<p dir="ltr">Ben,<br>
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.</p>
<p dir="ltr">DanB</p>
<div class="gmail_quote">Am 04.11.2013 11:06 schrieb <<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>>:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<br>
Today's Topics:<br>
<br>
1. SIMPLE IM in UTF-16 (Alex Chan)<br>
2. Re: Raspberry Pi, mod_gsmopen GSM dongle (Jayanth Acharya)<br>
3. Re: 481 Call Does Not Exist (Miha)<br>
4. Re: Scalability and Bridge (DanB)<br>
5. Skype is likely to stop supporting desktop API by Dec. 2013<br>
(Henry Huang)<br>
6. Re: Scalability and Bridge (Ben Langfeld)<br>
7. Re: Execute_on_answer and ${strepoch} (Gregor Nanger)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Mon, 4 Nov 2013 09:46:32 +0800<br>
From: Alex Chan <<a href="mailto:alex@slackadmin.com">alex@slackadmin.com</a>><br>
Subject: [Freeswitch-users] SIMPLE IM in UTF-16<br>
To: <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
Message-ID:<br>
<CAG=<a href="mailto:KnHW2BkP1k-PXGt4N8-t5DCH5Kye8bNtM%2BaqyauiMZtvtug@mail.gmail.com">KnHW2BkP1k-PXGt4N8-t5DCH5Kye8bNtM+aqyauiMZtvtug@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hi folks,<br>
<br>
I'm using FreeSWITCH as a SIP IM B2BUA. When the content-type was in<br>
UTF-16BE, the message body will be dammed at where the zero byte 0x00 first<br>
appears. Any ideas?<br>
<br>
Thanks in advance.<br>
<br>
Alex<br>
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<br>
Message: 2<br>
Date: Mon, 4 Nov 2013 08:25:57 +0530<br>
From: Jayanth Acharya <<a href="mailto:jayachar88@gmail.com">jayachar88@gmail.com</a>><br>
Subject: Re: [Freeswitch-users] Raspberry Pi, mod_gsmopen GSM dongle<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Message-ID:<br>
<<a href="mailto:CAHDL-swz9nR7Cz-p3pbB_rg69JDDp2BSKz4yRR6tdDWWv_cakQ@mail.gmail.com">CAHDL-swz9nR7Cz-p3pbB_rg69JDDp2BSKz4yRR6tdDWWv_cakQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Afraid that this may not have much to do with the distro and the weak-link<br>
may indeed be mod_gsmopen. Have experienced similar things on regular PC on<br>
Debian based systems. The chan_dongle development is a bit more mainstream<br>
and bit more active. More importantly - the developers seem to be actively<br>
using it. It is a shame that mod_gsmopen isn't getting more development...<br>
The working of module that works with 3G dongles isn't complex, especially<br>
after one understands AT commands. However without proficiency in general<br>
FS development, it is hard to contribute meaningfully.<br>
On Nov 4, 2013 6:52 AM, "Steve Underwood" <<a href="mailto:steveu@coppice.org">steveu@coppice.org</a>> wrote:<br>
<br>
> On 10/29/2013 03:05 PM, Sergey Zhuravlov wrote:<br>
> > Hi!<br>
> ><br>
> > Does somebody have a positive experience with Raspberry Pi,<br>
> > mod_gsmopen GSM dongle ???<br>
> > I use Raspberry PiB and e1550.<br>
> ><br>
> > It's a shame that with an asterisk all works fine -- RasPBX.<br>
> > <a href="http://www.raspberry-asterisk.org/" target="_blank">http://www.raspberry-asterisk.org/</a><br>
> ><br>
> > But I prefer the FreeSWITCH ;-)<br>
> > On the basis of experience, I thought that PS will work best for this<br>
> > modest hardware faster and require fewer resources.<br>
> ><br>
> > But it turns out it is not. Asterisk with a web server, MySQL<br>
> > database, and other, less load than FS.<br>
> > In this connection, other parameters may be used starting FS to save<br>
> > resources?<br>
> ><br>
> > And the main question! Silence an incoming call on 5000 (IVR) as well<br>
> > as an outgoing in gsm network. Those are not sound.<br>
> ><br>
> This may have nothing to do with FreeSwitch. Is the same Linux distro<br>
> being used for Asterisk and FreeSwitch? It makes a huge difference on a<br>
> Raspberry Pi. The obvious performance difference is that some distros<br>
> use software floating point, and some use hardware floating point. Its<br>
> more that, though. When I tried various distributions I found compile<br>
> times vary by a factor of 5 or more.<br>
><br>
> Regards,<br>
> Steve<br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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><br>
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><br>
> FreeSWITCH-users mailing list<br>
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<br>
Message: 3<br>
Date: Mon, 04 Nov 2013 08:43:34 +0100<br>
