February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

We need help with the wiki, looking for new administrators
by Diego Viola 08 Feb '11

08 Feb '11

Hello, As some of you might know, I'm one of the administrators of the FreeSWITCH wiki; for the past months I've been trying to keep the wiki clean from spammers and related abuses, which means I spent my time blocking spammers, deleting spam pages, and improving the wiki, etc. Recently I got a job which occupies most of my time so I won't have the time that I had before for doing such things, so if someone have the time for this task please reply on this thread so we can grant you administrator rights. This task isn't hard and it's basically as follows: keep an eye at "Recent changes" here: http://wiki.freeswitch.org/wiki/Special:RecentChanges -- there is also a "Recent changes" link in the main page and in every wiki page, then when you find someone spammed the wiki, block that user with indefinite time, and then delete his/her spam page or the affected pages. Please contact us if you're interested in helping. Sincerely, Diego Viola

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UPDATE: Buy the devs dinner
by Michael Collins 08 Feb '11

08 Feb '11

Hello all! Just an update: all the FreeSWITCH developers are together this week and will be going to dinner. It's still not too late for those who wish to pitch in and help feed them. Just hit the Paypal link on the main page and drop a few dollars! Thanks, MC

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Conference / Page usage
by Jon Young 08 Feb '11

08 Feb '11

I utilized the "Mad Boss" intercom section of the sample dialplan to set up a paging scenario. This worked great on some slower PC's (and slightly older FS builds). I wa utilizing apx 25 stations on a PIII 650Mhz PC. It takes a few seconds for the call setup, but otherwise works and sounds great. I Recently I attempted to move my paging setup (XML files) from the PIII to a 2xXeon 3.4GHz server running VM. My results surprised me. The sound quality is great, but the call setup changed significantly. While the PIII took a while to setup, I was able to communicate soon after the call setup. On the VM machine, the call setup goes quickly and then I hear 25 conf. entrance tones on all of the destinations as the conf. is set up. Can I eliminate the conference announce tones and have just one announce tone...or none? The following is one of the paging groups I defined: <extension name="paging01"> <condition field="destination_number" expression="^1999$"> <action application="set" data="conference_auto_outcall_caller_id_name=Page1"/> <action application="set" data="conference_auto_outcall_caller_id_number=1999"/> <action application="set" data="conference_auto_outcall_timeout=60"/> <action application="set" data="conference_auto_outcall_flags=mute"/> <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/> <action application="set" data="sip_exclude_contact=${network_addr}"/> <action application="conference_set_auto_outcall" data="${group_call(pagezone01)}"/> <action application="conference" data="paging1@default+flags{endconf|deaf}"/> </condition> </extension>

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Embedded Freeswitch .NET Example
by Mitch Capper 08 Feb '11

08 Feb '11

It took awhile but I have updated the Embedded Freeswitch wiki page ( http://wiki.freeswitch.com/wiki/Embedded_FreeSWITCH) with the notes from the presentation a few weeks ago, at the bottom is also a link to a c# example project that demos a softphone is around 100 lines of code that makes use of most of what is talked about on that page. ~Mitch

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Re: [Freeswitch-users] play_and_get_digits question
by Madovsky 08 Feb '11

08 Feb '11

or maybe this channel var read_result is the result of play_and_get_digits ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:43 PM Subject: Re: play_and_get_digits question Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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Routing calls acording to the divertion header
by Meftah Tayeb 08 Feb '11

08 Feb '11

guys, i have a divertion header like this: Diversion:<sip:213983200193@12.13.14.15;user=phone>;reason=unknown;counter=1 and another like: Diversion:<sip:15148001395@12.13.14.15;user=phone>;reason=unknown;counter=1 how can i diferentiate calls acording to the divertion header? thanks -- Meftah Tayeb inum: +883510001288000 phone: +13477595883

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Re: [Freeswitch-users] Using mod_opal with MSN Messenger
by Peter Olsson 08 Feb '11

08 Feb '11

The use for these modules are to communicate with other h323 stacks/PBX'es. MSN is not one of them AFAIK. /Peter ----- Reply message ----- Från: "mazilo" <Nabble(a)slickdeals.endjunk.com> Datum: mån, feb 7, 2011 22:17 Rubrik: [Freeswitch-users] Using mod_opal with MSN Messenger Till: "freeswitch-users(a)lists.freeswitch.org" <freeswitch-users(a)lists.freeswitch.org> Rhosyn, Thanks for your response. So, it looks like there is no way one can configure an FS system using either mod_opal nor mod_h323 to place/receive VoIP calls to/from any MSN/HotMail (Live) users. So, what is the use for either mod_opal or mod_h323? ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Using-mod-opal-with-MSN-Messe… Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4d500cac32762891562358!

