January 2010 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] How to use FSComm to make a call?
by Paul Li 23 Jan '10

23 Jan '10

I have built FSComm on Windows with Visual Studio 2008. I am able to register my VoIP service provider, but I am receiving an error "NO_ROUTE_DESTINATION" when trying to call my home number with "0019876543210". I have tried FreeSwitch with X-Lite before and was able to make the same call. I am wondering where I need to modify my configuration so that I can make a similar call. Any screen print or example of calling is welcome.

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[Freeswitch-dev] How to delay playback during an originate call?
by Paul Li 22 Jan '10

22 Jan '10

I really need some help/advice on how to delay playback during the following originate calls: $ originate {ignore_early_media=true}sofia/gateway/name/number &playback(message.wav) $ originate sofia/example/1000(a)somewhere.com &javascript(test.js) I am hoping playback of message.wav or application (calling test.js) is invoked only after the call is established (connected). Is it possible to add some flag or channel variable, like media_bug_answer_req, to delay playback or call the JavaScript, Lua script, etc.?

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[Freeswitch-dev] IAX2 Support Removed.
by Brian West 22 Jan '10

22 Jan '10

Due to lack of support for the libiax2 being updated to support the newer protocol changes and the lack of interest from anyone willing to actually work on it. I have moved mod_iax to unsupported where it will stay until someone steps up to rewrite a new IAX2 lib. Thanks, Brian

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[Freeswitch-dev] FreeSWITCH Weekly Call Starting in 30 min
by Michael Collins 22 Jan '10

22 Jan '10

Hello all, The weekly conference call will be starting in a bit. Here's the agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_01_22 I'll be on just after 9am my time as I have to get my kids off to school. Talk to you all soon. -Michael

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[Freeswitch-dev] Problem with event_socket application when serving as a consultative transfer target
by Mardy Marshall 21 Jan '10

21 Jan '10

I'm having a problem with an event_socket application which is the transfer target of a consultative transfer. When the transferor begins the transfer process, the outbound event_socket session is created and everything looks normal. When the transfer is completed, switch_ivr_bridge fires a CHANNEL_EXECUTE / "uuid_bridge" event, supplying a new UUID for the "Other-Leg" which corresponds to the transferee. From this point on, I no longer receive events specific to this new UUID, such as DTMF events or CHANNEL_EXECUTE_COMPLETE events after issuing "playback" commands. Regarding the "playback" command, if I direct the command to the "Other-Leg" UUID, I get audio, but no events when it is done. On the other hand, if I direct the command to the original UUID, the one corresponding to the transferrer call leg, I get no audio, but do get completion events. I have tried re-issuing a "myevents" command using the new UUID, but the command fails due to it only being allowed to be called once per session. Am I missing something in order to handle the transfer or have I run into a bug? Thanks, -Mardy

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[Freeswitch-dev] fscomm with celt
by vip killa 21 Jan '10

21 Jan '10

has anyone built fscomm with the celt codec for windows?

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[Freeswitch-dev] FSComm VS2008 Project file - updated
by Paul Li 21 Jan '10

21 Jan '10

The present FSComm VS2008 Project file need to be updated as attached in order to build it with Visual Studio 2008 for the issues: (1) prefaccounts.h/prefaccounts.cpp need to be added to the project. (2) "Generated Files" need to be filtered properly for Debug/Release configuration, so that the proper files are used for the corresponding configuration. (3) Link to proper "FreeSwitchCore.lib", based on the selected configuration. (4) All intermediate files now go to $(IntDir), except for "ui_*.h". All these issues have been corrected with the attached .proj file. Thanks for sharing the app!

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[Freeswitch-dev] Help on switch_channel_expand_variables(switch_channel_t *channel, const char *in)
by Alex To 21 Jan '10

21 Jan '10

Hi, Could somebody shed some lights on how the mentioned method works ? What does it do exactly ? My best guess is it takes in a channel, a string and does something to the string (escape the string ? split the string ? or something like that? But why need a channel parameter?) If there is already an explanation somewhere about this, please point it out for me or possibly just tell me what configuration should I make to trigger this method to be called. I can use debugger and figure it out by myself. Any help would be greatly appreciated. Thanks very much Regards Alex To

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[Freeswitch-dev] NOTICE: mod_iax slated for unsupported on Feb. 5
by Michael Collins 19 Jan '10

19 Jan '10

To everyone in the FreeSWITCH community: We would like to put a call out and let everyone know that unless we find someone who wants to be the maintainer for mod_iax it will be moved to unsupported status as of Friday February 5th. Anyone who wishes to maintain the module will need to be in charge of all bug reports, updates, patches, and technical support questions. If are in a position to maintain mod_iax then please contact Brian West offlist: brian(a)freeswitch.org. -Michael

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[Freeswitch-dev] Openzap and intermitent ringing
by devel＠thom.fr.eu.org 19 Jan '10

19 Jan '10

So I finally could find a fix. Then problem is the ring_on and ring_off length as defined by wp_globals in ozmod_wanpipe.c For France (and I guess all other European countries), the ring on and off length shall be 1700ms and 3300ms. That correspond (at least for Europe) to the length of the ringing tone. So I was wondering I it would not be a good idea to take these values from the tones.conf file. I wonder if that is possible in wp_open_range function in ozmod_wanpipe.c. Could anybody indicate if there is any diagram of how each module in openzap calls the other, then that could be a good starting point (I dont really get the interactions/calls between ozmod_wanpipe.c and ozmod_analog.c). Also, I would need to know if this tone generation should apply to span types other than FXS. Thanks François

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