[Freeswitch-users] Setting WAV File as rinback

[Max Bridgewater]

Hi,

I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a
call. Unfortunately, it doesn't seem to work; the RINGING teletone is being
used instead of my audio. I have debug level enabled on Freeswitch but i
don't find anything suspicious.

is there anything specific i should be taking into consideration?


Thanks,
max.
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Re: [Freeswitch-users] G729 transcoding workaround

[Raymond Chandler]

On Aug 17, 2009, at 4:30 PM, Justin Miller wrote:
Is there a way to end the transcoded call legs, and bridge to the  
phone from the original call leg?  This would free up the M1000, and  
just seems like a better way to do things.



You might consider using a REFER if your endpoints support it.

Check out the "deflect" app


Raymond Chandler
http://freeswitchsolutions.com
http://cluecon.com
http://cudatel.com

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Re: [Freeswitch-users] Eavesdrop getting killed after being answered

[Lars Zeb]

Thanks, Michael, it's working at 14548.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, August 17, 2009 2:22 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop getting killed after being
answered

 

 

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb  wrote:

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC
 

Thanks Lars


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Re: [Freeswitch-users] JAVA ESL

[Michael Jerris]

Thanks, I'll get that merged in and clear from this we need to start  
using proper autoconf checks for java.


mike

On Aug 17, 2009, at 5:35 PM, Fernando Testa wrote:


Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185

Testa

On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris   
wrote:
can someone post a patch to that makefile to jira.freeswitch.org  
please.


Mike

On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:


I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a  
libesl.so with a cmd like this:

g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;

class Test
{
  public static void main(String[] args)
  {
System.loadLibrary("esl");
System.out.println("hello");
  }
}



On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite > wrote:

Thank you Anthony.

--- On Thu, 7/30/09, Anthony Minessale  
 wrote:


From: Anthony Minessale 
Subject: Re: [Freeswitch-users] JAVA ESL
To: freeswitch-users@lists.freeswitch.org
Received: Thursday, July 30, 2009, 9:00 PM


it might be a build issue, I was not exactly sure how to build it  
etc.

so it may need some help from a java expert

I wrote all of that with swig and never was able to test it.


On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite > wrote:

Hello,

I built libesl and JAVA mod. (make and make javamod). But when I  
try to run a JAVA script with the following code


ESLconnection connection = new ESLconnection("127.0.0.1", "9000",  
"");

ESLevent events = connection.getInfo();
System.out.println(events.toString());

I get the following error message:

Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0

at java.lang.ClassLoader$NativeLibrary.load(Native Method)
at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
at java.lang.Runtime.loadLibrary0(Runtime.java:823)
at java.lang.System.loadLibrary(System.java:1030)
at ivr.IVRServer.(IVRServer.java:18)

Any help would be appreciated.

Thanks.


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Voice Technology Ltda
+55 11 35882166

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Re: [Freeswitch-users] Cluecon 2009

[Rodrigo P. Telles]

Hi FS Team,

Thanks for your great work at Cluecon, we came back to Brazil with some 
good ideas (+ Snom 360 + Sangoma B600) ;-)

See you again next year!

Rodrigo Telles
Devel-IT

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Re: [Freeswitch-users] JAVA ESL

[Fernando Testa]

Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185

Testa

On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris  wrote:

