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.<br>
<br><br><br><br><div class="gmail_quote">On Wed, Nov 25, 2009 at 4:18 PM, Adam Ford <span dir="ltr"><<a href="mailto:lists@redbonez.net">lists@redbonez.net</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Samuel,<br>
<br>
FreeSWITCH has a Skype module that uses Skype client instances to connect to<br>
the Skype network, you can read about it at<br>
<a href="http://wiki.freeswitch.org/wiki/Skypiax" target="_blank">http://wiki.freeswitch.org/wiki/Skypiax</a><br>
<br>
As far as an official Skype module for non-Asterisk PBX-es, it looks like it<br>
is in beta right now -<br>
<a href="http://www.skype.com/business/products/pbx-systems/sip/" target="_blank">http://www.skype.com/business/products/pbx-systems/sip/</a><br>
<font color="#888888"><br>
-AF<br>
</font><div class="im"><br>
<br>
-----Original Message-----<br>
From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
[mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of Samuel<br>
Mukoti<br>
</div><div><div></div><div class="h5">Sent: Wednesday, November 25, 2009 1:17 PM<br>
Cc: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
Subject: Re: [Freeswitch-users] Grandstream gateways<br>
<br>
Thank you for those tips,<br>
<br>
I do have some small setups using gxw4108 they work or, except CID<br>
doesn't seem to work. I will try the channel bank route - just don't<br>
know too much about the setup options or how you'd purchase the<br>
correct config, eg. For 150 FXS channel bank, can I get a single PCI<br>
card for that?<br>
<br>
I may end up using the grandstream fxs gateways then use the T1<br>
channel bank from sangoma,<br>
<br>
Thank you all..<br>
<br>
Lastly, I know asterisk now has an offical skype_ module, Is there<br>
anything similar I could use?<br>
<br>
<br>
On 25 Nov,2009, at 9:52 PM, Cory Andrews <<a href="mailto:cory@voipsupply.com">cory@voipsupply.com</a>> wrote:<br>
<br>
> Samuel - you could go with FXS gateways or channel banks. If you go<br>
> the gateway route Grandstream or Audiocodes would work fine.<br>
> Audiocodes are a bit more telco grade. If you have 25 POTS incoming<br>
> you could use a 24FXO channel bank cross connected with Rhino T1<br>
> cards, or individual FXO gateways but you may have a hard time<br>
> finding 24 ports of FXO in a single GW. Best performing T1 cards in<br>
> my experience (thousands of deployments) are Sangoma. Your server<br>
> configuration looks fine.<br>
><br>
> Cory J. Andrews<br>
> Director New Market Initiatives<br>
><br>
> Sayers Media Group<br>
> VoIP Supply, LLC<br>
> 454 Sonwil Drive<br>
> Buffalo, NY 14225<br>
> 716-250-3402 OFFICE<br>
> 716-630-1548 FAX<br>
> 716-601-4474 MOBILE<br>
> <a href="mailto:candrews@sayersmedia.com">candrews@sayersmedia.com</a><br>
><br>
><br>
> Have I exceeded your expectations? Please share your experience<br>
> with my boss, Benjamin P. Sayers, CEO<br>
><br>
> NOTICE: The information contained in this email and any document<br>
> attached hereto is intended only for the named recipient(s). It is<br>
> the property of the VoIP Supply, LLC and shall not be used,<br>
> disclosed or reproduced without the express written consent of VoIP<br>
> Supply, LLC. If you are not the intended recipient, nor the employee<br>
> or agent responsible for delivering this message in confidence to<br>
> the intended recipient(s), you are hereby notified that you have<br>
> received this transmittal in error, and any review, dissemination,<br>
> distribution or copying of this transmittal or its attachments is<br>
> strictly prohibited. If you have received this transmittal and/or<br>
> attachments in error, please notify me immediately by reply e-mail<br>
> or telephone and then delete this message, including any<br>
> attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY<br>
> 14225 USA.<br>
><br>
><br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a><br>
> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</a>] On Behalf Of<br>
> Samuel Mukoti<br>
> Sent: Wednesday, November 25, 2009 2:40 PM<br>
> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> Subject: [Freeswitch-users] Grandstream gateways<br>
><br>
> Hi all,<br>
