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Hello,<BR>
<BR>
I noticed some guys had develop the Skype module while there is a policy from Skype(Ulimited call planso call FAP: ):"Each subscription is to be used by one person only and is not to be shared with any other user (whether via a PBX, call centre, computer or any other means)" , which means once you use Skype unlimited calls plan into PBX, Skype will frozen your account without any money return. That's a big risk for anybody to use Skype unlimited call plan.<BR>
<BR>
My question is how do we avoid such kind of risk?<BR>
<BR>
Thanks<BR>
Daniel Zeng<BR>
<BR>From: freeswitch-users-request@lists.freeswitch.org<BR>Subject: FreeSWITCH-users Digest, Vol 42, Issue 193<BR>To: freeswitch-users@lists.freeswitch.org<BR>Date: Mon, 21 Dec 2009 08:21:08 -0800<BR><BR><PRE>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> <A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><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></PRE><BR><BR>--附转发的邮件--<BR>From: anthony.minessale@gmail.com<BR>To: freeswitch-users@lists.freeswitch.org<BR>Date: Mon, 21 Dec 2009 09:42:30 -0600<BR>Subject: Re: [Freeswitch-users] Difference between ESL execute() andexecuteAsync()<BR><BR>if you run the socket in async mode, every call to execute is async<BR>if you don't specify async in the socket app in FS all calls are synchronous but you can send async calls with te asyncExecute<BR><BR><BR>
<DIV class=ecxgmail_quote><BR>On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod <SPAN dir=ltr><<A href="mailto:ron.freeswitch@mcleodnet.com">ron.freeswitch@mcleodnet.com</A>></SPAN> wrote:<BR>
<BLOCKQUOTE class=ecxgmail_quote style="PADDING-LEFT: 1ex"><BR>Here's the ES network trace:<BR><BR><BR><BR>Content-Length: 1502<BR><BR>Content-Type: text/event-plain<BR><BR>Event-Name: CHANNEL_STATE<BR><BR>Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f<BR><BR>FreeSWITCH-Hostname: ron-laptop<BR><BR>FreeSWITCH-IPv4: 192.168.100.132<BR><BR>FreeSWITCH-IPv6: %3A%3A1<BR><BR>Event-Date-Local: 2009-12-19%2019%3A12%3A09<BR><BR>Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT<BR><BR>Event-Date-Timestamp: 1261278729767397<BR><BR>Event-Calling-File: switch_channel.c<BR><BR>Event-Calling-Function: switch_channel_perform_set_running_state<BR><BR>Event-Calling-Line-Number: 1024<BR><BR>Channel-State: CS_ROUTING<BR><BR>Channel-State-Number: 2<BR><BR>Channel-Name: sofia/internal/699%40192.168.100.132<BR><BR>Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f<BR><BR>Call-Direction: inbound<BR><BR>Presence-Call-Direction: inbound<BR><BR>Answer-State: ringing<BR><BR>Channel-Read-Codec-Name: PCMU<BR><BR>Channel-Read-Codec-Rate: 8000<BR><BR>Channel-Write-Codec-Name: PCMU<BR><BR>Channel-Write-Codec-Rate: 8000<BR><BR>Caller-Username: 699<BR><BR>Caller-Dialplan: XML<BR><BR>Caller-Caller-ID-Name: Ron%20Soft%20Phone<BR><BR>Caller-Caller-ID-Number: 699<BR><BR>Caller-Network-Addr: 192.168.100.3<BR><BR>Caller-Destination-Number: 444<BR><BR>Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f<BR><BR>Caller-Source: mod_sofia<BR><BR>Caller-Context: mytest<BR><BR>Caller-Channel-Name: sofia/internal/699%40192.168.100.132<BR><BR>Caller-Profile-Index: 1<BR><BR>Caller-Profile-Created-Time: 1261278729764077<BR><BR>Caller-Channel-Created-Time: 1261278729764077<BR><BR>Caller-Channel-Answered-Time: 0<BR><BR>Caller-Channel-Progress-Time: 0<BR><BR>Caller-Channel-Progress-Media-Time: 0<BR><BR>Caller-Channel-Hangup-Time: 0<BR><BR>Caller-Channel-Transfer-Time: 0<BR><BR>Caller-Screen-Bit: true<BR><BR>Caller-Privacy-Hide-Name: false<BR><BR>Caller-Privacy-Hide-Number: false<BR><BR><BR><BR><BR><BR>sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f<BR><BR>call-command: execute<BR><BR>execute-app-name: answer<BR><BR>execute-app-arg:<BR><BR><BR><BR><BR><BR>Content-Type: command/reply<BR><BR>Reply-Text: +OK<BR><BR><BR><BR><BR><BR>sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f<BR><BR>call-command: execute<BR><BR>execute-app-name: playback<BR><BR>execute-app-arg: /tmp/ann.wav<BR><BR><BR><BR><BR><BR>Content-Type: command/reply<BR><BR>Reply-Text: +OK<BR><BR>
