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Scheduled hangup from javascript
by vidhya sagar dixit 10 Sep '08

10 Sep '08

Hi All, How to use scheduled hangup from within a java script. Here is the flow. User dials 1212 from softphone call goes to default dialplan and calls test.js in test.js i am doing like this ==================== session.answer(); //....... //Do some stuff like ivr play etc ....... //...... var dialstr = "sofia/external/911234567890(a)11.22.33.44"; // now originate a new session new_session = new Session(); new_session.setCallerData("caller_id_name", "Vidhya"); new_session.setCallerData("caller_id_number", "911140500000"); new_session.originate(session, dialstr,60); bridge(session, new_session); new_session.hangup(); session.hangup(); =================== Now i want to hangup new_session after lets say 2 minutes. I can not use scheduled hangup in dialplan as this hangup time may vary and will come from database. Any help will be appreciated. -- Thanks and Regards Vidhya Sagar Dixit

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Recording and Playing Audio
by Robert Clayton 10 Sep '08

10 Sep '08

All, What command denotes the end of audio recording with recordFile so that the recorded file can be played back to the user by streamFile? Bob

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Seeking in recordFIle
by Robert Clayton 10 Sep '08

10 Sep '08

All, Where streamFile allows you to seek forward and backward by samples. Is there anyway to seek backward and play a file to a user allowing the user to pause and continue recording (I assume with recordFile) a file at an intermediate point (not insert). Bob

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listing gateways
by Gayatri Kulkarni 10 Sep '08

10 Sep '08

like "sofia status profile internal" lists the users registered to FS at the moment, is there a command to list out only gateways registered at the moment? I know, "sofia status" lists gateways alongwith other users registered - but what if i want only gateways to be listed? Regards, Gayatri Kulkarni

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registration problem
by Gayatri Kulkarni 10 Sep '08

10 Sep '08

Hi Guys, I have FS installed on 58.86.117.43 I have the X-lite on 68.86.117.44 The X-Lite cannot register to the FS - can you guess why? SIP account settings for Xlite: Username: 1002 Password: 1234 Auth user name: 1002 domain: 58.86.117.43 Topology: IP address: Discover at runtime STUN server: Use specific - 58.86.117.43 I am using a default extension of FS, I dont think i need a gateway here - am i correct? Please reply soon. Appreciate all your help, -- Regards, Gayatri Kulkarni

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How to divert a virtual PSTN line to another server ?
by Henk Oegema 10 Sep '08

10 Sep '08

I use a 'virtual' PSTN line (voip trunk) from (http://www.voxbone.com) as incoming external line to my Asterisk server (192.168.1.100) In my router I have have : Application Start End Protocol IP Address ----------------------------------------------------------------------------- SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.100 RTP 5090 to 5100 UDP 192.168.1.100 That works OK. Now I want to divert that PSTN line from Asterisk to my Freeswitch server (192.168.1.101) So I changed in my router the ip addreese from 192.168.1.100 to 192.168.1.101 Application Start End Protocol IP Address ----------------------------------------------------------------------------- SIP 5004 to 5082 Both(UDP&TCP) 192.168.1.101 RTP 5090 to 5100 UDP 192.168.1.101 But.....when an external call comes in, it still goes to Asterisk. Am I on the wrong track or ....... (?) Rgds Henk

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telnet window
by Jonas Gauffin 10 Sep '08

10 Sep '08

Hello I'm running FS as a NT service. Is it possible to access the FS console through telnet or something like that? //Jonas

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mod_esf and mod_fsv
by Jonas Gauffin 09 Sep '08

09 Sep '08

Hello What does mod_esf and mod_fsv do? mod_esf just says "Extra SIP Functionality" in the wiki. mod_fsv is not documented at all. Regards, Jonas

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Login & Password for mod_portaudio
by Евгений Золотов 09 Sep '08

