Hello,
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.
My question is how do we avoid such kind of risk?
Thanks
Daniel Zeng
From: freeswitch-users-requ...@lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 42, Issue 193
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 21 Dec 2009 08:21:08 -0800
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--附转发的邮件--
From: anthony.miness...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 21 Dec 2009 09:42:30 -0600
Subject: Re: [Freeswitch-users] Difference between ESL execute()
andexecuteAsync()
if you run the socket in async mode, every call to execute is async
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
On Sat, Dec 19, 2009 at 9:16 PM, Ron McLeod <ron.freeswi...@mcleodnet.com>
wrote:
Here's the ES network trace:
Content-Length: 1502
Content-Type: text/event-plain
Event-Name: CHANNEL_STATE
Core-UUID: bb9ea62a-ed02-11de-91b1-8b7cb185f66f
FreeSWITCH-Hostname: ron-laptop
FreeSWITCH-IPv4: 192.168.100.132
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-12-19%2019%3A12%3A09
Event-Date-GMT: Sun,%2020%20Dec%202009%2003%3A12%3A09%20GMT
Event-Date-Timestamp: 1261278729767397
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_perform_set_running_state
Event-Calling-Line-Number: 1024
Channel-State: CS_ROUTING
Channel-State-Number: 2
Channel-Name: sofia/internal/699%40192.168.100.132
Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: ringing
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 699
Caller-Dialplan: XML
Caller-Caller-ID-Name: Ron%20Soft%20Phone
Caller-Caller-ID-Number: 699
Caller-Network-Addr: 192.168.100.3
Caller-Destination-Number: 444
Caller-Unique-ID: 76021ab2-ed15-11de-91b1-8b7cb185f66f
Caller-Source: mod_sofia
Caller-Context: mytest
Caller-Channel-Name: sofia/internal/699%40192.168.100.132
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1261278729764077
Caller-Channel-Created-Time: 1261278729764077
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f
call-command: execute
execute-app-name: answer
execute-app-arg:
Content-Type: command/reply
Reply-Text: +OK
sendmsg 76021ab2-ed15-11de-91b1-8b7cb185f66f
call-command: execute
execute-app-name: playback
execute-app-arg: /tmp/ann.wav
Content-Type: command/reply
Reply-Text: +OK
> -----Original Message-----
> From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-
> users-boun...@lists.freeswitch.org] On Behalf Of Ron McLeod
> Sent: Saturday, December 19, 2009 5:30 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] Difference between ESL execute()
> andexecuteAsync()
>
> I don't notice any different in behavior between execute() and
> executeAsync(). I was expecting that executeAsync() would return
> right-away, and that execute() would only return after the specified
> application runs to completion (CHANNEL_EXECUTE_COMPLETE event).
>
> Running the sample app below, I see the "About to call execute(playback)"
> and "returned" displayed one right-after the other, even though the file
> being played takes about 4 minutes to play-out.
>
> Do I have this wrong, or is there something incorrect in my app?
>
> APP:
> #!/usr/bin/php
> <?php
> require_once "ESL.php";
>
> $eventSocket = New ESLconnection('192.168.100.132', '8021', 'ClueCon');
> $eventSocket->events('plain', 'CHANNEL_STATE');
> $eventSocket->filter('channel-state', 'CS_ROUTING');
>
> // Wait for new call attempts
> while($eventSocket->connected()){
> $event = $eventSocket->recvEvent();
> $serializedBody = $event->serialize();
> $listOfLines = toArrayOfLines($serializedBody);
> $nameValuePairs = toArrayOfNameValuePairs($listOfLines);
>
> $uuid = $nameValuePairs['Caller-Unique-ID'];
> printf("New call from uuid: $uuid\n");
>
> // answer the caller and play announcement
> $eventSocket->execute('answer', Null ,$uuid);
>
> printf("About to call execute(playback)\n");
> $eventSocket->execute('playback', '/tmp/ann.wav', $uuid);
> printf("returned\n");
> }
> ?>
>
>
> DIALPLAN:
> <?xml version="1.0" encoding="utf-8"?>
> <include>
> <context name="mytest">
> <extension name="mytest">
> <condition field="destination_number" expression="^.*$">
> <action application="park"/>
> </condition>
> </extension>
> </context>
> </include>
>
>
> _______________________________________________
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> --
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--附转发的邮件--
From: and...@hijacked.us
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 21 Dec 2009 11:07:08 -0500
Subject: Re: [Freeswitch-users] [ANN] Spice Telephony 0.9.5 released (FS based
callcenter)
On Sun, Dec 20, 2009 at 11:03:46PM -0800, ram wrote:
> Hi
>
> its good to hear
>
> any compare document between Vicidial and this project
>
No document, but briefly:
* More focused on inbound than on outbound (at least for the moment)
vicidial is more geared for outbound.
* Handles email in queue (and soon chat), vicidial is only voice.
* wrapup time is per-call not static per-'campaign'
* license is a little more liberal
* can operate as a distributed system
* doesn't need asterisk ;)
Andrew
--附转发的邮件--
From: j...@consiglia.dk
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 21 Dec 2009 17:13:21 +0100
Subject: [Freeswitch-users] mod_xml_curl and gateways
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?
All the best /Jon
--附转发的邮件--
From: mrene_li...@avgs.ca
To: freeswitch-users@lists.freeswitch.org
Date: Mon, 21 Dec 2009 11:21:04 -0500
Subject: Re: [Freeswitch-users] mod_xml_curl and gateways
Hi,
All gateways are common, putting them in a user only serves the purpose of
grouping related information together in the XML files.
This said, you can bind to the "configuration" section and return those
gateways as part of the sip profile's xml data.
Cheers,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 21-Dec-09, at 11:13 AM, Jon Bruel wrote:
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?
All the best /Jon
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