[Freeswitch-users] Grandstream gateways

Samuel Mukoti samuelmukoti at gmail.com
Wed Nov 25 11:40:00 PST 2009


Hi all,

I'm wanting to try out a my first large scale setup at the office, 200  
extensions and 24 POTS incoming, also a T1 line once the telco guys  
are ready.  I wanted assistance with choosing the most appropriate  
hardware.  We already have about 150 analogue phones, and I was  
wondering what's best? A couple of grandstream FXS GXW4024? Also for  
my POTS lines, gxw4108  FXO gateway or is it better to buy a sangoma  
or digium card? The best voice quality is paramount. Lastly for T1  
what cards are recommeded,

I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM,  
would that perform? Or do I need hardware transcoding?

Thank you,

Sam

Twitter: twitter.com/samuelmukoti


On 25 Nov,2009, at 8:05 PM, freeswitch-users-request at lists.freeswitch.org 
  wrote:

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> Today's Topics:
>
>   1. Re: mod_conference kick to abort invitations (Michael Jerris)
>   2. Re: Handling the 302 Moved Temporarily response    from
>      JavaScript (Michael Jerris)
>   3. Re: No NOTIFY MWI when registering via proxy. (Brian West)
>   4. Re: remote_media_ip variable not set (Michael Jerris)
>   5. Re: How to find whether the destination    extension supports
>      encryption (Michael Jerris)
>   6. Re: Bypass_media and re_invite (srinivasula reddy)
>   7. Re: Handling the 302 Moved Temporarily response    from
>      JavaScript (Stephen Crosby)
>   8. Re: Handling the 302 Moved Temporarily response    from
>      JavaScript (Tihomir Culjaga)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Wed, 25 Nov 2009 12:44:46 -0500
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] mod_conference kick to abort
>    invitations
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B at jerris.com>
> Content-Type: text/plain; charset="windows-1252"
>
> Its a feature we don't have, patches welcome.
>
> Mike
>
> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:
>
>> Hi members,
>> I?m controlling freeswitch with the conference module via xmlrpc.
>>
>> Is it desired that the kick command can only kick users that are  
>> connected to the conference?
>> Is there no chance abort an  invitation?
>> The kick command has no effect until the person I invited with the  
>> dial command is connected.
>
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> ------------------------------
>
> Message: 2
> Date: Wed, 25 Nov 2009 12:45:50 -0500
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>    response    from JavaScript
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <A8FA625F-16D2-4A9F-B8C4-13343A488777 at jerris.com>
> Content-Type: text/plain; charset=us-ascii
>
> In trunk there is a sofia profile setting to allow dialplan  
> processing of 302 responses.  This won't get you back into your same  
> javascript, but you can probably do something clever from there.
>
> Mike
>
> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>
>>
>> I have considered writing JavaScript code to bridge two calls  
>> together. However, I would like to perform custom handling of the  
>> 302 Moved Temporarily response. How do I handle the 302 Moved  
>> Temporarily response if I use JavaScript?
>>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 25 Nov 2009 11:46:05 -0600
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via
>    proxy.
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6 at freeswitch.org>
> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes
>
> Yes an alias will be required for every domain you run on the profile
> so it can find it.
>
> /b
>
> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:
>
>> Try an alias on the sip profile.
>>
>> Mike
>
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 25 Nov 2009 12:47:37 -0500
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] remote_media_ip variable not set
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <DF3ECA04-0247-40BB-A810-2468F9C4D805 at jerris.com>
> Content-Type: text/plain; charset=us-ascii
>
> It's possible it does not.  I just added some code to set it on auto- 
> adjust so it might be there sometimes now.  You might need to add  
> some code in mod_sofia to add it other times.  Maybe it makes sense  
> to move that var setting down to switch_rtp.c.  Patches for this  
> would be welcome.
>
> Thanks
>
> Mike
>
> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote:
>
>> Hi,
>>
>> In the case of proxy_media=true, does it gets set at all then?
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 25 Nov 2009 12:48:39 -0500
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] How to find whether the destination
>    extension supports encryption
> To: freeswitch-users at lists.freeswitch.org
> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40 at jerris.com>
> Content-Type: text/plain; charset=us-ascii
>
> You can send the call with secure enabled and if it supports it it  
> will use it.
>
> Mike
>
> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:
>
>> Hello,
>>
>>  We have a mix of phones that support RTP encryption and those that  
>> do not. I have to support both types in the meanwhile, and would  
>> like to have encryption enabled on the relevant leg, even if the  
>> other leg does not support it (why? one of our ATAs either must  
>> have it unencrypted or have it encrypted, but cannot have both).
>>
>> How do I find whether the destination supports encryption? I do not  
>> want to manage an additional table in the database...
>>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 25 Nov 2009 23:25:01 +0530
> From: srinivasula reddy <srinivas.ksvreddy at gmail.com>
> Subject: Re: [Freeswitch-users] Bypass_media and re_invite
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>    <f8af5740911250955x62d66f55h9584582beba76ba0 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> HI,
> thanks for your reply, my requirement is i am doing failover stuff  
> with
> freeswitch. i dont want cut the calls when freeswitch dies, when  
> failover
> happens mean one freeswitch dies we are going to start the second
> freeswitch, i dont want close call intiated by the  first  