> can someone post a patch to that makefile to jira.freeswitch.org please.
> Mike
>
> On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
>
> I found this same issue on my machine.
> If you could compile a esl_wrap.o then you have to generate a libesl.so
> with a cmd like this:
> g++ -shared esl_wrap.o -o libesl.so
> Then in your code, do something like this:
> /* Test.java */
> import org.freeswitch.esl.*;
>
> class Test
> {
>   public static void main(String[] args)
>   {
> System.loadLibrary("esl");
> System.out.println("hello");
>   }
> }
>
>
>
> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <
> hyppolit...@yahoo.com> wrote:
>
>>  Thank you Anthony.
>>
>> --- On *Thu, 7/30/09, Anthony Minessale *wrote:
>>
>>
>> From: Anthony Minessale 
>> Subject: Re: [Freeswitch-users] JAVA ESL
>> To: freeswitch-users@lists.freeswitch.org
>> Received: Thursday, July 30, 2009, 9:00 PM
>>
>>
>> it might be a build issue, I was not exactly sure how to build it etc.
>> so it may need some help from a java expert
>>
>> I wrote all of that with swig and never was able to test it.
>>
>>
>> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
>> hyppolit...@yahoo.com
>> > wrote:
>>
>>>   Hello,
>>>
>>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>>> run a JAVA script with the following code
>>>
>>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>>> ESLevent events = connection.getInfo();
>>> System.out.println(events.toString());
>>>
>>> I get the following error message:
>>>
>>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>>> __gxx_personality_v0
>>> at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>> at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>> at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>> at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>> at java.lang.System.loadLibrary(System.java:1030)
>>> at ivr.IVRServer.(IVRServer.java:18)
>>>
>>> Any help would be appreciated.
>>>
>>> Thanks.
>>>
>>>
>>> --
>>> The new Internet Explorer® 8 - Faster, safer, easier. Optimized for
>>> Yahoo! *Get it Now for 
>>> Free!*
>>>
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>>> 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
>> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:8...@conference.freeswitch.org
>> iax:gu...@conference.freeswitch.org/888
>> googletalk:conf+...@conference.freeswitch.org
>> pstn:213-799-1400
>>
>> -Inline Attachment Follows-
>>
>> ___
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>>
>>
>>   --
>>
>> *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark
>> your favourite sites. Download it now! 
>>
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>>
>
>
> --
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> Voice Technology Ltda
> +55 11 35882166
>
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Re: [Freeswitch-users] Eavesdrop getting killed after being answered

[Michael Collins]

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb  wrote:

>  I used to be able to dial 88+extension to eavesdrop, but now it is killed
> right after the call is answered by the extension. Can anyone tell me what I
> have done wrong?
>
>
>
> I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
> 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.
>
>
>
> http://pastebin.freeswitch.org/10025
>
>
>
Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC


> Thanks Lars
>
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Re: [Freeswitch-users] Better results from mod_vmd

[Eric des Courtis]

Matt,

You must first capture the audio beeps and verify that they are sine
waves. If not, simply tweaking the algorithm will not give you better
results.

It might be possible to use FFT and I would be happy to help you
implement such a solution but keep in mind FFT is very very demanding
on the hardware. Ideally what you want to find out is what functions
was use to generate the beep in the first place so that it can be
detected. Is it two sines waves like in DTMF? Or something more
complex?

Anyway my email is eric.des.cour...@benbria.ca.

Cheers.

Eric des Courtis

On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong wrote:
> I tried emailed Eric, seeking advice on this, but his email (the one in the
> source code) is bouncing email (invalid user), so thought I would ask here
> instead. If anyone has eric's new email address, I'd be interesting in it.
>
> I did some tests with mod_vmd this afternoon, but I'm only finding about 33%
> of the voice mail beeps and did have 1 false-positive in my test of 7 voice
> mail machines. I've recorded the audio of the session in .wav files that
> were both successful and not, as a comparison. I can upload the .wav files
> if they would be useful.
> mod_vmd works great for voicemails of Skype Users, and kall8.com, but has
> issues dealing with mobile phone carriers.
> sprint - not successful
> tmobile - not successful
> verizon - not successful
> panasonic home answering machine system - not successful
> kall8 - SUCCESS
> skype - SUCCESS
> I'm wondering if you can recommend a simple fix, like changing some of the
> constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
> file, or if better success requires more complex analysis.  Do you have any
> recommendations on how this might be done? Listening to the .wav's
> its apparent the beeps are not as loud for the mobile phone carriers as they
> are with skype and kall8. Any guidance would be greatly appreciated.
> --matt
> hello hunter
> http://www.hellohunter.com
> voice broadcasting & hosted dialer
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Re: [Freeswitch-users] Better results from mod_vmd

[Eric des Courtis]

Matt,

Okay the good news is vmd should be able to handle these cases. The
bad news is for whatever reason they are not getting detected at the
moment.

vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.

The following seem to use the same beep:

vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise

You can try to play with these values:

POINTS 32
VALID 22
MAX_CHIRP 22

If that doesn't work let me know I will try to improve the algorithm
to detect the providers.

Cheers!

