<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Thanks so much Anthony but I have one more question:<br><br>I was checking the source file sofia_reg.c and it seems that the code had been written iin such a way that FreeSWITCH can authenticate SIP agents based on RFC2069 and RFC2617. Is that conclusion correct?<br><br>Thanks in advance, <br><br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"><pre><br>Message: 2<br>Date: Thu, 5 Feb 2009 10:46:54 -0600<br>From: Anthony Minessale <anthony.minessale@gmail.com><br>Subject: Re: [Freeswitch-users] SIP Authentication<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <191c3a030902050846o60047c30pa2890707eae386d6@mail.gmail.com><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>It's optional for us but it's mandatory for the client if we exercise<br>the<br>option which we have opted to
always do =D<br>There is no way in the code to disable sending it because we prefer the more<br>secure version of SIP auth.<br>So again it's a bug in the client for not following the protocol. It would<br>be considered a feature in FreeSWITCH to support limping for the sake of<br>this broken client and we currently do not have any plans for implementing<br>this feature.<br><br><br><br><br><br>On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie<br><kerrada2003@yahoo.com>wrote:<br><br>><br>> We're using HelpCaster softphone.<br>><br>> The issue here is that in Digest Authentication, if the server sends the<br>> parameter "qop" in the challenge then the client should respond<br>with the<br>> "cnonce" parameter. The parameter "qop" is optional in<br>Digest Auth. So the<br>> question here is that, can we configure FreeSWITCH so that it will not<br>send<br>> "qop" in the challenge?<br>><br>> Thanks!<br>><br>> --- On *Wed,
2/4/09, freeswitch-users-request@lists.freeswitch.org <<br>> freeswitch-users-request@lists.freeswitch.org>* wrote:<br>><br>> From: freeswitch-users-request@lists.freeswitch.org <<br>> freeswitch-users-request@lists.freeswitch.org><br>> Subject: Freeswitch-users Digest, Vol 32, Issue 39<br>> To: freeswitch-users@lists.freeswitch.org<br>> Date: Wednesday, February 4, 2009, 2:05 PM<br>><br>> Send Freeswitch-users mailing list submissions to<br>> freeswitch-users@lists.freeswitch.org<br>><br>> To subscribe or unsubscribe via the World Wide Web, visit<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> or, via email, send a message with subject or body 'help' to<br>> freeswitch-users-request@lists.freeswitch.org<br>><br>> You can reach the person managing the list at<br>> freeswitch-users-owner@lists.freeswitch.org<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: SIP Authentication (Brian West)<br>> 2. Re: origainate through sofia gateway (Michael Collins)<br>> 3. Recording background music and voice is out of sync (Daniel Liang)<br>> 4. Re: Q931 decoding Update (Gopalakrishnan A.N)<br>> 5. mod_limit (Chav Paskov)<br>> 6. Re: mod_limit (Michael Collins)<br>> 7. Re: mod_limit (Chav<br>> Paskov)<br>> 8. Re: mod_limit (Michael Collins)<br>><br>><br>> ----------------------------------------------------------------------<br>><br>> Message: 1<br>> Date: Wed, 4 Feb 2009 10:52:45 -0600<br>> From: Brian West <brian@freeswitch.org><br>> Subject: Re: [Freeswitch-users] SIP Authentication<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID:
<7DAC91F6-2FD3-464D-AA81-321EBCADC8C0@freeswitch.org><br>> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes<br>><br>> What client is this? I also notice we receive port 3458 and reply to<br>> port 1059...<br>><br>> /b<br>><br>> On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote:<br>><br>> > What I have noted is that the client does not send the values for<br>> > "cnonce" and "nc" in the response. I'm not<br>sure if<br>> this is the<br>> > reason, however how this problem can be solved?<br>> ><br>> > Thanks,<br>> ><br>> > Ali<br>><br>><br>><br>><br>> ------------------------------<br>><br>> Message:<br>> 2<br>> Date: Wed, 4 Feb 2009 09:41:07 -0800<br>> From: Michael Collins <msc@freeswitch.org><br>> Subject: Re: [Freeswitch-users] origainate through sofia gateway<br>> To: freeswitch-users@lists.freeswitch.org<br>>
