From: Milena <testeador01@gmail.com>
Subject: Re: [Freeswitch-users] Grandstream gateways
To: freeswitch-users@lists.freeswitch.org
Date: Wednesday, November 25, 2009, 4:00 PM

Hello,

Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media.

Everyone: What FXO devices do you currently use / recommend?

2009/11/25 Chris Chen <chris.chen2004@gmail.com>

You haven't really put it into production for more than one year. The issue with GXW4108 is that after some time, say a couple of months, either all FXO ports not working, or worse, some FXO ports not working, but after power recycling, they will come back to work for some time until on strike again at some time you have no control.

This had been reported for a couple of years without improvement. Go google search you will find out, this has happened to many GXW4108 users.

On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti <samuelmukoti@gmail.com> wrote:

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@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
> 716-601-4474 MOBILE
> candrews@sayersmedia.com
>
>
> Have I exceeded your expectations?  Please share your experience
> with my boss,  Benjamin P. Sayers, CEO
>
> NOTICE: The information contained in this email and any document
> attached hereto is intended only for the named recipient(s). It is
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>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces@lists.freeswitch.org
> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of
> Samuel Mukoti
> Sent: Wednesday, November 25, 2009 2:40 PM
> To: freeswitch-users@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@lists.freeswitch.org
>  wrote:
>
>> Send FreeSWITCH-users mailing list submissions to
>>   freeswitch-users@lists.freeswitch.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> or, via email, send a message with subject or body 'help' to
>>   freeswitch-users-request@lists.freeswitch.org
>>
>> You can reach the person managing the list at
>>   freeswitch-users-owner@lists.freeswitch.org
>>
>> 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@jerris.com>
>> Subject: Re: [Freeswitch-users] mod_conference kick to abort
>>   invitations
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@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.
>>
>> -------------- next part --------------
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>>
>> ------------------------------
>>
>> Message: 2
>> Date: Wed, 25 Nov 2009 12:45:50 -0500
>> From: Michael Jerris <mike@jerris.com>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>>   response    from JavaScript
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <A8FA625F-16D2-4A9F-B8C4-13343A488777@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@freeswitch.org>
>> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via
>>   proxy.
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@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@jerris.com>
>> Subject: Re: [Freeswitch-users] remote_media_ip variable not set
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <DF3ECA04-0247-40BB-A810-2468F9C4D805@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@jerris.com>
>> Subject: Re: [Freeswitch-users] How to find whether the destination
>>   extension supports encryption
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID: <38C9574B-EA25-4B8F-9AF6-21861D0FDA40@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@gmail.com>
>> Subject: Re: [Freeswitch-users] Bypass_media and re_invite
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID:
>>   <f8af5740911250955x62d66f55h9584582beba76ba0@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@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@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
>> -------------- next part --------------
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>>
>> ------------------------------
>>
>> Message: 7
>> Date: Wed, 25 Nov 2009 10:01:14 -0800
>> From: Stephen Crosby <stevecrozz@gmail.com>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>>   response    from JavaScript
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID:
>>   <11990ade0911251001t1e04447aq6aeaf4b14e9c101e@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@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@lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>> -------------- next part --------------
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>>
>> ------------------------------
>>
>> Message: 8
>> Date: Wed, 25 Nov 2009 19:04:56 +0100
>> From: Tihomir Culjaga <tculjaga@gmail.com>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily
>>   response    from JavaScript
>> To: freeswitch-users@lists.freeswitch.org
>> Message-ID:
>>   <65d96fc80911251004l401d5efbl8df3a2ac920207b8@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@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@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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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>> End of FreeSWITCH-users Digest, Vol 41, Issue 189
>> *************************************************
>
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