Eric des Courtis



On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong wrote:
> Hi Eric,
> Thanks for the response. I had tried emailing you @brenbria.com and the
> email had bounced, thanks for responding to my mail.
> If you'd be interested I .zipped up my sample voicemail beeps
> at http://bandcon.hellohunter.com/vmd_wav.zip
> I'm relatively new to telephony, but can you point me in the right direction
> for figuring out if the beeps are sinewaves. About as far as I've come with
> audio is being able to open the .wav files in audacity. Any website
>  recommendations I can read? Thanks so much.
> --matt
>
> On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>  wrote:
>>
>> Matt,
>>
>> You must first capture the audio beeps and verify that they are sine
>> waves. If not, simply tweaking the algorithm will not give you better
>> results.
>>
>> It might be possible to use FFT and I would be happy to help you
>> implement such a solution but keep in mind FFT is very very demanding
>> on the hardware. Ideally what you want to find out is what functions
>> was use to generate the beep in the first place so that it can be
>> detected. Is it two sines waves like in DTMF? Or something more
>> complex?
>>
>> Anyway my email is eric.des.cour...@benbria.ca.
>>
>> Cheers.
>>
>> Eric des Courtis
>>
>> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong wrote:
>> > I tried emailed Eric, seeking advice on this, but his email (the one in
>> > the
>> > source code) is bouncing email (invalid user), so thought I would ask
>> > here
>> > instead. If anyone has eric's new email address, I'd be interesting in
>> > it.
>> >
>> > I did some tests with mod_vmd this afternoon, but I'm only finding about
>> > 33%
>> > of the voice mail beeps and did have 1 false-positive in my test of
>> > 7 voice
>> > mail machines. I've recorded the audio of the session in .wav files that
>> > were both successful and not, as a comparison. I can upload the .wav
>> > files
>> > if they would be useful.
>> > mod_vmd works great for voicemails of Skype Users, and kall8.com, but
>> > has
>> > issues dealing with mobile phone carriers.
>> > sprint - not successful
>> > tmobile - not successful
>> > verizon - not successful
>> > panasonic home answering machine system - not successful
>> > kall8 - SUCCESS
>> > skype - SUCCESS
>> > I'm wondering if you can recommend a simple fix, like changing some of
>> > the
>> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
>> > file, or if better success requires more complex analysis.  Do you have
>> > any
>> > recommendations on how this might be done? Listening to the .wav's
>> > its apparent the beeps are not as loud for the mobile phone carriers as
>> > they
>> > are with skype and kall8. Any guidance would be greatly appreciated.
>> > --matt
>> > hello hunter
>> > http://www.hellohunter.com
>> > voice broadcasting & hosted dialer
>> > ___
>> > 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
>> >
>> >
>
>

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[Freeswitch-users] G729 transcoding workaround

[Justin Miller]

To overcome the G729 transcoding issue with voicemail, I'm using an Audiocodes 
Mediant 1000 for transcoding. Our SIP trunk provider and all of our phones use 
G729 exclusively.
When a call needs to go to voicemail, the call is bridged to the M1000, which 
transcodes to G711, and returns the call to Freeswitch on another port (5090). 
This seems to be working well.

I'm now working on getting the IVR working so that I can start using the PBX 
functionality of Freeswitch for our office. When a call needs to hit the IVR, 
it's bridged to the M1000 the same as voicemail. I hear the custom menu I've 
created, and I can choose options successfully, except that the call is now 
operating in G711. When I bridge to a phone, I get the transcoding error. One 
option would be to transcode through the M1000 yet again, but this would take 
up 2 more DSP sessions in the M1000, and I would be running a total of 6 call 
legs in Freeswitch for this one call.

Is there a way to end the transcoded call legs, and bridge to the phone from 
the original call leg? This would free up the M1000, and just seems like a 
better way to do things.

Thank you
Justin

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[Freeswitch-users] Eavesdrop getting killed after being answered

[Lars Zeb]

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Thanks Lars

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Re: [Freeswitch-users] JAVA ESL

[Michael Jerris]

can someone post a patch to that makefile to jira.freeswitch.org please.

Mike

On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:


I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a  
libesl.so with a cmd like this:

g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;

class Test
{
  public static void main(String[] args)
  {
System.loadLibrary("esl");
System.out.println("hello");
  }
}



On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite > wrote:

Thank you Anthony.

--- On Thu, 7/30/09, Anthony Minessale   
wrote:


From: Anthony Minessale 
Subject: Re: [Freeswitch-users] JAVA ESL
To: freeswitch-users@lists.freeswitch.org
Received: Thursday, July 30, 2009, 9:00 PM


it might be a build issue, I was not exactly sure how to build it etc.
so it may need some help from a java expert

I wrote all of that with swig and never was able to test it.