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<DIV class=h5><BR><BR><BR><BR>> -----Original Message-----<BR><BR>> From: <A href="mailto:freeswitch-users-bounces@lists.freeswitch.org">freeswitch-users-bounces@lists.freeswitch.org</A> [mailto:<A href="mailto:freeswitch-">freeswitch-</A><BR><BR>> <A href="mailto:users-bounces@lists.freeswitch.org">users-bounces@lists.freeswitch.org</A>] On Behalf Of Ron McLeod<BR><BR>> Sent: Saturday, December 19, 2009 5:30 PM<BR><BR>> To: <A href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A><BR><BR>> Subject: [Freeswitch-users] Difference between ESL execute()<BR><BR>> andexecuteAsync()<BR><BR>><BR><BR>> I don't notice any different in behavior between execute() and<BR><BR>> executeAsync(). I was expecting that executeAsync() would return<BR><BR>> right-away, and that execute() would only return after the specified<BR><BR>> application runs to completion (CHANNEL_EXECUTE_COMPLETE event).<BR><BR>><BR><BR>> Running the sample app below, I see the "About to call execute(playback)"<BR><BR>> and "returned" displayed one right-after the other, even though the file<BR><BR>> being played takes about 4 minutes to play-out.<BR><BR>><BR><BR>> Do I have this wrong, or is there something incorrect in my app?<BR><BR>><BR><BR>> APP:<BR><BR>> #!/usr/bin/php<BR><BR>> <?php<BR><BR>> require_once "ESL.php";<BR><BR>><BR><BR>> $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');<BR><BR>> $eventSocket->events('plain', 'CHANNEL_STATE');<BR><BR>> $eventSocket->filter('channel-state', 'CS_ROUTING');<BR><BR>><BR><BR>> // Wait for new call attempts<BR><BR>> while($eventSocket->connected()){<BR><BR>> $event = $eventSocket->recvEvent();<BR><BR>> $serializedBody = $event->serialize();<BR><BR>> $listOfLines = toArrayOfLines($serializedBody);<BR><BR>> $nameValuePairs = toArrayOfNameValuePairs($listOfLines);<BR><BR>><BR><BR>> $uuid = $nameValuePairs['Caller-Unique-ID'];<BR><BR>> printf("New call from uuid: $uuid\n");<BR><BR>><BR><BR>> // answer the caller and play announcement<BR><BR>> $eventSocket->execute('answer', Null ,$uuid);<BR><BR>><BR><BR>> printf("About to call execute(playback)\n");<BR><BR>> $eventSocket->execute('playback', '/tmp/ann.wav', $uuid);<BR><BR>> printf("returned\n");<BR><BR>> }<BR><BR>> ?><BR><BR>><BR><BR>><BR><BR>> DIALPLAN:<BR><BR>> <?xml version="1.0" encoding="utf-8"?><BR><BR>> <include><BR><BR>> <context name="mytest"><BR><BR>> <extension name="mytest"><BR><BR>> <condition field="destination_number" expression="^.*$"><BR><BR>> <action application="park"/><BR><BR>> </condition><BR><BR>> </extension><BR><BR>> </context><BR><BR>> </include><BR><BR>><BR><BR>><BR><BR>> _______________________________________________<BR><BR>> FreeSWITCH-users mailing list<BR><BR>> <A href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</A><BR><BR>> <A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR><BR>> UNSUBSCRIBE:<A href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</A><BR><BR>> <A href="http://www.freeswitch.org/">http://www.freeswitch.org</A><BR><BR>><BR><BR></DIV></DIV>> --<BR><BR>> This email was Anti Virus checked by Astaro Security Gateway.<BR><BR>> <A href="http://www.astaro.com/">http://www.astaro.com</A><BR><BR>
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<DIV class=h5><BR><BR><BR><BR>_______________________________________________<BR><BR>FreeSWITCH-users mailing list<BR><BR><A href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</A><BR><BR><A href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</A><BR><BR>UNSUBSCRIBE:<A href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</A><BR><BR><A href="http://www.freeswitch.org/">http://www.freeswitch.org</A><BR><BR></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><BR>Twitter: <A href="http://twitter.com/FreeSWITCH_wire">http://twitter.com/FreeSWITCH_wire</A><BR><BR>AIM: anthm<BR><A href="mailto:MSN:anthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</A><BR>GTALK/JABBER/<A href="mailto:PAYPAL:anthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</A><BR><BR>IRC: <A href="http://irc.freenode.net/">irc.freenode.net</A> #freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A href="mailto:sip:888@conference.freeswitch.org">sip:888@conference.freeswitch.org</A><BR><A href="http://conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</A><BR><BR><A href="mailto:googletalk:conf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</A><BR>pstn:+19193869900<BR><BR><BR><BR>--附转发的邮件--<BR>From: andrew@hijacked.us<BR>To: freeswitch-users@lists.freeswitch.org<BR>Date: Mon, 21 Dec 2009 11:07:08 -0500<BR>Subject: Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based callcenter)<BR><BR><PRE>On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote:<BR>> Hi<BR>> <BR>> its good to hear<BR>> <BR>> any compare document between Vicidial and this project<BR>><BR> <BR>No document, but briefly:<BR> <BR>* More focused on inbound than on outbound (at least for the moment)<BR> vicidial is more geared for outbound.