> freeswtich, they
> are communicating with meida(bypass media). when one endpoing try to  
> end the
> call at that time i want to close the call for the other end also.
>
>
> srinivas
>
> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
>> FreeSWITCH will kill the calls when you shut it down, if you  
>> intentionally
>> kill the network without shutting down FreeSWITCH the only thing  
>> you can do
>> is enable session timers or rtp timers in the soft phones to kill  
>> the call
>> when FreeSWITCH dies or when the call is over.
>>
>> Mike
>>
>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote:
>>
>>> Hi All,
>>>
>>> goodmorning to all, i have a scenario, two pjsua clients are  
>>> connected
>> with Freeswitch and they are in call and bypass_media=true.  i  
>> close the
>> Freeswitch server, still they are in call, again i started the  
>> Freeswitch,
>> and registerd these two endpoints, now how can i end the call 
>> (estabilished
>> by the first Freeswitch)? if i call re_invite will it estabilish  
>> the call
>> between two endpoints?
>>> any idea?
>>
>>
>> _______________________________________________
>> 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
>>
>
>
>
> -- 
> Srinivasula Reddy K
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> ------------------------------
>
> Message: 7
> Date: Wed, 25 Nov 2009 10:01:14 -0800
> From: Stephen Crosby <stevecrozz at gmail.com>
> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>    response    from JavaScript
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>    <11990ade0911251001t1e04447aq6aeaf4b14e9c101e at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Surprisingly, I've found no way to access the HTTP response status  
> code
> using mod_spidermonkey_curl. I'd love to see this feature added or  
> discussed
> if it already exists and I'm missing it.
>
> --Stephen
>
> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris <mike at jerris.com>  
> wrote:
>
>> In trunk there is a sofia profile setting to allow dialplan  
>> processing of
>> 302 responses.  This won't get you back into your same javascript,  
>> but you
>> can probably do something clever from there.
>>
>> Mike
>>
>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>>
>>>
>>> I have considered writing JavaScript code to bridge two calls  
>>> together.
>> However, I would like to perform custom handling of the 302 Moved
>> Temporarily response. How do I handle the 302 Moved Temporarily  
>> response if
>> I use JavaScript?
>>>
>>
>> _______________________________________________
>> 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
>>
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> ------------------------------
>
> Message: 8
> Date: Wed, 25 Nov 2009 19:04:56 +0100
> From: Tihomir Culjaga <tculjaga at gmail.com>
> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>    response    from JavaScript
> To: freeswitch-users at lists.freeswitch.org
> Message-ID:
>    <65d96fc80911251004l401d5efbl8df3a2ac920207b8 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> this is how i do it from the dialplan:
>
>
>
>
>   <extension name="ServiceLookup">
>      <condition field="destination_number"
> expression="^(300030)(.*)|^\+(300030)(.*)">
>
>         <action application="set" data="bPfx=$1$3"/>
>         <action application="set" data="bNum=$2$4"/>
>
>         <action inline="true" application="set"
> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/>
>         <action application="set"
> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number: 
> 1:32} :
> ${caller_id_number})}"/>
>
>         <action inline="true" application="set"
> data="aPfx=${caller_id_number:0:6}"/>
>         <action inline="true" application="set"
> data="aNum=${caller_id_number:6:16}"/>
>         <action inline="true" application="set"
> data="IP_ADDR=${network_addr}:5060"/>
>
>         <action application="lookup_service_destination" data="in $ 
> {aNum},
>                                                                in $ 
> {aPfx},
>                                                                in $ 
> {bNum},
>                                                                in $ 
> {bPfx},
>                                                                in
> ${IP_ADDR},
>                                                                out
> redContact,
>                                                                out
> authResult"/>
>
>         <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>         <action application="log" data="INFO ########################
> contact = '${redContact}' ##############\n"/>
>         <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>         <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>
>         <action application="execute_extension" data="doRedirect XML
> public"/>
>        </condition>
>   </extension>
>
>
>   <extension name="doRedirect">
>      <condition field="destination_number" expression="^doRedirect$"/>
>      <condition field="${authResult}" expression="^0$|">
>         <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>         <action application="redirect" data="${red_contact}"/>
>         <anti-action application="log" data="INFO  
> ########################
> RADIUS auth NOK!! ##########\n"/>
>         <anti-action application="respond" data="403 Forbidden"/>
>      </condition>
>
>   </extension>
>
>
>
>
> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris <mike at jerris.com>  
> wrote:
>
>> In trunk there is a sofia profile setting to allow dialplan  
>> processing of
>> 302 responses.  This won't get you back into your same javascript,  
>> but you
>> can probably do something clever from there.
>>
>> Mike
>>
>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:
>>
>>>
>>> I have considered writing JavaScript code to bridge two calls  
>>> together.
>> However, I would like to perform custom handling of the 302 Moved
>> Temporarily response. How do I handle the 302 Moved Temporarily  
>> response if
>> I use JavaScript?
>>>
>>
>> _______________________________________________
>> 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
>>
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