On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite > wrote:

Hello,

I built libesl and JAVA mod. (make and make javamod). But when I try  
to run a JAVA script with the following code


ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
ESLevent events = connection.getInfo();
System.out.println(events.toString());

I get the following error message:

Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0

at java.lang.ClassLoader$NativeLibrary.load(Native Method)
at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
at java.lang.Runtime.loadLibrary0(Runtime.java:823)
at java.lang.System.loadLibrary(System.java:1030)
at ivr.IVRServer.(IVRServer.java:18)

Any help would be appreciated.

Thanks.


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Re: [Freeswitch-users] JAVA ESL

[Fernando Testa]

I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a libesl.so with
a cmd like this:
g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;

class Test
{
  public static void main(String[] args)
  {
System.loadLibrary("esl");
System.out.println("hello");
  }
}



On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite  wrote:

> Thank you Anthony.
>
> --- On *Thu, 7/30/09, Anthony Minessale *wrote:
>
>
> From: Anthony Minessale 
> Subject: Re: [Freeswitch-users] JAVA ESL
> To: freeswitch-users@lists.freeswitch.org
> Received: Thursday, July 30, 2009, 9:00 PM
>
>
> it might be a build issue, I was not exactly sure how to build it etc.
> so it may need some help from a java expert
>
> I wrote all of that with swig and never was able to test it.
>
>
> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
> hyppolit...@yahoo.com
> > wrote:
>
>>   Hello,
>>
>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>> run a JAVA script with the following code
>>
>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>> ESLevent events = connection.getInfo();
>> System.out.println(events.toString());
>>
>> I get the following error message:
>>
>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>> __gxx_personality_v0
>> at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>> at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>> at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>> at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>> at java.lang.System.loadLibrary(System.java:1030)
>> at ivr.IVRServer.(IVRServer.java:18)
>>
>> Any help would be appreciated.
>>
>> Thanks.
>>
>>
>> --
>> The new Internet Explorer® 8 - Faster, safer, easier. Optimized for Yahoo!
>> *Get it Now for Free!* 
>>
>> ___
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>
>
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>
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>
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Re: [Freeswitch-users] ext sip & rtp IP trouble

[kokoska rokoska]

Let me apologize to waste your time, Brian (and others too).

I'm still not 100% sure where problem is, but I'm nearly sure it is not
related to FreeSWITCH.

I found similar problems with SMB - with IPv6 active some clients (even
with only IPv4 stack) can't connect to server. Disabling IPv6 solves the
issue => it has to be network problem...

Thank you very very much for your help!

Best regards,

kokoska.rokoska



kokoska rokoska napsal(a):
> 
> 
> Brian West napsal(a):
>> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
>>
>> /b
>>
> 
> OK, I do it.
> But wait few minutes, please:
> 
> 1. I should asleep my chidren :-)
> 2. Before a while I discovered that if I disable IPv6 networking in my
> CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
> => it looks like it may be network layer/config (iptables, ip6tables,
> roueting etc.) error not related to FS.
> 
> I'll be back in an hour or so and let you know what is current status.
> 
> Many thanks, Brian, for your valuable help.
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
>> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
>>
>>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>>
>>> If you be so kind to look at that machine, I'll be very glad.
>>> I still think it will be some trivial, stupid, misconfiguration, but
>>> can't find what I did wrong. Especially if all worked fine before FS
>>> update...
>>> What public ssh key should I install (shinzon, freeswitch) or should I
>>> get another one?
>>>
>>> Thanks once more for your help!
>>>
>>> Best regards,
>>>
>>> kokoska.rokoska
>>
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Re: [Freeswitch-users] ext sip & rtp IP trouble

[kokoska rokoska]

Brian West napsal(a):
> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
> 
> /b
> 

OK, I do it.
But wait few minutes, please:

1. I should asleep my chidren :-)
2. Before a while I discovered that if I disable IPv6 networking in my
CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
=> it looks like it may be network layer/config (iptables, ip6tables,
roueting etc.) error not related to FS.

I'll be back in an hour or so and let you know what is current status.

Many thanks, Brian, for your valuable help.

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
> 
>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>
>> If you be so kind to look at that machine, I'll be very glad.
>> I still think it will be some trivial, stupid, misconfiguration, but
>> can't find what I did wrong. Especially if all worked fine before FS
>> update...
>> What public ssh key should I install (shinzon, freeswitch) or should I
>> get another one?
>>
>> Thanks once more for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
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Re: [Freeswitch-users] ext sip & rtp IP trouble

[Brian West]

shinzon.pub and get on IRC and msg bkw_ and i'll take a look.