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FS ESL in Adobe Flex
by Francois Barrouin 08 Feb '11

08 Feb '11

Hello, I would like to use Adobe Flex to access FS event socket. I read in the (excellent!) “FreeSWITCH 1.0.6” book that FS ESL is based on swig and unfortunately Adobe Flex is not a language supported by this project. I wonder if anyone has already developed a Flex ESL for FS? Thanks François

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mod_dinglaing errors and google voice
by Daniel Wells 08 Feb '11

08 Feb '11

I sent this earlier but am not sure if it went though (I never got it in my inbox). I apologize if you have already seen it. ____________________________________________________________________________________________ So I finally have what appears to be a working version of mod_dingaling on my windows box. I say working only because the console starts and I see evidence in the logs that freeswitch is communicating with my google account though mod_dingaling (I see presence information about some of my contacts). However I am not sure my dialplan is working (just a guess since this my first experience with freeswitch-or any PBX for that matter). I have been trying to follow the instructions as found here: http://wiki.freeswitch.org/wiki/Google_Voice I have a client connected using the default user 1000. I created a dingaling profile called gv1000.xml (I am using the second example of the dialplan on the above site and inferred that the client profile needed to renamed). My profile is identical to the one in the example with the exception of the username, password and "exten" (which is set to 1000). I put a copy of the dial plan in the "default" directory. But I am getting an error when trying to make outbound calls. Does this mean I have done something wrong, or is google blocking this now? freeswitch@dw-laptop> 2011-02-07 10:31:44.361514 [WARNING] sofia_reg.c:1247 SIP auth challenge (INVITE) on sofia profile 'internal' for [18XXXXXXXXX(a)192.168.2.97] from ip 192.168.2.97 2011-02-07 10:31:44.477521 [NOTICE] switch_channel.c:811 New Channel sofia/internal/1000(a)192.168.2.97 [cfda05d4-2f61-4ab0-8ca6-f759cfac53b7] 2011-02-07 10:31:44.689533 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->18XXXXXXXXX in context default 2011-02-07 10:31:44.689533 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:697 Ring Ready sofia/internal/1000(a)192.168.2.97! 2011-02-07 10:31:44.689533 [NOTICE] mod_dingaling.c:721 Close Channel N/A [CS_NEW] 2011-02-07 10:31:44.689533 [ERR] switch_ivr_originate.c:2638 Cannot create outgoing channel of type [dingaling] cause: [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.689533 [INFO] mod_dptools.c:2621 Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2011-02-07 10:31:44.689533 [NOTICE] mod_dptools.c:2684 Hangup sofia/internal/1000(a)192.168.2.97 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/1000(a)192.168.2.97) Ended 2011-02-07 10:31:44.900545 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1000(a)192.168.2.97 [CS_DESTROY] Sincerely, Daniel Wells

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C# problem for calling "freeswitch.switch_core_session_read_frame()"
by Frankie Yiu 08 Feb '11

08 Feb '11

Hi there, I would like to dial a phone number, and read the RTP package back when connected so that we can analyze the data; (to determine when we should play an audio at the right time--human or answer machine.) We found a code for IVR test (http://docs.freeswitch.org/switch__ivr_8c-source.html) Function name: switch_ivr_sound_test We would like to do the same but with C# code, but we encountered a problem when calling the function "freeswitch.switch_core_session_read_frame( ) " < in swig.cs switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session, SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id) > where the second parameter--frame is a pointer to pointer of switch_frame type and in C# code, it is having a difficulty passing an object to the C++ side and keep the pointer place holder before going deeper into the C code (switch_core_io.c) 1) So is there any way I can call this function in C#? 2) Is there another function or routine that you can suggest me to for reading RTP package? Thanks, Frank

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