Am 04.11.2013 11:06 schrieb <freeswitch-users-request@lists.freeswitch.org>:

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Today's Topics:

   1. SIMPLE IM in UTF-16 (Alex Chan)
   2. Re: Raspberry Pi, mod_gsmopen GSM dongle (Jayanth Acharya)
   3. Re: 481 Call Does Not Exist (Miha)
   4. Re: Scalability and Bridge (DanB)
   5. Skype is likely to stop supporting desktop API    by Dec. 2013
      (Henry Huang)
   6. Re: Scalability and Bridge (Ben Langfeld)
   7. Re: Execute_on_answer and ${strepoch} (Gregor Nanger)


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Message: 1
Date: Mon, 4 Nov 2013 09:46:32 +0800
From: Alex Chan <alex@slackadmin.com>
Subject: [Freeswitch-users] SIMPLE IM in UTF-16
To: FreeSWITCH-users@lists.freeswitch.org
Message-ID:
        <CAG=KnHW2BkP1k-PXGt4N8-t5DCH5Kye8bNtM+aqyauiMZtvtug@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Hi folks,

I'm using FreeSWITCH as a SIP IM B2BUA. When the content-type was in
UTF-16BE, the message body will be dammed at where the zero byte 0x00 first
appears. Any ideas?

Thanks in advance.

Alex
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Message: 2
Date: Mon, 4 Nov 2013 08:25:57 +0530
From: Jayanth Acharya <jayachar88@gmail.com>
Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Message-ID:
        <CAHDL-swz9nR7Cz-p3pbB_rg69JDDp2BSKz4yRR6tdDWWv_cakQ@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Afraid that this may not have much to do with the distro and the weak-link
may indeed be mod_gsmopen. Have experienced similar things on regular PC on
Debian based systems. The chan_dongle development is a bit more mainstream
and bit more active. More importantly - the developers seem to be actively
using it. It is a shame that mod_gsmopen isn't getting more development...
The working of module that works with 3G dongles isn't complex, especially
after one understands AT commands. However without proficiency in general
FS development, it is hard to contribute meaningfully.
On Nov 4, 2013 6:52 AM, "Steve Underwood" <steveu@coppice.org> wrote:

> On 10/29/2013 03:05 PM, Sergey Zhuravlov wrote:
> > Hi!
> >
> > Does somebody have a positive experience with Raspberry Pi,
> > mod_gsmopen GSM dongle ???
> > I use Raspberry PiB and e1550.
> >
> > It's a shame that with an asterisk all works fine -- RasPBX.
> > http://www.raspberry-asterisk.org/
> >
> > But I prefer the FreeSWITCH ;-)
> > On the basis of experience, I thought that PS will work best for this
> > modest hardware faster and require fewer resources.
> >
> > But it turns out it is not. Asterisk with a web server, MySQL
> > database, and other, less load than FS.
> > In this connection, other parameters may be used starting FS to save
> > resources?
> >
> > And the main question! Silence an incoming call on 5000 (IVR) as well
> > as an outgoing in gsm network. Those are not sound.
> >
> This may have nothing to do with FreeSwitch. Is the same Linux distro
> being used for Asterisk and FreeSwitch? It makes a huge difference on a
> Raspberry Pi. The obvious performance difference is that some distros
> use software floating point, and some use hardware floating point. Its
> more that, though. When I tried various distributions I found compile
> times vary by a factor of 5 or more.
>
> Regards,
> Steve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
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> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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Message: 3
Date: Mon, 04 Nov 2013 08:43:34 +0100
From: Miha <miha@softnet.si>
Subject: Re: [Freeswitch-users] 481 Call Does Not Exist
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Message-ID: <52775026.6040000@softnet.si>
Content-Type: text/plain; charset=windows-1252; format=flowed

Hi,

could some please help me with figuring out why FS sends 481.

tnx

miha

Dne 10/24/2013 9:47 AM, pi?e Miha:
> Hi,
>
> I need a little help with figuring out why FS sends "481".
>
> This is happing when call has been forward (302).
>
> All calls are relayed throught opensips as I am using it for
> registrations and load_balancing.
>
> http://pastebin.freeswitch.org/21555
>
> tnx!
>
> miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>




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

Message: 4
Date: Mon, 04 Nov 2013 09:13:05 +0100
From: DanB <danb.lists@gmail.com>
Subject: Re: [Freeswitch-users] Scalability and Bridge
To: freeswitch-users@lists.freeswitch.org
Message-ID: <52775711.3050901@gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hi Valter,

There are multiple approaches to this issue.
I personally go for the following: monitor active calls in a pool of FS
boxes somewhere in a database (NoSQL preferred by me) and issue a 302
redirect from the FS box where the call hits initially towards the FS
box where the conference was already established. If the conference is
not active anywhere, start it in without redirect.
Another scenario I know is using a SIP proxy (Kamailio/OpenSIPS) in
front of a pool of FS boxes and monitor there where the conference was
established directly proxying the call to the active one.

