Re: [Freeswitch-users] Requesting testing.
please follow the procedures http://wiki.freeswitch.org/wiki/Reporting_Bugs to report bugs at http://jira.freeswitch.org. Also, you will need to provide far more detail than in this email for anyone to be able to have a possibility of fixing it. Please include details such as, what file is missing, what errors and warnings you get. How to reproduce it and preferably a patch to correct the problem if you can create one. Mike On Nov 24, 2009, at 2:49 AM, RobertT wrote: I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. ___ 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] os x compile failure
Please retest this with current svn trunk fresh checkout. Thanks Mike On Nov 23, 2009, at 9:47 PM, Brian West wrote: Ok 32bit... we are currently working on that as I type. /b On Nov 23, 2009, at 8:44 PM, James Budge wrote: 2GHz Intel Core Duo OS 10.6.2 Xcode 3.2.1 Updated to revision 15648. ___ 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 proxy media and bypass media in FreeSWITCH
This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote: I was using revision 15586. From: br...@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml: ___ 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] DTMF javasript
Your not telling anything to call your callback.
On Nov 24, 2009, at 1:03 AM, Baskar wrote:
Hi,
I want to check value given to the javascript with conditions whether it is
voicefile, extension or mobile Number when i press the dtmf value.
Steps i need to check in javascript:
When i Press the DTMF value 1 it should check the 3 condition
If the Value for argv[2]=vfsurya means it is a voice file so it should play
the Voice file
If the Value for argv[2]=1001 means it is a extension. The call should Bridge
the extension
If the Value for argv[2]=9841799874 means it is a Mobile number. The call
should Bridge the Mobile number
var exit = false;
var dtmf_digits = ;
var repeat = 0;
var argv[2]=vfsurya; // or var argv[2]=1001 or var argv[2]=Mobile Number
function onInput( session, type, data, arg )
{
if ( type == dtmf )
{
console_log( info, Got digit + data.digit + \n );
if ( data.digit == 1 )
{
if(argv[2].startswith(vf))
{
var voice2=voice.substring(2)+br /
session.streamFile(/usr/local/freeswitch/sounds/en/us/callie/+voice2+.wav,
onInput );
}
else if(argv[2].length==4)
{
console_log( info, Got voicefile + argv[2] + \n
);
session.execute(bridge,
sofia/internal/+argv[2]+%192.168.1.2, onInput );
}
else
{
session.execute(bridge,
sofia/default/sip:+argv[2]+@192.168.1.135:5066, onInput );
}
}
}
}
But if 1 is pressed there is no event trigger but it get the dtmf value as 1
in freeswitch console.
can any one specify what is the error or correct me where i am wrong.
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Re: [Freeswitch-users] DTMF Event is not coming while using playback terminators.
async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: Dear All, I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ 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] DTMF Event is not coming while using playback terminators.
Yes, I am using async mode only.. On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote: async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: Dear All, I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ 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] User who answer the bridge in a execute_answer
Hi,
thanks for the suggestion! In the end i updated freeswitch using lastest
source in the trunk and callee_id_number worked!
Best Regard,
Daniele
Michael Jerris ha scritto:
Try running the info app there and see if the info is anywhere in that output .
Mike
On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
i'm writing some dialplan parts that get executed on execute_on_answer. In this
dialplan that get executed i need to make a directory to handle recordings for
record_session and my folder structure is:
USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav
--
action application=system data=mkdir -p
$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/ /
action application=bind_meta_app data=1 a s
record_session::$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/${strftime(%H_%M_%S)}-${caller_id_number}.wav
/
--
The call flow is:
Call from external - IVR - Transfer to Group - Execute on Answer -
system/bind_meta_app
Pratically, i need the number (or better the user) that answered the call: what
variable should i check?
I tried with sip_from_user, callee_id_number and some other.
Thank for your help,
Best Regards,
Daniele
attachment: info.vcf___
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[Freeswitch-users] Call Transfer Help Please
Hi,
I'm trying to setup call transfer for a phone without a transfer button. I was
on IRC last night and got some pointers to how this is setup in dialplan.xml
and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the transfer, in
the default config, can only be initiated by the b leg of the call, I was
able to make this work as configured in the defaults, i.e, to initiate a
transfer (for an internal call) from the dialled extension to a new extension.
Now the problem . . .
I have an incoming PSTN line that rings a group of extensions, what I want to
be able to do is to give whoever answers the PSTN call ability to transfer the
call on to another extension.
There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch
extension of 1000, it rings the extension phones in the group.
I'd hoped that the default transfer setup would handle this without
modification - the incoming call on extension 1000 would be the a leg, the
answering extension would be the b leg and a transfer from b would work as
per the default config. This does not work for me though.
I'm struggling a bit with the bind meta app options and can't seem to make it
do what I want.
Could someone please confirm that what I'm trying to do is feasible and perhaps
suggest the right parameters to use in dialplan.xml and features.xml please ?
Relevant section in the is_transfer section in features.xml
action application=transfer data=-bleg ${digits} XML default/
And in default.xml from
action application=bind_meta_app data=1 b s execute_extension::dx XML
features/ to
I've tried posting a call log to the Pastebin (11252/3) but there was an error
- it looks like the dump was too big. Not sure what the maximum size on
pastebin dumps is ?
My understanding (or lack of) of a and b are in the scenario described is
not helping ...
Is the a leg the call coming in on the PSTN line (on Ext 1000) ?
Is the answering extension the b leg ?
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
What is the correct transfer data string in features.xml ?
Or am I totally on the wrong track here ?
If it is possible to do what I want, and changes are required to the
dialplan.xml and/or features.xml files, is it possible to have different logic
in there such that the actions are different whether it is the a leg or b
leg that's requesting the transfer ?
regards
Dave
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Re: [Freeswitch-users] Callback to the user in ESL
Yes Mr. Collins, I've tried with shed_api. But I was not able to control, if the user reject the call. I made a shed_api to originate a call to 1000 and If it is answered, I'll transfer the call to 9097 (So it comes to my program, refer the dialplan in my question). But what happens if the user 1000, reject the call. I can't control that. If the user 1000, reject the call, I need to call the user after some time. Any way to do this!! On Mon, Nov 23, 2009 at 11:21 PM, Michael Collins m...@freeswitch.orgwrote: On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy lakindi...@gmail.com wrote: Hi, I'm using perl ESL to control the call in freeswitch. I'm having the following scenario, but not able to get it right. Dialplan: extension name=outbound_soc condition field=destination_number expression=^9097$ action application=set data=continue_on_fail=true/ action application=socket data=192.168.1.222:8447 async full/ /condition /extension 1. User A calls to an extention (1000). 2. My ESL program will be running, and it answers the call. 3. Then the program will get a number from the user. 4. It will hangup the call. 5. The program has to call to the number that was given by the user. In the above scenario, I was able to do until the 4th step. After hangup the call, if I say originate it is not working. Any ideas on how to do this in ESL. I want to make sure I understand what the script is supposed to be doing. The caller will key in a phone number to your script and your script will collect those digits. The script will then hangup on the caller and originate a completely new call? Perhaps you could use sched_api to schedule a new originate command for a few seconds into the future and then hangup? -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 ___ 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] Execute on Answer with JavaScript
Hi Mike, I understand. I just need to not use the session.answer(). Many thanks. Michael Jerris wrote: This is done automatically when you bridge 2 sessions together. Mike On Nov 23, 2009, at 6:45 AM, Oscav wrote: How can we send the answer to the caller only when the callee answers, in JavaScript?? ___ 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 -- View this message in context: http://old.nabble.com/Execute-on-Answer-with-JavaScript-tp26476532p26494996.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Callback to the user in ESL
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);
$con-api(originate,user/1000 park());
print Hai\n;
}
print
Bye\n--\n;
close($new_sock);
}
Output:
Connected call 6b713588-d8c5-11de-8d50-596fac84e59e, from 1000 to 9097
1000
Connection not exists
Hai
Bye
--
The freeswitch log is in
http://pastebin.freeswitch.org/11258
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??