/b

On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:

>
> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>
> If you be so kind to look at that machine, I'll be very glad.
> I still think it will be some trivial, stupid, misconfiguration, but
> can't find what I did wrong. Especially if all worked fine before FS
> update...
> What public ssh key should I install (shinzon, freeswitch) or should I
> get another one?
>
> Thanks once more for your help!
>
> Best regards,
>
> kokoska.rokoska


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Re: [Freeswitch-users] ext sip & rtp IP trouble

[kokoska rokoska]

Brian West napsal(a):
> if all else fails get me access to the machine please.
> 
> /b
> 

I'm sorry, but even after fresh svn checkout all goes wrong :-)

If you be so kind to look at that machine, I'll be very glad.
I still think it will be some trivial, stupid, misconfiguration, but
can't find what I did wrong. Especially if all worked fine before FS
update...
What public ssh key should I install (shinzon, freeswitch) or should I
get another one?

Thanks once more for your help!

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:
> 
>> Still no luck - no RTPs from FreeSWITCH.
>> For sure I'll make fresh svn checkout in a minute and let you know  
>> if it
>> helps...
>>
>> Thank you very much for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
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Re: [Freeswitch-users] how to set different action for different cause code

[Michael Collins]

On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson wrote:

> Hello,
>
> I find hangup_hook, but I would like to define different actions for
> different hangup codes.  Is there anyway to do that?
>
>

I can think of at least two ways you could do this: one that uses only the
dialplan and one that uses a script. If you don't mind using a scripting
language then you can make it very clean:



Then have your Lua script handle all the if-then-else or case stuff.

Question: are you trying to transfer the a-leg to some other destination if
the b-leg hangup is a specific cause, or are you just doing some external
cleanup stuff? Just curious...

-MC


> Woody
>
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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

[Michael Jerris]

On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:


hi MikeJ,
i prefer creating MSI file that is easy to mintin$


How is this useful vs. something that is already maintained?

unstid of using inno setup or advanced installer (not free), we can  
use WIX (Windows installer XML) that is a open source one


All the tools being used are free, however all are not open source.

we can create a customised MSI that fully install mor features,  
including Sounds / MOH/...


This installer already does all that and more.

and we can edit XML files easyly to let users chouse each module to  
install during setup, for example conferencing, voice mail, Event  
Socket and ...

also the WAMP server is not fully stable
i prefer Uniform Server, that mintin Full Compatibility with Unix  
like platform including for example, perl files that need this  
path: /usr/...
apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql  
and php is in /usr/local/php

and perl is in /usr/bin
php my admin is in /etc/phpmyadmin
and mor featurs, including multiple instance, virtual hosting and  
full management using control pannel
this will help enterprise users that need that in windows unstid of  
WAMP

 ---
if you don't want to mintin it, jive me a chance to mintin it
  ---
thanks


I just don't want to split efforts here, every different way we do  
this is yet another way we have to support.  I see no compelling  
argument here other than personal preference.  If I am missing  
anything someone please chime in and correct me.


Mike


Contact Me


SIP
INUM
PSTN
Email

Michael Jerris wrote:


I think sticking with standard WAMP is preferable.  What is the  
advantage to creating yet another installer over the one that we  
have already done and maintained?


Mike

On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:


hello
i'm rewriting this executable file in MSI format
i can use:
MakeMSI
WIX (Windows installer XML)
what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called  
uniform server
this server have a conventional Unix Path configuration, for  
example, web files can by stored in /www/ and apache is in /usr/ 
local/apache2

this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:


http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe






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Re: [Freeswitch-users] Transporting SIP over TCP

[Brian West]

Well if you append ;transport=tcp on the bridge lines it will use TCP .

/b

On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:

>> FreeSWITCH works very well as a client :P
> I am currently porting it into iPhone and Symbian, I am almost  
> done ;-)
>
> anyway, seriously now, can one point to a wiki page about this?
> How do I do that?
> I would need 3 server instances to place a call, right?


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Re: [Freeswitch-users] ext sip & rtp IP trouble

[Brian West]

if all else fails get me access to the machine please.

