From moises.silva at gmail.com Sun Jan 1 02:24:15 2012 From: moises.silva at gmail.com (Moises Silva) Date: Sat, 31 Dec 2011 17:24:15 -0600 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> References: <4EF942F0.4010404@googlemail.com> <1325000092695-7130382.post@n2.nabble.com> <4EFA1527.4030309@googlemail.com> <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> Message-ID: On Wed, Dec 28, 2011 at 1:40 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > The procedure to build FS on Windows is usually quite simple. Follow the > instructions on the wiki, especially how to configure git before checking > out the code. Then you just need to open the solution file and build. > > modules.conf doesn't exist, it builds most modules by default. If you > build freetdm stuff uoy will need to install the Sangome provided wanpipe > and sng_isdn packages, and also make sure that the include/library-patchs > will find the files provided by these packages. Also, freetdm is a separate > package, the solution file can be found under libs\freetdm (it not included > in FS mail build solution, since it's a standalone project). > > Some helpful instructions have been posted by Sangoma tech support: http://wiki.sangoma.com/fs-windows-freeswitch-compile-isdn *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111231/76623127/attachment.html From th982a at googlemail.com Sun Jan 1 04:01:27 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 01 Jan 2012 02:01:27 +0100 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: References: <4EF942F0.4010404@googlemail.com> <1325000092695-7130382.post@n2.nabble.com> <4EFA1527.4030309@googlemail.com> <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> Message-ID: <4EFFB067.7020103@googlemail.com> I allredy posted sangoma an answer. if you make a new account on the wiki site, and you want to access the data, you get displayed on the screen: "You are logged in but do not have access to this content. Login as a different user or go back." do a test account yourself, to get this message displayed. Tamer Am 01.01.2012 00:24, schrieb Moises Silva: > On Wed, Dec 28, 2011 at 1:40 AM, Peter Olsson > > wrote: > > The procedure to build FS on Windows is usually quite simple. Follow > the instructions on the wiki, especially how to configure git before > checking out the code. Then you just need to open the solution file > and build. > > modules.conf doesn't exist, it builds most modules by default. If > you build freetdm stuff uoy will need to install the Sangome > provided wanpipe and sng_isdn packages, and also make sure that the > include/library-patchs will find the files provided by these > packages. Also, freetdm is a separate package, the solution file can > be found under libs\freetdm (it not included in FS mail build > solution, since it's a standalone project). > > > Some helpful instructions have been posted by Sangoma tech support: > > http://wiki.sangoma.com/fs-windows-freeswitch-compile-isdn > > *Moises Silva > **/Software Engineer, Development Manager/*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > > > ** > > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter > `| > | YouTube > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sherifomran2000 at yahoo.com Sun Jan 1 04:26:11 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 31 Dec 2011 17:26:11 -0800 (PST) Subject: [Freeswitch-users] freeswitch b-leg data logging Message-ID: <1325381171.46859.YahooMailClassic@web110803.mail.gq1.yahoo.com> Hi I have a? problem with freeswitch and don't get the b-leg information although it is enabled in autoload_configs/xml_cdr.conf.xml file and would appreciate if you could help. waiting your reply kind regards, Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111231/b3c999c2/attachment.html From bblister at gmail.com Sun Jan 1 14:01:20 2012 From: bblister at gmail.com (BBLister) Date: Sun, 1 Jan 2012 13:01:20 +0200 Subject: [Freeswitch-users] Help with Javascript "minuteminder.js" Message-ID: <20120101110120.GA71465@bigb5.homeftp.net> Hi and happy new year, I am trying to implement a simple minute minder javascript, that will play a beep every minute in telephone call. I am using execute_on_answer and I have specified before the bridge the following: In the file minuteminder.js I have this code: --------------------------------------------------------------------------------------- function beep(nr_or_tries,sv_uuid) { console_log("Action: Broadcasting beep on " + sv_uuid + "\n" ); if (session.ready()) apiExecute("uuid_broadast" , sv_uuid + " b5/beep.wav"); } var nr_or_tries = 10; curDate = Date(); console_log("Action: Entering loop at "+curDate); var sv_uuid = session.uuid; console_log("Action: Activating minute minder on session "+sv_uuid+"\n"); session.answer(); while (session.ready()) { console_log("Action: Entering while loop\n"); while (nr_or_tries > 0 ) { console_log("Action: Loop id" + nr_or_tries + "\n"); beep(nr_or_tries--,sv_uuid); session.sleep(60000); } } console_log("Action: Finished\n"); exit(); --------------------------------------------------------------------------------------- I have put the console_log for debuging. When I make a call I see in my logfile the message from console_log ("Action: Broadcasting beep....") and the session number and after this the error that Session is Not active: 2011-12-31 23:54:26.616866 [DEBUG] minuteminder.js:1 Action: Broadcasting beep on d78cf701-fa33-e111-ab5a-00241d71e483 2011-12-31 23:54:26.616866 [ERR] inline:1 Session is not active! The Beep is not heard. I have tried many combinations to fix this but I always get "Session is not active". What can I do in order to fix this? Do I need session.answer() or session.read() or do I assume that the session is ready because this script was called on "exeute_on_answer"? Thank you very much in advance. From jpablolorenzetti at gmail.com Sun Jan 1 19:53:46 2012 From: jpablolorenzetti at gmail.com (Juan Pablo L) Date: Sun, 1 Jan 2012 10:53:46 -0600 Subject: [Freeswitch-users] freeswitch as MGW Message-ID: Hello all, i was wondering if its possible the make a configuration in such a way that you have one box, box1, with a freeswitch instance to deal with all the sip related tasks and a next box, box2, that deals with all the audio related tasks being controlled by box1 by means of MGCP, or any other protocol for that purpose, acting as a MGW (that could also be used by other nodes in the network) ? From th982a at googlemail.com Mon Jan 2 01:41:20 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 01 Jan 2012 23:41:20 +0100 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> References: <4EF942F0.4010404@googlemail.com> <1325000092695-7130382.post@n2.nabble.com> <4EFA1527.4030309@googlemail.com> <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> Message-ID: <4F00E110.9000308@googlemail.com> Hi Peter! I have tried it once before, and there were more then 30 errors on the screen. I want to build FS 64Bit, as well freetdm. Therfor I will post you the results, the next time I will give it a try these days. Tamer Am 28.12.2011 08:40, schrieb Peter Olsson: > The procedure to build FS on Windows is usually quite simple. Follow the instructions on the wiki, especially how to configure git before checking out the code. Then you just need to open the solution file and build. > > modules.conf doesn't exist, it builds most modules by default. If you build freetdm stuff uoy will need to install the Sangome provided wanpipe and sng_isdn packages, and also make sure that the include/library-patchs will find the files provided by these packages. Also, freetdm is a separate package, the solution file can be found under libs\freetdm (it not included in FS mail build solution, since it's a standalone project). > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tamer Higazi > Skickat: den 27 december 2011 19:58 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! > > Hi Jeff! > Thank you for your support- > I saw there are batch files for 2008 and 2010 available. The batch file for 2008 processed with errors at the end. As well, I want to compile the freeswitch on 64 bit windows, and not on 32bit. what he did by default. > > I modified the patch file with the /p flag to use the "x64" bit profile. > However, 53 errors were at the end available. > > I will post here what I did wrong. > > Are the procedures of compiling freeswitch on windows the same as on Linux?! I ment, in the modules.conf to comment out the stuff I want to have?! > > Do I have to be careful when compiling freeswitch along mod_freetdm ?! > > > Tamer > > Am 27.12.2011 16:34, schrieb Jeff Lenk: >> Thats a pretty open ended question. I can tell you that it is >> supported and it works well. You will have to build from source and >> you can use the freely available version of Visual Studio 2010 >> Express. The projects and solutions to do this are already built and >> available in Git. If you ask some more specific questions I will try to help. >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-fr >> eetdm-on-windows-tp7129722p7130382.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4efa5a9a32765031432928! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jeff at jefflenk.com Mon Jan 2 07:42:06 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 1 Jan 2012 20:42:06 -0800 (PST) Subject: [Freeswitch-users] problem with date-time In-Reply-To: <811071325234531@web151.yandex.ru> References: <811071325234531@web151.yandex.ru> Message-ID: <1325479326847-7143070.post@n2.nabble.com> check git head for a correction -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/problem-with-date-time-tp7138683p7143070.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Mon Jan 2 10:19:43 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 2 Jan 2012 07:19:43 +0000 Subject: [Freeswitch-users] Help with Javascript "minuteminder.js" In-Reply-To: <20120101110120.GA71465@bigb5.homeftp.net> References: <20120101110120.GA71465@bigb5.homeftp.net> Message-ID: I suggest using sched_broadcast instead. I don't think your javascript will ever work, since the loop will stop the calls from bridging correctly. The only thing that will happen on the b-leg is your loop, instead of sending RTP frames for the call. /Peter ----- Reply message ----- Fr?n: "BBLister" Datum: s?n, jan 1, 2012 23:49 Rubrik: [Freeswitch-users] Help with Javascript "minuteminder.js" Till: "freeswitch-users at lists.freeswitch.org" Hi and happy new year, I am trying to implement a simple minute minder javascript, that will play a beep every minute in telephone call. I am using execute_on_answer and I have specified before the bridge the following: In the file minuteminder.js I have this code: --------------------------------------------------------------------------------------- function beep(nr_or_tries,sv_uuid) { console_log("Action: Broadcasting beep on " + sv_uuid + "\n" ); if (session.ready()) apiExecute("uuid_broadast" , sv_uuid + " b5/beep.wav"); } var nr_or_tries = 10; curDate = Date(); console_log("Action: Entering loop at "+curDate); var sv_uuid = session.uuid; console_log("Action: Activating minute minder on session "+sv_uuid+"\n"); session.answer(); while (session.ready()) { console_log("Action: Entering while loop\n"); while (nr_or_tries > 0 ) { console_log("Action: Loop id" + nr_or_tries + "\n"); beep(nr_or_tries--,sv_uuid); session.sleep(60000); } } console_log("Action: Finished\n"); exit(); --------------------------------------------------------------------------------------- I have put the console_log for debuging. When I make a call I see in my logfile the message from console_log ("Action: Broadcasting beep....") and the session number and after this the error that Session is Not active: 2011-12-31 23:54:26.616866 [DEBUG] minuteminder.js:1 Action: Broadcasting beep on d78cf701-fa33-e111-ab5a-00241d71e483 2011-12-31 23:54:26.616866 [ERR] inline:1 Session is not active! The Beep is not heard. I have tried many combinations to fix this but I always get "Session is not active". What can I do in order to fix this? Do I need session.answer() or session.read() or do I assume that the session is ready because this script was called on "exeute_on_answer"? Thank you very much in advance. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f00e0cf32761527225430! From yehavi.bourvine at gmail.com Mon Jan 2 11:13:59 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 2 Jan 2012 10:13:59 +0200 Subject: [Freeswitch-users] Debugging intermitent BLF problems Message-ID: Hello, I am "stuck" with FreeSWITCH version from 15-Oct since all newer vesions that I tried have intermitent BLF problems. The BLF gets stuck once in a while (stuck in Ringing, busy or free) until the next SUBSCRIBE refresh (once every hour). It is not determenistic, and it happens on a profile on which I have over 180 extensions (it does not happen on a test profile I built and put one phone on it). Before I open a JIRA: 1. Has anyone noticed this? 2. What debugs shall I enable so there will be a reasonable amount of data? BTW, most phones are Polycoms (and one SNOM...). Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/32bf97f2/attachment.html From benkokakao at gmail.com Mon Jan 2 16:43:07 2012 From: benkokakao at gmail.com (Christian Benke) Date: Mon, 2 Jan 2012 14:43:07 +0100 Subject: [Freeswitch-users] Debugging intermitent BLF problems In-Reply-To: References: Message-ID: > 1. Has anyone noticed this? Yes, i noticed occassional issues - however i was not aware that this didn't happen in earlier version. I also don't know how to reproduce the issues and did not investigate on the problem so far. m2c Christian From potxoka at gmail.com Mon Jan 2 16:46:04 2012 From: potxoka at gmail.com (Anto) Date: Mon, 2 Jan 2012 14:46:04 +0100 Subject: [Freeswitch-users] Mod_SPANDSP Message-ID: hello I'm adapting the files are in http://wiki.freeswitch.org/wiki/Mod_spandsp to make faxing my business. I wanted to know whether to send the fax, an error occurs (destination busy, etc.), this is still retrying successive times or is there a way to handle this. I've been looking fax_result_code and fax_result_text, to make a correct treatment but have not found information on them. Does anyone have any examples or can guide me? Thank you very much Best Regards Anto From yehavi.bourvine at gmail.com Mon Jan 2 17:43:52 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 2 Jan 2012 16:43:52 +0200 Subject: [Freeswitch-users] Debugging intermitent BLF problems In-Reply-To: References: Message-ID: It was rock-solid until mid-October (and that's why I have to stay at that version in the meantime). I am willing to help debugging this, but need to know which debugs to add which will be usefull... Thanks, __Yehavi: 2012/1/2 Christian Benke > > 1. Has anyone noticed this? > > Yes, i noticed occassional issues - however i was not aware that this > didn't happen in earlier version. I also don't know how to reproduce > the issues and did not investigate on the problem so far. > > m2c > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/24123f53/attachment.html From frank at impactfax.com Mon Jan 2 18:01:12 2012 From: frank at impactfax.com (Frank @ Impact) Date: Mon, 2 Jan 2012 10:01:12 -0500 Subject: [Freeswitch-users] audo sync issues with record_session to mp3 In-Reply-To: Message-ID: <6AEC73649FA6431CA9EF84A54F378594@ws4> We have the same problem. We are running git from 12/30/11. our aleg is a sip channel coming to FS and the bleg is a sip channels leaving FS. I noticed this problem really when we started using mp3 instead of wav. With wav, it really was not noticeable for us in a 10-15minute call. But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so far out of sync, it sounds like 2 different calls. The relevant dialplan is
