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Freeswitch and Google PUSH (C2DM)
by Nacho 10 Jul '11

10 Jul '11

Hello all. My name is Jose. This is my first post here. I joined the list because I was wondering about developing a feature for Freeswitch and I would like to know your opinion. The idea is about making a non-persistent client for cell phones that works with PUSH technology (C2DM and Android for example). These clients wouldn't be connected to the Freeswitch server at first, but if there is an incoming call, the Freeswitch server would send a PUSH message to these clients, the client would process it and, if accepted by the user, the client would awake, connect to Freeswitch and then receive the the call invite message and accept it automatically. I don't know if this idea is plausible due to the real time restrictions we have to face in phone calls. If the delay introduced by C2DM delivery is high, the waiting time for the caller is probably something that he isn't willing to accept in order to get his call answered. Another question is about the battery life time it would save because the application woulnd't need to be answering the "still alive" ACKS from Freeswitch. I'm new in freeswitch developing and still learning about it, but I'm really interested in this, so any help would be kindly appreciated. -- Aquellos que hablan son esclavos de sus palabras y los que callan dueños de su silencio.

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How to Submit patches?
by Tihomir Culjaga 07 Jul '11

07 Jul '11

is there a procedure to do that ?

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Chat Dialplan
by Kurtis Heimerl 07 Jul '11

07 Jul '11

Hello Freeswitch-dev! I've recently begin modifying Sofia to support a chat "dialplan", very similar to the voice dialplan. Basically, a chat/sms will come in at a certain number/address and be routed via the dialplan. My current plan for doing this is as follows: 1) Add message events to sofia: I want sofia (and eventually all chat clients) to send message events through FS's event framework when an event arrives. 2) Remove message routing from Sofia: Sofia should not be routing messages itself. Instead, it should just generate the event saying a message has arrived and someone else should do the routing (namely my dialplan) 3) Implement a simple chat dialplan module that listens for message events and routes them according to an XML dialplan. I have some questions for the FS developer community. First, building my own module is conceptually simple. Modifying Sofia is not. What's the best way to go about making these changes? I could submit extension tickets (e.g., a configuration variable that adds messages and one that removes routing), or submit patches, or both. I think this could be broadly useful, and I want to try to contribute to the community. Secondly, I'm having a little difficulty finding the place where messages are routed in sofia. That's probably the place to insert both 1 & 2. Any pointers would be appreciated. Thanks!

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Flex RTMP Client Added To Git Tree!
by Michael Collins 06 Jul '11

06 Jul '11

I thought many of you would be interested in this story about the RTMP Flex client: http://www.freeswitch.org/node/332 The sample Flex client has been added to tree, so feel free to start experimenting! -Michael

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ClueCon News: Free OpenSIPS Training, Sponsor Logo - Last Chance, Party RSVP
by marketing＠cluecon.com 06 Jul '11

06 Jul '11

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FreeSWITCH Conference Call Today
by Michael Collins 06 Jul '11

06 Jul '11

C'mon down! http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_06 Lots of new stuff to talk about! -MC

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mod_avmd with wav file or wav buffer
by laurent.habib＠voila.fr 06 Jul '11

06 Jul '11

Hello, Do you have any exemple for detecting beep in wav file with your mod_avmd module ? (used langage C) Thanks a lot Laurent ___________________________________________________________ Les dernières infos et exclus de stars sont sur Voila people http://people.voila.fr

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api_hangup_hook and channel variable
by Dome Charoenyost 06 Jul '11

06 Jul '11

This is my lua script session:setVariable("session_in_hangup_hook", "true"); session:setVariable("api_hangup_hook", "lua hangup/report.lua "..macct); session:transfer("dialnow_wh", "XML", "wh"); It's work fine before hangup i cal see session vation in report.lua but my quession is how to manipulate or create new variable in report.lua ? i want to see variable in cdr BG Dome C.

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Trouble with bind_digit_action and exec:execute_extension
by Michael Collins 06 Jul '11

06 Jul '11

Hello, I know that some have recently reported having trouble with bind_digit_action and exec:execute_extension. I have not been able to reproduce any of the reported symptoms. I've produced a simple test dialplan that you can drop into conf/dialplan/default/ and just reloadxml for a quick test. I've attached it and also pasted it here for reference. If you've been having BDA trouble please try this out and report back. Be sure to use pastebin.freeswitch.org for reporting your logs. -MC Dialplan: <extension name="BDA Test"> <condition field="destination_number" expression="^77(\d\d\d\d)$"> <action application="answer"/> <action application="bind_digit_action" data="test1,*1,exec:execute_extension,BDA_TARGET_1 XML default"/> <action application="bind_digit_action" data="test1,*2,exec:execute_extension,BDA_TARGET_2 XML default"/> <action application="bind_digit_action" data="test1,*3,exec:execute_extension,BDA_TARGET_3 XML default"/> <action application="bind_digit_action" data="test1,*4,exec:execute_extension,BDA_TARGET_4 XML default"/> <action application="bind_digit_action" data="test1,*5,exec:execute_extension,BDA_TARGET_5 XML default"/> <action application="bind_digit_action" data="test1,*6,exec:execute_extension,BDA_TARGET_6 XML default"/> <action application="digit_action_set_realm" data="test1"/> <action application="playback" data="${hold_music}"/> </condition> </extension> <extension name="BDA Target"> <condition field="destination_number" expression="^BDA_TARGET_(\d)$"> <action application="say" data="en number pronounced $1"/> </condition> </extension> Just dial 77xxxx and then press *1, *2, etc. You will hear the number voiced, i.e. *1 will say "one", *2 will say "two" and so forth.

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Removing SDP from 183/180
by Artur Cichocki 05 Jul '11

05 Jul '11

Hi All. I just wrote this patch. It ads 'sip_remove_183sdp' flag, which removes SDP from 183 and returns 180 (it also removes SDP from 180, because FS changes 180 with SDP to 183). To turn it on add {sip_remove_183sdp=true} to the dial string. Why? Because in some situations we don't want to transfer voice between clients before a real call establishment (and yes, I know that early media in this stage is allowed by RFC). Real case: Try to setup two clients, one of them using Sipdroid. Turn on bypass mode. Try to call to Sipdroid client, you will hear caller voice together with the ring (because Sipdroid sends 180 with SDP, and puts voice on the speaker). Patch attached. -- Artur Cichocki

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