On Tue, Nov 24, 2009 at 1:10 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
or open a new outbound connection at the end of your script so you can send
your originate command.
Since the channel hanging up will close your existing connection since it's
only an outbound single session socket.
On Mon, Nov 23, 2009 at 11:51 AM, Michael Collins m...@freeswitch.orgwrote:
On Mon, Nov 23, 2009 at 3:25 AM, lakshmanan ganapathy
lakindi...@gmail.com wrote:
Hi,
I'm using perl ESL to control the call in freeswitch.
I'm having the following scenario, but not able to get it right.
Dialplan:
extension name=outbound_soc
condition field=destination_number expression=^9097$
action application=set data=continue_on_fail=true/
action application=socket data=192.168.1.222:8447 async
full/
/condition
/extension
1. User A calls to an extention (1000).
2. My ESL program will be running, and it answers the call.
3. Then the program will get a number from the user.
4. It will hangup the call.
5. The program has to call to the number that was given by the user.
In the above scenario, I was able to do until the 4th step. After hangup
the call, if I say originate it is not working.
Any ideas on how to do this in ESL.
I want to make sure I understand what the script is supposed to be doing.
The caller will key in a phone number to your script and your script will
collect those digits. The script will then hangup on the caller and
originate a completely new call? Perhaps you could use sched_api to schedule
a new originate command for a few seconds into the future and then hangup?
-MC
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Re: [Freeswitch-users] sched_broadcast doesn't execute
Hi Anthony,
Now it works very well. Thank you so much for your help. I'm having a lot of
fun with this platform.
Regards.
Anthony Minessale-2 wrote:
is that your exact code?
${uuid} will not be expanded by javascript
var uuid = session.getVariable(uuid);
new_session.execute(sched_broadcast, +20 alloted_timeout + uuid +
playback:ivr-welcome_to_freeswitch.wav);
On Wed, Nov 18, 2009 at 10:07 AM, Oscav os...@hotmail.fr wrote:
Hi,
I'm writing a script in Javascript that plays a message during a bridge.
I'm
trying to use a sched_broadcast to do it. The job is scheduled and then
deleted but I never hear the wav file and I don't get the OK Message
Scheduled in the log. It even doesn't display any error message if I
specify a wrong file name. Someone could help me on this issue ??
new_session.execute(sched_broadcast, +20 alloted_timeout ${uuid}
playback:ivr-welcome_to_freeswitch.wav);
I already did some posts but I got no answer. This is very difficult to
progress without help.
Many thanks
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[Freeswitch-users] How to find whether the destination extension supports encryption
Hello, We have a mix of phones that support RTP encryption and those that do not. I have to support both types in the meanwhile, and would like to have encryption enabled on the relevant leg, even if the other leg does not support it (why? one of our ATAs either must have it unencrypted or have it encrypted, but cannot have both). How do I find whether the *destination* supports encryption? I do not want to manage an additional table in the database... Thanks! __Yehavi: ___ 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 do I know the destination profile name?
Hello Anthony,
Indeed I see the reference to this channel variable in the code, but when
trying to access it from the dial plan it is empty... I try to get the value
of ${sip_profile_name} and it is empty.
Thanks! __Yehavi:
2009/11/23 Anthony Minessale anthony.miness...@gmail.com
Let's just do this:
r15629 or higher
look for sip_profile_name
On Tue, Nov 17, 2009 at 3:03 AM, Eli Hayun eliha...@gmail.com wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks
Eli
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Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP
Hi Jeff, All is good, I have looked at the x64 related changes you made and will merge them back to UniMRCP tree most probably during the next week. Thanks, Arsen. From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Mon, November 23, 2009 9:01:25 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP Hi Arsen, I have merged your changes in now - thank you. Would you perhaps be able to look at the x64 changes I made to the projects and merge them back into your code to ease the future updating. Thanks Jeff Arsen Chaloyan wrote: Hi Jeff, Your input would be very helpful, I just wanted to understand where the problem is and contribute the way I can. I see you're the assignee, so please go ahead and let me know if there is anything left I can help with. Arsen. From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Mon, November 23, 2009 8:16:28 AM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP Hi Arsen, I would be happy to help with the FS integration if you want - please do put your patch in a Jira. Jeff Date: Sun, 22 Nov 2009 10:09:41 -0800 From: [hidden email] To: [hidden email] Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk [hidden email] To: [hidden email] Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: Hi Everyone, Please help freeswitch experts... !!! i have been working on freeswitch from last 2 days. i have downloaded freeswitch and unimrcp (server + client) for windows. I tested the unimrcp client and server, which is running fine with the command: run synth and run recog. I got both synth.pcm recog.pcm files. But my objective is to call Freeswitch through x-lite, where freeswitch should call unimrcp client and return the PCM files. I tried it alot, but unable to do it. after lots of reading i found that i do not have mod_unimrcp. i do not know from where to download it and how to merge it into freeswitch. I would be very thankful if you may help. Thanks, ss -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list [hidden email] 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 [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org View message @ http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4047148.html To unsubscribe from Re: need help !! Problem with freeswitch uniMRCP, click here. Hotmail: Trusted email with powerful SPAM protection. Sign up now. View this message in context: RE: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 -- View this message in context:
[Freeswitch-users] Noise with openzap
Hi, I have an Ubuntu box running FS1.0.4 which has been processing a good volume of calls between local users with soft phones (Xlite) and GSM handsets via a number or Portech gateways, this has worked very well for some time and audio quality is very good. I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, configured openzap and I can originate and answer calls on the the openzap lines fine, however these calls via opezap all seem to suffer from significant noise, the audio path works fine in both directions but noise seems particularly bad at the local soft phone end. Quality of all other calls through the box is fine though, any ideas appreciated ?, NB A regular handset plugged directly into the PSTN lines has no problems though Thanks Steve ___ 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] FS cluster and how to get sofia gateway health status?
