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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Samuel Mukoti Sent: Wednesday, November 25, 2009 1:17 PM Cc: freeswitch-users@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@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.
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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:
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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)
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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.