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