Hi everyone, I'm setting up FS cluster In my application, I plan to use two FS server as front and four FS as backend, the incoming calls first send to the front FS, then the front FS forward the call to backend FS server by return 302 to invite message. The front FS need to known the backend FS's status, so it won't forward calls to a server if it's down. The question is, how to check the backend FS's status. As I known, fs can add gateways to sofia profile, the endpoint will check gateways's state by send ping message, I think it is the function what I need if I can get the gateways's status from fs, does someone known how to do it? or can someone give me some suggestion about how to setup FS cluster? -- Lei.Tang lei.tl...@gmail.com ___ 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] need help !! Problem with freeswitch uniMRCP
Hi Can we enable passive recording in freeswitch ,wanpipe ,openzap , we are using a sangoma tapping system with freeswitch. Thanks Imthiyaz ___ 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] SoftPhone
Hello, I have been going through FreeSWITCH for quite sometime now. I would like to develop my own SIP Client soft-phone in Java/etc., how do I start?. Will I get any SDK/APIs for this. Please assist. Thanks, Rex -- View this message in context: http://n2.nabble.com/SoftPhone-tp4058292p4058292.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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] How to run IVR application
Hi to all, I am very new this platform . I just downloaded freeswitch to my windows xp machine , compiled successfully and run. After that I dont have any idea what to do :( . I am not finding simple kind of tutorial on the net. could you please suggest me, how I have to proceed. My requirement is; I need to run IVR application on machine using SIP phone. I am very sorry to my bad English. Thanks and Regards, Venkat. -- 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
[Freeswitch-users] remote_media_ip variable not set
Hi, I tried to use the variable remote_media_ip from within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, jb ___ 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] SoftPhone
Suggestion: Be one the first to integrate QuteCOm -E Gpro.ws ___ 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] need help !! Problem with freeswitch uniMRCP
What does this have to do with uniMRCP? Mike On Nov 24, 2009, at 9:54 AM, Imthiyaz Ahmed wrote: Hi Can we enable passive recording in freeswitch ,wanpipe ,openzap , we are using a sangoma tapping system with freeswitch. ___ 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] remote_media_ip variable not set
It gets set whenever the codec is negotiated. So it'll be NULL until (pre_)answer if you have late-negotiation on. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 24-Nov-09, at 10:22 AM, Juan Backson wrote: Hi, I tried to use the variable remote_media_ip from within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, jb ___ 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] remote_media_ip variable not set
Hi, In the case of proxy_media=true, does it gets set at all then? thanks, jb On Tue, Nov 24, 2009 at 11:46 PM, Mathieu Rene mrene_li...@avgs.ca wrote: It gets set whenever the codec is negotiated. So it'll be NULL until (pre_)answer if you have late-negotiation on. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 24-Nov-09, at 10:22 AM, Juan Backson wrote: Hi, I tried to use the variable remote_media_ip from within dialplan, but it is not returning anything. Does anyone know when this variable gets set and how to have this variable to be set as soon as an INVITE hit freeswitch? Thanks, jb ___ 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 ___ 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] Problem while playing more than 10 voice files using playback
1) Did you ever supply a log of your problem? 2) Are you using ESL lib or did you make your own event socket client, (if you did maybe you implemented the protocol client wrong) You are not supplying any specific information like traces of the connection or the version of the code you are using, weather you have tried the latest release or not etc. And lastly you are not using the events I told you about to tell exactly when the commands in question are being executed. getting a variable in a loop is a non-scalable memory consuming bad idea in how to program over a socket. On Mon, Nov 23, 2009 at 11:56 PM, Thangappan.M thangappan...@gmail.comwrote: The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. Why this command event is occurring? How can I restrict this? What are the information it has? How can I get all the information in it ? ( If command event has info) Help me On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M thangappan...@gmail.comwrote: I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?. When I play back lesser than ten voice files the variable has been set properly. What could be the reason? -- Forwarded message -- From: Thangappan.M thangappan...@gmail.com Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users freeswitch-users@lists.freeswitch.org Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: = Set the playback_delimiter variable = Set the playback_sleep_val variable = Set the event lock as true = Set the freeswitch ( my own) variable as zero = Wait in the loop until the variable is been set as zero = Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) = Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). = Wait in the loop until the variable is been set as one. = Set the Event lock as false = Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me? Thanks in advance. -- Regards, Thangappan.M -- Regards, Thangappan.M -- Regards, Thangappan.M ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
[Freeswitch-users] Patch to allow gateways to be defined without the password parameter set
I have modified sofia.c in mod_sofia so that I can define gateways without having to specify the password parameter. This is because I am using a SIP gateway that does not require SIP registration. The modified version still requires the password to be set on any gateway for which register is set to true. Attached is the diff file for these changes. _ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurantsform=MFESRPpubl=WLHMTAGcrea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 sofia_password_patch.diff Description: Binary data ___ 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] Patch to allow gateways to be defined without the password parameter set
John, If the remote end doesn't require a username or password then you don't need to create a gateway to send a call to that endpoint. You can simply do sofia/profile/num...@remoteip and it'll work. Also can you put the patch on jira via http://jira.freeswitch.org /b On Nov 24, 2009, at 1:41 PM, John Platts wrote: I have modified sofia.c in mod_sofia so that I can define gateways without having to specify the password parameter. This is because I am using a SIP gateway that does not require SIP registration. The modified version still requires the password to be set on any gateway for which register is set to true. Attached is the diff file for these changes. _ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search? q = restaurants form = MFESRP publ = WLHMTAG crea = TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 sofia_password_patch.diff ___ 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] DTMF Event is not coming while using playback terminators.
1. can you supply a trace of this esl communications. 2. is it inband or rfc2833 dtmf ? MIke On Nov 24, 2009, at 3:59 AM, velusamy velu wrote: Yes, I am using async mode only.. On Tue, Nov 24, 2009 at 2:12 PM, Michael Jerris m...@jerris.com wrote: async? On Nov 24, 2009, at 2:22 AM, velusamy velu wrote: Dear All, I am using Perl ESL::IVR module to develop a simple IVR. I have filtered DTMF events. I have also set playback_terminators to cut the playback when giving the digits. I have faced problem that DTMF event has not come if DTMF given while playing voice files. I have received 'COMMAND' event. I have the following questions. Why the 'COMMAND' event came event filter is on? How to avoid this event in ESL? ___ 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] Call Transfer Help Please
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
Hi,
I'm trying to setup call transfer for a phone without a transfer button. I
was on IRC last night and got some pointers to how this is setup in
dialplan.xml and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the transfer, in
the default config, can only be initiated by the b leg of the call, I was
able to make this work as configured in the defaults, i.e, to initiate a
transfer (for an internal call) from the dialled extension to a new extension.
Now the problem . . .
I have an incoming PSTN line that rings a group of extensions, what I want to
be able to do is to give whoever answers the PSTN call ability to transfer
the call on to another extension.
There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch
extension of 1000, it rings the extension phones in the group.
I'd hoped that the default transfer setup would handle this without
modification - the incoming call on extension 1000 would be the a leg, the
answering extension would be the b leg and a transfer from b would work
as per the default config. This does not work for me though.
I'm struggling a bit with the bind meta app options and can't seem to make
it do what I want.