><br>
> I'm wanting to try out a my first large scale setup at the office, 200<br>
> extensions and 24 POTS incoming, also a T1 line once the telco guys<br>
> are ready. I wanted assistance with choosing the most appropriate<br>
> hardware. We already have about 150 analogue phones, and I was<br>
> wondering what's best? A couple of grandstream FXS GXW4024? Also for<br>
> my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma<br>
> or digium card? The best voice quality is paramount. Lastly for T1<br>
> what cards are recommeded,<br>
><br>
> I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM,<br>
> would that perform? Or do I need hardware transcoding?<br>
><br>
> Thank you,<br>
><br>
> Sam<br>
><br>
> Twitter: <a href="http://twitter.com/samuelmukoti" target="_blank">twitter.com/samuelmukoti</a><br>
><br>
><br>
> On 25 Nov,2009, at 8:05 PM, <a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a><br>
> wrote:<br>
><br>
>> Send FreeSWITCH-users mailing list submissions to<br>
>> <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>><br>
>> To subscribe or unsubscribe via the World Wide Web, visit<br>
>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>> or, via email, send a message with subject or body 'help' to<br>
>> <a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a><br>
>><br>
>> You can reach the person managing the list at<br>
>> <a href="mailto:freeswitch-users-owner@lists.freeswitch.org">freeswitch-users-owner@lists.freeswitch.org</a><br>
>><br>
>> When replying, please edit your Subject line so it is more specific<br>
>> than "Re: Contents of FreeSWITCH-users digest..."<br>
>><br>
>><br>
>> Today's Topics:<br>
>><br>
>> 1. Re: mod_conference kick to abort invitations (Michael Jerris)<br>
>> 2. Re: Handling the 302 Moved Temporarily response from<br>
>> JavaScript (Michael Jerris)<br>
>> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West)<br>
>> 4. Re: remote_media_ip variable not set (Michael Jerris)<br>
>> 5. Re: How to find whether the destination extension supports<br>
>> encryption (Michael Jerris)<br>
>> 6. Re: Bypass_media and re_invite (srinivasula reddy)<br>
>> 7. Re: Handling the 302 Moved Temporarily response from<br>
>> JavaScript (Stephen Crosby)<br>
>> 8. Re: Handling the 302 Moved Temporarily response from<br>
>> JavaScript (Tihomir Culjaga)<br>
>><br>
>><br>
>> ---<br>
>> -------------------------------------------------------------------<br>
>><br>
>> Message: 1<br>
>> Date: Wed, 25 Nov 2009 12:44:46 -0500<br>
>> From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> Subject: Re: [Freeswitch-users] mod_conference kick to abort<br>
>> invitations<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID: <<a href="mailto:1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@jerris.com">1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B@jerris.com</a>><br>
>> Content-Type: text/plain; charset="windows-1252"<br>
>><br>
>> Its a feature we don't have, patches welcome.<br>
>><br>
>> Mike<br>
>><br>
>> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote:<br>
>><br>
>>> Hi members,<br>
>>> I?m controlling freeswitch with the conference module via xmlrpc.<br>
>>><br>
>>> Is it desired that the kick command can only kick users that are<br>
>>> connected to the conference?<br>
>>> Is there no chance abort an invitation?<br>
>>> The kick command has no effect until the person I invited with the<br>
>>> dial command is connected.<br>
>><br>
>> -------------- next part --------------<br>
>> An HTML attachment was scrubbed...<br>
>> URL:<br>
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288d63a0/attachment-0001.html</a><br>
>><br>
>> ------------------------------<br>
>><br>
>> Message: 2<br>
>> Date: Wed, 25 Nov 2009 12:45:50 -0500<br>
>> From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily<br>
>> response from JavaScript<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID: <<a href="mailto:A8FA625F-16D2-4A9F-B8C4-13343A488777@jerris.com">A8FA625F-16D2-4A9F-B8C4-13343A488777@jerris.com</a>><br>
>> Content-Type: text/plain; charset=us-ascii<br>
>><br>
>> In trunk there is a sofia profile setting to allow dialplan<br>
