From: freeswitch-users-request@lists.freeswitch.org <freeswitch-users-request@lists.freeswitch.org>
Subject: Freeswitch-users Digest, Vol 32, Issue 39
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, February 4, 2009, 2:05 PM

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Today's Topics:

   1. Re: SIP Authentication (Brian West)
   2. Re: origainate through sofia gateway (Michael Collins)
   3. Recording background music and voice is out of	sync (Daniel Liang)
   4. Re: Q931 decoding Update (Gopalakrishnan A.N)
   5. mod_limit (Chav Paskov)
   6. Re: mod_limit (Michael Collins)
   7. Re: mod_limit (Chav
 Paskov)
   8. Re: mod_limit (Michael Collins)


----------------------------------------------------------------------

Message: 1
Date: Wed, 4 Feb 2009 10:52:45 -0600
From: Brian West <brian@freeswitch.org>
Subject: Re: [Freeswitch-users] SIP Authentication
To: freeswitch-users@lists.freeswitch.org
Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org>
Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes

What client is this?  I also notice we receive port 3458 and reply to  
port 1059...

/b

On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote:

> What I have noted is that the client does not send the values for  
> "cnonce" and "nc" in the response. I'm not sure if
this is the  
> reason, however how this problem can be solved?
>
> Thanks,
>
> Ali




------------------------------

Message:
 2
Date: Wed, 4 Feb 2009 09:41:07 -0800
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] origainate through sofia gateway
To: freeswitch-users@lists.freeswitch.org
Message-ID:
	<87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

I'll make sure the substance of this is in the wiki and I'll look for
references to the deprecated way and remove those.
-MC

On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:
> Where did you learn how to use js this way?
> session.originate is being misused here and is depricated and may be
> removed.
>
> the first arg to session.originate is either undefined or a *different*
> session (the a leg)
>
> session1 = new Session();
> session1.originate(undefined,
>
 "{ignore_early_media=true}user/1008@192.168.1.122");
>
>
session1.setVariable("effective_caller_id_number","fixed0248b");
>
> //once you have session1 when you originate session2 you pass session1 as
> the arg
> // the effective_caller_id is taken from session1
>
> session2 = new Session();
> session2.originate(session1,
"sofia/gateway/halonet/0225490317");
>
> Anyway this whole code is depricated in favor of this:
>
> session1 = new
Session("{ignore_early_media=true}user/1008@192.168.1.122");
> if (session1.ready()) {
>  
session1.setVariable("effective_caller_id_number","fixed0248b");
>   session2 = new Session("sofia/gateway/halonet/0225490317",
session1);
> }
>
> and could be further refactored down to this:
>
> session1 = new
Session("{ignore_early_media=true}user/1008@192.168.1.122");
> if
 (session1.ready()) {
>  
session1.setVariable("effective_caller_id_number","fixed0248b");
>   session1.execute("bridge",
"sofia/gateway/halonet/0225490317");
> }
>
> or down to this one line of code that will setup the call detached from
the
> script and exit.
>
> var result = apiExecute("originate",
>
"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122
> bridge:sofia/gateway/halonet/0225490317 inline");
>
> if you dont care about the result and want to exit even before the call is
> completed.
>
> var result = apiExecute("bgapi", "originate
>
{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122
> bridge:sofia/gateway/halonet/0225490317 inline");
>
>
>
> On Wed, Feb 4, 2009 at
 2:51 AM, Jacek Sokulski
<jsokulski@dotsystems.pl>
> wrote:
>>
>> We have tried setting both effective_caller_id_number and
>> origination_caller_id_number:
>>
>>
>>
session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15);
>>  but the problem still exists. The solution we have found for the case
>> when we originate two calls, local and external, is as follow:
>>
>> session1 = new Session();
>>
session1.originate(session1,"user/1003@192.168.1.122",15);//local
>> if(session1.ready()) {
>>    session1.execute("execute_extension","00930691688627
XML
>> default");//external
>> }
>>
>> so the external call goes through the dialplan.
>> It does not work if both calls are external. One possible solution
could
>>
 be
>> to pass the originating call through dialplan (loopback?) but we have
not
>> managed
>> to figure out how to do it.
>>
>> Thanks
>> Jacek
>>
>> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze:
>> > Oops! Well, fortunately I don't use that voip provider
anymore (nor the
>> > script).
>> >
>> > Thanks Brian.
>> >
>> > Nicolas
>> >
>> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West
<brian@freeswitch.org> wrote:
>> > > YOU should NEVER use this method or call setCallerData at
all  you
>> > > should use the correct methods to override the callerid.
>> > >
>> > > If its a B-Leg born from an A-Leg you use these on the on
the A-Leg:
>> > >
>> > >
>> >
 >
http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name
>> > >
>> > >
http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number
>> > >
>> > > If you're originating you use this:
>> > >
>> > >
>> > >
http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name
>> > >
>> > >
http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number
>> > >
>> > > /b
>> >
>> > _______________________________________________
>> > Freeswitch-users mailing list
>> > Freeswitch-users@lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
 >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale@hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888@conference.freeswitch.org
>
 iax:guest@conference.freeswitch.org/888
> googletalk:conf+888@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
>
>



------------------------------

Message: 3
Date: Wed, 4 Feb 2009 09:43:10 -0800
From: "Daniel Liang" <Daniell@airg.com>
Subject: [Freeswitch-users] Recording background music and voice is
	out of	sync
To: <freeswitch-users@lists.freeswitch.org>
Message-ID:
	<0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg>
Content-Type: text/plain; charset="us-ascii"

 What I did was the following:
 
First, I sent the
 playback command:
 
call-command: execute
execute-app-name: playback
execute-app-arg: <filename>
 
Then I send uuid_record (Sorry, it was not Record command):
 
api uuid_record <uuid> start <filename> 120
 
I also tried replacing the playback command with:
api uuid_displace <uuid> start <filename> 0 mux
 
But the end results are the same. The recorded user's voice is about 0.5
second behind the expected result.
 