Could someone please confirm that what I'm trying to do is feasible and
perhaps suggest the right parameters to use in dialplan.xml and features.xml
please ?
Relevant section in the is_transfer section in features.xml
action application=transfer data=-bleg ${digits} XML default/
And in default.xml from
action application=bind_meta_app data=1 b s execute_extension::dx XML
features/ to
I've tried posting a call log to the Pastebin (11252/3) but there was an
error - it looks like the dump was too big. Not sure what the maximum size on
pastebin dumps is ?
My understanding (or lack of) of a and b are in the scenario described is
not helping ...
Is the a leg the call coming in on the PSTN line (on Ext 1000) ?
Yes, the calling leg
Is the answering extension the b leg ?
Yes
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
I don't understand this question
What is the correct transfer data string in features.xml ?
ditto
Or am I totally on the wrong track here ?
You should just need to make sure that the bind meta is called in this scenario
so the b leg is able to do it, thats it.
If it is possible to do what I want, and changes are required to the
dialplan.xml and/or features.xml files, is it possible to have different
logic in there such that the actions are different whether it is the a leg
or b leg that's requesting the transfer ?
regards
Dave
FreeSwitch Version 1.0.4 (14460)
also, try the latest 1.0.5. pre release or svn trunk to confirm this is not an
issue that has already been fixed.
Mike
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set param name=manage-presence value=true/ in the internal profile. When a phone is set up I get the following Snom Phone REGISTER = OpenSIPS= Freeswitch Freeswitch OK = OpenSIPS=Snom Phone Snom Phone SUBSCRIBE = OpenSIPS= Freeswitch Freeswitch 202 Accepted = OpenSIPS=Snom Phone Snom Phone PUBLISH = OpenSIPS= Freeswitch Freeswitch 200 OK = OpenSIPS=Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswi...@sip11.mydomain.com nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:1...@sip1.mydomain.com SIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_so...@192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_so...@192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x8fc3d0): sent signal r_respond nua(0x8fc3d0): recv signal r_respond 200 OK nua: nua_handle_destroy: entering nua: nua_stack_set_params: entering nua(0x8fc3d0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (7)
[Freeswitch-users] Call forwarding problem
I was having trouble doing call forwarding from my SIP phone that is connected to FreeSWITCH. It turns out that my SIP phone is actually sending 302 Moved Temporarily responses, but my SIP gateway does not support 302 Moved Temporarily or SIP REFER messages. How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? Here is the SIP debug from our gateway: Received: INVITE sip:+19725357...@ipipgw.ipdimensions.com:5060;user=phone;transport=UDP;maddr=168.75.202.246 SIP/2.0 v: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0 Record-Route: sip:65.243.172.245:5060;lr v: SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236 record-route: sip:63.77.76.236;lr f: sip:+19729555...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3ec95a-3ad03068-3ec95a t: sip:+19725357...@63.77.76.236:5060;user=phone i: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Max-Forwards: 18 k: 100rel, replaces allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK v: SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 m: sip:199.173.101.208:5060;transport=UDP c: application/SDP l: 210 P-Asserted-Identity: sip:9729555...@63.77.76.236;user=phone Privacy: none v=0 o=- 540754816 540754816 IN IP4 199.173.111.141 s=- c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: SIP/2.0 100 Trying Via: SIP/2.0/UDP 65.243.172.245:5060;branch=z9hG4bKe19865e46222056ca70435e66fde4127.19be3eb0,SIP/2.0/UDP 63.77.76.236:5060;branch=z9hG4bK3f49bc4eb4ac163ffa354de0e6384d30.12e7ffbd;received=63.77.76.236,SIP/2.0/UDP DAL4:5060;maddr=199.173.101.208;branch=z9hG4bK-3ec95a-f54289ba-139ab2d1;received=199.173.101.208 From: sip:+19729555...@199.173.101.208:5060;user=phone;tag=dc7-13c4-3ec95a-3ad03068-3ec95a To: sip:+19725357...@63.77.76.236:5060;user=phone Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: a1f37fb0d065adc713c43ec95af54289baa8ec2034c293850-0569-7989 CSeq: 1 INVITE Allow-Events: telephone-event Server: Cisco-SIPGateway/IOS-12.x Content-Length: 0 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: INVITE sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212 Date: Tue, 24 Nov 2009 21:08:00 GMT Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 Supported: timer,resource-priority,replaces Min-SE: 1800 Cisco-Guid: 1208058493-3631485406-2892683743-874095366 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Timestamp: 1259096880 Contact: sip:19729555...@168.75.202.246:5060 Expires: 180 Allow-Events: telephone-event Max-Forwards: 17 P-Asserted-Identity: sip:19729555...@168.75.202.246 Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 314 v=0 o=CiscoSystemsSIP-GW-UserAgent 2925 1780 IN IP4 168.75.202.246 s=SIP Call c=IN IP4 199.173.111.141 t=0 0 m=audio 30056 RTP/AVP 18 0 8 101 c=IN IP4 199.173.111.141 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Nov 24 15:08:00.367 CST: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212 Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE Timestamp: 1259096880 0.000342 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Content-Length: 0 Nov 24 15:08:00.419 CST: //-1//SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212;tag=49aF8vtgHme2c Call-ID: 4802bacc-d87411de-ac70d9df-3419a...@168.75.202.246 CSeq: 101 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15654M Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=16;text=NORMAL_CLEARING Content-Length: 0 P-Asserted-Identity: 19725357722 sip:19725357...@168.75.202.212 Nov 24 15:08:00.427 CST: //-1//SIP/Msg/ccsipDisplayMsg: Sent: ACK sip:19725357...@168.75.202.212:5062 SIP/2.0 Via: SIP/2.0/UDP 168.75.202.246:5060;branch=z9hG4bK6821870 From: sip:19729555...@168.75.202.246;tag=14E93594-2488 To: sip:19725357...@168.75.202.212;tag=49aF8vtgHme2c Date: Tue,
Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH
I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. From: m...@jerris.com Date: Tue, 24 Nov 2009 03:39:16 -0500 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH This was fixed in trunk yesterday about 8 hrs before you sent this message. (15619). Please update and try again. Mike On Nov 23, 2009, at 11:33 PM, John Platts wrote: I was using revision 15586. From: br...@freeswitch.org Date: Mon, 23 Nov 2009 18:25:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problems with proxy media and bypass media in FreeSWITCH What rev exactly? /b On Nov 23, 2009, at 6:19 PM, John Platts wrote: I actually checked out the latest version of FreeSWITCH in the SVN repository. I have the following configured in /usr/local/freeswitch/conf/ dialplan/default.xml: _ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ 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] Call forwarding problem
You'll have to hairpin the media thru your machine usually if they won't accept either of those. /b On Nov 24, 2009, at 3:05 PM, John Platts wrote: How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? ___ 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 proxy media and bypass media in FreeSWITCH
Are you sure you did a make current? and can you outline the issue in more detail? /b On Nov 24, 2009, at 3:28 PM, John Platts wrote: I actually checked out revision 15654 today, and I was still getting problems with proxy media and bypass media in FreeSWITCH. ___ 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] No NOTIFY MWI when registering via proxy.
connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX prometheus...@gmx.net wrote: Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set param name=manage-presence value=true/ in the internal profile. When a phone is set up I get the following Snom Phone REGISTER = OpenSIPS= Freeswitch Freeswitch OK = OpenSIPS=Snom Phone Snom Phone SUBSCRIBE = OpenSIPS= Freeswitch Freeswitch 202 Accepted = OpenSIPS=Snom Phone Snom Phone PUBLISH = OpenSIPS= Freeswitch Freeswitch 200 OK = OpenSIPS=Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswi...@sip11.mydomain.com nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 401 Unauthorized for REGISTER (6) nua(0x905a80): recv signal r_destroy nta_leg_destroy((nil)) nta: received PUBLISH sip:1...@sip1.mydomain.comsip%3a...@sip1.mydomain.comSIP/2.0 (CSeq 3) nta: PUBLISH (3) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905f10): event i_publish 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905f10): sent signal r_respond nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x905f10): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua(0x905f10): sent signal r_destroy nta: sent 200 OK for PUBLISH (3) nua(0x905f10): recv signal r_destroy nta_leg_destroy((nil)) nta: received SUBSCRIBE sip:mod_so...@192.168.178.200:5062 SIP/2.0 (CSeq 2) nta: canonizing sip:mod_so...@192.168.178.200:5062 with contact nta: SUBSCRIBE (2) going to existing leg nua: nua_stack_process_request: entering nta: sent 200 OK for SUBSCRIBE (2) nua(0x905560): event i_subscribe 200 OK nua: nua_application_event: entering nta: received REGISTER sip:sip1.mydomain.com SIP/2.0 (CSeq 8) nta: REGISTER (8) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5dc073ba0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5dc073ba0): sent signal r_respond nua(0x7fd5dc073ba0): recv signal r_respond 200 OK nua: nua_stack_set_params: entering nua: nua_handle_destroy: entering nua(0x7fd5dc073ba0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nta: sent 200 OK for REGISTER (8) nua(0x7fd5dc073ba0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER
[Freeswitch-users] Handling the 302 Moved Temporarily response from JavaScript
I have considered writing JavaScript code to bridge two calls together. However, I would like to perform custom handling of the 302 Moved Temporarily response. How do I handle the 302 Moved Temporarily response if I use JavaScript? _ Bing brings you maps, menus, and reviews organized in one place. http://www.bing.com/search?q=restaurantsform=MFESRPpubl=WLHMTAGcrea=TEXT_MFESRP_Local_MapsMenu_Resturants_1x1 ___ 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] Noise with openzap
you may want to try the latest release of both wanpipe and FS openzap is still a moving target since its in constant development from both the hardware and software end On Tue, Nov 24, 2009 at 7:25 AM, Steven Brown st...@justfone.com wrote: Hi, I have an Ubuntu box running FS1.0.4 which has been processing a good volume of calls between local users with soft phones (Xlite) and GSM handsets via a number or Portech gateways, this has worked very well for some time and audio quality is very good. I've now added a Sangoma A200 with 4 ports hooked up to 4 PSTN lines, configured openzap and I can originate and answer calls on the the openzap lines fine, however these calls via opezap all seem to suffer from significant noise, the audio path works fine in both directions but noise seems particularly bad at the local soft phone end. Quality of all other calls through the box is fine though, any ideas appreciated ?, NB A regular handset plugged directly into the PSTN lines has no problems though Thanks Steve ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Call Transfer Help Please
Hi Mike,
thanks for the reply. I am using the pre-compiled Windows binary - is there a
1.0.5 pre-release of that yet ?
FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I was
sure that I had previously loaded a later SVN Version, but just did it again,
unless I'm not doing it right, the version number does not seem to be getting
updated. The current build in the precompiled binaries area is reported to be
15604 and I've downloaded and installed that - although when the installer runs
it tells me that it is version 15376. Either way, the Version command in
FreeSwitch reports 1.0.4 (14460).
The Transfer still does not work for me from the extension which answers the
call.
Sorry if my earlier questions were unclear ...
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
What is the correct transfer data string in features.xml ?
I don't understand this question(s)
I was looking for clarification of the second two arguments in the
bind_meta_app data call, i.e, that the b and s were the correct values and
also that the is transfer transfer data argument was -bleg
That is, that the arguments in the default dialplan are correct for this
scenario - which they appear to be based on your previous reply to my query.
So, is there anything else that I can check to see why this is not working ?
regards
Dave
- Original Message -
From: Michael Jerris
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, November 24, 2009 8:19 PM
Subject: Re: [Freeswitch-users] Call Transfer Help Please
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
Hi,
I'm trying to setup call transfer for a phone without a transfer button. I
was on IRC last night and got some pointers to how this is setup in
dialplan.xml and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the transfer,
in the default config, can only be initiated by the b leg of the call, I was
able to make this work as configured in the defaults, i.e, to initiate a
transfer (for an internal call) from the dialled extension to a new extension.
Now the problem . . .
I have an incoming PSTN line that rings a group of extensions, what I want
to be able to do is to give whoever answers the PSTN call ability to transfer
the call on to another extension.
There is an ATA (Linksys SPA3101) set up on the PSTN line with a FreeSwitch
extension of 1000, it rings the extension phones in the group.
I'd hoped that the default transfer setup would handle this without
modification - the incoming call on extension 1000 would be the a leg, the
answering extension would be the b leg and a transfer from b would work as
per the default config. This does not work for me though.
I'm struggling a bit with the bind meta app options and can't seem to
make it do what I want.
Could someone please confirm that what I'm trying to do is feasible and
perhaps suggest the right parameters to use in dialplan.xml and features.xml
please ?
Relevant section in the is_transfer section in features.xml
action application=transfer data=-bleg ${digits} XML default/
And in default.xml from
action application=bind_meta_app data=1 b s execute_extension::dx XML
features/ to
I've tried posting a call log to the Pastebin (11252/3) but there was an
error - it looks like the dump was too big. Not sure what the maximum size on
pastebin dumps is ?
My understanding (or lack of) of a and b are in the scenario described
is not helping ...
Is the a leg the call coming in on the PSTN line (on Ext 1000) ?
Yes, the calling leg
Is the answering extension the b leg ?
Yes
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
I don't understand this question
What is the correct transfer data string in features.xml ?
ditto
Or am I totally on the wrong track here ?
You should just need to make sure that the bind meta is called in this
scenario so the b leg is able to do it, thats it.
If it is possible to do what I want, and changes are required to the
dialplan.xml and/or features.xml files, is it possible to have different logic
in there such that the actions are different whether it is the a leg or b
leg that's requesting the transfer ?
regards
Dave
FreeSwitch Version 1.0.4 (14460)
also, try the latest 1.0.5. pre release or svn trunk to confirm this is not
an issue that has already been fixed.
Mike
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Re: [Freeswitch-users] No NOTIFY MWI when registering via proxy.