Message-ID:<br>> <87f2f3b90902040941r61d669aaie949aa7cc8578a9a@mail.gmail.com><br>> Content-Type: text/plain; charset=ISO-8859-1<br>><br>> I'll make sure the substance of this is in the wiki and I'll look<br>for<br>> references to the deprecated way and remove those.<br>> -MC<br>><br>> On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale<br>> <anthony.minessale@gmail.com> wrote:<br>> > Where did you learn how to use js this way?<br>> > session.originate is being misused here and is depricated and may be<br>> > removed.<br>> ><br>> > the first arg to session.originate is either undefined or a<br>*different*<br>> > session (the a leg)<br>> ><br>> > session1 = new Session();<br>> > session1.originate(undefined,<br>> ><br>> "{ignore_early_media=true}user/1008@192.168.1.122");<br>> ><br>>
><br>><br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>> ><br>> > //once you have session1 when you originate session2 you pass<br>session1 as<br>> > the arg<br>> > // the effective_caller_id is taken from session1<br>> ><br>> > session2 = new Session();<br>> > session2.originate(session1,<br>> "sofia/gateway/halonet/0225490317");<br>> ><br>> > Anyway this whole code is depricated in favor of this:<br>> ><br>> > session1 = new<br>> Session("{ignore_early_media=true}user/1008@192.168.1.122");<br>> > if (session1.ready()) {<br>> ><br>><br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>> > session2 = new<br>Session("sofia/gateway/halonet/0225490317",<br>> session1);<br>> > }<br>> ><br>> > and could be further refactored down to this:<br>> ><br>> > session1 = new<br>>
Session("{ignore_early_media=true}user/1008@192.168.1.122");<br>> > if<br>> (session1.ready()) {<br>> ><br>><br>session1.setVariable("effective_caller_id_number","fixed0248b");<br>> > session1.execute("bridge",<br>> "sofia/gateway/halonet/0225490317");<br>> > }<br>> ><br>> > or down to this one line of code that will setup the call detached<br>from<br>> the<br>> > script and exit.<br>> ><br>> > var result = apiExecute("originate",<br>> ><br>><br>"{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122<br>> > bridge:sofia/gateway/halonet/0225490317 inline");<br>> ><br>> > if you dont care about the result and want to exit even before the<br>call is<br>> > completed.<br>> ><br>> > var result = apiExecute("bgapi", "originate<br>>
><br>><br>{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008@192.168.1.122<br>> > bridge:sofia/gateway/halonet/0225490317 inline");<br>> ><br>> ><br>> ><br>> > On Wed, Feb 4, 2009 at<br>> 2:51 AM, Jacek Sokulski<br>> <jsokulski@dotsystems.pl><br>> > wrote:<br>> >><br>> >> We have tried setting both effective_caller_id_number and<br>> >> origination_caller_id_number:<br>> >><br>> >><br>> >><br>><br>session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15);<br>> >> but the problem still exists. The solution we have found for the<br>case<br>> >> when we originate two calls, local and external, is as follow:<br>> >><br>> >> session1 = new Session();<br>>
>><br>><br>session1.originate(session1,"user/1003@192.168.1.122",15);//local<br>> >> if(session1.ready()) {<br>> >> <br>session1.execute("execute_extension","00930691688627<br>> XML<br>> >> default");//external<br>> >> }<br>> >><br>> >> so the external call goes through the dialplan.<br>> >> It does not work if both calls are external. One possible<br>solution<br>> could<br>> >><br>> be<br>> >> to pass the originating call through dialplan (loopback?) but we<br>have<br>> not<br>> >> managed<br>> >> to figure out how to do it.<br>> >><br>> >> Thanks<br>> >> Jacek<br>> >><br>> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner<br>pisze:<br>> >> > Oops! Well, fortunately I don't use that voip provider<br>> anymore (nor the<br>> >> > script).<br>>