-----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel Gunderson Sent: Thursday, October 20, 2011 3:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3 On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott wrote: > Longer calls, after about 10 minutes start to introduce sync issues > between the A-leg and the B-leg. > > We are running record_session on the A-leg, and it seems to get ahead of > the B-leg. > > For example the caller on the A-leg will be heard to answer a question > whilst the person on the B-leg is asking it. What's on the other end of each leg? That might help us figure this out. Gabe FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From juanito1982 at gmail.com Mon Jan 2 18:42:29 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Mon, 2 Jan 2012 16:42:29 +0100 Subject: [Freeswitch-users] Finish LUA script Message-ID: Hello! Is there any way to finish a LUA script before reaching its end line? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/0ce73529/attachment.html From brian.wiese.freeswitch at gmail.com Mon Jan 2 19:56:42 2012 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Mon, 2 Jan 2012 10:56:42 -0600 Subject: [Freeswitch-users] Global Variable Substitution Message-ID: Hi Everyone. I thought I read somewhere that this was possible, but I can't find it now... I need a way to find-and-replace within a variable. So, for example, I want to take variables that have values like this: 123abc123abc abc123abc123abc ...and do a find/replace of the "abc" with "xyz" so the variables would now return: 123xyz123xyz xyz123xyz123xyz The use case I've run into is that I need to add leg variables to group_call. In my case, group_call can return any number of members, so I figured I would just replace the first "[" with "[variable-I-want-to-set=...". Thanks for the help! ~Brian From brandon.mcginty at gmail.com Mon Jan 2 20:03:28 2012 From: brandon.mcginty at gmail.com (Brandon McGinty) Date: Mon, 02 Jan 2012 12:03:28 -0500 Subject: [Freeswitch-users] mod_sofia not binding to port 5060 Message-ID: <4F01E360.9010007@gmail.com> Good afternoon. We've got a freeswitch machine, no external SIP gateways for outside calls, eight extensions, two conferences, and a completely public IP for the machine in question (no NAT weirdness), running on Debian 6 (weezy/sid). I woke up this morning, to find that sip_external_port 5060 was not listening, though it was enabled in config/vars.xml. The internal port, 5080, is listening, and I have folks confirming that they can speak using that port. I've set netcat to listen via 5080, so I know it isn't firewalled, or not at least ina conventional manner. Any help you all can provide, things I can check, would be greatly appreciated. Sincerely, Brandon McGinty-Carroll From notlikeme75 at yahoo.com Mon Jan 2 20:16:41 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Mon, 2 Jan 2012 09:16:41 -0800 (PST) Subject: [Freeswitch-users] mod shout win32 In-Reply-To: References: Message-ID: <1325524601.99361.YahooMailNeo@web65302.mail.ac2.yahoo.com> I have build mod_shout on win32 and placed the following files in the c:\program files\freeswitch\mod\ folder mod_shout.dll mod_shout.exp mod_shout.lib mod_shout.pdb and I have put the mpg123 folder at \\Winserver2008\c\Program Files\FreeSWITCH\mpg123-1.13.4-static-x86 but when I go to my extension i get the following errors in console: err mod_shout.c:859 error: mpg123 error at mod_shout.c:659 err mod_shout.c:862 error from mpg123: Invalid mpg123 handle. (code 10) did i not put the mod_shout files in correct folders or is it not knowing where to look for mpg123?? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, December 27, 2011 6:56 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 176 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: sangoma A200 with mod_freetdm on windows?! (Jeff Lenk) ? 2. Re: mod shout win32 (Jeff Lenk) ? 3. prevention of duplicate calls. (Rodney) ? 4. Re: sangoma A200 with mod_freetdm on windows?! (Moises Silva) ? 5. micrsoft voip plug in for fax (Darcy) ? 6. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) ? 7. Re: Building Freeswitch on OSX Lion - ./configure openssl ? ? ? error (James) Thats a pretty open ended question. I can tell you that it is supported and it works well. You will have to build from source and you can use the freely available version of Visual Studio 2010 Express. The projects and solutions to do this are already built and available in Git. If you ask some more specific questions I will try to help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-freetdm-on-windows-tp7129722p7130382.html Sent from the freeswitch-users mailing list archive at Nabble.com. Have you seen http://wiki.freeswitch.org/wiki/Installation_for_Windows otherwise post some more specific questions on what you are having problems with. The Git supplied VS projects will build mod_shout but are building some of the dependent libraries with an older version than on linux and this may cause you to see differences in behavior or other. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-win32-tp7130183p7130388.html Sent from the freeswitch-users mailing list archive at Nabble.com. is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up? and continuing the first call. On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi wrote: Hi people! >I got my A200 board running with 1 FXS module on Linux along with >mod_freetdm, but I am facing problems getting it to run on Windows. From >Sangoma I followed the instructions to set up the board on Windows7 >winpipe module, which works so far. > >How do I get freeswitch with mod_freetdm to run on Windows that I can >make use of the board (pbx) on a win machine? > > You may want to send an email to Sangoma support. They are working already in a wiki page for Windows setup, in the meantime they can help you with instructions via email. Moises Silva Software Engineer, Development Manager msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t.?? +1 800 388 2475 (N. America) t.?? +1 905 474 1990 x128 f.?? +1 905 474 9223 ? Products?|?Solutions?|?Events?|?Contact?|?Wiki?|?Facebook?|?Twitter`| |?YouTube ????????????????VegaStream is now part of Sangoma! ????????????????Ask us about both?Gateway Appliances?and?Internal Gateways Has anyone any experience using microsoft?s voip plug in for fax with the freeswitch, I only get about 1 in 10 completion, inbound works 100%, outbound will only work if the server is in the same building. ? DarcyHi Jeff! Thank you for your support- I saw there are batch files for 2008 and 2010 available. The batch file for 2008 processed with errors at the end. As well, I want to compile the freeswitch on 64 bit windows, and not on 32bit. what he did by default. I modified the patch file with the /p flag to use the "x64" bit profile. However, 53 errors were at the end available. I will post here what I did wrong. Are the procedures of compiling freeswitch on windows the same as on Linux?! I ment, in the modules.conf to comment out the stuff I want to have?! Do I have to be careful when compiling freeswitch along mod_freetdm ?! Tamer Am 27.12.2011 16:34, schrieb Jeff Lenk: > Thats a pretty open ended question. I can tell you that it is supported and > it works well. You will have to build from source and you can use the freely > available version of Visual Studio 2010 Express. The projects and solutions > to do this are already built and available in Git. If you ask some more > specific questions I will try to help. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-freetdm-on-windows-tp7129722p7130382.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks for the response, Seven - the openssl issue has been reported in JIRA already a couple of times. ?The workarounds there don't seem to work. ?See:?http://jira.freeswitch.org/browse/FS-3642 Your workaround that you mentioned did not appear to work for me either - I actually ended up getting the error that you originally reported in this other JIRA issue:?http://jira.freeswitch.org/browse/FS-3243 What do I need to do to fix that error? ?And by the way, I would say that last JIRA issue should be reopened, as it's happening on my Macbook Air (OSX Lion) too. ? On Tue, Dec 27, 2011 at 5:08 AM, Seven Du wrote: 1. report on jira.freeswitch.og > > >2. for a workaround, try ?mv libs/iksemel to /tmp before configure and move back before make if you don't need dingaling support. > > >--? >About: http://about.me/dujinfang >Blog: http://www.dujinfang.com >Proj: ?http://www.freeswitch.org.cn > >Sent with Sparrow > > > >On Tuesday, December 27, 2011 at 1:37 PM, James wrote: >I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: >>? >> >>./configure: line 12283: syntax error near unexpected token `openssl,' >>./configure: line 12283: ` ? ?PKG_CHECK_MODULES(openssl, openssl,' >>configure: error: ./configure.gnu failed for libs/iksemel >> >> >>./bootstrap.sh runs fine; ./configure eventually spits out the error above. ?Any tips for me? ?I am mainly trying to set it up for local development purposes, not production. ? >> >> >>Thanks. >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/a90e7925/attachment-0001.html From chad at apartmentlines.com Mon Jan 2 20:37:24 2012 From: chad at apartmentlines.com (Chad Phillips) Date: Mon, 2 Jan 2012 09:37:24 -0800 Subject: [Freeswitch-users] Finish LUA script In-Reply-To: References: Message-ID: <8D5A3F612B9242BD9F7C26BF2282A266@gmail.com> see http://www.lua.org/pil/4.4.html especially the very last thing mentioned. chad On Monday, January 2, 2012 at 7:42 AM, Juan Antonio Iba?ez Santorum wrote: > Hello! > > Is there any way to finish a LUA script before reaching its end line? > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/50e09efa/attachment.html From jeff at jefflenk.com Mon Jan 2 23:09:03 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 2 Jan 2012 12:09:03 -0800 (PST) Subject: [Freeswitch-users] mod shout win32 In-Reply-To: <1325524601.99361.YahooMailNeo@web65302.mail.ac2.yahoo.com> References: <1325524601.99361.YahooMailNeo@web65302.mail.ac2.yahoo.com> Message-ID: <1325534943091-7144706.post@n2.nabble.com> The only file that is needed is mod_shout.dll all the others are statically linked. Look for errors with the file or file location. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-win32-tp7144313p7144706.html Sent from the freeswitch-users mailing list archive at Nabble.com. From juanito1982 at gmail.com Mon Jan 2 23:45:28 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Mon, 2 Jan 2012 21:45:28 +0100 Subject: [Freeswitch-users] Finish LUA script In-Reply-To: <8D5A3F612B9242BD9F7C26BF2282A266@gmail.com> References: <8D5A3F612B9242BD9F7C26BF2282A266@gmail.com> Message-ID: Good reference! Regards 2012/1/2 Chad Phillips > see http://www.lua.org/pil/4.4.html especially the very last thing > mentioned. > > chad > > On Monday, January 2, 2012 at 7:42 AM, Juan Antonio Iba?ez Santorum wrote: > > Hello! > > Is there any way to finish a LUA script before reaching its end line? > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/2b882ec6/attachment.html From msc at freeswitch.org Tue Jan 3 00:13:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Jan 2012 13:13:06 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Schedule Message-ID: 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 at freeswitch.org so we can make arrangements. Thanks to everyone who makes FreeSWITCH such a great community! Michael S Collins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/39254bff/attachment.html From darknesslabs at gmail.com Tue Jan 3 00:56:55 2012 From: darknesslabs at gmail.com (Karol) Date: Mon, 2 Jan 2012 16:56:55 -0500 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 Message-ID: Anyone else see this?? http://pastebin.freeswitch.com/18078 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/898b298f/attachment.html From sos at sokhapkin.dyndns.org Tue Jan 3 01:06:13 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 02 Jan 2012 17:06:13 -0500 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: References: Message-ID: <6007889.WdBRkjEMWA@sos> Sure nobody else see this, because everybody have fail2ban running. On Monday 02 January 2012 16:56:55 Karol wrote: > Anyone else see this?? > > http://pastebin.freeswitch.com/18078 From avi at avimarcus.net Tue Jan 3 03:33:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jan 2012 02:33:06 +0200 Subject: [Freeswitch-users] Voicemail - see if user left a message? Message-ID: Can the xml_cdrs tell me if the user actually left a voicemail, rather than merely that VM picked up? I don't see where. I know it's on ESL, and it hits the voicemail email code.. what's the easiest way to integrate it with the CDRs? preferably with the xml_cdrs? Maybe a patch to the xml_cdr mod? Or the easiest is to have mod_voicemail set a channel variable once a voicemail gets marked as saved or the like? Is there currently a solution for this that I missed? Thanks, -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/3348bb94/attachment-0001.html From bdfoster at endigotech.com Tue Jan 3 02:26:27 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 2 Jan 2012 18:26:27 -0500 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: References: Message-ID: This is just part of having a switch open to the world. We get these registration attempts on a daily, almost hourly basis. It is your responsibility to make sure your system is secured. Please look at fail2ban and iptables as ways to protect your system. If you don't, it could cost you big money. On Jan 2, 2012 4:59 PM, "Karol" wrote: > Anyone else see this?? > > http://pastebin.freeswitch.com/18078 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/033b2b5a/attachment.html From philippe at ppmt.org Tue Jan 3 04:22:58 2012 From: philippe at ppmt.org (Philippe Le Toquin) Date: Mon, 02 Jan 2012 20:22:58 -0500 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: References: Message-ID: <4F025872.6060805@ppmt.org> Hello Karol, I had some issue like that as well and they completely disappear after I installed fail2ban and the correct "plugin" for freeswitch as other have mentioned I followed this instruction here. http://wiki.freeswitch.org/wiki/Fail2ban /Philippe On 12-01-02 04:56 PM, Karol wrote: > Anyone else see this?? > > http://pastebin.freeswitch.com/18078 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/d01271f5/attachment.html From th982a at googlemail.com Tue Jan 3 04:26:41 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 03 Jan 2012 02:26:41 +0100 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: References: Message-ID: <4F025951.2040009@googlemail.com> Or better! Make all clients behind static ip addresses available, and limit the registration with those ip-addresses. Tamer Am 03.01.2012 00:26, schrieb Brian Foster: > This is just part of having a switch open to the world. We get these > registration attempts on a daily, almost hourly basis. It is your > responsibility to make sure your system is secured. Please look at > fail2ban and iptables as ways to protect your system. If you don't, it > could cost you big money. > > On Jan 2, 2012 4:59 PM, "Karol" > wrote: > > Anyone else see this?? > > http://pastebin.freeswitch.com/18078 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From th982a at googlemail.com Tue Jan 3 04:34:08 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 03 Jan 2012 02:34:08 +0100 Subject: [Freeswitch-users] mod_sofia not binding to port 5060 In-Reply-To: <4F01E360.9010007@gmail.com> References: <4F01E360.9010007@gmail.com> Message-ID: <4F025B10.4030705@googlemail.com> Look at: conf/autoload_configs/switch.conf.xml read this: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files look at "sip-port" I am tired, I am going to bed. Tamer Am 02.01.2012 18:03, schrieb Brandon McGinty: > Good afternoon. > We've got a freeswitch machine, no external SIP gateways for outside > calls, eight extensions, two conferences, and a completely public IP for > the machine in question (no NAT weirdness), running on Debian 6 (weezy/sid). > I woke up this morning, to find that sip_external_port 5060 was not > listening, though it was enabled in config/vars.xml. > The internal port, 5080, is listening, and I have folks confirming that > they can speak using that port. > I've set netcat to listen via 5080, so I know it isn't firewalled, or > not at least ina conventional manner. > Any help you all can provide, things I can check, would be greatly > appreciated. > > Sincerely, > Brandon McGinty-Carroll > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From olimonkey at gmail.com Tue Jan 3 06:46:33 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Tue, 3 Jan 2012 11:46:33 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR Message-ID: I've been battling while creating an IVR using FreeSWITCH mod_managed and connecting through a CISCO 2811. Most things now work quite well, but I am having a few issues with the way the system answers calls (or doesn't answer calls...). I have FreeSWITCH running as a windows service on Windows Server 2008, which is connected via LAN to a CISCO 2811 with a 4 port FXO card, which is then connected to a POTS phone line. Take the following scenario: 1. Managed .NET application creates a call string and uses ESL to talk to freeswitch and originate a call: string callstring = "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)'"; eslConnection.API("originate", callstring); where 192.168.x.x is the CISCO IP. 2. The CISCO sees that the phone number (1091234567) starts with a 1 so it uses FXO port 1 and strips the 1 and uses the remaining phone number (091234567) to make the call. 3. My phone rings, I pick up and I can hear my IVR playing. These are my current problems: - IVR starts playing before I even pick up the phone. This means that if the system calls a mobile phone and the person doesn't pick up, the IVR will start playing and eventually the mobile phone will divert to voice mail. Obviously I then get a missed call and an sms saying I have a new voice mail, which is annoying. Instead I would like it to KNOW that no one has picked up, but I don't know how to do this. Somehow the CISCO needs to be able to tell FreeSWITCH that the call has not yet been answered. For some reason however as soon as the CISCO starts calling FreeSWITCH thinks the call is already connected. It doesn't know that the CISCO is actually still ringing. Maybe I'm doing originate the wrong way or something ... - The phone only rings for about 10 seconds before hanging up. I've tried "call_timeout", "bridge_answer_timeout". I've also tried setting CISCO "ring number". Nothing works, my phone still only rings for about 10 seconds. I don't know if this is a FreeSWITCH issue or a CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just starts playing even if no one answers the phone. CISCO Config for relevant FXO port: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 supplementary-service h450.12 no supplementary-service sip moved-temporarily no supplementary-service sip refer fax protocol cisco sip registrar server expires max 3600 min 3600 no update-callerid no call service stop voice-port 0/3/2 output attenuation -3 no comfort-noise cptone AU impedance complex1 caller-id enable ! dial-peer voice 100 pots preference 1 destination-pattern 1T port 0/3/2 ! Many Thanks, Oliver From nbhatti at gmail.com Tue Jan 3 10:02:20 2012 From: nbhatti at gmail.com (nbhatti) Date: Mon, 2 Jan 2012 23:02:20 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: <1325549865328-7145235.post@n2.nabble.com> References: <1325549865328-7145235.post@n2.nabble.com> Message-ID: Yes, it will support prepaid calling card and many more features soon. On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] wrote: > It looks great! > > Will vBilling support batch user/prepaid calling card creation? > > ________________________________ > If you reply to this email, your message will be added to the discussion > below: > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html > To unsubscribe from vBilling Beta Program!!, click here. > NAML -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120102/414ee64d/attachment.html From anita.hall at simmortel.com Tue Jan 3 14:53:27 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 3 Jan 2012 17:23:27 +0530 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs Message-ID: Hi! We have been playing with Intel IPP G729 code and its module for FreeSWITCH. As reported here, we have successfully compile the fs-g729 module after install IPP libraries. https://code.google.com/p/fs-g729/issues/detail?id=3#c1 However, to test the codec in a loopback SIP call (FreeSWITCH calling itself over SIP), I need some G729 encoded files (do I really?) FreeSWITCH gives errors with G729 encoded file from here http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-English-g729.tar.gz Dialplan [ERR] switch_core_file.c:122 Invalid file format [g729] for [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! And also gives errors with file encoded using Intel IPP sample G729 encoder. [ERR] switch_core_file.c:122 Invalid file format [g729] for [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]! Please note that the files are actually kept in the directory /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr sound files. # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729: data /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch.g729: RIFF (little-endian) data, WAVE audio, mono 8000 Hz I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 encoder) is not really G729. Any pointers will be most helpful. Thanks! Anita. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/7f5e0cc0/attachment-0001.html From krice at freeswitch.org Tue Jan 3 15:46:33 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Jan 2012 06:46:33 -0600 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: Message-ID: The IPP Codecs are NOT supported by the project. These are not useable in any production environment where the core developers reside due to Patent and other IP Law restrictions. If you really need G729 support you should look at either getting the proper licensing from freeswitch.org or look at other hardware options available. This is to protect the project in general and the livelihoods of the dev team. K On 1/3/12 5:53 AM, "Anita Hall" wrote: > Hi! > > We have been playing with Intel IPP G729 code and its module for FreeSWITCH. > > As reported here, we have successfully compile the fs-g729 module after > install IPP libraries. > https://code.google.com/p/fs-g729/issues/detail?id=3#c1 > > However, to test the codec in a loopback SIP call (FreeSWITCH calling itself > over SIP), I need some G729 encoded files (do I really?) > > FreeSWITCH gives errors with G729 encoded file from here > http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-English-g > 729.tar.gz > Dialplan > > > [ERR] switch_core_file.c:122 Invalid file format [g729] for > [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! > > And also gives errors with file encoded using Intel IPP sample G729 encoder. > > [ERR] switch_core_file.c:122 Invalid file format [g729] for > [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]> ! > > Please note that the files are actually kept in the directory > /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr sound > files. > > # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* > /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729:????????? > ????? data > /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch. > g729: RIFF (little-endian) data, WAVE audio, mono 8000 Hz > > I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 > encoder) is not really G729. > > Any pointers will be most helpful. > > Thanks! > Anita. > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/8e1f51c9/attachment.html From kerem.erciyes at gmail.com Tue Jan 3 15:55:18 2012 From: kerem.erciyes at gmail.com (Kerem Erciyes) Date: Tue, 3 Jan 2012 14:55:18 +0200 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: References: Message-ID: Didn't your mom teach you not to play with IPP Codecs. -- Plus I've never seen and IPP Codec that works without memory problems, FS commercial codec works well however. On Tue, Jan 3, 2012 at 1:53 PM, Anita Hall wrote: > Hi! > > We have been playing with Intel IPP G729 code and its module for > FreeSWITCH. > > As reported here, we have successfully compile the fs-g729 module after > install IPP libraries. > https://code.google.com/p/fs-g729/issues/detail?id=3#c1 > > However, to test the codec in a loopback SIP call (FreeSWITCH calling > itself over SIP), I need some G729 encoded files (do I really?) > > FreeSWITCH gives errors with G729 encoded file from here > http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-English-g729.tar.gz > Dialplan > > > [ERR] switch_core_file.c:122 Invalid file format [g729] for > [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! > > And also gives errors with file encoded using Intel IPP sample G729 > encoder. > > [ERR] switch_core_file.c:122 Invalid file format [g729] for > [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]! > > Please note that the files are actually kept in the directory > /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr > sound files. > > # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* > /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729: > data > /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch.g729: > RIFF (little-endian) data, WAVE audio, mono 8000 Hz > > I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 > encoder) is not really G729. > > Any pointers will be most helpful. > > Thanks! > Anita. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Kerem Erciyes - Sistem Danismani http://keremerciyes.