Anthony, thanks for the hint, I receive events like the following RECV EVENT Event-Name: MESSAGE_WAITING Core-UUID: e71632c8-d948-11de-942b-0138c6269e37 FreeSWITCH-Hostname: sip11.mydomain.com FreeSWITCH-IPv4: 192.168.178.200 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2009-11-24 23:33:13 Event-Date-GMT: Tue, 24 Nov 2009 22:33:13 GMT Event-Date-Timestamp: 1259101993918617 Event-Calling-File: mod_voicemail.c Event-Calling-Function: update_mwi Event-Calling-Line-Number: 1738 MWI-Messages-Waiting: yes MWI-Message-Account: 2...@sip1.mydomain.com MWI-Voice-Message: 4/1 (0/0) I think the problem may be the Freeswitch cluster we are working with. All phones register with realm (e.g. 2...@sip1.mydomain.com). The FS hostname is sip11.mydomain.com resp. sip12.mydomain.com on the other host. With xml_curl we ensure that for both domain names a directory entry is passed back. That way it works nicely with registering phones, receiving voicemails, recording voicemails etc. but not for MWI. For recording and querying voicemails we use the realm instead of the domain name and that way it works. When a voicemail has finished recording - and at the time the above message occurs - I see 2 directory xml_curl requests with Event-Calling-File=mod_voicemail.cEvent-Calling-Function=resolve_id One I expect is for retrieving the MWI data and the other one for sending the VM email (which is sucessfully sent). Any hint how we can workaround this? Or is there a parameter to tell mod_voicemail that is should use the realm instead of the local hostname for sending MWI? Best regards Peter Anthony Minessale schrieb: connect to FS with fs_cli Issue the command: /events MESSAGE_QUERY MESSAGE_WAITING then leave some voice mails probably you have a mis-configuration where the user/domain/profile cannot be resolved to the correct sofia profile to send the notify The event starts out as a freeswitch event and is translated into the notify by mod_sofia but only if it can match the event to a real sip user On Tue, Nov 24, 2009 at 2:54 PM, Peter P GMX prometheus...@gmx.net mailto:prometheus...@gmx.net wrote: Hello, I have a similar problem with Freeswitch behind OpenSIPS as a load balancer: When registering, Freeeswitch does not send a MWI NOTIFY message for a Phone which has voicemails. Even after recording a new voicemail there is no NOTIFY message sent. And there are no error messages on the console. I have explicitely set param name=manage-presence value=true/ in the internal profile. When a phone is set up I get the following Snom Phone REGISTER = OpenSIPS= Freeswitch Freeswitch OK = OpenSIPS=Snom Phone Snom Phone SUBSCRIBE = OpenSIPS= Freeswitch Freeswitch 202 Accepted = OpenSIPS=Snom Phone Snom Phone PUBLISH = OpenSIPS= Freeswitch Freeswitch 200 OK = OpenSIPS=Snom Phone So presence generally seems to work. But ngrepping the Network traffic there's no MWI NOTIFY message coming from Freeswitch to any phone. FreeSWITCH Version is 1.0.trunk (15648), so the patch discussed before should be already there. Any idea how to force the NOTIFY messages? Best regards Peter Here's the debug Level9 output for nta and nua when a phone with VMs registers, seems like there is no error in it: freeswi...@sip11.mydomain.com mailto:freeswi...@sip11.mydomain.com nta: received REGISTER sip:sip1.mydomain.com http://sip1.mydomain.com SIP/2.0 (CSeq 7) nta: REGISTER (7) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x7fd5d409c8f0): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x7fd5d409c8f0): sent signal r_respond nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): sent signal r_destroy nua: nua_handle_magic: entering nua: nua_handle_destroy: entering nua(0x7fd5d409c8f0): recv signal r_respond 401 Unauthorized nua: nua_stack_set_params: entering nta: sent 401 Unauthorized for REGISTER (7) nta: timer set to 32000 ms nua(0x7fd5d409c8f0): recv signal r_destroy nta_leg_destroy((nil)) nta: received REGISTER sip:sip1.mydomain.com http://sip1.mydomain.com SIP/2.0 (CSeq 6) nta: REGISTER (6) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering nua(0x905a80): event i_register 100 Trying nua: nua_application_event: entering nua: nua_respond: entering nua(0x905a80): sent signal r_respond nua: nua_handle_destroy: entering nua(0x905a80): recv signal r_respond 401 Unauthorized nua(0x905a80): sent signal r_destroy nua: nua_stack_set_params:
Re: [Freeswitch-users] Call forwarding problem
Is there any way to tell FreeSWITCH to do the following when 302 Moved Temporarily is sent to FreeSWITCH: - End the session between FreeSWITCH and the phone - Bridge the original session with the number that the call is forwarded to From: br...@freeswitch.org Date: Tue, 24 Nov 2009 15:32:44 -0600 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Call forwarding problem You'll have to hairpin the media thru your machine usually if they won't accept either of those. /b On Nov 24, 2009, at 3:05 PM, John Platts wrote: How do I get FreeSWITCH to forward calls without sending 302 Moved Temporarily or SIP REFER messages? ___ 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 _ Hotmail: Trusted email with Microsoft's powerful SPAM protection. http://clk.atdmt.com/GBL/go/177141664/direct/01/ http://clk.atdmt.com/GBL/go/177141664/direct/01/ ___ 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] mod_conference kick to abort invitations
Hi members, I'm controlling freeswitch with the conference module via xmlrpc. Is it desired that the kick command can only kick users that are connected to the conference? Is there no chance abort an invitation? The kick command has no effect until the person I invited with the dial command is connected. Thanks in advance! Jan ___ 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] Call Transfer Help Please
Hi again folks,
I have posted a dump into the Pastebin (11276), could someone have a look and
perhaps suggest where the problem might be please ?
I'm sure you'll be able to work it out, but the log is for a call where :-
incoming on PSTN Line (ext 1000)
Group exts, 111, 1001, 1001
Answered on 111 and requested transfer to 1001 with no success
regards
Dave
- Original Message -
From: Dave Stevenson
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, November 24, 2009 10:36 PM
Subject: Re: [Freeswitch-users] Call Transfer Help Please
Hi Mike,
thanks for the reply. I am using the pre-compiled Windows binary - is there a
1.0.5 pre-release of that yet ?
FreeSwitch reports its version as 1.0.4 (14460) but this is not correct, I
was sure that I had previously loaded a later SVN Version, but just did it
again, unless I'm not doing it right, the version number does not seem to be
getting updated. The current build in the precompiled binaries area is reported
to be 15604 and I've downloaded and installed that - although when the
installer runs it tells me that it is version 15376. Either way, the Version
command in FreeSwitch reports 1.0.4 (14460).
The Transfer still does not work for me from the extension which answers the
call.
Sorry if my earlier questions were unclear ...
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
What is the correct transfer data string in features.xml ?
I don't understand this question(s)
I was looking for clarification of the second two arguments in the
bind_meta_app data call, i.e, that the b and s were the correct values and
also that the is transfer transfer data argument was -bleg
That is, that the arguments in the default dialplan are correct for this
scenario - which they appear to be based on your previous reply to my query.
