April 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How does mod_lcr handle invalid ANIs?
by Max Clark 19 Apr '11

19 Apr '11

Hello, I am calling the lcr module from the dialplan like this: <action application="lcr" data="$2 default ${caller_id_number} intrastate"/> So I can get interstate vs. intrastate routing (I dropped the intralata column from the database). How does mod_lcr handle an invalid ANI? Which rate is used to route for this condition? Thanks, Max

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nibblebill cdr
by budi wibowo 19 Apr '11

19 Apr '11

hi is there any possibility that mod_nibblebill create cdr with call pricing? right now i just see the balance deducted, in fs_cli console i can see the call info regards budi

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BUG FIX: "Buffer size sanity check failed!" drops FAX receiving unneeded
by Markus Mueller 19 Apr '11

19 Apr '11

Hello FreeSwitch users and programmers, I found a problem on receiving faxes and want to share a working patch for this. The problem is that on receiving a fax, it is unneeded aborted by a sanity check. Sanity checks are fine, but a unneeded abort instead of a warning is in productive versions not the best solution. The message apearing is: 2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size sanity check failed! which is normaly aborting in receiving the fax. Simply decreasing this fault to a warning let the server receive the fax without any problems. After the patch the message apears up to five times per fax before the fax is beeing accepted. I am using libpri with the three HFC ISDN Cards and the DAHDI from Debian Squeeze 6.0. For more informations about my hardware just write me an email. Regards, Markus Mueller http://projekte.priv.de/ root@sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c* --- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100 +++ switch_core_codec.c.org 2011-03-14 10:47:02.000000000 +0100 @@ -657,9 +657,9 @@ uint32_t frames = encoded_data_len / codec->implementation->encoded_bytes_per_packet; if (frames && codec->implementation->decoded_bytes_per_packet * frames > *decoded_data_len) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n"); - // return SWITCH_STATUS_GENERR; + return SWITCH_STATUS_GENERR; } } if (codec->mutex) switch_mutex_lock(codec->mutex); root@sip:/usr/local/src/freeswitch/src#

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Voicemail params
by Joseph L. Casale 19 Apr '11

19 Apr '11

I am trying to change the domain of the vm email message. I added <param name="vm-mailfrom" value="1000(a)foo.com"/> the user but it is not modifying it, anyone know what else I am missing? It remains as the ip of the internal sofia domain. Thanks, jlc

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No Audio on Gateway Incoming
by William Alianto 19 Apr '11

19 Apr '11

I have recently setup a freeswitch server and configure with gateway for outbound/inbound call. I have succesfully done outbound call without any problem, but whenever I do inbound call, the phone ring but no audio at all. Has anybody encounter such problem before?

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Originated session callback.
by Mohammed Habib 19 Apr '11

19 Apr '11

I need help getting events from originated session. This is my lua script: function onInput_MainSession(s, type, obj) -- This one is working fine. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end function onInput_NewSession(s, type, obj) -- This one is never called. freeswitch.consoleLog("info", "Callback with type " .. type .. "\n"); end session:answer(); session:setInputCallback("onInput_MainSession"); session:sleep(200); session:execute("detect_speech", "unimrcp testgrammer trestgrammer"); newsession = freeswitch.Session("user/1002"); newsession:setInputCallback("onInput_NewSession"); newsession:sleep(200); newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer"); while ((session:ready() == true) ) and (newsession:ready() == true) do -- Loop sleep(200); end I am unable to capture any of the new session events or dtmf. Please help. Thank you, Mohammed Habib

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Playing tones
by Clive Lansink 19 Apr '11

19 Apr '11

Hello I am relatively new to Freeswitch; I've played with it for about two weeks. It is now fully installed as our home PABX system. I've also done some elementary IVR stuff using an external script communicating to event_socket. It can monitor Freeswitch events, answer calls, play and record files and get digits so I suppose it does all the basics. But I'm having difficulty playing tones. A simple tone is no problem like the UK busy tone, %(400,200,400,450);%(400,2000,400,450), but things go wrong when I try loops. In my external script when I want to play a tone, I sende a sendmsg request with the appropriate UUID. In the headers, I have: call-command: execute execute-app-name: gentones execute-app-arg: %(400,200,400,450);%(400,2000,400,450) ] That works, but if I want the ring tone to repeat for an indefinite period, the header then becomes: execute-app-arg: %(400,200,400,450);%(400,2000,400,450);loops=-1 When I do this, I hear one ring tone, then a single DTMF digit is played, presumably the 1 of loops=-1. I've also tried like loops=5 and got the same result. After seeing some examples, I then tried the following, again using the sendmsg command: call-command: execute execute-app-name: playback execute-app-arg: tone_stream://%(400,200,400,450);%(400,2000,400,450);loops=-1 ] This time it does at least seem to loop indefinitely, but after the first two or so loops, the speed increases (not the pitch) so that the tone repeats much more quickly than it should. Any help would be much appreciated. It seems to me that using playback in this way must be accessing the same raw functionality as gentones, so I don't understand why the result would be different. But in the end I guess I don't care which method I use but it would be good to understand why both methods currently don't work for me. By the way, I am using a recent version of Freeswitch, a pre-compiled version running on Windows XP. Cheers. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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Loading xml.so
by Roger Castaldo 19 Apr '11

19 Apr '11

Hi I have a freshly compiled install of freeswitch that appears to be working, but the cpu is sitting at 100% with it doing nothing, no logging messages, nothing. I ran some tests and tried to reload xml to see if there was a configuration error and it throws a critical message saying "/freeswitch/mod/xml.so: cannot open shared object file: No such file or directory". I have double checked my configurations and there should not be any loading of this that I can see, but the file is missing in the mod directory and I cannot seem to find a reference of it in the source. I am not sure if this is where my issue is with the application but I think this is a good start, to find out why it was not compiled in and what is using the library.

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Async packetization?
by "Корнев Е.С." 19 Apr '11

19 Apr '11

Hello Recently I had issue with IVR file - it sounded faster than original sound in audacity. After changing packetization on Cisco gateway from 80 to standard 160 isuue disappeared. Is that was a bug or misconfiguring situation? WBR Kornev E.S. Email secured by Check Point

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mod_h323
by Vedran Zeljeznak 19 Apr '11

19 Apr '11

hello, has anybody compiled and ran mod_h323 with success lately? I'm having trouble bridging two H.323 calls via mod_h323. First two calls bridge relatively successfully, eg. some problems occur when called side hangs up the call first, the calling side of the call doesn't hangup automatically. The third call however doesn't create any debuging messages in FS console. Looks like something remained locked in the first call. My FS version is: "FreeSWITCH Version 1.0.head (git-413d590 2011-04-14 19-56-45 -0500)" OpenH.323 version: h323plus-20100525 PTLib version: 2.8.2 - http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/ i've put freeswitch.log file to pastebin ( http://pastebin.freeswitch.org/16108) Do you have any suggestions on where did i go wrong or where to start looking for a bug? --- Vedran Zeljeznak

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