[Freeswitch-users] FreeSWITCH-users Digest, Vol 57, Issue 25

Thu Mar 3 00:48:35 MSK 2011

Thanks!

I am using C# managed function through swig.cs.  And I am calling the
function:

switch_ivr_menu_init(SWIGTYPE_p_p_switch_ivr_menu new_menu,.......)

With the type "SWIGTYPE_p_p_switch_ivr_menu", do they type ever work?  It
has a run time error saying "Attempted to read or write protected memory.
This is often an indication that other memory is corrupt." for this variable
basically it is dereferencing a Null pointer.

How do I resolve this if I want to call this in C#?

Thanks,

Frankie

>
> ---------- Forwarded message ----------
> From: Malay Thakershi <mthakershi at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 1 Mar 2011 16:42:47 -0600
> Subject: Re: [Freeswitch-users] How to create IVR application in C#?
> mod_managed could be an option.
>
> http://wiki.freeswitch.org/wiki/Mod_managed
>
> <http://wiki.freeswitch.org/wiki/Mod_managed>It allows you to use most
> native FS features from C# managed code.
>
> Malay
>
> On Tue, Mar 1, 2011 at 3:01 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:
>
>> Hi there,
>>
>> I am newbie to FreeSwitch and I have question about creating an IVR
>> application in C#, with a possibly of using VoiceXML.  Could someone please
>> points me to how I can get started or any example that I can look at?
>>
>> Thanks a lot!!!
>>
>> Frankie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: GillesToo <codecomplete at free.fr>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 1 Mar 2011 14:52:28 -0800 (PST)
> Subject: Re: [Freeswitch-users] Does FS handle three-way calling on the
> same POTS?
>
> mercutioviz wrote:
> > What are the instructions from the provider to create the second call?
> > Usually with an analog line you have to do a hook flash, then dial some
> > more digits, and then do another hook flash
>
> Thanks for the help. Asterisk doesn't allow using Dial() to call a second
> number after the original call was put on hold. A work-around is using
> SendDTMF() to dial the second number, but it appears there's no way to get
> call progress and know if the remote party has answered.
>
> Does FS provide a better way than this hack? I really need to use three-way
> calling on the same line since all calls are free this way.
>
> Thank you.
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Does-FS-handle-three-way-calling-on-the-same-POTS-tp6076537p6079027.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 1 Mar 2011 15:07:28 -0800
> Subject: Re: [Freeswitch-users] route inbound - based on sip account
> The FS on pfSense is pretty old, but if all you are working on is a simple
> routing issue your best bet is to add the "info" app in the public context.
> Somewhere near the top of public.xml just add this:
>
> <extension name="quick info dump" continue="true">
>   <condition>
>     <action application="info"/>
>   </condition>
> </extension>
>
> Save that, press F6 (or do reloadxml) and then make a test inbound call.
> Watch the console - you'll see a TON of information. Look through the pieces
> of data that are displayed. You should be able to find something to key off
> of. Once you've done that then go read up on creating your dialplan here:
>
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Condition
>
> In fact, that whole page is important in understanding the XML dialplan. I
> would read it more than once.
>
> -MC
>
> On Tue, Mar 1, 2011 at 11:16 AM, Andreas Tuerpe <andreas at tuerpe-net.de>wrote:
>
>> Hallo FreeSWITCH Users,
>>
>> I use FS V.0.9.6 on pfSense
>> see on -> [http://doc.pfsense.org/index.php/FreeSWITCH]
>>
>> Symptom:
>> German ISP "Portunity" forward inbound calls without any
>> destination_number.
>>
>>
>> ISP solution tip:
>> FS to register over a second sip account to the provider twice.
>> Any account has a separate number.
>> Based on the channel over which the call come in, I have to decide which
>> number is called.
>>
>> So I need help, which condition assignment must use - howto ???
>> - which fields can I use?
>> - which syntax I have to use?
>>
>>
>> <!--  here my intention ... ???
>>
>>   <extension name="test_did">
>>     <condition field="channel over which the call come in"
>> expression="sip-account-1">
>>       <action application="set" data="called_number=111111"/>
>>       <action application="transfer" data="1001 XML default"/>
>>     <condition field="channel over which the call come in"
>> expression="sip-account-2">
>>       <action application="set" data="called_number=222222"/>
>>       <action application="transfer" data="1002 XML default"/>
>>     </condition>
>>   </extension>
>>
>> ??? -->
>>
>> thanks in advance
>>   tuerpean
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> ---------- Forwarded message ----------
> From: mazilo <Nabble at slickdeals.endjunk.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Tue, 1 Mar 2011 17:48:49 -0800 (PST)
> Subject: Re: [Freeswitch-users] mod_dingaling and google voice
>
> Yossi Neiman wrote:
> > Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060.  Machine FS
> > runs on is an iptables NAT/Firewall/router.  SIP calls to/from the
> > outside world work fine otherwise.
> I am lost here. You have two FS machines and one on a public IP Address
> while the other on a private IP Address? Both are having problems with GV
> incoming calls?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice-tp6072163p6079399.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110302/bd638ffc/attachment.html 