>> ><br>> >> > Thanks Brian.<br>> >> ><br>> >> > Nicolas<br>> >> ><br>> >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West<br>> <brian@freeswitch.org> wrote:<br>> >> > > YOU should NEVER use this method or call setCallerData<br>at<br>> all you<br>> >> > > should use the correct methods to override the<br>callerid.<br>> >> > ><br>> >> > > If its a B-Leg born from an A-Leg you use these on the<br>on<br>> the A-Leg:<br>> >> > ><br>> >> > ><br>> >> ><br>> ><br>> http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name<br>> >> > ><br>> >> > ><br>><br>http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number<br>> >> > ><br>> >> > > If you're originating you use
this:<br>> >> > ><br>> >> > ><br>> >> > ><br>><br>http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name<br>> >> > ><br>> >> > ><br>><br>http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number<br>> >> > ><br>> >> > > /b<br>> >> ><br>> >> > _______________________________________________<br>> >> > Freeswitch-users mailing list<br>> >> > Freeswitch-users@lists.freeswitch.org<br>> >> ><br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >><br>> ><br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> > http://www.freeswitch.org<br>> >><br>> >><br>> >> _______________________________________________<br>> >> Freeswitch-users mailing
list<br>> >> Freeswitch-users@lists.freeswitch.org<br>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >><br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> http://www.freeswitch.org<br>> ><br>> ><br>> ><br>> ><br>> > --<br>> > Anthony Minessale II<br>> ><br>> > FreeSWITCH http://www.freeswitch.org/<br>> > ClueCon http://www.cluecon.com/<br>> ><br>> > AIM: anthm<br>> > MSN:anthony_minessale@hotmail.com<br><MSN%3Aanthony_minessale@hotmail.com><br>> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br><PAYPAL%3Aanthony.minessale@gmail.com><br>> > IRC: irc.freenode.net #freeswitch<br>> ><br>> > FreeSWITCH Developer Conference<br>> > sip:888@conference.freeswitch.org<br><sip%3A888@conference.freeswitch.org><br>> ><br>>
iax:guest@conference.freeswitch.org/888<br>> > googletalk:conf+888@conference.freeswitch.org<br><googletalk%3Aconf%2B888@conference.freeswitch.org><br>> > pstn:213-799-1400<br>> ><br>> > _______________________________________________<br>> > Freeswitch-users mailing list<br>> > Freeswitch-users@lists.freeswitch.org<br>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> ><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> ><br>> ><br>><br>><br>><br>> ------------------------------<br>><br>> Message: 3<br>> Date: Wed, 4 Feb 2009 09:43:10 -0800<br>> From: "Daniel Liang" <Daniell@airg.com><br>> Subject: [Freeswitch-users] Recording background music and voice is<br>> out of sync<br>> To: <freeswitch-users@lists.freeswitch.org><br>> Message-ID:<br>>
<0B02E756F603CC409EB553879B090CC80A23EBB5@HPEXCHVS01.exchange.airg><br>> Content-Type: text/plain; charset="us-ascii"<br>><br>> What I did was the following:<br>><br>> First, I sent the<br>> playback command:<br>><br>> call-command: execute<br>> execute-app-name: playback<br>> execute-app-arg: <filename><br>><br>> Then I send uuid_record (Sorry, it was not Record command):<br>><br>> api uuid_record <uuid> start <filename> 120<br>><br>> I also tried replacing the playback command with:<br>> api uuid_displace <uuid> start <filename> 0 mux<br>><br>> But the end results are the same. The recorded user's voice is about<br>0.5<br>> second behind the expected result.<br>><br>> Thanks,<br>> Daniel<br>><br>><br>> -----Original Message-----<br>> From: freeswitch-users-bounces@lists.freeswitch.org<br>>
[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of<br>> Brian West<br>> Sent: February 3, 2009 6:36 PM<br>> To: freeswitch-users@lists.freeswitch.org<br>> Subject: Re: [Freeswitch-users] Recording background music and voice is<br>> outof sync<br>><br>> Can you show us an example of how you're doing this? Playback and<br>> Record aren't async so<br>> you'll need to show us how you're doing<br>> this.<br>><br>> Also don't hijack threads you hit replay on the one "Re:<br>[Freeswitch-<br>> users] FreeSwitch setup as a "Dumb" SBC" as the subject,<br>deleted<br>> the<br>> subject and started a new body. That hijacks the thread and that can<br>> cause your problem to go ignored in some cases if people aren't<br>> interested in the thread topic depending on how their reader threads the<br>> emails.<br>><br>> Please click new message and type freeswitch-