>> processing of 302 responses. This won't get you back into your same<br>
>> javascript, but you can probably do something clever from there.<br>
>><br>
>> Mike<br>
>><br>
>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:<br>
>><br>
>>><br>
>>> I have considered writing JavaScript code to bridge two calls<br>
>>> together. However, I would like to perform custom handling of the<br>
>>> 302 Moved Temporarily response. How do I handle the 302 Moved<br>
>>> Temporarily response if I use JavaScript?<br>
>>><br>
>><br>
>><br>
>><br>
>> ------------------------------<br>
>><br>
>> Message: 3<br>
>> Date: Wed, 25 Nov 2009 11:46:05 -0600<br>
>> From: Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>><br>
>> Subject: Re: [Freeswitch-users] No NOTIFY MWI when registering via<br>
>> proxy.<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID: <<a href="mailto:0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org">0AB8A3A0-0E59-49A4-9CF0-0A1083ECD3E6@freeswitch.org</a>><br>
>> Content-Type: text/plain; charset=us-ascii; format=flowed; delsp=yes<br>
>><br>
>> Yes an alias will be required for every domain you run on the profile<br>
>> so it can find it.<br>
>><br>
>> /b<br>
>><br>
>> On Nov 25, 2009, at 11:39 AM, Michael Jerris wrote:<br>
>><br>
>>> Try an alias on the sip profile.<br>
>>><br>
>>> Mike<br>
>><br>
>><br>
>><br>
>><br>
>> ------------------------------<br>
>><br>
>> Message: 4<br>
>> Date: Wed, 25 Nov 2009 12:47:37 -0500<br>
>> From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> Subject: Re: [Freeswitch-users] remote_media_ip variable not set<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID: <<a href="mailto:DF3ECA04-0247-40BB-A810-2468F9C4D805@jerris.com">DF3ECA04-0247-40BB-A810-2468F9C4D805@jerris.com</a>><br>
>> Content-Type: text/plain; charset=us-ascii<br>
>><br>
>> It's possible it does not. I just added some code to set it on auto-<br>
>> adjust so it might be there sometimes now. You might need to add<br>
>> some code in mod_sofia to add it other times. Maybe it makes sense<br>
>> to move that var setting down to switch_rtp.c. Patches for this<br>
>> would be welcome.<br>
>><br>
>> Thanks<br>
>><br>
>> Mike<br>
>><br>
>> On Nov 24, 2009, at 10:56 AM, Juan Backson wrote:<br>
>><br>
>>> Hi,<br>
>>><br>
>>> In the case of proxy_media=true, does it gets set at all then?<br>
>><br>
>><br>
>><br>
>><br>
>> ------------------------------<br>
>><br>
>> Message: 5<br>
>> Date: Wed, 25 Nov 2009 12:48:39 -0500<br>
>> From: Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> Subject: Re: [Freeswitch-users] How to find whether the destination<br>
>> extension supports encryption<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID: <<a href="mailto:38C9574B-EA25-4B8F-9AF6-21861D0FDA40@jerris.com">38C9574B-EA25-4B8F-9AF6-21861D0FDA40@jerris.com</a>><br>
>> Content-Type: text/plain; charset=us-ascii<br>
>><br>
>> You can send the call with secure enabled and if it supports it it<br>
>> will use it.<br>
>><br>
>> Mike<br>
>><br>
>> On Nov 24, 2009, at 8:05 AM, Yehavi Bourvine wrote:<br>
>><br>
>>> Hello,<br>
>>><br>
>>> We have a mix of phones that support RTP encryption and those that<br>
>>> do not. I have to support both types in the meanwhile, and would<br>
>>> like to have encryption enabled on the relevant leg, even if the<br>
>>> other leg does not support it (why? one of our ATAs either must<br>
>>> have it unencrypted or have it encrypted, but cannot have both).<br>
>>><br>
>>> How do I find whether the destination supports encryption? I do not<br>
>>> want to manage an additional table in the database...<br>
>>><br>
>><br>
>><br>
>><br>
>> ------------------------------<br>
>><br>
>> Message: 6<br>
>> Date: Wed, 25 Nov 2009 23:25:01 +0530<br>
>> From: srinivasula reddy <<a href="mailto:srinivas.ksvreddy@gmail.com">srinivas.ksvreddy@gmail.com</a>><br>
>> Subject: Re: [Freeswitch-users] Bypass_media and re_invite<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID:<br>
>> <<a href="mailto:f8af5740911250955x62d66f55h9584582beba76ba0@mail.gmail.com">f8af5740911250955x62d66f55h9584582beba76ba0@mail.gmail.com</a>><br>
>> Content-Type: text/plain; charset="iso-8859-1"<br>
>><br>
>> HI,<br>
