April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to read RTP package?
by Frankie Yiu 13 May '11

13 May '11

Hi there, I am new to FreeSwitch, and I have a question to ask. If I want to read the RTP package in C++ or C#, what function should I call to get it? Currently I have can make a call to a phone and play an audio. Thanks in advance. Frankie ____________________________________________________________________________________ Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html

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mod_say in different codecs
by David Notivol 12 May '11

12 May '11

Hi all, I'm trying to setup an IVR server, and I'm using the session:say function from a LUA script. My question is if there's any way of having mod_say playing audios different than audio files; I mean audios encoded in G729, G711ulaw, G711alaw, etc. to avoid having FS making transcoding every time I run a say command. Checking the folders tree for the sounds, I can see the place for the different languages, voices, and wav qualities (8k, 16k...) is clearly specified; but is it a way to place files encoded in different codecs? Thanks in advance. Regards, David Notivol

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mod_crd_sqlite entry limit and sqlite segfaults on triggers
by Neven Boric 10 May '11

10 May '11

Hi, By looking at the code, I believe there is no limit on the number of rows mod_cdr_sqlite will add to the cdr table. This could lead to (eventually) eating all RAM if you DB is in tmpfs or eating all disk space if it is stored on disk. Is this correct or am I missing something? To correct this I tried to add a trigger to the table along the lines of: CREATE TRIGGER IF NOT EXISTS CDR_LIMIT AFTER INSERT ON cdr BEGIN DELETE FROM crd WHERE rowid <= (SELECT MAX(rowid) from cdr) - 100000; END and no matter how I try to add the trigger, I get a segmentation fault when the code calls sqlite3_exec in switch_core_db_exec. Currently I'm blaiming SQLite, as there is a somewhat similar issue involving triggers reported for versions older than 3.5.4 (FS is using 3.3.13) http://sqlite.org/cvstrac/wiki?p=DatabaseCorruption Of course I could just execute the delete statement manually after every insert, but it is uglier (more costly, non atomic). Has anyone any experience using triggers with the sqlite version included in FS? Or has anyone tried other solution to limit the number on entries on the cdr table? Thanks in advance Neven Boric

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Does FS supports speex with ptime=40?
by mayamatakeshi 10 May '11

10 May '11

I am testing making calls with x-lite 4. Although it doesn't send a ptime in its INVITE/SDP, i can see it is sending RTP using ptime=40 as I can count 25 UDP packets per second. When I try to bridge this to PSTN which responds with PCMU, at the PSTN side I hear trembling/metallic/robotic audio. I was suggested to set FS to use ptime=40, so I've tried setting codecs with speex@8000h@40i But it doesn't work. I get: "488 Not Acceptable Here" (Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION) So it seems only ptime=20 is supported as listed here: http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs Could mod_speex be changed to work with ptime=40 too? Or is there anything else that i can try? regards, takeshi

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Call state notifications
by Rogelio Perez 10 May '11

10 May '11

> Event sockets is what you want.... Thanks Ken, it looks like Event Sockets will do the job, but I'm not sure how. I'm testing with netcat listening on port 8084, and then I've managed to connect to it by inserting this line on the dialplan: <action application="socket" data="127.0.0.1:8084 async full"/> Since I dont need to control the call but only receive notification events about the channel state changes I assume I have to send the command "myevents\n\n\" and the pass the variable socket_resume:true back, but this doesnt seem to be working. sendmsg call-command: execute execute-app-name: myevents\n\n\ socket_resume:true This is my first attempt to write an application for FS so I need some guidance. Thanks!

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NEW skypopen installer and easy Skype client download, Skype calls on FreeSWITCH in one minute
by Giovanni Maruzzelli 03 May '11

03 May '11

Dear FreeSWITCHers, after a fair amount of effort, I ended up with a new way to install and use mod_skypopen on Linux. No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA. First, we can use the readily available Skype client for OSS. (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS driver, that's very easy to compile and install, and do not need to mess with the operating system installation.) Second, I wrote an installer that automatically do all the tedious work for you: download and install the skype client, create the config directory for Skype clients, create the config file for mod_skypopen, create the script that launches the Skype clients. I hope those improvements will lower the barriers for Skype calls on FreeSWITCH. Actually is ludicrously simple now, and after you compile FreeSWITCH, mod_skypopen and the skypopen.ko OSS driver it will take like less than one minute to have a complete installation of mod_skypopen ready to make and receive calls. All automatic, no more need to fiddle around with the Skype client download, configurations, authorization, etc. Is all well tested, but maybe there are still some bugs, and maybe the docs are not clear/easy enough. Please have a look at the new and improved wiki page and let me know what do you think about (and maybe test the procedures). http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Inter… You must update to the latest git to have all the goodies. Thank you all for your support, -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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mod_radius_cdr and Acct-Delay-Time
by Camila Troncoso 03 May '11

