May 2007 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Javascript
by Jonas Gauffin 12 May '07

12 May '07

Hello 1. How do I configure FS to allow DTMF to abort streamFile? (i.e. The sound files should stop playing if the user knows which key to press in the menu) 2. I'm trying to use TeleTone, I've looked at the answeringmachine example and are trying to generate the same tone. I get the following error: 2007-05-12 09:34:32 [DEBUG] mod_spidermonkey_teletone.c:119 teletone_construct() Raw Codec Activated Generate: (941.00+1477.00) [volume 3761; samples 480(60ms) x 1 channel; wait 0(0ms); decay_factor 0; decay_step 2; wrote 960 bytes] 2007-05-12 09:34:32 [ERR] switch_core_file.c:51 switch_core_file_open() Invalid Format My code: var tts = new TeleTone(session); tts.addTone("d", 350.0, 440.0, 0.0); tts.generate("v=4000;>=0;+=2;#(60,0)"); //Jonas

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Wikipedia Page on FreeSWITCH
by Brian West 11 May '07

11 May '07

http://en.wikipedia.org/wiki/FreeSWITCH I call on the community to go and add Notability to the Entry. Refer to this for guidance: http://en.wikipedia.org/wiki/Wikipedia:Notability Thanks, Brian West FreeSWITCH

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NAT traversal
by Cosimo Alfarano 11 May '07

11 May '07

Hi all, I cannot find such information on wiki/docs. I've a SIP device in my NAT'd LAN connecting to a SIP server outside the LAN. Today the server is an asterisk one, and I put nat=yes to my device's enry in sip.conf I tried to connect to a FS sever in the same scenario, and FS tries to answer to the 192.168 private IP of device. Is there a way to make FS acting in the same nat=<val> way asterisk acts? I cannot put a bridging SIP proxy on my LAN in a easy way, device does not support NAT traversal and IAX. thanks, cosimo. -- Cosimo Alfarano <kalfa(a)bononia.it>

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Channel events
by Jonas Gauffin 10 May '07

10 May '07

Hello I got two questions: a) How do I now if a UA is doing an outbound call, or if it's ringing on the UA? Tapi has two events for that: Offering and Alerting. The first one is when a UA is dialling someone, and the second when someone is ringing on the UA. b) How do I know which one that hangup? (If it was the UA or the calling user). Thanks, Jonas

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extracting sip subject
by Madhuri Patwardhan 08 May '07

08 May '07

Hi, I need to extract the sip subject from the incomng sip call and initialize a variable with this value, which then gets used further. I went through the documentation on Freeswitch wiki and haven't found a way to extract sip subject from the incoming sip call. Is there a way to do this? From the documentation I realized that I could use application set to initialize a variable with this value. Thanks, Madhuri ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

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.NET support
by Jason Steele 08 May '07

08 May '07

Hi all, I want to use Freeswitch for a fairly complex IVR application. I really don't want to do it in javascript and would much prefer to use C#. Can anyone tell me whether Freeswitch currently supports modules written in .NET languages? If so, how do I go about developing one and configuring Freeswitch to use it? If not is work currently in progress to provide this functionality? I see from previous posts that James Martelletti was pretty active in this area, are you still working on this James? I also noticed that mod_mono was recently removed from Freeswitch, but that there is a Freswitch.Net project within the solution. Has mod_mono been replaced by Freeswitch.NET? Many thanks, Jason Steele Jason Steele IT Director Office: +44 (0)1603 34 34 00 Support: +44 (0) 1603 34 00 00 email: jason.steele(a)rcsuk.co.uk <mailto:roy.ferris@rcsuk.co.uk> web site: www.rcsuk.co.uk This message and any attachments are intended for the addressee only and may contain confidential information. If you are not the intended recipient, please forward this message to info(a)rcsuk.co.uk, before destroying completely.

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Jitter problems
by Ivan C Myrvold 08 May '07

08 May '07

I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced. I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec: 1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch->ip24 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24->freeswitch There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2). What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24. Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg Ivan

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Wiki Updates
by Brian West 07 May '07

07 May '07

FreeSWITCHers, Now that I am seeing many of you starting to build and use FreeSWITCH could you take some time to look over the wiki (wiki.freeswitch.org) and see if their is anything you can help clear up documentation on. Thanks, Brian West

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external RTPs/SDP manipulation
by David Villasmil Govea 07 May '07

07 May '07

Hello all, I need a signalling server, what I need it to do is the following, there are two scenarios: The network structure is this: ________ |RTP BOX |____________ |________| __|________ | END POINT | |___________| ________ | | FS |____________| |________| Scenario 1: Generate a call to and end-point specifying dynamically the ip/port for rtps and report back the end-point's rtps's ip/port. Scenario 2: Receive INVITES and report them walong with its rtp's ip/port and answer with a dynamically assigned the ip/port. The rtp's ip/port assigment is done dinamically on another box which soulh communicate with FS via a socket or a script. Does anyone knows wether it is possible at this pont? if it is, how go about this, pointing me in the right direction? I would prefer to do this via scripts (Perl, php) Thanks everyone. David -- DVG -- Imagination is more important than knowledge Albert Einstein

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Re: [Freeswitch-users] Jitter problems
by Anthony Minessale 07 May '07

07 May '07

No what I mean is.... Email me (just me not the whole list) and tell me the login and password to your sip provider account that is not working. Then I can log in and reproduce and discover the problem from my end. Once I figure it out then you can change the pass to something else. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Ivan C Myrvold <ivan(a)myrvold.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, May 6, 2007 9:18:07 AM Subject: Re: [Freeswitch-users] Jitter problems You can call my mobile 91316356. (I am in Norway CET time). Ivan Den 6. mai. 2007 kl. 15:05 skrev Anthony Minessale: and a number i can call to test it _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front

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