field=destination_number expression=^100$
action application=answer/
action application=voicemail data=check $${voicemail_profile}
$${domain} 100/
/condition
/extension
Could anyone versed in xml and variables comment on this so it generically
checked
if the extension dialed was of your extension length, like ^(\d{3})$ then
if it matched
your caller_id_number go into the action so you could leave it as $1, not
100 in my case?
That way you could only have one of these plans work for all extensions.
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Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.
condition field=destination_number expression=^${caller_id_number}$
Of course:) Thank you!
jlc
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Re: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called.
freeswitch list wrote:
condition field=destination_number expression=^${caller_id_number}$
I knew this day would come. After the accumulation of all of the knowledge from
the list members, the list has finally achieved sentience and is now answering
questions by itself. :-)
--
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
I tried now with phones directly attached to the freeswitch (without an OpenSIPS in between). I also added the alias. But the behaviour is as before: No notify message from freeswitch, neither after register nor after a voicemail is recorded. Best regards Peter Brian West schrieb: Yes an alias will be required for every domain you run on the profile so it can find it. /b On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote: Try an alias on the sip profile. Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called.
LOL thats funny.
freeswitch, what is the meaning of life?
On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote:
freeswitch list wrote:
condition field=destination_number expression=^$
{caller_id_number}$
I knew this day would come. After the accumulation of all of the
knowledge from the list members, the list has finally achieved
sentience and is now answering questions by itself. :-)
--
Russell Mosemann
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Re: [Freeswitch-users] Re-routing calls to PSTN
On 11/26/2009 6:02 AM, Otis wrote: Can I get FS to re-route incoming-calls to PSTN. If this has been raised before could someone direct me to URL or link please Since I don't have a hard line, I do something like: include extension name=2800 condition field=destination_number expression=^2800$ action application=bridge data=sofia/gateway/YOURPROVIDER/18005551212/ /condition /extension /include -- Andrew Thompson ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Business/holiday hours routing
On Wed, Nov 25, 2009 at 11:21:25AM -0700, Adam Ford wrote: Awesome, thanks Andrew, I will have to keep an eye out for that patch. Here's my patch in its (probably) final form. http://eagle.bsd.st/~andrew/mweek2.diff It includes a usage example that covers all but 2 of the US federal holidays (memorial day is a real toughie). I'm just waiting on Tony to green light it for commit. If the patch looks like a mess in your browser, blame the XML :) Andrew ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dialplan rule to send the caller to voicemail when same extension is called.
Weird don't know how that got set to freeswitch list.
On Thu, Nov 26, 2009 at 9:27 PM, Rob Forman rob4manh...@gmail.com wrote:
LOL thats funny.
freeswitch, what is the meaning of life?
On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote:
freeswitch list wrote:
condition field=destination_number expression=^$
{caller_id_number}$
I knew this day would come. After the accumulation of all of the
knowledge from the list members, the list has finally achieved
sentience and is now answering questions by itself. :-)
--
Russell Mosemann
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Re: [Freeswitch-users] dialplan rule to send the caller to voice mail when same extension is called.(Working)
Thanks for all the help, here is what I put in the dialplan, I tested
this and it is working for me.
!-- User calling Self Goes To voicemail --
extension name=usertoselfvmain
condition field=destination_number
expression=^${caller_id_number}$
action application=answer/
action application=sleep data=1000/
action application=voicemail data=check default
${domain_name} ${sip_from_user}/
/condition
/extension
this was added just before the line extension name=Local_Extension
Orien Love.
Still learning, but getting there with help from all the great people on
this list :)
Subject:
Re: [Freeswitch-users] dialplan rule to send the caller to voice mail
when same extension is called.
From:
freeswitch list e...@chabotel.com
Date:
Thu, 26 Nov 2009 17:26:15 -0500
To:
freeswitch-users@lists.freeswitch.org
condition field=destination_number expression=^${caller_id_number}$
On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale
jcas...@activenetwerx.com mailto:jcas...@activenetwerx.com wrote:
Of course. Please read through the default configs and the
getting started guide and xml dialplan information on the wiki.
Mike
This is of interest to me as well, would that be something like this:
extension name=ext_100_vm
condition field=caller_id_number expression=^100$/
condition field=destination_number expression=^100$
action application=answer/
action application=voicemail data=check
$${voicemail_profile} $${domain} 100/
/condition
/extension
Could anyone versed in xml and variables comment on this so it
generically checked
if the extension dialed was of your extension length, like
^(\d{3})$ then if it matched
your caller_id_number go into the action so you could leave it as
$1, not 100 in my case?
That way you could only have one of these plans work for all
extensions.
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Re: [Freeswitch-users] How to run IVR application
Hi MC, I have created won sample application yesterday, It was working fine. Today, I checked that my local ip has changed. so, I changed the domain(IP) name in sip-account settings in my x-lite configuration. After that x-lite is not able to register with FS. I am getting error like Registration error 405 : Method not Allowed . Could you please tell me ,why its happening ? Thanks in advance, Venkat. On Thu, Nov 26, 2009 at 6:08 PM, ovvenkat ovvenkate...@gmail.com wrote: Thank you very much MC . Its working :) . I started loving FS ;) On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins m...@freeswitch.orgwrote: On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang lei.tl...@gmail.com wrote: you can do this in follow steps: 1.edit default.xml diaplan config file in your fs config directory(FS/conf/dialplan/default.xml), and section extension name=ivr_demo2 condition field=destination_number expression=^\*114$ action application=lua data=../ivr/test.lua/ /condition /extension 2. edit your ivr script, your can refer to http://wiki.freeswitch.org/wiki/Mod_lua for how to write ivr script in lua. 3. connect your sip phone to fs, and dial 114, this will launch your ivr application You can also do IVRs with static XML. I recommend you try out the demo IVR by dialing 5000. Now go look at the two main files that we used to build that IVR: conf/autoload_configs/ivr.conf.xml (menu structure) conf/lang/en/demo/demo-ivr.xml (phrase macros) it's overwhelming at first, however once you get the hang of it you'll appreciate how powerful it is. The wiki and the sample XML config files have lots of information so be sure to read as much as you can and try things. You can't break anything. :) -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problems with nat
Ok. I've been running this system since FS was a beta. It stopped working after a update. I'll switch to a single profile. What NAT settings should it have? I really want to get rid of the RECOVERY_ON_TIMER_EXPIRE error. On Thu, Nov 26, 2009 at 9:44 PM, Michael Jerris m...@jerris.com wrote: In this case you should not need 2 profiles either. On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote: It's a windowsserver which is behind a router. Which profile should local-network-acl be specified on? When I bridge calls to the outside world, should I use sofia/internal/phoneNumber@gateway or sofia/external/phoneNumber@gateway? On Thu, Nov 26, 2009 at 4:42 PM, Brian West br...@freeswitch.org wrote: Are you doing this all on a linux box thats acting as your router too? If not you don't need two profiles... you also don't need to set the local-network-acl on ANY profile that isn't do anything with nat. /b ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
