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i need help regarding freeswitch
by Faisal Maqsoodi 19 Nov '08

19 Nov '08

Dear all, Hi I ve just started reading on freeswitch. >From where should i start? Basically, i am am a graduate electronic engineer. Can i be a successful developer of freeswitch? Its my first experience of any telephonic system. Faisal rehman, Pakistan

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Anthony is all heart ...
by Gonzalo Servat 19 Nov '08

19 Nov '08

Hi All, We all like to be thanked and when someone does something for me, I like to show my gratitude towards them and I think what Anthony did for me today deserves a public show of gratitude. It goes like this: I have a single port FXO card in my home machine running FS and I needed to modify the tone_detect app so that it would not incorrectly detect busy tones during phone calls. If I call in and hang up on the remote end, FS would detect the busy tone and hang up correctly. The problem was that tone_detect would act as soon as it detected the requested tone (in my case, busy tone) so I might be in the middle of a phone call it would sometimes incorrectly detect a single busy tone and drop the call. I tried to ignore this at first but it started happening more often so I spoke to Anthony about the possibility of modifying tone_detect to have a "hit count" during a time window, so that it would have to detect X number of busy tones before hanging up the call. As you can all see, this is probably a minor issue on Anthony's priority list since there are many more important issues to fix or things to do on FS than fix tone_detect. Even so, he spent a good 6 hours on this issue and kept hacking at it until he got it fixed. Anthony, once again, fixing tone_detect really helped me but what I appreciate more than anything is the willingness to help and invest so much of your time on this .. so thank you once again! - Gonzalo (aka znoG) PS: I will soon update the wiki page http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect with details on how to use the hit count

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RFC 4028 - SIP Session Timers
by Iñaki Baz Castillo 18 Nov '08

18 Nov '08

Hi, I've read that FS supports/implements Session Timers to monitorice both legs of a call. How to enable it? I mean: alice ------- FS -------- bob - alice calls bob vía FS - FS calls bob. - bob answers (sends 200 OK). - "bypass_media" mode, no RTP through FS. - FS establishes a SIP dialog with alice and other one with bob. - From this moment FS starts sending periodically in-dialog INVITE/UPDATE to both legs in order to check if each SIP dialog is still alive in both endpoints. - In case alice crashes (looses dialog info), alice will reply "481 Call/Transaction doesn't exist" when the in-dialog INVITE/UPDATE arrives from FS, so FS will understand that alice has ended the dialog (or has crashed) and sends a BYE to bob. Is it possible with FS? how to enable it? -- Iñaki Baz Castillo <ibc(a)aliax.net>

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Relative timeout in Session.collectInput?
by mszlazak＠aol.com 18 Nov '08

18 Nov '08

Session.collectInput looks like it has a fixed timeout setting which blocks the channel until the timeout expires. I've been informed that appears to be the case from a brief examination of how speech is collected. SWITCH_DECLARE(switch_status_t) switch_ivr_collect_digits_callback? seems to use a fixed timeout to cancel recording. ?? Is it possible to set-up a "relative" timeout which is based on the time from when the last valid sound was recognized as in choosing toppings from a list of 20 toppings in the pizza demo? This situation involves an indeterminate number of valid sounds (toppings) and thus is unlike the dtmf example give in? http://wiki.freeswitch.org/wiki/Session_collectInput where a counter in the callback is also used to determine when to unblock the channel. Mark. ?

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Call answered by human or answerphone
by Birgit Arkesteijn 18 Nov '08

18 Nov '08

Hi, Our customer has different requirements for the situation where a person answered a call or a answerphone. I found a solution on: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation This solution requires a person to press a DTMF. However, I'd much rather do it the other way round, but didn't find anything when searching. I heard mentioned 'wait for silence', but I don't see how waiting for silence will give me information on human VS machine detection. Any ideas or solutions out there? Our implementation is done in Javascript, so ideally the solution should be tied in. Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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SIP Invite IP fragmentation issue
by Saurabh Aggarwal 18 Nov '08

18 Nov '08

I am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

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Problems with DTMF and * on inbound leg
by Dennis 18 Nov '08

18 Nov '08

i am using socket outbound and if an inbound call comes in, i answer the call and play a soundfile for the caller. if the caller presses the dtmf key * the playback is stopped and i receive an channel_execute_complete playback for this file. this only happens with the * - the other keys do nothing (as i whish). i there somewhere a default setting for this, which i can turn off? i looked through all conf-files, but could not find enything. for my tests i use the default 1000 user as inbound. in the ivr.conf.xml i found something with menu-top and digits *. i commented this line out, but this didn't change anything. thanks for your help.

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Javascript: stream, speak, stream - cepstral cut offs 2nd stream
by Birgit Arkesteijn 18 Nov '08

18 Nov '08

Hi, In javascript I doing the following: var consumer_name = "Birgit Arkesteijn"; var endpoint_url = 'sofia/gateway/westhawk/0662'; var mSession = new Session("{ignore_early_media=true,originate_timeout=8}" + endpoint_url); var ready = mSession.ready(); // log("merchant answered: " + ready); if (ready == true) { mSession.streamFile("westhawk/lead_waiting.wav", dummy); mSession.speak("cepstral", "Diane", "<break time='2s'/>" + consumer_name); // msleep(500); mSession.streamFile("westhawk/consumer_hungup.wav", dummy); exit(); } function dummy(session, type, data, arg){} No matter what I try, cepstral somehow cuts off the beginning two seconds of the second streamFile(). Adding a msleep(500) in between 'speak' and 'streamFile' makes it only worse. I'm running FreeSWITCH Version 1.0.trunk (597:10325M) Started. and Cepstral_Diane_x86-64-linux_5.1.0/ on Linux Suse 10.0 x86_64 Any ideas? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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WARNING mod_xml_cdr is not a valid FreeSWITCH module dir, skipping it.....
by ram 18 Nov '08

18 Nov '08

Hi iam installing SVN version of Freeswitch when i enable XML_CDR i get the following error WARNING mod_xml_cdr is not a valid FreeSWITCH module dir, skipping it..... what is the problem, where iam missing ? ram

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Gateway registration failed after recent trunk update?
by Johny Kadarisman 17 Nov '08

17 Nov '08

Hi, I experience gateway registration problem after recently update fs to latest trunk. I couldn't connect to broadvoice any more! Even after set the console log to debug, The only clue i see is these message, 2008-11-16 10:42:46 [NOTICE] sofia_reg.c:159 sofia_reg_check_gateway() Registering broadvoice.com 2008-11-16 10:42:46 [WARNING] sofia_reg.c:1088 sofia_reg_handle_sip_r_register() Registration Failed with status 900 I don't see any traffic from fs to broadvoice both with tport_log or ngrep. from comparing the sip profile config files, seems like it should work the way it was. Is there anyhing that I missed? Thanks, Johny K.

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