/b

On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:

> Still no luck - no RTPs from FreeSWITCH.
> For sure I'll make fresh svn checkout in a minute and let you know  
> if it
> helps...
>
> Thank you very much for your help!
>
> Best regards,
>
> kokoska.rokoska


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Re: [Freeswitch-users] Transporting SIP over TCP

[Tzury Bar Yochay]

> FreeSWITCH works very well as a client :P
I am currently porting it into iPhone and Symbian, I am almost done ;-)

anyway, seriously now, can one point to a wiki page about this?
How do I do that?
I would need 3 server instances to place a call, right?

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Re: [Freeswitch-users] ext sip & rtp IP trouble

[kokoska rokoska]

Brian West napsal(a):
> Update, Math refactored the functions that caused some of the problems.
> 
> /b
> 

Still no luck - no RTPs from FreeSWITCH.
For sure I'll make fresh svn checkout in a minute and let you know if it
helps...

Thank you very much for your help!

Best regards,

kokoska.rokoska



> On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:
> 
>> After some testing I found following:
>>
>> 1. When I try to bridge call between endpoints, there are always no  
>> RTPs
>> from FreeSWITCH (like I wrote before).
>> 2. When I just answer call on FS and play something localy (i.e.  
>> music)
>> everything works fine - FS sends RTPs like expected.
>>
>> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
>> and from public IPs, private IPs at foreign LAN, private IPs at the  
>> same
>> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
>> same LAN). And I found no difference.
>>
>> Any hint is really appreciated :-)
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
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Re: [Freeswitch-users] Transporting SIP over TCP

[Brian West]

FreeSWITCH works very well as a client :P

/b

On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:

> my conclusion is then:
> we are living in a sad world where there is no a production ready,
> cross platform, fully compliant, decent and open source sip client


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Re: [Freeswitch-users] SIPGATE Problem

[Brian West]

Well if you pay attention I told you in the last email... set the  
param from-domain on the gateway to sipgate.de


/b

On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:



Hi,

i have just taken some pictures.

http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG

the third picture is taken by a softphone that works.

To Brian:
If i set the effective_caller_id_name to 2395805 without @sipgate.de  
it´s

the same problem, because the freeswitch set in the from fielt:
2395...@139.13.37.160. And i can´t find the right parameter to change
"139.13.37.160" to "sipgate.de"

Which is the right parameter in the dialplan?



Thanks for your help



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Re: [Freeswitch-users] Transporting SIP over TCP

[Tzury Bar Yochay]

> They make you pay for TCP which is weird since the RFC says TCP is
> required.
>
> /b

my conclusion is then:
we are living in a sad world where there is no a production ready,
cross platform, fully compliant, decent and open source sip client

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Re: [Freeswitch-users] SIPGATE Problem

[NOx-WHV]

Hi,

i have just taken some pictures. 

http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG 
http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG 
http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG 

the third picture is taken by a softphone that works. 

To Brian: 
If i set the effective_caller_id_name to 2395805 without @sipgate.de it´s
the same problem, because the freeswitch set in the from fielt:
2395...@139.13.37.160. And i can´t find the right parameter to change
"139.13.37.160" to "sipgate.de"

Which is the right parameter in the dialplan?



Thanks for your help





Brian West-3 wrote:
> 
> You need to set from-domain on the gateway.  And set the  
> effective_caller_id_number to just the  number not the num...@host.
> 
> /b
> 
> On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have a problem using my freeswitch with a sipgate account. The  
>> gateway
>> entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
>> answer with
>> "403" (forbidden). I guess the string is incorrect.
>>
>> I modify the "effective_caller_id_number" to  
>> accountnum...@sipgate.de but
>> the trace show the strting from:
>> sip:accountnum...@sipgate.de@my-ip-address
>>
>> How can i modify the string to sip:accountnum...@sipgate.de without my
>> ip-address.
> 
> 
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Re: [Freeswitch-users] Transporting SIP over TCP

[Brian West]

They make you pay for TCP which is weird since the RFC says TCP is  
required.

/b

On Aug 17, 2009, at 7:18 AM, bakko wrote:

> Y make my tests with eyebeam.
>
> I thing X-lite dont't support TCP transport.
>
> BR


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Re: [Freeswitch-users] ext sip & rtp IP trouble

[Brian West]

Update, Math refactored the functions that caused some of the problems.

