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 <[email protected]> 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: [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<http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/%0A288d63a0/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<http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/%0Aec246f47/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<http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/%0Ab8ea2be6/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<http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/%0A638a2202/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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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