>> thanks for your reply, my requirement is i am doing failover stuff<br>
>> with<br>
>> freeswitch. i dont want cut the calls when freeswitch dies, when<br>
>> failover<br>
>> happens mean one freeswitch dies we are going to start the second<br>
>> freeswitch, i dont want close call intiated by the first<br>
>> freeswtich, they<br>
>> are communicating with meida(bypass media). when one endpoing try to<br>
>> end the<br>
>> call at that time i want to close the call for the other end also.<br>
>><br>
>><br>
>> srinivas<br>
>><br>
>> On Wed, Nov 25, 2009 at 11:14 PM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> wrote:<br>
>><br>
>>> FreeSWITCH will kill the calls when you shut it down, if you<br>
>>> intentionally<br>
>>> kill the network without shutting down FreeSWITCH the only thing<br>
>>> you can do<br>
>>> is enable session timers or rtp timers in the soft phones to kill<br>
>>> the call<br>
>>> when FreeSWITCH dies or when the call is over.<br>
>>><br>
>>> Mike<br>
>>><br>
>>> On Nov 25, 2009, at 11:53 AM, srinivasula reddy wrote:<br>
>>><br>
>>>> Hi All,<br>
>>>><br>
>>>> goodmorning to all, i have a scenario, two pjsua clients are<br>
>>>> connected<br>
>>> with Freeswitch and they are in call and bypass_media=true. i<br>
>>> close the<br>
>>> Freeswitch server, still they are in call, again i started the<br>
>>> Freeswitch,<br>
>>> and registerd these two endpoints, now how can i end the call<br>
>>> (estabilished<br>
>>> by the first Freeswitch)? if i call re_invite will it estabilish<br>
>>> the call<br>
>>> between two endpoints?<br>
>>>> any idea?<br>
>>><br>
>>><br>
>>> _______________________________________________<br>
>>> FreeSWITCH-users mailing list<br>
>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-</a><br>
>>> users<br>
>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
>><br>
>><br>
>><br>
>> --<br>
>> Srinivasula Reddy K<br>
>> -------------- next part --------------<br>
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>><br>
>> ------------------------------<br>
>><br>
>> Message: 7<br>
>> Date: Wed, 25 Nov 2009 10:01:14 -0800<br>
>> From: Stephen Crosby <<a href="mailto:stevecrozz@gmail.com">stevecrozz@gmail.com</a>><br>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily<br>
>> response from JavaScript<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID:<br>
>> <<a href="mailto:11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com">11990ade0911251001t1e04447aq6aeaf4b14e9c101e@mail.gmail.com</a>><br>
>> Content-Type: text/plain; charset="utf-8"<br>
>><br>
>> Surprisingly, I've found no way to access the HTTP response status<br>
>> code<br>
>> using mod_spidermonkey_curl. I'd love to see this feature added or<br>
>> discussed<br>
>> if it already exists and I'm missing it.<br>
>><br>
>> --Stephen<br>
>><br>
>> On Wed, Nov 25, 2009 at 9:45 AM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> wrote:<br>
>><br>
>>> In trunk there is a sofia profile setting to allow dialplan<br>
>>> processing of<br>
>>> 302 responses. This won't get you back into your same javascript,<br>
>>> but you<br>
>>> can probably do something clever from there.<br>
>>><br>
>>> Mike<br>
>>><br>
>>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:<br>
>>><br>
>>>><br>
>>>> I have considered writing JavaScript code to bridge two calls<br>
>>>> together.<br>
>>> However, I would like to perform custom handling of the 302 Moved<br>
>>> Temporarily response. How do I handle the 302 Moved Temporarily<br>
>>> response if<br>
>>> I use JavaScript?<br>
>>>><br>
>>><br>
>>> _______________________________________________<br>
>>> FreeSWITCH-users mailing list<br>
>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-</a><br>
>>> users<br>
>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
>> -------------- next part --------------<br>
>> An HTML attachment was scrubbed...<br>
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>><br>
>> ------------------------------<br>
>><br>
>> Message: 8<br>
>> Date: Wed, 25 Nov 2009 19:04:56 +0100<br>
>> From: Tihomir Culjaga <<a href="mailto:tculjaga@gmail.com">tculjaga@gmail.com</a>><br>
>> Subject: Re: [Freeswitch-users] Handling the 302 Moved Temporarily<br>