/b

On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:

>
> After some testing I found following:
>
> 1. When I try to bridge call between endpoints, there are always no  
> RTPs
> from FreeSWITCH (like I wrote before).
> 2. When I just answer call on FS and play something localy (i.e.  
> music)
> everything works fine - FS sends RTPs like expected.
>
> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
> and from public IPs, private IPs at foreign LAN, private IPs at the  
> same
> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
> same LAN). And I found no difference.
>
> Any hint is really appreciated :-)
>
> Best regards,
>
> kokoska.rokoska


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Re: [Freeswitch-users] SIPGATE Problem

[Brian West]

You need to set from-domain on the gateway.  And set the  
effective_caller_id_number to just the  number not the num...@host.


/b

On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:



Hi,

i have a problem using my freeswitch with a sipgate account. The  
gateway
entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
answer with

"403" (forbidden). I guess the string is incorrect.

I modify the "effective_caller_id_number" to  
accountnum...@sipgate.de but

the trace show the strting from: sip:accountnum...@sipgate.de@my-ip-address

How can i modify the string to sip:accountnum...@sipgate.de without my
ip-address.


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Re: [Freeswitch-users] SIPGATE Problem

[Chris Chen]

Hi, could you please check the destination number in your dial string? If it
is the right format, one of the reasons could be that number is not in
service
when you get "403" response from the SIP gateway.

Thanks,

Chris

On Mon, Aug 17, 2009 at 8:10 AM, NOx-WHV  wrote:

>
> Hi,
>
> i have a problem using my freeswitch with a sipgate account. The gateway
> entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer
> with
> "403" (forbidden). I guess the string is incorrect.
>
> I modify the "effective_caller_id_number" to accountnum...@sipgate.de but
> the trace show the strting from: 
> sip:accountnum...@sipgate.de
> @my-ip-address
>
> How can i modify the string to 
> sip:accountnum...@sipgate.dewithout my
> ip-address.
>
>
> --
> View this message in context:
> http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Transporting SIP over TCP

[bakko]

Y make my tests with eyebeam.

I thing X-lite dont't support TCP transport.

BR

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[Freeswitch-users] SIPGATE Problem

[NOx-WHV]

Hi,

i have a problem using my freeswitch with a sipgate account. The gateway
entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer with
"403" (forbidden). I guess the string is incorrect.

I modify the "effective_caller_id_number" to accountnum...@sipgate.de but
the trace show the strting from: sip:accountnum...@sipgate.de@my-ip-address

How can i modify the string to sip:accountnum...@sipgate.de without my
ip-address.


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View this message in context: 
http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
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Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

[Meftah Tayeb]

hi MikeJ,
i prefer creating MSI file that is easy to mintin$
unstid of using inno setup or advanced installer (not free), we can use 
WIX (Windows installer XML) that is a open source one
we can create a customised MSI that fully install mor features, 
including Sounds / MOH/...
and we can edit XML files easyly to let users chouse each module to 
install during setup, for example conferencing, voice mail, Event Socket 
and ...

also the WAMP server is not fully stable
i prefer Uniform Server, that mintin Full Compatibility with Unix like 
platform including for example, perl files that need this path: /usr/...
apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql and 
php is in /usr/local/php

and perl is in /usr/bin
php my admin is in /etc/phpmyadmin
and mor featurs, including multiple instance, virtual hosting and full 
management using control pannel

this will help enterprise users that need that in windows unstid of WAMP
---
if you don't want to mintin it, jive me a chance to mintin it
 --- 
thanks



   Contact Me


   * SIP 
   * INUM 
   * PSTN 
   * Email 


Michael Jerris wrote:
I think sticking with standard WAMP is preferable.  What is the 
advantage to creating yet another installer over the one that we have 
already done and maintained?


Mike

On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:


hello
i'm rewriting this executable file in MSI format
i can use:

* MakeMSI
* WIX (Windows installer XML)

what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called 
uniform server
this server have a conventional Unix Path configuration, for example, 
web files can by stored in /www/ and apache is in /usr/local/apache2

this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:

http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe





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Re: [Freeswitch-users] Transporting SIP over TCP

[Jason White]

Tzury Bar Yochay  wrote:
 
> I think I confused you with this TLS/TCP thing.
> For the sake of clarification, I am talking about TCP and _not_ about TLS.
> That is simply transporting the signaling packets over TCP instead of UDP.
> No TLS should be involved at this stage. It is simply communication layer
> matter.

I can confirm that transport=tcp works fine here from one FreeSWITCH to
another. (I'm doing it over IPv6, but that shouldn't make any difference).