users@lists.freeswitch.org<br>> in and then input your subject and body to start a new thread.<br>><br>> Thanks,<br>> Brian West<br>> FreeSWITCH.org<br>><br>><br>> On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote:<br>><br>> > Hi,<br>> ><br>> > I was trying to record a background music with a user's voice at<br>the<br>> > same time. I did a playback and started recording. But the recorded<br>> > user's voice and the background music is about 0.5 second out of<br>sync.<br>><br>> > I also tried<br>> to use uuid_displace instead of playback, but I got the<br>> > same result.<br>><br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>>
http://www.freeswitch.org <http://www.freeswitch.org/><br>><br>> -------------- next part --------------<br>> An HTML attachment was scrubbed...<br>> URL:<br>><br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/2d1124e2/attachment-0001.html<br>><br>><br>> ------------------------------<br>><br>> Message: 4<br>> Date: Wed, 4 Feb 2009 23:26:14 +0530<br>> From: "Gopalakrishnan A.N" <saigop@gmail.com><br>> Subject: Re: [Freeswitch-users] Q931 decoding Update<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID:<br>><br>> <2ea4d47e0902040956v75c5472foa4649c50b7340484@mail.gmail.com><br>> Content-Type: text/plain; charset="iso-8859-1"<br>><br>> Hi,<br>> Its a awesome. Can the packet capturing be done with event socket?<br>><br>> --<br>> Thank you with regards,<br>> Gopal,<br>> -------------- next part
--------------<br>> An HTML attachment was scrubbed...<br>> URL:<br>><br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090204/74f6434a/attachment-0001.html<br>><br>><br>> ------------------------------<br>><br>> Message: 5<br>> Date: Wed, 04 Feb 2009 09:59:48 -0800<br>> From: Chav Paskov <chavpaskov@shaw.ca><br>> Subject: [Freeswitch-users] mod_limit<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID: <4989D794.1010805@shaw.ca><br>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>><br>> Hi ,<br>> is it possible to use mod_limit in case if the end point is not<br>> registered / gateway for<br>> example/.<br>> Regards<br>> Chav<br>><br>><br>><br>> ------------------------------<br>><br>> Message: 6<br>> Date: Wed, 4 Feb 2009 10:06:52 -0800<br>> From: Michael Collins <msc@freeswitch.org><br>>
Subject: Re: [Freeswitch-users] mod_limit<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID:<br>> <87f2f3b90902041006v7d7d7ff2t9961274905d9d5b0@mail.gmail.com><br>> Content-Type: text/plain; charset=ISO-8859-1<br>><br>> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov <chavpaskov@shaw.ca><br>wrote:<br>> > Hi ,<br>> > is it possible to use mod_limit in case if the end point is not<br>> > registered / gateway for example/.<br>><br>> Could you add some detail to this question? What are you trying to do?<br>> (mod_limit may or may not work, but there might be another solution<br>> which is why I am asking.)<br>><br>> -MC<br>><br>> > Regards<br>> > Chav<br>> ><br>> > _______________________________________________<br>> > Freeswitch-users mailing list<br>> ><br>> Freeswitch-users@lists.freeswitch.org<br>> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> ><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> ><br>><br>><br>><br>> ------------------------------<br>><br>> Message: 7<br>> Date: Wed, 04 Feb 2009 10:54:56 -0800<br>> From: Chav Paskov <chavpaskov@shaw.ca><br>> Subject: Re: [Freeswitch-users] mod_limit<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID: <4989E480.1080105@shaw.ca><br>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>><br>> Michael Collins wrote:<br>> > On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov<br><chavpaskov@shaw.ca><br>> wrote:<br>> ><br>> >> Hi ,<br>> >> is it possible to use mod_limit in case if the end point is not<br>> >> registered / gateway for example/.<br>> >><br>> ><br>> >