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/bd8bbe1d/attachment.html From engineerzuhairraza at gmail.com Tue Jan 3 16:07:03 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 3 Jan 2012 17:07:03 +0400 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: References: Message-ID: LOL Regards, Zohair Raza On Tue, Jan 3, 2012 at 4:55 PM, Kerem Erciyes wrote: > Didn't your mom teach you not to play with IPP Codecs. > > -- Plus I've never seen and IPP Codec that works without memory problems, > FS commercial codec works well however. > > On Tue, Jan 3, 2012 at 1:53 PM, Anita Hall wrote: > >> Hi! >> >> We have been playing with Intel IPP G729 code and its module for >> FreeSWITCH. >> >> As reported here, we have successfully compile the fs-g729 module after >> install IPP libraries. >> https://code.google.com/p/fs-g729/issues/detail?id=3#c1 >> >> However, to test the codec in a loopback SIP call (FreeSWITCH calling >> itself over SIP), I need some G729 encoded files (do I really?) >> >> FreeSWITCH gives errors with G729 encoded file from here >> http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-English-g729.tar.gz >> Dialplan >> >> >> [ERR] switch_core_file.c:122 Invalid file format [g729] for >> [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! >> >> And also gives errors with file encoded using Intel IPP sample G729 >> encoder. >> >> [ERR] switch_core_file.c:122 Invalid file format [g729] for >> [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]! >> >> Please note that the files are actually kept in the directory >> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr >> sound files. >> >> # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* >> /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729: >> data >> /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch.g729: >> RIFF (little-endian) data, WAVE audio, mono 8000 Hz >> >> I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 >> encoder) is not really G729. >> >> Any pointers will be most helpful. >> >> Thanks! >> Anita. >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Kerem Erciyes - Sistem Danismani > http://keremerciyes.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/6b1e6ac7/attachment-0001.html From kaushalshriyan at gmail.com Tue Jan 3 16:47:24 2012 From: kaushalshriyan at gmail.com (Kaushal Shriyan) Date: Tue, 3 Jan 2012 19:17:24 +0530 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Message-ID: Hi, Please help me understand the following applications and what are its advantages if we compare between each of them. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. Regards Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/db290096/attachment.html From brandon.mcginty at gmail.com Tue Jan 3 16:48:25 2012 From: brandon.mcginty at gmail.com (Brandon McGinty) Date: Tue, 03 Jan 2012 08:48:25 -0500 Subject: [Freeswitch-users] mod_sofia not binding to port 5060 In-Reply-To: <4F025B10.4030705@googlemail.com> References: <4F01E360.9010007@gmail.com> <4F025B10.4030705@googlemail.com> Message-ID: <4F030729.1070208@gmail.com> Thanks for these. However, I checked all three of these resources. For some reason, this issue is only occuring when the init.d script for debian, is used to start freeswitch; from the command line, the issue goes away completely. All that is running is the -nc switch, so I'm not sure what could be causing the issue. Thanks again. Brandon McGinty-Carroll On 1/2/2012 8:34 PM, Tamer Higazi wrote: > Look at: > > conf/autoload_configs/switch.conf.xml > > read this: > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files > > > look at "sip-port" > > > I am tired, I am going to bed. > > > Tamer > > > Am 02.01.2012 18:03, schrieb Brandon McGinty: >> Good afternoon. >> We've got a freeswitch machine, no external SIP gateways for outside >> calls, eight extensions, two conferences, and a completely public IP for >> the machine in question (no NAT weirdness), running on Debian 6 (weezy/sid). >> I woke up this morning, to find that sip_external_port 5060 was not >> listening, though it was enabled in config/vars.xml. >> The internal port, 5080, is listening, and I have folks confirming that >> they can speak using that port. >> I've set netcat to listen via 5080, so I know it isn't firewalled, or >> not at least ina conventional manner. >> Any help you all can provide, things I can check, would be greatly >> appreciated. >> >> Sincerely, >> Brandon McGinty-Carroll >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From engineerzuhairraza at gmail.com Tue Jan 3 16:53:22 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 3 Jan 2012 17:53:22 +0400 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: all of them have a wiki page http://lmgtfy.com/?q=Asterisk http://lmgtfy.com/?q=freeswitch http://lmgtfy.com/?q=openser http://lmgtfy.com/?q=TrixBox Regards, Zohair Raza On Tue, Jan 3, 2012 at 5:47 PM, Kaushal Shriyan wrote: > Hi, > > Please help me understand the following applications and what are its > advantages if we compare between each of them. > > Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. > > Regards > > Kaushal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/f2beb914/attachment.html From gmaruzz at gmail.com Tue Jan 3 16:54:49 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 3 Jan 2012 14:54:49 +0100 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: http://www.freeswitch.org/node/117 On Tue, Jan 3, 2012 at 2:47 PM, Kaushal Shriyan wrote: > Hi, > > Please help me understand the following applications and what are its > advantages if we compare between each of them. > > Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. > > Regards > > Kaushal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From kaushalshriyan at gmail.com Tue Jan 3 16:58:12 2012 From: kaushalshriyan at gmail.com (Kaushal Shriyan) Date: Tue, 3 Jan 2012 19:28:12 +0530 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza wrote: > all of them have a wiki page > > http://lmgtfy.com/?q=Asterisk > http://lmgtfy.com/?q=freeswitch > http://lmgtfy.com/?q=openser > http://lmgtfy.com/?q=TrixBox > > > Regards, > Zohair Raza > > > Hi Zohair I was interested in some sort of comparison sheet and its advantages over each other. Regards Kaushal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/22108914/attachment.html From engineerzuhairraza at gmail.com Tue Jan 3 17:04:08 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 3 Jan 2012 18:04:08 +0400 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: Hi, This may help you. http://www.techistan.com/2010/05/31/difference-between-kamailio-and-freeswitch-or-asterisk-and-more-with-mierla/ Regards, Zohair Raza On Tue, Jan 3, 2012 at 5:58 PM, Kaushal Shriyan wrote: > > > On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza wrote: > >> all of them have a wiki page >> >> http://lmgtfy.com/?q=Asterisk >> http://lmgtfy.com/?q=freeswitch >> http://lmgtfy.com/?q=openser >> http://lmgtfy.com/?q=TrixBox >> >> >> Regards, >> Zohair Raza >> >> >> Hi Zohair > > I was interested in some sort of comparison sheet and its advantages over > each other. > > Regards > > Kaushal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/4f2ea04f/attachment.html From nvitaly at gmail.com Tue Jan 3 17:16:36 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Tue, 3 Jan 2012 09:16:36 -0500 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: Hello, These all completely different products. If you tell what exactly you are trying to accomplish you probably get better answer. Vitaly On Tue, Jan 3, 2012 at 8:47 AM, Kaushal Shriyan wrote: > Hi, > > Please help me understand the following applications and what are its > advantages if we compare between each of them. > > Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. > > Regards > > Kaushal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- Vitaly Nikolaev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/32f6e11f/attachment-0001.html From engineerzuhairraza at gmail.com Tue Jan 3 17:17:13 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 3 Jan 2012 18:17:13 +0400 Subject: [Freeswitch-users] Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. In-Reply-To: References: Message-ID: In general, Asterisk and freeswitch are telephony servers like traditional PBXs but with a lot of features and transport is carried over internet protocol (VOIP) TrixBox is a GUI version of Asterisk http://www.voipfon.com.ar/66-trixbox_status.jpg.jpg SER, OpenSER: these are Sip express routers that only deal with sip messages and can provide features like voicemail, call recording etc. They are often used for scaling Asterisk and Freeswitch at large (for thousands of calls) no good idea about others. Regards, Zohair Raza On Tue, Jan 3, 2012 at 5:58 PM, Kaushal Shriyan wrote: > > > On Tue, Jan 3, 2012 at 7:23 PM, Zohair Raza wrote: > >> all of them have a wiki page >> >> http://lmgtfy.com/?q=Asterisk >> http://lmgtfy.com/?q=freeswitch >> http://lmgtfy.com/?q=openser >> http://lmgtfy.com/?q=TrixBox >> >> >> Regards, >> Zohair Raza >> >> >> Hi Zohair > > I was interested in some sort of comparison sheet and its advantages over > each other. > > Regards > > Kaushal > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/283fbfb8/attachment.html From jdiaz at coinfru.com Tue Jan 3 18:03:20 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Tue, 3 Jan 2012 16:03:20 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle Message-ID: I was thinking if it could be possible to build a module like the asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place calls throught GSM networks I can build an asterisk server for this, but i am working continously with freeswitch and i have not too much knowledge of c/c++. I enjoy really work with freeswitch. I think that make this module could be nice to use it in a low cost sip to GSM gateway. We can use it to develop pbx systems with gsm, or trunking. Perhaps there is other module better than this made in freeswitch and i am asking an stupid thing. What i like from this chan dongle is that it is not needed to use any sound card or other device for this. Just only as a module cause the huawey have the voice mode activable. The web page for the chan is this: http://wiki.e1550.mobi/doku.php?id=installation It is open source. so we can use the sources without any issue. Sorry me if the information is not really exaustive or if i maid many mistakes not knowing completely the posibilities with other modules that can do this job. By the way, sorry me for my english. It is not really good. Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/da1d1a4a/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/da1d1a4a/attachment.jpe From gmaruzz at gmail.com Tue Jan 3 18:18:03 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 3 Jan 2012 16:18:03 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: Message-ID: Ciao Josue, I was planning of bringing up mod_gsmopen to use also the huawei module. I've not yet had the time for it, tough. I'm planning to do it starting end of February. Unfortunately I cannot do it before that time, but hopefully it will be ready by end of March. -giovanni On Tue, Jan 3, 2012 at 4:03 PM, Josue Diaz Cruz wrote: > ** > I was thinking if it could be possible to build a module like the asterisk > chan dongle. Uses the huawey USB GMS/UTMS modem to place calls throught GSM > networks > > I can build an asterisk server for this, but i am working continously with > freeswitch and i have not too much knowledge of c/c++. I enjoy really work > with freeswitch. I think that make this module could be nice to use it in a > low cost sip to GSM gateway. We can use it to develop pbx systems with > gsm, or trunking. > > Perhaps there is other module better than this made in freeswitch and i am > asking an stupid thing. What i like from this chan dongle is that it is not > needed to use any sound card or other device for this. Just only as a > module cause the huawey have the voice mode activable. > > The web page for the chan is this: > > http://wiki.e1550.mobi/doku.php?id=installation > > It is open source. so we can use the sources without any issue. > > Sorry me if the information is not really exaustive or if i maid many > mistakes not knowing completely the posibilities with other modules that > can do this job. > > By the way, sorry me for my english. It is not really good. > > * > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > jdiaz at coinfru.com > > > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.com > > > > > > * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/a14e685f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/a14e685f/attachment-0001.jpe From steveu at coppice.org Tue Jan 3 18:25:59 2012 From: steveu at coppice.org (Steve Underwood) Date: Tue, 03 Jan 2012 23:25:59 +0800 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: Message-ID: <4F031E07.8040900@coppice.org> Hi, Whilst you can use the code from chan_dongle privately, for your down purposes, with Freeswitch, it cannot be distributed with Freeswitch. The licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. Steve On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > I was thinking if it could be possible to build a module like the > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > calls throught GSM networks > I can build an asterisk server for this, but i am working continously > with freeswitch and i have not too much knowledge of c/c++. I enjoy > really work with freeswitch. I think that make this module could be > nice to use it in a low cost sip to GSM gateway. We can use it to > develop pbx systems with gsm, or trunking. > Perhaps there is other module better than this made in freeswitch and > i am asking an stupid thing. What i like from this chan dongle is that > it is not needed to use