Hope this helps!

DanB



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

Message: 5
Date: Mon, 4 Nov 2013 00:57:49 -0800
From: Henry Huang <red.rain.seven@gmail.com>
Subject: [Freeswitch-users] Skype is likely to stop supporting desktop
        API     by Dec. 2013
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Message-ID:
        <CAP_G395SFFdj+js-KxU2bEUbt4wP4yxeio1v3=2uhZGsXPzMtA@mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Here is the link:
http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated

I don't know if this mean that we'll no longer be able to interconnect with
Skype anymore. And I just want to share the news with this community.

Henry
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Message: 6
Date: Mon, 4 Nov 2013 07:58:41 -0200
From: Ben Langfeld <ben@langfeld.co.uk>
Subject: Re: [Freeswitch-users] Scalability and Bridge
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Message-ID:
        <CAAyX+KGe_9i9TKLpj08W2tG39kmn_rLGsQ9xGaY2jLMG965i3A@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Of course that approach has an implicit race condition. Are you able to do
anything about that?


On 4 November 2013 06:13, DanB <danb.lists@gmail.com> wrote:

> Hi Valter,
>
> There are multiple approaches to this issue.
> I personally go for the following: monitor active calls in a pool of FS
> boxes somewhere in a database (NoSQL preferred by me) and issue a 302
> redirect from the FS box where the call hits initially towards the FS
> box where the conference was already established. If the conference is
> not active anywhere, start it in without redirect.
> Another scenario I know is using a SIP proxy (Kamailio/OpenSIPS) in
> front of a pool of FS boxes and monitor there where the conference was
> established directly proxying the call to the active one.
>
> Hope this helps!
>
> DanB
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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Message: 7
Date: Mon, 4 Nov 2013 11:05:29 +0100
From: Gregor Nanger <gregor@infomedia.si>
Subject: Re: [Freeswitch-users] Execute_on_answer and ${strepoch}
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Message-ID:
        <CAC7ttFGO51SP7yO0wmw8J=fc=-s3MH7Eh8HNusMrDDfdCtAWmg@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"

Yes, Dave.

I solved it with mod_managed and create app that I call it on
execute_on_answer. Just wanted to know if there is some catch to formulate
variable so that FS is not evaluating it on dialplan parsing.

Thank you, Yosii. I know that channel has answer time, but I have more
complicated scenario and I need this functionality to execute something on
answer.

--Gregor


2013/11/3 I put the Who? in Mishehu <mishehu@freeswitch.org>

>  Possibly more to the point - the channel already has the answer time in
> the timetables.
>
> -Yossi
>
>
>
> On 11/03/2013 03:55 PM, Dave R. Kompel wrote:
>
> Rather then doing the set as the DP action, you can have it execut a
> script, that sets the var, or execute_extension from the dialplan that does
> it. That way the eval will be done at execute time.
>
> --Dave
>
>  ------------------------------
> *From:* Gregor Nanger [mailto:gregor@infomedia.si <gregor@infomedia.si>]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org<freeswitch-users@lists.freeswitch.org>
> ]
> *Sent:* Sun, 03 Nov 2013 13:28:53 -0800
> *Subject:* [Freeswitch-users] Execute_on_answer and ${strepoch}
>
>  One funny "by design" catch.
>
>  I want to set variable when channel is answered. execute_on_answer is
> perfect tool for this job. I have:
>
>
>  <action application="set" data="execute_on_answer=set
> myvariable=${strepoch()}" />
>
>  Everything is working, except that ${strepoch()} is evaluated during
> dialplan parsing, not when on answer command is executed. So I get time
> when diaplan is processed, not when user pickups.
>
>  Is there a notation to tell FS not to evaluate variable during dialplan
> parsing?
>
>  Best regards, Gregor
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
>
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>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting@freeswitch.org
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server
> http://www.cudatel.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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