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thesis on live sip calls migration
by stefano bossi 23 Jan '10

23 Jan '10

Hi all, as promised in #freeswitch-dev I'll try to explain better what I've done and I would like to achieve for my thesis. At the moment I successfully patched sofsip_cli and now it can do something near active call migration :) Thanks to the versatility of nua api I could achieve my objective with little effort. Basically I call someone, then I kill sofsip_cli and I restart it. I launch my "reinvite" command and the call can continue normally. The called client doesn't realize what happened. In this case I write call data in a file and when I launch the reinvite I read directly from this file. Sofsip_cli sends an INVITE message with some TAG appended to the nua_handle. So the called phone thinks to receive an in-session invite and simply refreshes the call, but Sofsip_cli instead allocates all the stuff for a new call. In this way I can avoid to modify directly the RTP part. It seems to be all simpler. I even wrote a little module for FS able to monitor(using the SIP OPTION message) the life of a specific sofia profile.. It's just a proof of concept but it can do failover(without live call migration) between 2 machines in about 50ms. This is possible with these actions: - registrations are shared with odbc - on the start of the module(present only on the backup machine): 1. I set an arp rule blocking the arp response for the virtual IP (set on the primary machine) 2. I set the virtual IP on the backup machine( but no one knows thanks to the arp rule and so I can bind to vIP) 3. I prepare and run the sip profile (on the backup)... loading here the profile permits to save a lot of time during reaction - on the reaction 1. I remove the arp rule 2. I send a gratuitous arp request As I said in about 50ms sip clients can call again. Maybe this time can decrease using NETLINK socket for the arp table. The union of these 2 works will give us a very fast "live profile migration" :D There some points to discuss: - switch_core_session_resurrect_channel: I didn't know this function, I need to understand what it offers, maybe tomorrow ;) - the propagation of call state variables: my first idea was to use the multicast events adding some headers to send all the necessary data (but anthm proposed XML) - I thought only to sofia aspects.. In next days I'll try to understand where to hook in FS to try these ideas. When I'll have something ready (and without very very big errors:) I'll be happy to send you the patch. This is my first work on something real like FS.. I'm opened to any kind of suggestions!* *please contact me for further clarifications Thanks Stefano Bossi

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FSComm Custom Parameters and Crashes
by Paul Li 23 Jan '10

23 Jan '10

Custom Parameter tab is having another problem and crashes under the following situation. If a gateway, gw1, has two custom params, and the second gateway, gw2, has zero or one custom param. After you first try to edit gw1, and then cancel to close, then you proceed to edit gw2, you will see blank row(s). When click OK, FSComm crashes, which might resulted from the blank row(s) in the table. João, it appears that you did not completely clears the entries for the Advance (custom) tab. Other than this problem, everything else looks good, thanks for your hard work. Next time, I will try to log any issue with Jira.

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FSComm preferences
by Dmitry M 23 Jan '10

23 Jan '10

Checking phones preferences I noticed Caller id Name and Number located in General Preferences. IMO it'll be more natural to move them into Account. I expect account may have different proto (SIP,IAX,H323,...), different caller id, different codec settings, etc. Dmitry

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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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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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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 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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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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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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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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