any sound card or other device for this. Just > only as a module cause the huawey have the voice mode activable. > The web page for the chan is this: > http://wiki.e1550.mobi/doku.php?id=installation > It is open source. so we can use the sources without any issue. > Sorry me if the information is not really exaustive or if i maid many > mistakes not knowing completely the posibilities with other modules > that can do this job. > By the way, sorry me for my english. It is not really good. > * > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > /jdiaz at coinfru.com/ > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.com > > // > > * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From justlikeef at gmail.com Tue Jan 3 18:57:22 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 10:57:22 -0500 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <4F031E07.8040900@coppice.org> References: <4F031E07.8040900@coppice.org> Message-ID: <201201031057.22767.justlikeef@gmail.com> While MPL is not compatible with GPLd code, I don't think that GPLd code is compatible with MPL, as MPL is slightly more restrictive. In fact, there are GPL2 projects included now in the form of automaticly built external libraries... On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > Hi, > > Whilst you can use the code from chan_dongle privately, for your down > purposes, with Freeswitch, it cannot be distributed with Freeswitch. The > licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. > > Steve > > On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > I was thinking if it could be possible to build a module like the > > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > calls throught GSM networks > > I can build an asterisk server for this, but i am working continously > > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > really work with freeswitch. I think that make this module could be > > nice to use it in a low cost sip to GSM gateway. We can use it to > > develop pbx systems with gsm, or trunking. > > Perhaps there is other module better than this made in freeswitch and > > i am asking an stupid thing. What i like from this chan dongle is that > > it is not needed to use any sound card or other device for this. Just > > only as a module cause the huawey have the voice mode activable. > > The web page for the chan is this: > > http://wiki.e1550.mobi/doku.php?id=installation > > It is open source. so we can use the sources without any issue. > > Sorry me if the information is not really exaustive or if i maid many > > mistakes not knowing completely the posibilities with other modules > > that can do this job. > > By the way, sorry me for my english. It is not really good. > > * > > > > Josue Diaz Cruz > > > > Departamento Tecnico y Soporte > > > > /jdiaz at coinfru.com/ > > > > C/ Balsicas 3 > > > > Alquerias | 30580 | Murcia > > > > www.coinfru.com > > > > // > > > > * > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/aa411bca/attachment.html From krice at freeswitch.org Tue Jan 3 19:05:47 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Jan 2012 10:05:47 -0600 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031057.22767.justlikeef@gmail.com> Message-ID: Which libs are these? They should all be LGPL2 The GPL is actually actually just as restrictive as the MPL, just in different ways. K On 1/3/12 9:57 AM, "Rob Hutton" wrote: > While MPL is not compatible with GPLd code, I don't think that GPLd code is > compatible with MPL, as MPL is slightly more restrictive. In fact, there are > GPL2 projects included now in the form of automaticly built external > libraries... > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > >> > Hi, > >> > > >> > Whilst you can use the code from chan_dongle privately, for your down > >> > purposes, with Freeswitch, it cannot be distributed with Freeswitch. The > >> > licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. > >> > > >> > Steve > >> > > >> > On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > >>> > > I was thinking if it could be possible to build a module like the > >>> > > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > >>> > > calls throught GSM networks > >>> > > I can build an asterisk server for this, but i am working continously > >>> > > with freeswitch and i have not too much knowledge of c/c++. I enjoy > >>> > > really work with freeswitch. I think that make this module could be > >>> > > nice to use it in a low cost sip to GSM gateway. We can use it to > >>> > > develop pbx systems with gsm, or trunking. > >>> > > Perhaps there is other module better than this made in freeswitch and > >>> > > i am asking an stupid thing. What i like from this chan dongle is that > >>> > > it is not needed to use any sound card or other device for this. Just > >>> > > only as a module cause the huawey have the voice mode activable. > >>> > > The web page for the chan is this: > >>> > > http://wiki.e1550.mobi/doku.php?id=installation > >>> > > It is open source. so we can use the sources without any issue. > >>> > > Sorry me if the information is not really exaustive or if i maid many > >>> > > mistakes not knowing completely the posibilities with other modules > >>> > > that can do this job. > >>> > > By the way, sorry me for my english. It is not really good. > >>> > > * > >>> > > > >>> > > Josue Diaz Cruz > >>> > > > >>> > > Departamento Tecnico y Soporte > >>> > > > >>> > > /jdiaz at coinfru.com/ > >>> > > > >>> > > C/ Balsicas 3 > >>> > > > >>> > > Alquerias | 30580 | Murcia > >>> > > > >>> > > www.coinfru.com > >>> > > > >>> > > // > >>> > > > >>> > > * > >>> > > > >>> > > > >>> > > >>> _________________________________________________________________________ > >>> > > Professional FreeSWITCH Consulting Services: > >>> > > consulting at freeswitch.org > >>> > > http://www.freeswitchsolutions.com > >>> > > > >>> > > > >>> > > > >>> > > > >>> > > Official FreeSWITCH Sites > >>> > > http://www.freeswitch.org > >>> > > http://wiki.freeswitch.org > >>> > > http://www.cluecon.com > >>> > > > >>> > > FreeSWITCH-users mailing list > >>> > > FreeSWITCH-users at lists.freeswitch.org > >>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > > http://www.freeswitch.org > >> > > >> > > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/9a3a6eea/attachment-0001.html From gmaruzz at gmail.com Tue Jan 3 19:07:17 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 3 Jan 2012 17:07:17 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031057.22767.justlikeef@gmail.com> References: <4F031E07.8040900@coppice.org> <201201031057.22767.justlikeef@gmail.com> Message-ID: Rob, no, you're wrong. You cannot include GPLd code in FreeSWITCH. That has been clarified multiple times in the mailing list too. Please do not spread wrong info, that can lead to people wasting their time. -giovanni On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > While MPL is not compatible with GPLd code, I don't think that GPLd code is > compatible with MPL, as MPL is slightly more restrictive. In fact, there are > GPL2 projects included now in the form of automaticly built external > libraries... > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > >> Hi, > >> > >> Whilst you can use the code from chan_dongle privately, for your down > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. The > >> licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. > >> > >> Steve > >> > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > >> > I was thinking if it could be possible to build a module like the > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > >> > calls throught GSM networks > >> > I can build an asterisk server for this, but i am working continously > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > >> > really work with freeswitch. I think that make this module could be > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > >> > develop pbx systems with gsm, or trunking. > >> > Perhaps there is other module better than this made in freeswitch and > >> > i am asking an stupid thing. What i like from this chan dongle is that > >> > it is not needed to use any sound card or other device for this. Just > >> > only as a module cause the huawey have the voice mode activable. > >> > The web page for the chan is this: > >> > http://wiki.e1550.mobi/doku.php?id=installation > >> > It is open source. so we can use the sources without any issue. > >> > Sorry me if the information is not really exaustive or if i maid many > >> > mistakes not knowing completely the posibilities with other modules > >> > that can do this job. > >> > By the way, sorry me for my english. It is not really good. > >> > * > >> > > >> > Josue Diaz Cruz > >> > > >> > Departamento Tecnico y Soporte > >> > > >> > /jdiaz at coinfru.com/ > >> > > >> > C/ Balsicas 3 > >> > > >> > Alquerias | 30580 | Murcia > >> > > >> > www.coinfru.com > >> > > >> > // > >> > > >> > * > >> > > >> > > >> > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From justlikeef at gmail.com Tue Jan 3 19:38:36 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 11:38:36 -0500 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: Message-ID: <201201031138.37334.justlikeef@gmail.com> Sorry, missed the first L. Everything is either under special licese or LGPL2. On Tuesday 03 January 2012 11:05:47 Ken Rice wrote: > Which libs are these? They should all be LGPL2 > > The GPL is actually actually just as restrictive as the MPL, just in > different ways. > > K > > > On 1/3/12 9:57 AM, "Rob Hutton" wrote: > > > While MPL is not compatible with GPLd code, I don't think that GPLd code is > > compatible with MPL, as MPL is slightly more restrictive. In fact, there are > > GPL2 projects included now in the form of automaticly built external > > libraries... > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > >> > Hi, > > > >> > > > > >> > Whilst you can use the code from chan_dongle privately, for your down > > > >> > purposes, with Freeswitch, it cannot be distributed with Freeswitch. The > > > >> > licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. > > > >> > > > > >> > Steve > > > >> > > > > >> > On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > >>> > > I was thinking if it could be possible to build a module like the > > > >>> > > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > >>> > > calls throught GSM networks > > > >>> > > I can build an asterisk server for this, but i am working continously > > > >>> > > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > >>> > > really work with freeswitch. I think that make this module could be > > > >>> > > nice to use it in a low cost sip to GSM gateway. We can use it to > > > >>> > > develop pbx systems with gsm, or trunking. > > > >>> > > Perhaps there is other module better than this made in freeswitch and > > > >>> > > i am asking an stupid thing. What i like from this chan dongle is that > > > >>> > > it is not needed to use any sound card or other device for this. Just > > > >>> > > only as a module cause the huawey have the voice mode activable. > > > >>> > > The web page for the chan is this: > > > >>> > > http://wiki.e1550.mobi/doku.php?id=installation > > > >>> > > It is open source. so we can use the sources without any issue. > > > >>> > > Sorry me if the information is not really exaustive or if i maid many > > > >>> > > mistakes not knowing completely the posibilities with other modules > > > >>> > > that can do this job. > > > >>> > > By the way, sorry me for my english. It is not really good. > > > >>> > > * > > > >>> > > > > > >>> > > Josue Diaz Cruz > > > >>> > > > > > >>> > > Departamento Tecnico y Soporte > > > >>> > > > > > >>> > > /jdiaz at coinfru.com/ > > > >>> > > > > > >>> > > C/ Balsicas 3 > > > >>> > > > > > >>> > > Alquerias | 30580 | Murcia > > > >>> > > > > > >>> > > www.coinfru.com > > > >>> > > > > > >>> > > // > > > >>> > > > > > >>> > > * > > > >>> > > > > > >>> > > > > > >>> > > > >>> _________________________________________________________________________ > > > >>> > > Professional FreeSWITCH Consulting Services: > > > >>> > > consulting at freeswitch.org > > > >>> > > http://www.freeswitchsolutions.com > > > >>> > > > > > >>> > > > > > >>> > > > > > >>> > > > > > >>> > > Official FreeSWITCH Sites > > > >>> > > http://www.freeswitch.org > > > >>> > > http://wiki.freeswitch.org > > > >>> > > http://www.cluecon.com > > > >>> > > > > > >>> > > FreeSWITCH-users mailing list > > > >>> > > FreeSWITCH-users at lists.freeswitch.org > > > >>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >>> > > http://www.freeswitch.org > > > >> > > > > >> > > > > >> > _________________________________________________________________________ > > > >> > Professional FreeSWITCH Consulting Services: > > > >> > consulting at freeswitch.org > > > >> > http://www.freeswitchsolutions.com > > > >> > > > > >> > > > > >> > > > > >> > > > > >> > Official FreeSWITCH Sites > > > >> > http://www.freeswitch.org > > > >> > http://wiki.freeswitch.org > > > >> > http://www.cluecon.com > > > >> > > > > >> > FreeSWITCH-users mailing list > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >> > http://www.freeswitch.org > > > >> > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/fed93b5f/attachment-0001.html From mashudi72 at gmail.com Tue Jan 3 19:57:06 2012 From: mashudi72 at gmail.com (Mashudi) Date: Tue, 03 Jan 2012 23:57:06 +0700 Subject: [Freeswitch-users] mod_sofia not binding to port 5060 Message-ID: Brandon McGinty wrote: >Thanks for these. However, I checked all three of these resources. >For some reason, this issue is only occuring when the init.d script for >debian, is used to start freeswitch; from the command line, the issue >goes away completely. >All that is running is the -nc switch, so I'm not sure what could be >causing the issue. >Thanks again. > >Brandon McGinty-Carroll > > >On 1/2/2012 8:34 PM, Tamer Higazi wrote: >> Look at: >> >> conf/autoload_configs/switch.conf.xml >> >> read this: >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >> >> >> look at "sip-port" >> >> >> I am tired, I am going to bed. >> >> >> Tamer >> >> >> Am 02.01.2012 18:03, schrieb Brandon McGinty: >>> Good afternoon. >>> We've got a freeswitch machine, no external SIP gateways for outside >>> calls, eight extensions, two conferences, and a completely public IP for >>> the machine in question (no NAT weirdness), running on Debian 6 (weezy/sid). >>> I woke up this morning, to find that sip_external_port 5060 was not >>> listening, though it was enabled in config/vars.xml. >>> The internal port, 5080, is listening, and I have folks confirming that >>> they can speak using that port. >>> I've set netcat to listen via 5080, so I know it isn't firewalled, or >>> not at least ina conventional manner. >>> Any help you all can provide, things I can check, would be greatly >>> appreciated. >>> >>> Sincerely, >>> Brandon McGinty-Carroll >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org From justlikeef at gmail.com Tue Jan 3 19:58:27 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 11:58:27 -0500 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: <201201031057.22767.justlikeef@gmail.com> Message-ID: <201201031158.28269.justlikeef@gmail.com> I'm not going to start a holy war on the list, nor spread fud, but: http://www.mozilla.org/MPL/MPL-1.1.html http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are defined as "means a work which combines Covered Code or portions thereof with code not governed by the terms of this License", and other licenses are specifially allowed under these provisions under the MPL. The problem is with including MPL code in works licensed under GNU licence, as covered http://www.gnu.org/licenses/license-list.html, and as long as you are including MPL2 code and have any of the GNU2 series license, you are OK that way, also. From Wikipedia: The MPL is GPL-incompatible because the GPL module cannot be legally linked with an MPL module. However, versions of the MPL such as MPL 1.1 have a provision that allows part of a program to offer the GNU GPL as an alternative choice, thereby allowing part of the program to have a GPL-compatible license. That doesn't mean that the project can't accept code, it means that that the project managers don't want to, for whatever reason, and that is completely up to them. But sending someone off with the misconception that it is not allowed by license is just going to lead to future problems for someone. Thanks, Rob On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > Rob, > > no, you're wrong. > > You cannot include GPLd code in FreeSWITCH. > > That has been clarified multiple times in the mailing list too. > > Please do not spread wrong info, that can lead to people wasting their time. > > -giovanni > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > > While MPL is not compatible with GPLd code, I don't think that GPLd code is > > compatible with MPL, as MPL is slightly more restrictive. In fact, there are > > GPL2 projects included now in the form of automaticly built external > > libraries... > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > >> Hi, > > > >> > > > >> Whilst you can use the code from chan_dongle privately, for your down > > > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. The > > > >> licence for chan_dongle is GPL v2. This is incompatible with Freeswitch. > > > >> > > > >> Steve > > > >> > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > >> > I was thinking if it could be possible to build a module like the > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > >> > calls throught GSM networks > > > >> > I can build an asterisk server for this, but i am working continously > > > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > >> > really work with freeswitch. I think that make this module could be > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > > > >> > develop pbx systems with gsm, or trunking. > > > >> > Perhaps there is other module better than this made in freeswitch and > > > >> > i am asking an stupid thing. What i like from this chan dongle is that > > > >> > it is not needed to use any sound card or other device for this. Just > > > >> > only as a module cause the huawey have the voice mode activable. > > > >> > The web page for the chan is this: > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > >> > It is open source. so we can use the sources without any issue. > > > >> > Sorry me if the information is not really exaustive or if i maid many > > > >> > mistakes not knowing completely the posibilities with other modules > > > >> > that can do this job. > > > >> > By the way, sorry me for my english. It is not really good. > > > >> > * > > > >> > > > > >> > Josue Diaz Cruz > > > >> > > > > >> > Departamento Tecnico y Soporte > > > >> > > > > >> > /jdiaz at coinfru.com/ > > > >> > > > > >> > C/ Balsicas 3 > > > >> > > > > >> > Alquerias | 30580 | Murcia > > > >> > > > > >> > www.coinfru.com > > > >> > > > > >> > // > > > >> > > > > >> > * > > > >> > > > > >> > > > > >> > > >> > _________________________________________________________________________ > > > >> > Professional FreeSWITCH Consulting Services: > > > >> > consulting at freeswitch.org > > > >> > http://www.freeswitchsolutions.com > > > >> > > > > >> > > > > >> > > > > >> > > > > >> > Official FreeSWITCH Sites > > > >> > http://www.freeswitch.org > > > >> > http://wiki.freeswitch.org > > > >> > http://www.cluecon.com > > > >> > > > > >> > FreeSWITCH-users mailing list > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >> > http://www.freeswitch.org > > > >> > > > >> > > > >> _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >> http://www.freeswitch.org > > > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/f348c9cc/attachment-0001.html From curriegrad2004 at gmail.com Tue Jan 3 20:04:44 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 3 Jan 2012 09:04:44 -0800 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: References: Message-ID: Well, at least this is our first person who's asked us about the IPP G729 codec this year. On Tue, Jan 3, 2012 at 5:07 AM, Zohair Raza wrote: > LOL > > Regards, > Zohair Raza > > > On Tue, Jan 3, 2012 at 4:55 PM, Kerem Erciyes > wrote: >> >> Didn't your mom teach you not to play with IPP Codecs. >> >> -- Plus I've never seen and IPP Codec that works without memory problems, >> FS commercial codec works well however. >> >> On Tue, Jan 3, 2012 at 1:53 PM, Anita Hall >> wrote: >>> >>> Hi! >>> >>> We have been playing with Intel IPP G729 code and its module for >>> FreeSWITCH. >>> >>> As reported here, we have successfully compile the fs-g729 module after >>> install IPP libraries. >>> https://code.google.com/p/fs-g729/issues/detail?id=3#c1 >>> >>> However, to test the codec in a loopback SIP call (FreeSWITCH calling >>> itself over SIP), I need some G729 encoded files (do I really?) >>> >>> FreeSWITCH gives errors with G729 encoded file from here >>> http://www.enicomms.com/cutglassivr/audiofiles/Alison_Keenan-British-English-g729.tar.gz >>> Dialplan >>> >>> >>> [ERR] switch_core_file.c:122 Invalid file format [g729] for >>> [/usr/local/freeswitch/sounds/en/us/callie/ivr/demo-thanks.g729]! >>> >>> And also gives errors with file encoded using Intel IPP sample G729 >>> encoder. >>> >>> [ERR] switch_core_file.c:122 Invalid file format [g729] for >>> [/usr/local/freeswitch/sounds/en/us/callie/ivr/ivr-welcome_to_freeswitch.g729]! >>> >>> Please note that the files are actually kept in the directory >>> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000 as are other demo_ivr >>> sound files. >>> >>> # file /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/*.g72* >>> >>> /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/demo-thanks.g729: >>> data >>> >>> /usr/local/freeswitch/sounds/en/us/callie//ivr/8000/ivr-welcome_to_freeswitch.g729: >>> RIFF (little-endian) data, WAVE audio, mono 8000 Hz >>> >>> I suspect that ivr-welcome_to_freeswitch.g729 (produced by Intel IPP G729 >>> encoder) is not really G729. >>> >>> Any pointers will be most helpful. >>> >>> Thanks! >>> Anita. >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Kerem Erciyes - Sistem Danismani >> http://keremerciyes.com >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveu at coppice.org Tue Jan 3 20:03:52 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 04 Jan 2012 01:03:52 +0800 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031158.28269.justlikeef@gmail.com> References: <201201031057.22767.justlikeef@gmail.com> <201201031158.28269.justlikeef@gmail.com> Message-ID: <4F0334F8.1080005@coppice.org> On 01/04/2012 12:58 AM, Rob Hutton wrote: Can you explain how..... > > From Wikipedia: > > The MPL is GPL-incompatible because the GPL module cannot be legally > linked with an MPL module. However, versions of the MPL such as MPL > 1.1 have a provision that allows part of a program to offer the GNU > GPL as an alternative choice, thereby allowing part of the program to > have a GPL-compatible license. > .... leads you to the following conclusion? > > That doesn't mean that the project can't accept code, it means that > that the project managers don't want to, for whatever reason, and that > is completely up to them. But sending someone off with the > misconception that it is not allowed by license is just going to lead > to future problems for someone. > > Steve From garbytrash at gmail.com Tue Jan 3 12:18:11 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 3 Jan 2012 10:18:11 +0100 Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: References: <1325549865328-7145235.post@n2.nabble.com> Message-ID: looks promising, but the user login does not work. Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to all freeswitchers! On 1/3/12, nbhatti wrote: > Yes, it will support prepaid calling card and many more features soon. > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > wrote: >> It looks great! >> >> Will vBilling support batch user/prepaid calling card creation? >> >> ________________________________ >> If you reply to this email, your message will be added to the discussion >> below: >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >> To unsubscribe from vBilling Beta Program!!, click here. >> NAML > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > Sent from the freeswitch-users mailing list archive at Nabble.com. From eli at teltechcorp.com Tue Jan 3 17:16:07 2012 From: eli at teltechcorp.com (Eli Finkelman) Date: Tue, 3 Jan 2012 09:16:07 -0500 Subject: [Freeswitch-users] MP3 playback sounds bad in Chrome, encoding issue? Message-ID: Hi There, Whenever we playback an MP3 recording using the native Chrome MP3 player, it sounds really horrible, it jumps and crackles. When listening to it in Firefox/Safari it sounds just fine. Could there be an issue with the way FS is encoding the files? Is this just an issue with Chrome? We're using the native record_session method to record the MP3. switch_event.c:1521 Parsing variable [execute_on_answer]=[record_session /tmp/recordings/RE91fec223f65143e2aa5246b1f3a071a8.mp3] Here's a sample of what the audio sounds like, try playing this in Chrome in Firefox http://recordings.telapi.com/RB1cd90a0383114455830be5755c1891e6/RE91fec223f65143e2aa5246b1f3a071a8.mp3 Best Regards Eli -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/1740be1d/attachment-0001.html From daggelinckxmichel at gmail.com Tue Jan 3 18:14:59 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Tue, 3 Jan 2012 16:14:59 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: Message-ID: Sounds like a nice feature, especialy if it can be expanded to support a large number of usb gsm modems. On Tue, Jan 3, 2012 at 4:03 PM, Josue Diaz Cruz wrote: > ** > I was thinking if it could be possible to build a module like the asterisk > chan dongle. Uses the huawey USB GMS/UTMS modem to place calls throught GSM > networks > > I can build an asterisk server for this, but i am working continously with > freeswitch and i have not too much knowledge of c/c++. I enjoy really work > with freeswitch. I think that make this module could be nice to use it in a > low cost sip to GSM gateway. We can use it to develop pbx systems with > gsm, or trunking. > > Perhaps there is other module better than this made in freeswitch and i am > asking an stupid thing. What i like from this chan dongle is that it is not > needed to use any sound card or other device for this. Just only as a > module cause the huawey have the voice mode activable. > > The web page for the chan is this: > > http://wiki.e1550.mobi/doku.php?id=installation > > It is open source. so we can use the sources without any issue. > > Sorry me if the information is not really exaustive or if i maid many > mistakes not knowing completely the posibilities with other modules that > can do this job. > > By the way, sorry me for my english. It is not really good. > > * > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > jdiaz at coinfru.com > > > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.com > > > > > > * > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/8b427fd7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/8b427fd7/attachment-0001.jpe From anthony.minessale at gmail.com Tue Jan 3 20:43:43 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jan 2012 11:43:43 -0600 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031158.28269.justlikeef@gmail.com> References: <201201031057.22767.justlikeef@gmail.com> <201201031158.28269.justlikeef@gmail.com> Message-ID: The reason MPL is incompatible with GPL is because the GPL is designed to keep free software free by making it mandatory for its code and code combined with it to fall under its