From: Miha <<a href="mailto:miha@softnet.si">miha@softnet.si</a>><br>
Subject: Re: [Freeswitch-users] 481 Call Does Not Exist<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Message-ID: <<a href="mailto:52775026.6040000@softnet.si">52775026.6040000@softnet.si</a>><br>
Content-Type: text/plain; charset=windows-1252; format=flowed<br>
<br>
Hi,<br>
<br>
could some please help me with figuring out why FS sends 481.<br>
<br>
tnx<br>
<br>
miha<br>
<br>
Dne 10/24/2013 9:47 AM, pi?e Miha:<br>
> Hi,<br>
><br>
> I need a little help with figuring out why FS sends "481".<br>
><br>
> This is happing when call has been forward (302).<br>
><br>
> All calls are relayed throught opensips as I am using it for<br>
> registrations and load_balancing.<br>
><br>
> <a href="http://pastebin.freeswitch.org/21555" target="_blank">http://pastebin.freeswitch.org/21555</a><br>
><br>
> tnx!<br>
><br>
> miha<br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
><br>
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> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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><br>
> FreeSWITCH-users mailing list<br>
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> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
><br>
<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Mon, 04 Nov 2013 09:13:05 +0100<br>
From: DanB <<a href="mailto:danb.lists@gmail.com">danb.lists@gmail.com</a>><br>
Subject: Re: [Freeswitch-users] Scalability and Bridge<br>
To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Message-ID: <<a href="mailto:52775711.3050901@gmail.com">52775711.3050901@gmail.com</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
Hi Valter,<br>
<br>
There are multiple approaches to this issue.<br>
I personally go for the following: monitor active calls in a pool of FS<br>
boxes somewhere in a database (NoSQL preferred by me) and issue a 302<br>
redirect from the FS box where the call hits initially towards the FS<br>
box where the conference was already established. If the conference is<br>
not active anywhere, start it in without redirect.<br>
Another scenario I know is using a SIP proxy (Kamailio/OpenSIPS) in<br>
front of a pool of FS boxes and monitor there where the conference was<br>
established directly proxying the call to the active one.<br>
<br>
Hope this helps!<br>
<br>
DanB<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 5<br>
Date: Mon, 4 Nov 2013 00:57:49 -0800<br>
From: Henry Huang <<a href="mailto:red.rain.seven@gmail.com">red.rain.seven@gmail.com</a>><br>
Subject: [Freeswitch-users] Skype is likely to stop supporting desktop<br>
API by Dec. 2013<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Message-ID:<br>
<CAP_G395SFFdj+js-KxU2bEUbt4wP4yxeio1v3=<a href="mailto:2uhZGsXPzMtA@mail.gmail.com">2uhZGsXPzMtA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
Here is the link:<br>
<a href="http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated" target="_blank">http://www.engadget.com/2013/11/02/skype-killing-old-api/?ncid=rss_truncated</a><br>
<br>
I don't know if this mean that we'll no longer be able to interconnect with<br>
Skype anymore. And I just want to share the news with this community.<br>
<br>
Henry<br>
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<br>
Message: 6<br>
Date: Mon, 4 Nov 2013 07:58:41 -0200<br>
From: Ben Langfeld <<a href="mailto:ben@langfeld.co.uk">ben@langfeld.co.uk</a>><br>
Subject: Re: [Freeswitch-users] Scalability and Bridge<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Message-ID:<br>
<<a href="mailto:CAAyX%2BKGe_9i9TKLpj08W2tG39kmn_rLGsQ9xGaY2jLMG965i3A@mail.gmail.com">CAAyX+KGe_9i9TKLpj08W2tG39kmn_rLGsQ9xGaY2jLMG965i3A@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Of course that approach has an implicit race condition. Are you able to do<br>
anything about that?<br>
<br>
<br>
On 4 November 2013 06:13, DanB <<a href="mailto:danb.lists@gmail.com">danb.lists@gmail.com</a>> wrote:<br>
<br>
> Hi Valter,<br>
><br>
> There are multiple approaches to this issue.<br>
> I personally go for the following: monitor active calls in a pool of FS<br>
> boxes somewhere in a database (NoSQL preferred by me) and issue a 302<br>
> redirect from the FS box where the call hits initially towards the FS<br>
> box where the conference was already established. If the conference is<br>
> not active anywhere, start it in without redirect.<br>
