- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Epoch current time difference
by Glaucio Vaz de Mello 10 Jan '12

10 Jan '12

Hello, I noticed a difference between FreeSWITCH time and system clock time. Is this a standard behaviour? or something is wrong in my system? More details of the issue: Inside FreeSWITCH I look for current epoch time using this function: time_t now = switch_epoch_time_now(NULL); Then I look for system clock current time using: #include <time.h> time_t now = time(NULL); Between both results I noticed a difference of 9 seconds. This can also be viewed using Unix commands. Unix date command from console: # date +%s 1314741992 Then, I start a call and make a DB query: # sqlite3 /var/freeswitch/db/core.db "SELECT call_created_epoch FROM calls;" 1314742000 Then I repeat unix date command: # date +%s 1314741994 Any comments are welcome. Regards, Glaucio

10 22

voicemail - Message is less than minimum record length
by Oleg Khovayko 10 Jan '12

10 Jan '12

Hi, I have a problem, maybe someone know, how to fix it. OS: FreeBSD v 7.2 Freeswitch: GIT master/head, 2012-01-05 I receive call from Dingaling (google voice), answer. Make bridge, and go to voicemail, if bridge was unsuccessful. Voicemail talking prompt, invite to recording... But instantly says "Message is less than minimum record length, try again". After 2-4 fails/iterations, I able to write voicemail. 2012-01-09 20:24:41.050331 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2012-01-09 20:24:41.154669 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels 20ms 2012-01-09 20:24:45.843641 [DEBUG] switch_ivr_play_say.c:1678 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2012-01-09 20:24:46.956400 [DEBUG] switch_ivr_play_say.c:585 Raw Codec Activated 2012-01-09 20:24:46.956400 [DEBUG] switch_core_codec.c:116 DingaLing/new Push codec L16:70 2012-01-09 20:24:48.963089 [DEBUG] switch_core_codec.c:141 DingaLing/new Restore previous codec PCMU:0. 2012-01-09 20:24:48.963089 [DEBUG] mod_voicemail.c:1198 Message is less than minimum record length: 3, discarding it. 2012-01-09 20:24:48.963089 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] . . . .(thereafter ~ 20 lines like following) . . . 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it 2012-01-09 20:24:48.963089 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it

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FreeSWITCH Conference Call This Week: Sangoma
by Michael Collins 09 Jan '12

09 Jan '12

Greetings all! This week's conference call will feature Moises Silva from Sangoma. Moises (IRC: moy) has lots of experience with TDM communications and is the perfect person to present information on configuring PRI cards and FreeTDM. He also has some interesting information about new GSM stuff from Sangoma. We look forward to his presentation this coming Wednesday, Jan 11. The agenda page is in the usual place<http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_11>on the wiki. Talk to you all on Wednesday! -Michael

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[Minor fix] switch_core_session.c
by Visciano Giovanni 09 Jan '12

09 Jan '12

Minor stuff, missing comma after "T38_DESCRIPTION". switch_core_session.c, line 633 Regards, Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

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bridging calls using mod_managed
by Michael Lutz 09 Jan '12

09 Jan '12

Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike.

3 6

Answered channel
by Juan Antonio Ibañez Santorum 08 Jan '12

08 Jan '12

Hello, Is there any way to know if a channel has been answered having a reference to that channel? Channel profile answered time seems to be setted up as soon early media is received. Regards

1 1

FreeBSD build problem: no tz_t
by Oleg Khovayko 07 Jan '12

07 Jan '12

Hi, I tried pull recent GIT copy, and build by bootstrap.sh + make all, on the FreeBSD v 7.2. *See error:* In file included from /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/home/olegh/freeswitch/GIT/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' gmake[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 gmake: *** [all] Error 2 *I assume, tz_t is not defined:* /*! Set the time zone for the time stamp in page header lines. If this function is not used the current time zone of the program's environment is used. \brief Set the header timezone. \param s The T.4 context. \param tz A time zone descriptor. */ SPAN_DECLARE(void) t4_tx_set_header_tz(t4_state_t *s, tz_t *tz); # man tz_t No manual entry for tz_t *How to define it correctly?* Thanks, Oleg

