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: [email protected] [mailto:[email protected]] On Behalf Of Samuel Mukoti Sent: Wednesday, November 25, 2009 1:17 PM Cc: [email protected] 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 <[email protected]> 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 > [email protected] > > > 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: [email protected] > [mailto:[email protected]] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: [email protected] > 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, [email protected] > wrote: > >> Send FreeSWITCH-users mailing list submissions to >> [email protected] >> >> 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 >> [email protected] >> >> You can reach the person managing the list at >> [email protected] >> >> 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 <[email protected]> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: [email protected] >> Message-ID: <[email protected]> >> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 288d63a0/attachment-0001.html >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 25 Nov 2009 12:45:50 -0500 >> From: Michael Jerris <[email protected]> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: [email protected] >> Message-ID: <[email protected]> >> 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 <[email protected]> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> proxy. >> To: [email protected] >> Message-ID: <[email protected]> >> 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 <[email protected]> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> To: [email protected] >> Message-ID: <[email protected]> >> 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 <[email protected]> >> Subject: Re: [Freeswitch-users] How to find whether the destination >> extension supports encryption >> To: [email protected] >> Message-ID: <[email protected]> >> 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 <[email protected]> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> To: [email protected] >> Message-ID: >> <[email protected]> >> 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 <[email protected]> >> 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 >>> [email protected] >>> 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: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ ec246f47/attachment-0001.html >> >> ------------------------------ >> >> Message: 7 >> Date: Wed, 25 Nov 2009 10:01:14 -0800 >> From: Stephen Crosby <[email protected]> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: [email protected] >> Message-ID: >> <[email protected]> >> 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 <[email protected]> >> 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 >>> [email protected] >>> 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... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ b8ea2be6/attachment-0001.html >> >> ------------------------------ >> >> Message: 8 >> Date: Wed, 25 Nov 2009 19:04:56 +0100 >> From: Tihomir Culjaga <[email protected]> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> response from JavaScript >> To: [email protected] >> Message-ID: >> <[email protected]> >> 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 <[email protected]> >> 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 >>> [email protected] >>> 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... >> URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/ 638a2202/attachment.html >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> [email protected] >> 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 > [email protected] > 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 [email protected] 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 [email protected] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