Could you add some detail to this question? What are you trying to<br>do?<br>> ><br>> (mod_limit may or may not work, but there might be another solution<br>> > which is why I am asking.)<br>> ><br>> > -MC<br>> ><br>> ><br>> >> Regards<br>> >> Chav<br>> >><br>> >> _______________________________________________<br>> >> Freeswitch-users mailing list<br>> >> Freeswitch-users@lists.freeswitch.org<br>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >><br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> http://www.freeswitch.org<br>> >><br>> >><br>> ><br>> > _______________________________________________<br>> > Freeswitch-users mailing list<br>> > Freeswitch-users@lists.freeswitch.org<br>> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> ><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> ><br>> ><br>> i have few gateways under my ACL that<br>> are allowed to send calls to FS,<br>> but i want to be able to enforce "capacity" policy on the<br>traffic<br>> coming from any one of them depending on total termination capacity on<br>> my termination end.<br>> Let say GW 1 has to be limited to make 10 simultaneous calls while GW2<br>> could make up to 30 and so on.<br>> Regards<br>> Chav<br>><br>><br>><br>> ------------------------------<br>><br>> Message: 8<br>> Date: Wed, 4 Feb 2009 11:05:09 -0800<br>> From: Michael Collins <msc@freeswitch.org><br>> Subject: Re: [Freeswitch-users] mod_limit<br>> To: freeswitch-users@lists.freeswitch.org<br>> Message-ID:<br>>
<87f2f3b90902041105l50f51f08t230bab8d69eefb4e@mail.gmail.com><br>> Content-Type: text/plain; charset=ISO-8859-1<br>><br>> On Wed, Feb 4, 2009 at 10:54 AM, Chav Paskov <chavpaskov@shaw.ca><br>wrote:<br>> > Michael Collins wrote:<br>> >> On Wed, Feb 4, 2009 at 9:59 AM, Chav Paskov<br><chavpaskov@shaw.ca><br>> wrote:<br>> >><br>> >>> Hi<br>> ,<br>> >>> is it possible to use mod_limit in case if the end point is<br>not<br>> >>> registered / gateway for example/.<br>> >>><br>> >><br>> >> Could you add some detail to this question? What are you trying<br>to do?<br>> >> (mod_limit may or may not work, but there might be another<br>solution<br>> >> which is why I am asking.)<br>> >><br>> >> -MC<br>> >><br>> >><br>> >>> Regards<br>> >>> Chav<br>>
>>><br>> >>> _______________________________________________<br>> >>> Freeswitch-users mailing list<br>> >>> Freeswitch-users@lists.freeswitch.org<br>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >>><br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >>> http://www.freeswitch.org<br>> >>><br>> >>><br>> >><br>> >> _______________________________________________<br>> >><br>> Freeswitch-users mailing list<br>> >> Freeswitch-users@lists.freeswitch.org<br>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >><br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> http://www.freeswitch.org<br>> >><br>> >><br>> > i have few gateways under my ACL that
are allowed to send calls to<br>FS,<br>> > but i want to be able to enforce "capacity" policy on the<br>> traffic<br>> > coming from any one of them depending on total termination capacity<br>on<br>> > my termination end.<br>> > Let say GW 1 has to be limited to make 10 simultaneous calls while<br>GW2<br>> > could make up to 30 and so on.<br>><br>> I'm sure that this is possible. I don't personally have a way to<br>test<br>> all of this but I know that a number of our users are doing things<br>> like this currently. Can you hop on to the IRC channel? #freeswitch on<br>> irc.freenode.net. A lot of people there can help with<br>> this one.<br>><br>> -MC (IRC: mercutioviz)<br>><br>> > Regards<br>> > Chav<br>> ><br>> > _______________________________________________<br>> > Freeswitch-users mailing list<br>> > Freeswitch-users@lists.freeswitch.org<br>> >
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> ><br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> ><br>><br>><br>><br>> ------------------------------<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br>><br>> End of Freeswitch-users Digest, Vol 32, Issue 39<br>> ************************************************<br>><br>><br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>>
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