Hello, Samuel: We also have some GXW4104 gateways, in small production/testing environments; our caller id works fine and none of them has failed in over a year of being used. The thing that i dislike about the GXW series is that it has no support for early media.
Everyone: What FXO devices do you currently use / recommend? 2009/11/25 Chris Chen <chris.chen2...@gmail.com> > You haven't really put it into production for more than one year. The issue > with GXW4108 is that after some time, say a couple of months, either all FXO > ports not working, or worse, some FXO ports not working, but after power > recycling, they will come back to work for some time until on strike again > at some time you have no control. > > This had been reported for a couple of years without improvement. Go google > search you will find out, this has happened to many GXW4108 users. > > > > On Wed, Nov 25, 2009 at 3:16 PM, Samuel Mukoti <samuelmuk...@gmail.com>wrote: > >> 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 <c...@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 >> > candr...@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-boun...@lists.freeswitch.org >> > [mailto:freeswitch-users-boun...@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-requ...@lists.freeswitch.org >> > wrote: >> > >> >> Send FreeSWITCH-users mailing list submissions to >> >> freeswitch-users@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-requ...@lists.freeswitch.org >> >> >> >> You can reach the person managing the list at >> >> freeswitch-users-ow...@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 <m...@jerris.com> >> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> >> invitations >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: <1ccc981c-9f4a-4d97-acea-a6dfb906c...@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: >> 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 <m...@jerris.com> >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: <a8fa625f-16d2-4a9f-b8c4-13343a488...@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 <br...@freeswitch.org> >> >> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via >> >> proxy. >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: <0ab8a3a0-0e59-49a4-9cf0-0a1083ecd...@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 <m...@jerris.com> >> >> Subject: Re: [Freeswitch-users] remote_media_ip variable not set >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: <df3eca04-0247-40bb-a810-2468f9c4d...@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 <m...@jerris.com> >> >> Subject: Re: [Freeswitch-users] How to find whether the destination >> >> extension supports encryption >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: <38c9574b-ea25-4b8f-9af6-21861d0fd...@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.ksvre...@gmail.com> >> >> Subject: Re: [Freeswitch-users] Bypass_media and re_invite >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: >> >> <f8af5740911250955x62d66f55h9584582beba76...@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 <m...@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@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: >> 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 <stevecr...@gmail.com> >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: >> >> <11990ade0911251001t1e04447aq6aeaf4b14e9c1...@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 <m...@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@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... >> >> 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 <tculj...@gmail.com> >> >> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily >> >> response from JavaScript >> >> To: freeswitch-users@lists.freeswitch.org >> >> Message-ID: >> >> <65d96fc80911251004l401d5efbl8df3a2ac92020...@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 <m...@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@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... >> >> URL: >> http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20091125/638a2202/attachment.html >> >> >> >> ------------------------------ >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users@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@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@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@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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