license. MPL only requires changes to be shared with the author of the code they are working with. MPL has no restriction on distribution or trying to keep the resulting work free. We keep our code free because we want to and do not use the GPL to enforce it. We intend for our software to be made into products and we put enough work into the project that we feel those who benefit from our software will come back and contribute to our community which is much stronger and able to maintain the code than any private entity and anyone who can't see that probably cannot be convinced otherwise. There is a severe lack of education about open source licenses. Its very political and most of the people fighting about it don't even actually understand the details. I'm not accusing anyone on this thread of not understanding but it's important to realize that a large number of people who even use the GPL just default to it assuming its the only choice for open source and that it actually means open source. GPL tries very hard to underscore that FREE means FREE as in liberty and not FREE as in "no cost". I choose not to license my software under GPL but that does not mean I do not completely understand its intent or appreciate anything that has come from its existence. Anyway, We can't use GPL code. We can't release any of FS as GPL or it would break some of the other licenses, but you can ask the author of the GPL code to make it LGPL which is compat since it allows linking without contamination. You can also read the other code to see how it works and make your own implementation. On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton wrote: > ** > > I'm not going to start a holy war on the list, nor spread fud, but: > > > http://www.mozilla.org/MPL/MPL-1.1.html > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > defined as "means a work which combines Covered Code or portions thereof > with code not governed by the terms of this License", and other licenses > are specifially allowed under these provisions under the MPL. > > > The problem is with including MPL code in works licensed under GNU > licence, as covered http://www.gnu.org/licenses/license-list.html, and as > long as you are including MPL2 code and have any of the GNU2 series > license, you are OK that way, also. > > > From Wikipedia: > > The MPL is GPL-incompatible because the GPL module cannot be legally > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > have a provision that allows part of a program to offer the GNU GPL as an > alternative choice, thereby allowing part of the program to have a > GPL-compatible license. > > > > That doesn't mean that the project can't accept code, it means that that > the project managers don't want to, for whatever reason, and that is > completely up to them. But sending someone off with the misconception that > it is not allowed by license is just going to lead to future problems for > someone. > > > Thanks, > > Rob > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > Rob, > > > > > > no, you're wrong. > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > Please do not spread wrong info, that can lead to people wasting their > time. > > > > > > -giovanni > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > code is > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > there are > > > > GPL2 projects included now in the form of automaticly built external > > > > libraries... > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > >> Hi, > > > > > > > >> > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your down > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. > The > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > Freeswitch. > > > > > > > >> > > > > > > > >> Steve > > > > > > > >> > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > >> > I was thinking if it could be possible to build a module like the > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > > > > > >> > calls throught GSM networks > > > > > > > >> > I can build an asterisk server for this, but i am working > continously > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > > > > > >> > really work with freeswitch. I think that make this module could be > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > >> > Perhaps there is other module better than this made in freeswitch > and > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle is > that > > > > > > > >> > it is not needed to use any sound card or other device for this. > Just > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > >> > The web page for the chan is this: > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > many > > > > > > > >> > mistakes not knowing completely the posibilities with other modules > > > > > > > >> > that can do this job. > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > >> > * > > > > > > > >> > > > > > > > > >> > Josue Diaz Cruz > > > > > > > >> > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > >> > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > >> > > > > > > > > >> > C/ Balsicas 3 > > > > > > > >> > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > >> > > > > > > > > >> > www.coinfru.com > > > > > > > >> > > > > > > > > >> > // > > > > > > > >> > > > > > > > > >> > * > > > > > > > >> > > > > > > > > >> > > > > > > > > >> > > > > >> > > _________________________________________________________________________ > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > >> > consulting at freeswitch.org > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > >> > > > > > > > > >> > > > > > > > > >> > > > > > > > > >> > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > >> > http://www.freeswitch.org > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > >> > http://www.cluecon.com > > > > > > > >> > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > >> > http://www.freeswitch.org > > > > > > > >> > > > > > > > >> > > > > > > > >> > _________________________________________________________________________ > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > >> consulting at freeswitch.org > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > >> > > > > > > > >> > > > > > > > >> > > > > > > > >> > > > > > > > >> Official FreeSWITCH Sites > > > > > > > >> http://www.freeswitch.org > > > > > > > >> http://wiki.freeswitch.org > > > > > > > >> http://www.cluecon.com > > > > > > > >> > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > >> http://www.freeswitch.org > > > > > > > >> > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/ae6cf37c/attachment-0001.html From justlikeef at gmail.com Tue Jan 3 20:59:18 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 12:59:18 -0500 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <4F0334F8.1080005@coppice.org> References: <201201031158.28269.justlikeef@gmail.com> <4F0334F8.1080005@coppice.org> Message-ID: <201201031259.19231.justlikeef@gmail.com> Sorry if this is coming across sarcastic, I honestly don't mean it to be, licensing is just not as cut and dry as it is or it isn't. There are 1000 conflicting legal opinions about this stuff, which is why the licenses keep getting revised. - MPL specifically allows other licenses (starting at v1.1) - Includion in MPL projects is specifically allowed in all versions of GPL3 and most GPL2 code (as discussed here http://www.gnu.org/licenses/license-list.html) as you are allowed to derive works under whatever license you want as long as the GPLd code and any contributions you make to them are made available in source form and the license is included. ...so it doesn't appear to me to be a licensing issue with most current code, at least. Certainly there have been issues in the past If it has been clarified on the list many times that GPLd code can't be included in FreeSWITCH, and it's not a licensing issue, then who would have made the decision not to allow it? Thanks, Rob On Tuesday 03 January 2012 12:03:52 Steve Underwood wrote: > On 01/04/2012 12:58 AM, Rob Hutton wrote: > Can you explain how..... > > > > From Wikipedia: > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > linked with an MPL module. However, versions of the MPL such as MPL > > 1.1 have a provision that allows part of a program to offer the GNU > > GPL as an alternative choice, thereby allowing part of the program to > > have a GPL-compatible license. > > > .... leads you to the following conclusion? > > > > That doesn't mean that the project can't accept code, it means that > > that the project managers don't want to, for whatever reason, and that > > is completely up to them. But sending someone off with the > > misconception that it is not allowed by license is just going to lead > > to future problems for someone. > > > > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/8b47504d/attachment.html From sherifomran2000 at yahoo.com Tue Jan 3 21:08:39 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 3 Jan 2012 10:08:39 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: Message-ID: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> Can you give more details how it does not work? I have the same situation. I can reach the frontpage and when i give the username i get 404. Is this the case you have? Do you have Centos or Redhat ? regards, Sherif --- On Tue, 1/3/12, Zenny wrote: From: Zenny Subject: Re: [Freeswitch-users] vBilling Beta Program!! To: "FreeSWITCH Users Help" Date: Tuesday, January 3, 2012, 11:18 AM looks promising, but the user login does not work. Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to all freeswitchers! On 1/3/12, nbhatti wrote: > Yes, it will support prepaid calling card and many more features soon. > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > wrote: >> It looks great! >> >> Will vBilling support batch user/prepaid calling card creation? >> >> ________________________________ >> If you reply to this email, your message will be added to the discussion >> below: >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >> To unsubscribe from vBilling Beta Program!!, click here. >> NAML > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/d095c1c5/attachment.html From nbhatti at gmail.com Tue Jan 3 21:09:58 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 3 Jan 2012 21:09:58 +0300 Subject: [Freeswitch-users] vBilling Announcement - First Beta Release Message-ID: Hello everyone, Though a few days late, but happy new year to everyone. Last year was tough and we worked hard to get vBilling up and running. So, here we are, announcing the first beta release. vBilling is the first Open Source billing platform for FreeSWITCH. Take a look at http://www.vbilling.org/ for more details. Some of the major features includes: Both Pre-Paid and Post-Paid model Multiple administration access levels Multiple reseller level Easy rate/price management Route management Separate user interface to view their CDR(s) and billing information Authentication by IP/ANI and SIP registration Codec management for both user and switch CDR statistics Gateway statistics Admin/Reseller/User management Switch management from 1 GUI Balance and payment information I would like to invite everyone to download and install using the installer scripts available at http://vbilling.org/get-started/ We need your contribution and help to make this a success project. Don't forget to visit our forum at http://forum.vbilling.org/ and also the Customer support http://support.vbilling.org/ if required. All the source code is also available http://github.com/digitallinx/vBilling/ So, take a look, download, install and have fun. Thanks, Muhammad Naseer (Goni) From nbhatti at gmail.com Tue Jan 3 21:16:09 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 3 Jan 2012 21:16:09 +0300 Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> References: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: Sherif, have you installed manually? If so, you would have to enable mod_rewrite in your apache configuration. On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: > Can you give more details how it does not work? I have the same situation. > I can reach the frontpage and when i give the username i get 404. > > Is this the case you have? > > Do you have Centos or Redhat ? > > > regards, > Sherif > > > --- On *Tue, 1/3/12, Zenny * wrote: > > > From: Zenny > Subject: Re: [Freeswitch-users] vBilling Beta Program!! > To: "FreeSWITCH Users Help" > Date: Tuesday, January 3, 2012, 11:18 AM > > > looks promising, but the user login does not work. > > Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to > all freeswitchers! > > On 1/3/12, nbhatti > > wrote: > > Yes, it will support prepaid calling card and many more features soon. > > > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > > > > wrote: > >> It looks great! > >> > >> Will vBilling support batch user/prepaid calling card creation? > >> > >> ________________________________ > >> If you reply to this email, your message will be added to the discussion > >> below: > >> > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html > >> To unsubscribe from vBilling Beta Program!!, click here. > >> NAML > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/d3b41c9c/attachment-0001.html From justlikeef at gmail.com Tue Jan 3 21:18:09 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 13:18:09 -0500 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: <201201031158.28269.justlikeef@gmail.com> Message-ID: <201201031318.10145.justlikeef@gmail.com> Thank you for the clarification Anthony!!! I just didn't want someone walking away with the absolute and finding out someone else that there are other opinions and getting pissed off because they got something else from what they were told here. What I was trying to say was that the leaders of any project have the absolute right to decide what they will and will not accept. There