2 1

Bridge calls in mod_managed
by Michael Lutz 04 Jan '12

04 Jan '12

Hi, I've been working with FS for some time now, and are progressing. I am working on a platform (in/ivr) which is handled by .net managed code completely. I am using just 1 application which is always launched on incomming calls. This application immedeately connects with my MSSQL server and checks validity of gateways, inbound number etc. Also a cdr is written and updated periodically while the call is active. I have als set the HandleHangUp handler to finalize the cdr and call when the calller disconnects. This whole mechanism works as a charm. This application then plays voicefiles, getdigits, and everything as I would expect. Hoewever at some point I want to dialout to an external number and connect these two calls together. I do not want to leave the originating .net app as long as my incomming caller is still online (to update the cdr and finalize the call). While these two are connected I need to update my database periodicaly (keepalive). I have been looking the internet to find what I need but i can't get it to work 100%. In fact everything works, but when I bridge the two calls together it fails. I use the following and this whole peace of code actually works, it dials out, aborts when someone hangsup and when the b_leg picks up they both get audiofile turn by turn.... It even checks outbuond status and maps this to the original call. (so really acts like a switch). Debug("Got a route=>"+route+", Let the network play the ringtone..."); leg_a.Execute("ring_ready", ""); Debug("Creating the new session..."); ManagedSession leg_b = new ManagedSession("sofia/external/"+route); // route holds {destinationNumber}@{myExternalGateway} if (leg_b.Ready()) { Debug("Session_B ready!"); string fsuuid_b = leg_b.GetVariable("uuid"); Debug("Waiting for B-Leg to answer and A-Leg not to hangup..."); while (!leg_b.answered() && leg_a.Ready() ) { /* Ringing... */ } if (leg_b.answered()) { Debug("...B-Leg is answered"); AnswerCall(leg_a); while (!leg_a.answered()) { } Debug("...A-Leg is answered"); // ****** BRIDGE SHOULD BE HERE ****** Debug("As long as both legs are Ready(), we wait......"); while (leg_b.Ready() && leg_a.Ready()) { Play(leg_b, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED Play(leg_a, "Test\\dir-no_matching_results.wav"); // REMOVE WHEN BRIDGED } Debug("There is one leg no longer ready..."); if (leg_a.Ready()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready...");} if (leg_b.Ready()) { Debug("Session B still ready..."); } else { Debug("Session B NO LONGER ready...");} } else { Debug("A-Leg aborted while dialing..."); } if (leg_b.Ready()) { Debug("B-Leg still here, so hanging up B-Leg..."); leg_b.Hangup("NORMAL_CALL_CLEARING"); leg_b.destroy(); } } else { string obCause = leg_b.hangupCause(); leg_a.HangUp(obCause); Debug("Route failed with: "+obCause); } return true; Of course I need to bridge the two calls together so they can speak to each other.. I am trying to bridge the call at "// ****** BRIDGE SHOULD BE HERE ****** " like this: leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); Debug("apiResult="+apiResult); The console log shows "+OK {guid of leg_a} but at the very same moment the incomming calls is beging hung up, (I can see the Debug message of my HandleHangUp() at exactly the as me time as the apiResult of the bridge. I have checked without bridge to check if codecs are the the same and they both are PCMU 8000. Please any help is appreciated! I have been trying to make this work for over a week now, but it won''t work. note. I have also tried originate with the api, but gives me the same problem when I try to bridge these two legs together... Best Regards, Mike.

1 0

Working on building cross platform packages
by Rob Hutton 02 Jan '12

02 Jan '12

I am working on an RPM that will build the various pieces (modules, languages, etc.) each as a seperate packags and compile cross platform, right now on OpenSuse Build service. I have run into two issues, and before I dive deep into figuring something out, I wanted to ask for opinions/information: 1) Suse and Fedora won't allow Lame in "official" packages. How hard would it be to add a configure option or a build variable or something that would exclude the mp3 code or whatever depends on Lame? The goal being to make it avaiable in the base distribution, then have a non-open license version available on one of the "community" repositories that are not as strict. Lame is the only package I can find at this point that is specifically excluded. Others, like opal, will use FFMPEG, etc. if available, but will compile without them, also... 2) Compile checking has become more stringent in recent distributions, and I am getting overflow/underflow, buffer, etc. warning, mainly from the add on libraries (codec2, openldap2) that have been fixed since the versions that are "included". Is there a method in place to easily use system packages for those things? Thanks, Rob

1 0

FreeSWITCH Community Conference Call Schedule
by Michael Collins 02 Jan '12

02 Jan '12

Hello all! I wanted to send out a heads up with the tentative schedule for the upcoming FreeSWITCH community conference calls. We have some good topics scheduled for the next several weeks and I wanted to make sure everyone has ample opportunity to check their calendars and make sure they are able to join us on the days when their favorite topics are being discussed. Here are the upcoming presentations: Jan 4 - Dave Kompel et. al., discussing CDR parsing and rating Jan 11 - Moises Silva from Sangoma, discussing FreeTDM and how to configure PRI cards; also he will be presenting some new GSM goodies from Sangoma Jan 18 - Muhammed Naseer (IRC: Goni) discusses his FreeSWITCH billing application: vBilling Jan 25 - Marc Olivier Chouinard (IRC: moc) will talk about a new module he is working on: mod_voicemail_ivr Feb 1 - Robert Daniel from George Washington University will share some information on how they are using games to teach FreeSWITCH/telecommunications Feb 8 - OPEN Feb 15 - Darren Schreiber et. al. from the 2600hz project will be sharing with the community some updates on 2600hz, Whistle, and blue.box Feb 22 - OPEN We definitely look forward to all of these great presentations over the next few months. If you have a suggestion for a conference call presentation then please contact me at msc(a)freeswitch.org so we can make arrangements. Thanks to everyone who makes FreeSWITCH such a great community! Michael S Collins

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