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Re: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 & v1.0.4)

[Bruce McAlister]

Hi All,

Shall I log a JIRA for this issue?

Thanks
Bruce

Bruce McAlister wrote:
> Hi All,
> 
> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
> 1.0.4.
> 
> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
> 
> The build fails with:
> 
> --- snip ---
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> ---
> 
> Looking back through the build I can see the following error:
> 
> --- snip ---
> creating libfreeswitch.la
> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
> libfreeswitch.la)
> /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
> freeswitch-switch.o  ./.libs/libfreeswitch.so 
> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib
>  
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a
>  
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
> Undefined   first referenced
>   symbol in file
> herror  ./.libs/libfreeswitch.so
> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
> *** Error code 1
> The following command caused the error:
> `if test -z "" ; then echo /bin/bash 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
> --tag=CC --mode=link /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
> --- snip ---
> 
> Then a little above this error, there is the following warning that is 
> displayed (I'm not sure if it is related):
> 
> --- snip ---
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> --- snip ---
> 
> My configure line is as follows:
> 
> ---
> ./configure --prefix=/opt/freeswitch
> ---
> 
> I have the complete configure and make output if anyone needs them.
> 
> Any help/pointers would be greatly appreciated.
> 
> Thanks
> Bruce
> 
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> 

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Re: [Freeswitch-users] Transporting SIP over TCP

[Tzury Bar Yochay]

> I'm not sure whether that's supposed to use TLS. I suspect not.


Jason,

I think I confused you with this TLS/TCP thing.
For the sake of clarification, I am talking about TCP and _not_ about TLS.
That is simply transporting the signaling packets over TCP instead of UDP.
No TLS should be involved at this stage. It is simply communication layer
matter.
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Re: [Freeswitch-users] Transporting SIP over TCP

[Jason White]

Tzury Bar Yochay  wrote:
 
> Well, as I said at the beginning of this thread, TLS works fine for me.
> The problem is when using TCP (transport=tcp and not transport=tls)

I'm not sure whether that's supposed to use TLS. I suspect not.


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Re: [Freeswitch-users] Transporting SIP over TCP

[Tzury Bar Yochay]

> I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
> FreeSWITCH itself as the softphone).

Well, as I said at the beginning of this thread, TLS works fine for me.
The problem is when using TCP (transport=tcp and not transport=tls)

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Re: [Freeswitch-users] Transporting SIP over TCP

[Jason White]

Tzury Bar Yochay  wrote:
> Brian/Bakko,
> 
> Would you please tell me which softphone are you using?
> As you know, my own one is not working and when I tried tcp with xlite
> (providing transport=tls) I see in wireshark that it is still
> transporting it over udp(!)

I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
FreeSWITCH itself as the softphone).

Snom 320 phones are known to work as well if you set up SRV records for the
TLS.

I haven't tried other softphones because, basically, FreeSWITCH is a better
phone than anything else I can find.

To debug this, try wireshark or tshark to find out whether your softphone is
trying to connect over the TLS port at all.


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Re: [Freeswitch-users] Transporting SIP over TCP

[Tzury Bar Yochay]

Brian/Bakko,

Would you please tell me which softphone are you using?
As you know, my own one is not working and when I tried tcp with xlite
(providing transport=tls) I see in wireshark that it is still
transporting it over udp(!)

thanks allot in advance,
Tzury

On Sun, Aug 16, 2009 at 5:11 PM, Brian West  wrote:
>
> Works fine here.
>
> /b
>
> On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:
>
> > Hi Brian,
> >
> > Just for your information here is a mail I got from a colleague of
> > mine which I consider as an experienced freeswitch integrator
> >
> >> "... TCP used to work, and I had about 20-30 client phones
> >> connecting with it.
> >> About 3 months ago I did an upgrade and TCP no longer worked.  I
> >> was in panic mode, so I just ended up walking my
> >> clients through reconfiguring their phones to use UDP.
> >>
> >> It is definitely a bug in Freeswitch.  I can help you work up a
> >> jira bug report if you like.
> >> The bug happened somewhere between svn version 9800ish and 13223.
> >> 13223 is the version I'm running on my systems now, older, but
> >> stable for me..."
> >
> > It would be just great if you can confirm the TCP functionality in the
> > latest release of FreeSwitch
> >
> >
> > /t
> >
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>
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--
Tzury Bar Yochay

Regulus Labs ltd.
http://reguluslabs.com
+972 52 5133399

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