[Freeswitch-users] Skypiax: Skype account frozen

Mon Dec 21 10:16:14 PST 2009

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-request at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 42, Issue 193
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 21 Dec 2009 08:21:08 -0800

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--附转发的邮件--
From: anthony.minessale at gmail.com
To: freeswitch-users at 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.freeswitch at 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-bounces at lists.freeswitch.org [mailto:freeswitch-

> users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod

> Sent: Saturday, December 19, 2009 5:30 PM

> To: freeswitch-users at 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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--附转发的邮件--
From: andrew at hijacked.us
To: freeswitch-users at 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: jbr at consiglia.dk
To: freeswitch-users at 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_lists at avgs.ca
To: freeswitch-users at 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
mrene at 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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