09 Sep '08

On http://wiki.freeswitch.org/wiki/Freeswitch_softphone is written: > Web GUI > mod_xml_rpc loaded portaudio supplies it's own little gui http://your.box:8080/api/pa We have made: $ make mod_portaudio $ make mod_portaudio-install On running FreeSWITCH does: freeswitch(a)opos8.altron.lan> load mod_portaudio PortAudio version number = 1899 PortAudio version text = 'PortAudio V19-devel (built Sep 9 2008)' Number of devices = 7 2008-09-09 15:39:52 [ERR] mod_portaudio.c:878 load_config() Invalid ring device configured using output device! 2008-09-09 15:39:53 [CONSOLE] switch_loadable_module.c:778 switch_loadable_module_load_file() Successfully Loaded [mod_portaudio] 2008-09-09 15:39:53 [NOTICE] switch_loadable_module.c:139 switch_loadable_module_process() Adding Endpoint 'portaudio' 2008-09-09 15:39:53 [NOTICE] switch_loadable_module.c:257 switch_loadable_module_process() Adding API Function 'pa' freeswitch(a)opos8.altron.lan> load mod_xml_rpc 2008-09-09 15:43:35 [CONSOLE] switch_loadable_module.c:778 switch_loadable_module_load_file() Successfully Loaded [mod_xml_rpc] API CALL [load(mod_xml_rpc)] output: +OK freeswitch(a)opos8.altron.lan> 2008-09-09 15:43:35 [NOTICE] mod_xml_rpc.c:825 mod_xml_rpc_runtime() Starting HTTP Port 8080, DocRoot [/usr/local/freeswitch/htdocs] Now when we put URL in browser (Opera 9.51) : http://opos8:8080/api/pa - we receive request: Login: Password: What should be Login/Password by default?

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What means by *.SPEEX?
by Евгений Золотов 09 Sep '08

09 Sep '08

We have some number of questions in one letter: We make registration seance's audiostream simultaneously into a 2 different files, which have an extensions "wav" and "SPEEX". We use for this extension about such sort: <extension name="20024"> <condition field="destination_number" expression="^20024$"> <action application="set" data="bypass_media=false" /> <action application="set" data="RECORD_ANSWER_REQ="true" /> <action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_COPYRIGHT=(c) 2008, Altron Inc." /> <action application="set" data="RECORD_SOFTWARE=FreeSwitch" /> <action application="set" data="RECORD_ARTIST=Registration" /> <action application="set" data="RECORD_COMMENT=!!!" /> <action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/> <action application="set" data="RECORD_STEREO=true"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="record_session" data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.SPEEX"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="sofia/internal/${dialed_ext}@192.168.2.107:5062" /> <action application="answer" /> </condition> </extension> As result we receive simultaneously 2 files ( audio files are enclosed to the letter) $ ls -l 2008* -rw-r--r-- 1 olej olej 26880 Сен 5 14:13 2008-09-05-14-13-08_20024_1010.SPEEX -rw-r--r-- 1 olej olej 107752 Сен 5 14:13 2008-09-05-14-13-08_20024_1010.wav 1. File *.wav, as one would expect, contains RIFF header with metadata and can be played back with any matching tool, for example we used "sox" *.SPEEX file, in that sort as it is written with FreeSWITCH, contains, as we think looking for its size, compressed by speex codec. Is it really so? Project SPEEX community develops own tools (speexenc, speexdec) plaining, that their speex stream is located into OGG container and thus can be converted into *.wav or another ( with speexdec) and played back ( for example, with sox). 2. As it seems, *.SPEEX file can be played back with another extension by using FreeSWITCH playback application. Is it really so? If it isn't so, than with which 3-rd part tool we could play back *.SPEEX file for the control? 3. And final question: <action application="set" data="RECORD_STEREO=true"/> - does it pack 2 channels from A-leg and B-leg of bridge into one *.wav stream? Can we register only A-leg or B-leg with record_session application?? Thanks in advance! Evgeniy/

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