[Freeswitch-users] Scalability and Bridge

Dan-Cristian Bogos danb.lists at gmail.com
Mon Nov 4 13:37:14 MSK 2013 

Ben,
You are right about possible race here. One way I could sort of fix it is
by checking at the establishment time for a reservation token in the
database - redis in my case (valid for the probability time of the race).
This tocken mechanism would be only used if there has been no prio
conference active. I did not hit many race conditions so far but also not
many concurrent setups in a conference to be honest.

DanB
Am 04.11.2013 11:06 schrieb <freeswitch-users-request at 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)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Mon, 4 Nov 2013 09:46:32 +0800
> From: Alex Chan <alex at slackadmin.com>
> Subject: [Freeswitch-users] SIMPLE IM in UTF-16
> To: FreeSWITCH-users at lists.freeswitch.org
> Message-ID:
>         <CAG=
> KnHW2BkP1k-PXGt4N8-t5DCH5Kye8bNtM+aqyauiMZtvtug at 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 at gmail.com>
> Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Message-ID:
>         <
> CAHDL-swz9nR7Cz-p3pbB_rg69JDDp2BSKz4yRR6tdDWWv_cakQ at 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 at 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 at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
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> > 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 at softnet.si>
> Subject: Re: [Freeswitch-users] 481 Call Does Not Exist
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Message-ID: <52775026.6040000 at 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 at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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 at gmail.com>
> Subject: Re: [Freeswitch-users] Scalability and Bridge
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <52775711.3050901 at 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 at gmail.com>
> Subject: [Freeswitch-users] Skype is likely to stop supporting desktop
>         API     by Dec. 2013
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Message-ID:
>         <CAP_G395SFFdj+js-KxU2bEUbt4wP4yxeio1v3=
> 2uhZGsXPzMtA at 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 at langfeld.co.uk>
> Subject: Re: [Freeswitch-users] Scalability and Bridge
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Message-ID:
>         <
> CAAyX+KGe_9i9TKLpj08W2tG39kmn_rLGsQ9xGaY2jLMG965i3A at 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 at 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 at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
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> ------------------------------
>
> Message: 7
> Date: Mon, 4 Nov 2013 11:05:29 +0100
> From: Gregor Nanger <gregor at infomedia.si>
> Subject: Re: [Freeswitch-users] Execute_on_answer and ${strepoch}
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Message-ID:
>         <CAC7ttFGO51SP7yO0wmw8J=fc=-
> s3MH7Eh8HNusMrDDfdCtAWmg at 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 at 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 at infomedia.si <gregor at infomedia.si>]
> > *To:* FreeSWITCH Users Help [mailto:
> freeswitch-users at lists.freeswitch.org<
> freeswitch-users at 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
> >
> > http://
> www.cudatel.com
> >
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> >
> > 
> > 
> >
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