>> response from JavaScript<br>
>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
>> Message-ID:<br>
>> <<a href="mailto:65d96fc80911251004l401d5efbl8df3a2ac920207b8@mail.gmail.com">65d96fc80911251004l401d5efbl8df3a2ac920207b8@mail.gmail.com</a>><br>
>> Content-Type: text/plain; charset="iso-8859-1"<br>
>><br>
>> this is how i do it from the dialplan:<br>
>><br>
>><br>
>><br>
>><br>
>> <extension name="ServiceLookup"><br>
>> <condition field="destination_number"<br>
>> expression="^(300030)(.*)|^\+(300030)(.*)"><br>
>><br>
>> <action application="set" data="bPfx=$1$3"/><br>
>> <action application="set" data="bNum=$2$4"/><br>
>><br>
>> <action inline="true" application="set"<br>
>> data="intf=${regex(${caller_id_number}|^i\+(......)(.*) |%1)}"/><br>
>> <action application="set"<br>
>> data="caller_id_number=${cond(${intf}==true ? ${caller_id_number:<br>
>> 1:32} :<br>
>> ${caller_id_number})}"/><br>
>><br>
>> <action inline="true" application="set"<br>
>> data="aPfx=${caller_id_number:0:6}"/><br>
>> <action inline="true" application="set"<br>
>> data="aNum=${caller_id_number:6:16}"/><br>
>> <action inline="true" application="set"<br>
>> data="IP_ADDR=${network_addr}:5060"/><br>
>><br>
>> <action application="lookup_service_destination" data="in $<br>
>> {aNum},<br>
>> in $<br>
>> {aPfx},<br>
>> in $<br>
>> {bNum},<br>
>> in $<br>
>> {bPfx},<br>
>> in<br>
>> ${IP_ADDR},<br>
>> out<br>
>> redContact,<br>
>> out<br>
>> authResult"/><br>
>><br>
>> <action application="log" data="INFO ########################<br>
>> ServiceLookup ########################\n"/><br>
>> <action application="log" data="INFO ########################<br>
>> contact = '${redContact}' ##############\n"/><br>
>> <action application="log" data="INFO ########################<br>
>> CallerNum = '${caller_id_number:6:16}' ##########\n"/><br>
>> <action application="log" data="INFO ########################<br>
>> RADIUS auth = '${authResult}' ##########\n"/><br>
>><br>
>> <action application="execute_extension" data="doRedirect XML<br>
>> public"/><br>
>> </condition><br>
>> </extension><br>
>><br>
>><br>
>> <extension name="doRedirect"><br>
>> <condition field="destination_number" expression="^doRedirect$"/><br>
>> <condition field="${authResult}" expression="^0$|"><br>
>> <action application="log" data="INFO ########################<br>
>> RADIUS auth OK!!!' ##########\n"/><br>
>> <action application="redirect" data="${red_contact}"/><br>
>> <anti-action application="log" data="INFO<br>
>> ########################<br>
>> RADIUS auth NOK!! ##########\n"/><br>
>> <anti-action application="respond" data="403 Forbidden"/><br>
>> </condition><br>
>><br>
>> </extension><br>
>><br>
>><br>
>><br>
>><br>
>> On Wed, Nov 25, 2009 at 6:45 PM, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>><br>
>> wrote:<br>
>><br>
>>> In trunk there is a sofia profile setting to allow dialplan<br>
>>> processing of<br>
>>> 302 responses. This won't get you back into your same javascript,<br>
>>> but you<br>
>>> can probably do something clever from there.<br>
>>><br>
>>> Mike<br>
>>><br>
>>> On Nov 24, 2009, at 5:04 PM, John Platts wrote:<br>
>>><br>
>>>><br>
>>>> I have considered writing JavaScript code to bridge two calls<br>
>>>> together.<br>
>>> However, I would like to perform custom handling of the 302 Moved<br>
>>> Temporarily response. How do I handle the 302 Moved Temporarily<br>
>>> response if<br>
>>> I use JavaScript?<br>
>>>><br>
>>><br>
>>> _______________________________________________<br>
>>> FreeSWITCH-users mailing list<br>
>>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
>>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
>>> UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-</a><br>
>>> users<br>
>>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>>><br>
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>> users<br>
>> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
>><br>
>><br>
>> End of FreeSWITCH-users Digest, Vol 41, Issue 189<br>
>> *************************************************<br>
><br>
> _______________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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