[Freeswitch-users] Grandstream gateways
Anthony Minessale
anthony.minessale at gmail.com
Wed Nov 25 14:40:45 PST 2009
Skype for SIP is just a SIP account you can get from skype that is somehow
tied to skype, probably using the scary 2 year commerical endeavor to make
skype work in asterisk. We should all thank Giovanni Maruzzelli for giving
us a free solution.
On Wed, Nov 25, 2009 at 4:18 PM, Adam Ford <lists at redbonez.net> wrote:
> Samuel,
>
> FreeSWITCH has a Skype module that uses Skype client instances to connect
> to
> the Skype network, you can read about it at
> http://wiki.freeswitch.org/wiki/Skypiax
>
> As far as an official Skype module for non-Asterisk PBX-es, it looks like
> it
> is in beta right now -
> http://www.skype.com/business/products/pbx-systems/sip/
>
> -AF
>
>
> -----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 1:17 PM
> Cc: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Grandstream gateways
>
> 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
> > 716-601-4474 MOBILE
> > candrews at 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
> > the property of the VoIP Supply, LLC and shall not be used,
> > disclosed or reproduced without the express written consent of VoIP
> > Supply, LLC. If you are not the intended recipient, nor the employee
> > or agent responsible for delivering this message in confidence to
> > the intended recipient(s), you are hereby notified that you have
> > received this transmittal in error, and any review, dissemination,
> > distribution or copying of this transmittal or its attachments is
> > strictly prohibited. If you have received this transmittal and/or
> > attachments in error, please notify me immediately by reply e-mail
> > or telephone and then delete this message, including any
> > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY
> > 14225 USA.
> >
> >
> >
> > -----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
> >> freeswitch-users at 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 at lists.freeswitch.org
> >>
> >> You can reach the person managing the list at
> >> freeswitch-users-owner at 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 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.
> >>
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
>
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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
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
> >> URL:
>
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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
> >>>
> >> -------------- next part --------------
> >> An HTML attachment was scrubbed...
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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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> >>
> >> ------------------------------
> >>
> >> _______________________________________________
> >> 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
> >>
> >>
> >> End of FreeSWITCH-users Digest, Vol 41, Issue 189
> >> *************************************************
> >
> > _______________________________________________
> > 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
>
> _______________________________________________
> 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
>
>
> _______________________________________________
> 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
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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