[Freeswitch-users] Grandstream gateways
Samuel Mukoti
samuelmukoti at gmail.com
Wed Nov 25 12:16:39 PST 2009
Thank you for those tips,
I do have some small setups using gxw4108 they work or, except CID
doesn't seem to work. I will try the channel bank route - just don't
know too much about the setup options or how you'd purchase the
correct config, eg. For 150 FXS channel bank, can I get a single PCI
card for that?
I may end up using the grandstream fxs gateways then use the T1
channel bank from sangoma,
Thank you all..
Lastly, I know asterisk now has an offical skype_ module, Is there
anything similar I could use?
On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory at voipsupply.com> wrote:
> Samuel - you could go with FXS gateways or channel banks. If you go
> the gateway route Grandstream or Audiocodes would work fine.
> Audiocodes are a bit more telco grade. If you have 25 POTS incoming
> you could use a 24FXO channel bank cross connected with Rhino T1
> cards, or individual FXO gateways but you may have a hard time
> finding 24 ports of FXO in a single GW. Best performing T1 cards in
> my experience (thousands of deployments) are Sangoma. Your server
> configuration looks fine.
>
> Cory J. Andrews
> Director New Market Initiatives
>
> Sayers Media Group
> VoIP Supply, LLC
> 454 Sonwil Drive
> Buffalo, NY 14225
> 716-250-3402 OFFICE
> 716-630-1548 FAX
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> candrews at sayersmedia.com
>
>
> Have I exceeded your expectations? Please share your experience
> with my boss, Benjamin P. Sayers, CEO
>
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>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Samuel Mukoti
> Sent: Wednesday, November 25, 2009 2:40 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Grandstream gateways
>
> 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:
>
>> Send FreeSWITCH-users mailing list submissions to
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>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of FreeSWITCH-users digest..."
>>
>>
>> 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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>>
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>>
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>> *************************************************
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