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@gmail.com> To: FreeSWITCH Users Help <freeswitch-users@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@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

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---------- Forwarded message ---------- From: GillesToo <codecomplete@free.fr> To: freeswitch-users@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.

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---------- Forwarded message ---------- From: Michael Collins <msc@freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users@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@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

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---------- Forwarded message ---------- From: mazilo <Nabble@slickdeals.endjunk.com> To: freeswitch-users@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?

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