Thanks,
Daniel
 
 
-----Original Message-----
From: freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
Brian West
Sent: February 3, 2009 6:36 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Recording background music and voice is
outof sync
 
Can you show us an example of how you're doing this?  Playback and
Record aren't async so
 you'll need to show us how you're doing
this.
 
Also don't hijack threads you hit replay on the one "Re: [Freeswitch-
users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted
the
subject and started a new body.  That hijacks the thread and that can
cause your problem to go ignored in some cases if people aren't
interested in the thread topic depending on how their reader threads the
emails.
 
Please click new message and type freeswitch- users@lists.freeswitch.org
in and then input your subject and body to start a new thread.
 
Thanks,
Brian West
FreeSWITCH.org
 

On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote:
 
> Hi,
>
> I was trying to record a background music with a user's voice at the 
> same time. I did a playback and started recording. But the recorded 
> user's voice and the background music is about 0.5 second out of sync.

> I also tried
 to use uuid_displace instead of playback, but I got the 
> same result.
 

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------------------------------

Message: 4
Date: Wed, 4 Feb 2009 23:26:14 +0530
From: "Gopalakrishnan A.N" <saigop@gmail.com>
Subject: Re: [Freeswitch-users] Q931 decoding Update
To: freeswitch-users@lists.freeswitch.org
Message-ID:

	<2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi,
    Its  a awesome. Can the packet capturing be done with event socket?

-- 
Thank you  with regards,
Gopal,
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------------------------------

Message: 5
Date: Wed, 04 Feb 2009 09:59:48 -0800
From: Chav Paskov <chavpaskov@shaw.ca>
Subject: [Freeswitch-users] mod_limit
To: freeswitch-users@lists.freeswitch.org
Message-ID: <4989D794.1010805@shaw.ca>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi ,
is it possible to use mod_limit in case if the end point is not 
registered / gateway for
 example/.
Regards
Chav



------------------------------

Message: 6
Date: Wed, 4 Feb 2009 10:06:52 -0800
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] mod_limit
To: freeswitch-users@lists.freeswitch.org
Message-ID:
	<87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca> wrote:
> Hi ,
> is it possible to use mod_limit in case if the end point is not
> registered / gateway for example/.

Could you add some detail to this question? What are you trying to do?
(mod_limit may or may not work, but there might be another solution
which is why I am asking.)

-MC

> Regards
> Chav
>
> _______________________________________________
> 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
>



------------------------------

Message: 7
Date: Wed, 04 Feb 2009 10:54:56 -0800
From: Chav Paskov <chavpaskov@shaw.ca>
Subject: Re: [Freeswitch-users] mod_limit
To: freeswitch-users@lists.freeswitch.org
Message-ID: <4989E480.1080105@shaw.ca>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Michael Collins wrote:
> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca>
wrote:
>   
>> Hi ,
>> is it possible to use mod_limit in case if the end point is not
>> registered / gateway for example/.
>>     
>
> Could you add some detail to this question? What are you trying to do?
>
 (mod_limit may or may not work, but there might be another solution
> which is why I am asking.)
>
> -MC
>
>   
>> Regards
>> Chav
>>
>> _______________________________________________
>> 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
>
>   
i have few gateways under my ACL that 
 are allowed to send calls to FS, 
but i want to be able to enforce "capacity" policy  on the traffic 
coming from any one of them depending on total termination capacity on  
my termination end.
Let say GW 1 has to be limited to make  10 simultaneous calls  while GW2 
could make up to 30 and so on.
Regards
Chav



------------------------------

Message: 8
Date: Wed, 4 Feb 2009 11:05:09 -0800
From: Michael Collins <msc@freeswitch.org>
Subject: Re: [Freeswitch-users] mod_limit
To: freeswitch-users@lists.freeswitch.org
Message-ID:
	<87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1

On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca> wrote:
> Michael Collins wrote:
>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca>
wrote:
>>
>>> Hi
 ,
>>> is it possible to use mod_limit in case if the end point is not
>>> registered / gateway for example/.
>>>
>>
>> Could you add some detail to this question? What are you trying to do?
>> (mod_limit may or may not work, but there might be another solution
>> which is why I am asking.)
>>
>> -MC
>>
>>
>>> Regards
>>> Chav
>>>
>>> _______________________________________________
>>> 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
>>
>>
> i have few gateways under my ACL that  are allowed to send calls to FS,
> but i want to be able to enforce "capacity" policy  on the
traffic
> coming from any one of them depending on total termination capacity on
> my termination end.
> Let say GW 1 has to be limited to make  10 simultaneous calls  while GW2
> could make up to 30 and so on.

I'm sure that this is possible. I don't personally have a way to test
all of this but I know that a number of our users are doing things
like this currently. Can you hop on to the IRC channel? #freeswitch on
irc.freenode.net. A lot of people there can help with
 this one.

-MC (IRC: mercutioviz)

> Regards
> Chav
>
> _______________________________________________
> 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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End of Freeswitch-users Digest, Vol 32, Issue 39
************************************************