03 May '11

Hi, I´m working with radius accounting using the mod_radius_cdr module. I´m having trouble with the date format that Radius server receive. An example of what the radius server receive is: “Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"” This date format is very difficult to read and I want to change it to make the accounting easier. I search all around for some param or configuration that allows me to do so, but I only find that this date format is define in mod_radius_cdr.c module. I´m also having problems with the parameter Acct-Delay-Time, it is not increasing when the client resend the radius packet. I read the buildreq.c code but didn’t find the problem. Please some help with this matter. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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Call quality issues
by Eric Beard 02 May '11

02 May '11

Hello, I ran into some major call quality issues this week, and I'm trying to figure out how to troubleshoot things. I've been running FreeSwitch for a few weeks, and suddenly a few days ago my call quality dropped drastically. I had been running more than 100 concurrent calls, with the CPU at less than 20%, but now at 20 concurrent calls, the CPU is still at a little less than 20%, and the call quality is bad - any higher and calls go almost completely silent. There is a direct correlation between the number of simultaneous calls and call quality. I have tested against multiple gateways, same results against each, so it's not an issue with the gateway. I have captured packets on the machine and analyzed them with Wireshark. It seems like the inbound packets are all fine, no jitter or loss. But the packets being sent by FreeSwitch are degraded. One sample call showed: Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%) I'm not sure what that meant, but Wireshark's analysis and playback was very accurate, sounding exactly like what I hear when I test a call. (Freeswitch recordings are by comparison very clean sounding) So what's my next step? Where do I go to find the problem? I know FreeSwitch is capable of handling this traffic. I last updated from git yesterday. I'm running on a dual-quad 64 bit openSuse machine with 8Gb RAM. Thanks! ----------------------- Eric Z. Beard, CTO Loop LLC w (877) 850-2010 x9249 m (727) 776-2768 eric(a)loopfx.com<mailto:eric@loopfx.com>

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Reviewers needed: FreeSWITCH Cookbook
by Michael Collins 02 May '11

02 May '11

Hello, As you know we are working on a FreeSWITCH Cookbook for Packt Publishing. We are in need of 3 or 4 capable reviewers. Please contact me off list if you want to help and meet these qualifications: * Read and write English fluently * Have at least one instance of FreeSWITCH (or a sandbox) where you can test each recipe * Have time to read and try all the recipes in the book * Can offer useful feedback and proofreading * Preferably have a later version of MS Word for editing If you meet the above qualifications and would like to help out then please contact me off list for more information. Thanks, MC

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Trunking between Lync and FreeSWITCH
by David Ponzone 01 May '11

01 May '11

Well, I thought I could start a thread about that. I am trying to accomplish this, and I guess I am not the only one. Lync 2010 seems to have quite some success (compared to OCS), so there are probably business opportunities around this. My first attempt is a half success, as I got calls from Lync to FreeSWITCH working (need some extensive testing, though), and calls from FreeSWITCH to Lync are connecting ok, but they get disconnected after circa 30 seconds. I am trying to pinpoint the reason for this disconnection, but untll now, I had no luck. RTCP requirement was disabled on the Lync side (RTCPActiveCalls), but anyway I have enabled RTCP on the FreeSWITCH, so it should not be that. RTP is flowing normally, it's not a call on hold or muted. Refer was disabled on the Lync side. SessionTimer is enabled on the Lync side (but I dont think I ever saw a REINVITE caused by the session timer). I just suddenly receive a BYE from Lync, after 30-35 seconds. On the Lync side, as you can guess, it's not easy to access any meaningful logs. I think our partner managing the Lync will have to escalate that to the SVP Product Engineering to get the right command to enable the interesting debug mode :) I've been told that some folks at MS were claiming a such trunking was not possible. Well, I would tend to say otherwise, as I am not far to get this working, except of course if they just hardcoded some forbidden Owner Usernames in the SDP, like FreeSWITCH, in order to save this business from the mean OpenSource world and leave it to commercial SBC vendors. I can hardly imagine they would dare to do that in 2011, so the question remains opened: what may be closing that call after 30 seconds ? I would be glad to discuss the subject, here or privately, with anyone who is involved on a such project, or planning to be soon. David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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