So, is there anything else that I can check to see why this is not working ?
regards
Dave
- Original Message -
From: Michael Jerris
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, November 24, 2009 8:19 PM
Subject: Re: [Freeswitch-users] Call Transfer Help Please
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
Hi,
I'm trying to setup call transfer for a phone without a transfer button.
I was on IRC last night and got some pointers to how this is setup in
dialplan.xml and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the transfer,
in the default config, can only be initiated by the b leg of the call, I was
able to make this work as configured in the defaults, i.e, to initiate a
transfer (for an internal call) from the dialled extension to a new extension.
Now the problem . . .
I have an incoming PSTN line that rings a group of extensions, what I
want to be able to do is to give whoever answers the PSTN call ability to
transfer the call on to another extension.
There is an ATA (Linksys SPA3101) set up on the PSTN line with a
FreeSwitch extension of 1000, it rings the extension phones in the group.
I'd hoped that the default transfer setup would handle this without
modification - the incoming call on extension 1000 would be the a leg, the
answering extension would be the b leg and a transfer from b would work as
per the default config. This does not work for me though.
I'm struggling a bit with the bind meta app options and can't seem to
make it do what I want.
Could someone please confirm that what I'm trying to do is feasible and
perhaps suggest the right parameters to use in dialplan.xml and features.xml
please ?
Relevant section in the is_transfer section in features.xml
action application=transfer data=-bleg ${digits} XML default/
And in default.xml from
action application=bind_meta_app data=1 b s execute_extension::dx XML
features/ to
I've tried posting a call log to the Pastebin (11252/3) but there was an
error - it looks like the dump was too big. Not sure what the maximum size on
pastebin dumps is ?
My understanding (or lack of) of a and b are in the scenario
described is not helping ...
Is the a leg the call coming in on the PSTN line (on Ext 1000) ?
Yes, the calling leg
Is the answering extension the b leg ?
Yes
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml ?
I don't understand this question
What is the correct transfer data string in features.xml ?
ditto
Or am I totally on the wrong track here ?
You should just need to make sure that the bind meta is called in this
scenario so the b leg is able to do it, thats it.
If it is possible to do what I want, and changes are required to the
dialplan.xml and/or features.xml files, is it possible to have different logic
in there such that the actions are different whether
Re: [Freeswitch-users] How to run IVR application
you can do this in follow steps: 1.edit default.xml diaplan config file in your fs config directory(FS/conf/dialplan/default.xml), and section extension name=ivr_demo2 condition field=destination_number expression=^\*114$ 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 2009/11/24 ovvenkat ovvenkate...@gmail.com Hi to all, I am very new this platform . I just downloaded freeswitch to my windows xp machine , compiled successfully and run. After that I dont have any idea what to do :( . I am not finding simple kind of tutorial on the net. could you please suggest me, how I have to proceed. My requirement is; I need to run IVR application on machine using SIP phone. I am very sorry to my bad English. Thanks and Regards, Venkat. -- 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 -- Lei.Tang lei.tl...@gmail.com ___ 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] Call Transfer Help Please
I do not see the meta app getting added in your log
-
Dialplan: sofia/internal/1...@192.168.1.50 Action bind_meta_app(*
Without this no meta actions will occur
Dave Stevenson wrote:
Hi again folks,
I have posted a dump into the Pastebin (11276), could someone have a look
and perhaps suggest where the problem might be please ?
I'm sure you'll be able to work it out, but the log is for a call where :-
incoming on PSTN Line (ext 1000)
Group exts, 111, 1001, 1001
Answered on 111 and requested transfer to 1001 with no success
regards
Dave
- Original Message -
From: Dave Stevenson
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, November 24, 2009 10:36 PM
Subject: Re: [Freeswitch-users] Call Transfer Help Please
Hi Mike,
thanks for the reply. I am using the pre-compiled Windows binary - is
there a 1.0.5 pre-release of that yet ?
FreeSwitch reports its version as 1.0.4 (14460) but this is not correct,
I was sure that I had previously loaded a later SVN Version, but just did
it again, unless I'm not doing it right, the version number does not seem
to be getting updated. The current build in the precompiled binaries area
is reported to be 15604 and I've downloaded and installed that - although
when the installer runs it tells me that it is version 15376. Either way,
the Version command in FreeSwitch reports 1.0.4 (14460).
The Transfer still does not work for me from the extension which answers
the call.
Sorry if my earlier questions were unclear ...
What are the correct LISTEN_TO and RESPOND_ON entries in dialplan.xml
?
What is the correct transfer data string in features.xml ?
I don't understand this question(s)
I was looking for clarification of the second two arguments in the
bind_meta_app data call, i.e, that the b and s were the correct values
and also that the is transfer transfer data argument was -bleg
That is, that the arguments in the default dialplan are correct for this
scenario - which they appear to be based on your previous reply to my
query.
So, is there anything else that I can check to see why this is not
working ?
regards
Dave
- Original Message -
From: Michael Jerris
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, November 24, 2009 8:19 PM
Subject: Re: [Freeswitch-users] Call Transfer Help Please
On Nov 24, 2009, at 5:29 AM, Dave Stevenson wrote:
Hi,
I'm trying to setup call transfer for a phone without a transfer
button. I was on IRC last night and got some pointers to how this is setup
in dialplan.xml and features.xml and what bind meta app does.
Once it became clear how the transfer is initiated and that the
transfer, in the default config, can only be initiated by the b leg of
the call, I was able to make this work as configured in the defaults, i.e,
to initiate a transfer (for an internal call) from the dialled extension
to a new extension.
Now the problem . . .
I have an incoming PSTN line that rings a group of extensions, what
I want to be able to do is to give whoever answers the PSTN call ability
to transfer the call on to another extension.
There is an ATA (Linksys SPA3101) set up on the PSTN line with a
FreeSwitch extension of 1000, it rings the extension phones in the group.
I'd hoped that the default transfer setup would handle this without
modification - the incoming call on extension 1000 would be the a leg,
the answering extension would be the b leg and a transfer from b would
work as per the default config. This does not work for me though.
I'm struggling a bit with the bind meta app options and can't seem
to make it do what I want.
Could someone please confirm that what I'm trying to do is feasible
and perhaps suggest the right parameters to use in dialplan.xml and
features.xml please ?
Relevant section in the is_transfer section in features.xml
action application=transfer data=-bleg ${digits} XML default/
And in default.xml from
action application=bind_meta_app data=1 b s
execute_extension::dx XML features/ to
I've tried posting a call log to the Pastebin (11252/3) but there
was an error - it looks like the dump was too big. Not sure what the
maximum size on pastebin dumps is ?
My understanding (or lack of) of a and b are in the scenario
described is not helping ...
Is the a leg the call coming in on the PSTN line (on Ext 1000) ?
Yes, the calling leg
Is the answering extension the b leg ?
Yes
What are the correct LISTEN_TO and RESPOND_ON entries in
dialplan.xml ?