> Another scenario I know is using a SIP proxy (Kamailio/OpenSIPS) in<br>
> front of a pool of FS boxes and monitor there where the conference was<br>
> established directly proxying the call to the active one.<br>
><br>
> Hope this helps!<br>
><br>
> DanB<br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
><br>
> Official FreeSWITCH Sites<br>
> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
> <a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
> <a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
><br>
> FreeSWITCH-users mailing list<br>
> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
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><br>
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<br>
Message: 7<br>
Date: Mon, 4 Nov 2013 11:05:29 +0100<br>
From: Gregor Nanger <<a href="mailto:gregor@infomedia.si">gregor@infomedia.si</a>><br>
Subject: Re: [Freeswitch-users] Execute_on_answer and ${strepoch}<br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
Message-ID:<br>
<CAC7ttFGO51SP7yO0wmw8J=fc=-<a href="mailto:s3MH7Eh8HNusMrDDfdCtAWmg@mail.gmail.com">s3MH7Eh8HNusMrDDfdCtAWmg@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Yes, Dave.<br>
<br>
I solved it with mod_managed and create app that I call it on<br>
execute_on_answer. Just wanted to know if there is some catch to formulate<br>
variable so that FS is not evaluating it on dialplan parsing.<br>
<br>
Thank you, Yosii. I know that channel has answer time, but I have more<br>
complicated scenario and I need this functionality to execute something on<br>
answer.<br>
<br>
--Gregor<br>
<br>
<br>
2013/11/3 I put the Who? in Mishehu <<a href="mailto:mishehu@freeswitch.org">mishehu@freeswitch.org</a>><br>
<br>
> Possibly more to the point - the channel already has the answer time in<br>
> the timetables.<br>
><br>
> -Yossi<br>
><br>
><br>
><br>
> On 11/03/2013 03:55 PM, Dave R. Kompel wrote:<br>
><br>
> Rather then doing the set as the DP action, you can have it execut a<br>
> script, that sets the var, or execute_extension from the dialplan that does<br>
> it. That way the eval will be done at execute time.<br>
><br>
> --Dave<br>
><br>
> ------------------------------<br>
> *From:* Gregor Nanger [mailto:<a href="mailto:gregor@infomedia.si">gregor@infomedia.si</a> <<a href="mailto:gregor@infomedia.si">gregor@infomedia.si</a>>]<br>
> *To:* FreeSWITCH Users Help [mailto:<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>
> ]<br>
> *Sent:* Sun, 03 Nov 2013 13:28:53 -0800<br>
> *Subject:* [Freeswitch-users] Execute_on_answer and ${strepoch}<br>
><br>
> One funny "by design" catch.<br>
><br>
> I want to set variable when channel is answered. execute_on_answer is<br>
> perfect tool for this job. I have:<br>
><br>
><br>
> <action application="set" data="execute_on_answer=set<br>
> myvariable=${strepoch()}" /><br>
><br>
> Everything is working, except that ${strepoch()} is evaluated during<br>
> dialplan parsing, not when on answer command is executed. So I get time<br>
> when diaplan is processed, not when user pickups.<br>
><br>
> Is there a notation to tell FS not to evaluate variable during dialplan<br>
> parsing?<br>
><br>
> Best regards, Gregor<br>
><br>
><br>
><br>
><br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://<a href="http://www.freeswitchsolutions.com" target="_blank">www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://<a href="http://www.cudatel.com" target="_blank">www.cudatel.com</a><br>
><br>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://<a href="http://www.cluecon.com" target="_blank">www.cluecon.com</a><br>
><br>
> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
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><br>
><br>
><br>
> _________________________________________________________________________<br>
> Professional FreeSWITCH Consulting Services:<br>
> <a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
> <a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
><br>
> FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
> <a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
><br>
> Official FreeSWITCH Sites<br>
> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
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><br>
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End of FreeSWITCH-users Digest, Vol 89, Issue 4<br>
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