<BR>* Handles email in queue (and soon chat), vicidial is only voice.<BR>* wrapup time is per-call not static per-'campaign'<BR>* license is a little more liberal<BR>* can operate as a distributed system<BR>* doesn't need asterisk ;)<BR> <BR>Andrew<BR> <BR> <BR></PRE><BR><BR>--附转发的邮件--<BR>From: jbr@consiglia.dk<BR>To: freeswitch-users@lists.freeswitch.org<BR>Date: Mon, 21 Dec 2009 17:13:21 +0100<BR>Subject: [Freeswitch-users] mod_xml_curl and gateways<BR><BR><BR><BR><BR><BR><BR><BR><BR>
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<P class=ecxMsoNormal><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">I wonder if it is possible to define common gateways<BR>(not user specific gateways) by xml_curl, and if so, the bindings and syntax to<BR>use?</SPAN></FONT></P><BR><BR>
<P class=ecxMsoNormal><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"> </SPAN></FONT></P><BR><BR>
<P class=ecxMsoNormal><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">All the best /Jon</SPAN></FONT></P><BR><BR>
<P class=ecxMsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> </SPAN></FONT></P><BR><BR>
<P class=ecxMsoNormal><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> </SPAN></FONT></P><BR><BR></DIV><BR><BR><BR><BR><BR><BR><BR>--附转发的邮件--<BR>From: mrene_lists@avgs.ca<BR>To: freeswitch-users@lists.freeswitch.org<BR>Date: Mon, 21 Dec 2009 11:21:04 -0500<BR>Subject: Re: [Freeswitch-users] mod_xml_curl and gateways<BR><BR>Hi,
<DIV><BR>All gateways are common, putting them in a user only serves the purpose of grouping related information together in the XML files.
<DIV><BR></DIV>
<DIV>This said, you can bind to the "configuration" section and return those gateways as part of the sip profile's xml data.</DIV>
<DIV><BR></DIV>
<DIV>Cheers, </DIV>
<DIV><BR></DIV>
<DIV><SPAN class=ecxApple-style-span style="WORD-SPACING: 0px; FONT: medium Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate">
<DIV style="WORD-WRAP: break-word">
<DIV><SPAN class=ecxApple-style-span style="FONT-SIZE: 12px">
<DIV>Mathieu Rene</DIV>
<DIV>Avant-Garde Solutions Inc</DIV>
<DIV>Office: + 1 (514) 664-1044 x100</DIV>
<DIV>Cell: +1 (514) 664-1044 x200</DIV>
<DIV><A href="mailto:mrene@avgs.ca">mrene@avgs.ca</A></DIV>
<DIV><BR></DIV>
<DIV><BR></DIV></SPAN></DIV></DIV></SPAN><BR class=ecxApple-interchange-newline></DIV><BR>
<DIV>
<DIV>On 21-Dec-09, at 11:13 AM, Jon Bruel wrote:</DIV><BR class=ecxApple-interchange-newline>
<BLOCKQUOTE><SPAN class=ecxApple-style-span style="WORD-SPACING: 0px; FONT: medium Helvetica; TEXT-TRANSFORM: none; COLOR: rgb(0,0,0); TEXT-INDENT: 0px; WHITE-SPACE: normal; LETTER-SPACING: normal; BORDER-COLLAPSE: separate">
<DIV lang=DA>
<DIV class=ecxSection1>
<DIV style="FONT-SIZE: 12pt; MARGIN-BOTTOM: 0pt; MARGIN-LEFT: 0cm; MARGIN-RIGHT: 0cm; FONT-FAMILY: 'Times New Roman'"><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">I wonder if it is possible to define common gateways (not user specific gateways) by xml_curl, and if so, the bindings and syntax to use?</SPAN></FONT></DIV>
<DIV style="FONT-SIZE: 12pt; MARGIN-BOTTOM: 0pt; MARGIN-LEFT: 0cm; MARGIN-RIGHT: 0cm; FONT-FAMILY: 'Times New Roman'"><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial"> </SPAN></FONT></DIV>
<DIV style="FONT-SIZE: 12pt; MARGIN-BOTTOM: 0pt; MARGIN-LEFT: 0cm; MARGIN-RIGHT: 0cm; FONT-FAMILY: 'Times New Roman'"><FONT face=Arial size=2><SPAN lang=EN-GB style="FONT-SIZE: 10pt; FONT-FAMILY: Arial">All the best /Jon</SPAN></FONT></DIV>
<DIV style="FONT-SIZE: 12pt; MARGIN-BOTTOM: 0pt; MARGIN-LEFT: 0cm; MARGIN-RIGHT: 0cm; FONT-FAMILY: 'Times New Roman'"><FONT face="Times New Roman" size=3><SPAN style="FONT-SIZE: 12pt"> </SPAN></FONT></DIV>
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