I don't understand this question
What is the correct transfer data string in features.xml ?
ditto
Or am I totally on the wrong track here ?
You should just need
Re: [Freeswitch-users] register timeout / cisco 7960
People commonly use 60 sec registration refreshes to keep NAT routers happy Phillip Jones-2 wrote: hi there, I have set up some cisco 7960 up with fs. They work fine - but the only way I can keep them registered is to set the timer_register_expires in the Cisco cfg file to something really short like 10s. Does anyone know the default register timeout for fs? And where I might change this in fs? Thanks! Phil ___ 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 -- View this message in context: http://n2.nabble.com/register-timeout-cisco-7960-tp4054546p4062958.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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] Problem while playing more than 10 voice files using playback
The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M thangappan...@gmail.com FreeSWITCH version: freeswitch 1.0.4 I am using ESL library I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) I supplied the log , Here I attached the output of the ESL log. (Output.txt) Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. But in the output I got only 2,4,5,4 ( DTMF 1 is missed) Output of Perl code could be like Wait for response time out EVENT [COMMAND] Wait for response time out EVENT [DTMF] DTMF digit 2 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 5 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out Buffer: 2454 BYE Why the first digit(1) is missed here? In ESL log there is no digit called 1 why? Why the COMMAND event is received instead of DTMF? How can I get all DTMF digits? On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M thangappan...@gmail.comwrote: The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. Why this command event is occurring? How can I restrict this? What are the information it has? How can I get all the information in it ? ( If command event has info) Help me On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M thangappan...@gmail.comwrote: I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?. When I play back lesser than ten voice files the variable has been set properly. What could be the reason? -- Forwarded message -- From: Thangappan.M thangappan...@gmail.com Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users freeswitch-users@lists.freeswitch.org Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: = Set the playback_delimiter variable = Set the playback_sleep_val variable = Set the event lock as true = Set the freeswitch ( my own) variable as zero = Wait in the loop until the variable is been set as zero = Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) = Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). = Wait in the loop until the variable is been set as one. = Set the Event lock as false = Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. *DID I MAKE ANY MISTAKE IN THE ABOVE STEPS?* *NOTE*: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me? Thanks in advance. -- Regards, Thangappan.M -- Regards, Thangappan.M -- Regards, Thangappan.M -- Regards, Thangappan.M -- Regards, Thangappan.M ___ 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] Problem while playing more than 10 voice files using playback
you should use execute_complete events to tell when a command you tried to execute has finished and not poll the channel for a variable to be set because FreeSWITCH is an asynchronous application in the mode you are describing and you can never be sure of the timing. You are STILL polling for the variable. If you want help, perhaps you should at least attempt what is being suggested? Mike On Nov 25, 2009, at 1:18 AM, Thangappan.M wrote: The example script is there in the following link http://pastebin.com/f332f2fda In the previous post I have attached it. But it was not shown. 2009/11/25 Thangappan.M thangappan...@gmail.com FreeSWITCH version: freeswitch 1.0.4 I am using ESL library I attached the example Perl script which does the same steps that I posted already. ( Sample.pl) I supplied the log , Here I attached the output of the ESL log. (Output.txt) Through the softphone(Twinkle) I have given 1,2,4,5,4 as a DTMF digits. But in the output I got only 2,4,5,4 ( DTMF 1 is missed) Output of Perl code could be like Wait for response time out EVENT [COMMAND] Wait for response time out EVENT [DTMF] DTMF digit 2 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 5 (2000) Wait for inter digit time out EVENT [DTMF] DTMF digit 4 (2000) Wait for inter digit time out Buffer: 2454 BYE Why the first digit(1) is missed here? In ESL log there is no digit called 1 why? Why the COMMAND event is received instead of DTMF? How can I get all DTMF digits? On Tue, Nov 24, 2009 at 11:26 AM, Thangappan.M thangappan...@gmail.com wrote: The reason for waiting only for DTMF event is to handle the time outs in the IVR concept like response and inter digit time out. Using our own logic we 10 voice files in each play back if the voice files are more than 10. Now it works fine. Now the new problem has been raised. The problem is we are filtering only for DTMF events but we are getting COMMAND event . Because of this the DTMF digits are missing at the time . I am not able to proceed further. We are in the critical situation. Why this command event is occurring? How can I restrict this? What are the information it has? How can I get all the information in it ? ( If command event has info) Help me On Mon, Nov 23, 2009 at 10:04 AM, Thangappan.M thangappan...@gmail.com wrote: I am waiting only for DTMF events. That's why I am setting freeswitch variable for knowing whether the playback has done. My question is why this freeswitch variable is not setting properly when I play back more than 10 files using playback_delimiter option?. When I play back lesser than ten voice files the variable has been set properly. What could be the reason? -- Forwarded message -- From: Thangappan.M thangappan...@gmail.com Date: Sat, Nov 21, 2009 at 2:52 PM Subject: Problem while playing more than 10 voice files using playback To: freeswitch-users freeswitch-users@lists.freeswitch.org Dear all, I am in the process of implementing IVR using event outbound socket (async mode). I have implemented using Perl language. I did the following steps: = Set the playback_delimiter variable = Set the playback_sleep_val variable = Set the event lock as true = Set the freeswitch ( my own) variable as zero = Wait in the loop until the variable is been set as zero = Playback the voice files ( Here I combined the voice files with the delimiter value if more than one voice files are there) = Set the freeswitch(my own) variable as true ( This is used to identify whether the voice files are played successfully). = Wait in the loop until the variable is been set as one. = Set the Event lock as false = Trying to get the DTMF digits ( Have a assurance that all the voice files are played). The problem is, The above steps are working fine when the voice file count is lesser than or equal to 10. After the voice files are played only the variable(my own freeswitch) is set. Based on the variable I am doing further things. But when I tried to give the voice files count of more than 10 the variable has been set while starting to play back the first voice file itself . Because of this I am not able to proceed further. DID I MAKE ANY MISTAKE IN THE ABOVE STEPS? NOTE: I also referred mod_file_string documentation. In that they specified 128 files can be used to play back the voice files using playback_delimiter option. Please help me? Thanks in advance. -- Regards,
[Freeswitch-users] How to connect SIP phone to freeswitch
Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? -- 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] How to connect SIP phone to freeswitch
http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Nov 25, 2009, at 1:36 AM, ovvenkat wrote: Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established or not? ___ 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] ATA that supports TLS/SRTP w FS
Hi there Itamar, Does the SPA3102 support TLS or only SRTP? And what about Brians comments that 'It uses a sick twisted method of doing SRTP'. Do you have it working using SRTP together with FS? What about TLS? Otherwise are there any other ATA's that support TLS SRTP? On Sun, Nov 22, 2009 at 8:41 PM, Itamar Reis Peixoto ita...@ispbrasil.com.br wrote: it's support SRTP On Sun, Nov 22, 2009 at 7:21 AM, Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Do LInksys devices support TLS and SRTP that FS supports? 3102 at least doesn't according to this post -- Itamar Reis Peixoto e-mail/msn/google talk/sip: ita...@ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 ___ 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