has been a great amount of work in making the licenses compatable with each other to address this exact situation. Many projects are multiple licensed at this point, and many projects that are MPL'd include GPL code as a dependency, although the opposite is largely not true, for many reasons. Modules under other licenses can be included and distributed without modifying either their license or the FreeSWITCH licensing as long as the terms of that license are adhered to. For this project, it needs to be an LGPL or other "free without restriction" license like OpenLDAP, etc. Not knowing the history, I took Steve's original comment to mean that this was a licensing incompatability. Sorry to ruffle feathers!!! On Tuesday 03 January 2012 12:43:43 Anthony Minessale wrote: > The reason MPL is incompatible with GPL is because the GPL is designed to > keep free software free by making it mandatory for its code and code > combined with it to fall under its license. > > MPL only requires changes to be shared with the author of the code they are > working with. MPL has no restriction on distribution or trying to keep the > resulting work free. > > We keep our code free because we want to and do not use the GPL to enforce > it. We intend for our software to be made into products and we put enough > work into the project that we feel those who benefit from our software will > come back and contribute to our community which is much stronger and able > to maintain the code than any private entity and anyone who can't see that > probably cannot be convinced otherwise. > > There is a severe lack of education about open source licenses. Its very > political and most of the people fighting about it don't even actually > understand the details. I'm not accusing anyone on this thread of not > understanding but it's important to realize that a large number of people > who even use the GPL just default to it assuming its the only choice for > open source and that it actually means open source. GPL tries very hard to > underscore that FREE means FREE as in liberty and not FREE as in "no cost". > > I choose not to license my software under GPL but that does not mean I do > not completely understand its intent or appreciate anything that has come > from its existence. > > Anyway, We can't use GPL code. We can't release any of FS as GPL or it > would break some of the other licenses, but you can ask the author of the > GPL code to make it LGPL which is compat since it allows linking without > contamination. > > You can also read the other code to see how it works and make your own > implementation. > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton wrote: > > > ** > > > > I'm not going to start a holy war on the list, nor spread fud, but: > > > > > > http://www.mozilla.org/MPL/MPL-1.1.html > > > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > > defined as "means a work which combines Covered Code or portions thereof > > with code not governed by the terms of this License", and other licenses > > are specifially allowed under these provisions under the MPL. > > > > > > The problem is with including MPL code in works licensed under GNU > > licence, as covered http://www.gnu.org/licenses/license-list.html, and as > > long as you are including MPL2 code and have any of the GNU2 series > > license, you are OK that way, also. > > > > > > From Wikipedia: > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > > have a provision that allows part of a program to offer the GNU GPL as an > > alternative choice, thereby allowing part of the program to have a > > GPL-compatible license. > > > > > > > > That doesn't mean that the project can't accept code, it means that that > > the project managers don't want to, for whatever reason, and that is > > completely up to them. But sending someone off with the misconception that > > it is not allowed by license is just going to lead to future problems for > > someone. > > > > > > Thanks, > > > > Rob > > > > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > > > Rob, > > > > > > > > > > no, you're wrong. > > > > > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > > > > > Please do not spread wrong info, that can lead to people wasting their > > time. > > > > > > > > > > -giovanni > > > > > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > > > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > > code is > > > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > > there are > > > > > > GPL2 projects included now in the form of automaticly built external > > > > > > libraries... > > > > > > > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > > > > > >> Hi, > > > > > > > > > > > >> > > > > > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your down > > > > > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. > > The > > > > > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > > Freeswitch. > > > > > > > > > > > >> > > > > > > > > > > > >> Steve > > > > > > > > > > > >> > > > > > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > > > > > >> > I was thinking if it could be possible to build a module like the > > > > > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > > > > > > > > > >> > calls throught GSM networks > > > > > > > > > > > >> > I can build an asterisk server for this, but i am working > > continously > > > > > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > > > > > > > > > >> > really work with freeswitch. I think that make this module could be > > > > > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > > > > > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > > > > > >> > Perhaps there is other module better than this made in freeswitch > > and > > > > > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle is > > that > > > > > > > > > > > >> > it is not needed to use any sound card or other device for this. > > Just > > > > > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > > > > > >> > The web page for the chan is this: > > > > > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > > many > > > > > > > > > > > >> > mistakes not knowing completely the posibilities with other modules > > > > > > > > > > > >> > that can do this job. > > > > > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > Josue Diaz Cruz > > > > > > > > > > > >> > > > > > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > > > > > >> > > > > > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > > > > > >> > > > > > > > > > > > > >> > C/ Balsicas 3 > > > > > > > > > > > >> > > > > > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > > > > > >> > > > > > > > > > > > > >> > www.coinfru.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > // > > > > > > > > > > > >> > > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > >> > > > _________________________________________________________________________ > > > > > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> > consulting at freeswitch.org > > > > > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > > > > > >> > http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > _________________________________________________________________________ > > > > > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> consulting at freeswitch.org > > > > > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> Official FreeSWITCH Sites > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> http://wiki.freeswitch.org > > > > > > > > > > > >> http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/599e1d90/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 3 22:01:12 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jan 2012 13:01:12 -0600 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031318.10145.justlikeef@gmail.com> References: <201201031158.28269.justlikeef@gmail.com> <201201031318.10145.justlikeef@gmail.com> Message-ID: I think my last email actually was written while you were writing your 2nd to last reply but it was not a direct response to anything you actually said I just felt the need to give our official position on the matter after seeing the thread. No feathers ruffled, I never get excited about licensing, I try not to fight about anything that is based on preference and simply state our policy =p On Tue, Jan 3, 2012 at 12:18 PM, Rob Hutton wrote: > ** > > Thank you for the clarification Anthony!!! I just didn't want someone > walking away with the absolute and finding out someone else that there are > other opinions and getting pissed off because they got something else from > what they were told here. > > > What I was trying to say was that the leaders of any project have the > absolute right to decide what they will and will not accept. > > > There has been a great amount of work in making the licenses compatable > with each other to address this exact situation. Many projects are multiple > licensed at this point, and many projects that are MPL'd include GPL code > as a dependency, although the opposite is largely not true, for many > reasons. Modules under other licenses can be included and distributed > without modifying either their license or the FreeSWITCH licensing as long > as the terms of that license are adhered to. > > > For this project, it needs to be an LGPL or other "free without > restriction" license like OpenLDAP, etc. Not knowing the history, I took > Steve's original comment to mean that this was a licensing incompatability. > Sorry to ruffle feathers!!! > > > > On Tuesday 03 January 2012 12:43:43 Anthony Minessale wrote: > > > The reason MPL is incompatible with GPL is because the GPL is designed to > > > keep free software free by making it mandatory for its code and code > > > combined with it to fall under its license. > > > > > > MPL only requires changes to be shared with the author of the code they > are > > > working with. MPL has no restriction on distribution or trying to keep > the > > > resulting work free. > > > > > > We keep our code free because we want to and do not use the GPL to > enforce > > > it. We intend for our software to be made into products and we put enough > > > work into the project that we feel those who benefit from our software > will > > > come back and contribute to our community which is much stronger and able > > > to maintain the code than any private entity and anyone who can't see > that > > > probably cannot be convinced otherwise. > > > > > > There is a severe lack of education about open source licenses. Its very > > > political and most of the people fighting about it don't even actually > > > understand the details. I'm not accusing anyone on this thread of not > > > understanding but it's important to realize that a large number of people > > > who even use the GPL just default to it assuming its the only choice for > > > open source and that it actually means open source. GPL tries very hard > to > > > underscore that FREE means FREE as in liberty and not FREE as in "no > cost". > > > > > > I choose not to license my software under GPL but that does not mean I do > > > not completely understand its intent or appreciate anything that has come > > > from its existence. > > > > > > Anyway, We can't use GPL code. We can't release any of FS as GPL or it > > > would break some of the other licenses, but you can ask the author of the > > > GPL code to make it LGPL which is compat since it allows linking without > > > contamination. > > > > > > You can also read the other code to see how it works and make your own > > > implementation. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton > wrote: > > > > > > > ** > > > > > > > > I'm not going to start a holy war on the list, nor spread fud, but: > > > > > > > > > > > > http://www.mozilla.org/MPL/MPL-1.1.html > > > > > > > > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > > > > > > > > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > > > > defined as "means a work which combines Covered Code or portions > thereof > > > > with code not governed by the terms of this License", and other > licenses > > > > are specifially allowed under these provisions under the MPL. > > > > > > > > > > > > The problem is with including MPL code in works licensed under GNU > > > > licence, as covered http://www.gnu.org/licenses/license-list.html, > and as > > > > long as you are including MPL2 code and have any of the GNU2 series > > > > license, you are OK that way, also. > > > > > > > > > > > > From Wikipedia: > > > > > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > > > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > > > > have a provision that allows part of a program to offer the GNU GPL as > an > > > > alternative choice, thereby allowing part of the program to have a > > > > GPL-compatible license. > > > > > > > > > > > > > > > > That doesn't mean that the project can't accept code, it means that > that > > > > the project managers don't want to, for whatever reason, and that is > > > > completely up to them. But sending someone off with the misconception > that > > > > it is not allowed by license is just going to lead to future problems > for > > > > someone. > > > > > > > > > > > > Thanks, > > > > > > > > Rob > > > > > > > > > > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > > > > > > > Rob, > > > > > > > > > > > > > > > > > > no, you're wrong. > > > > > > > > > > > > > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > > > > > > > > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > > > > > > > > > > > > > Please do not spread wrong info, that can lead to people wasting > their > > > > time. > > > > > > > > > > > > > > > > > > -giovanni > > > > > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton > wrote: > > > > > > > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > > > > code is > > > > > > > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > > > > there are > > > > > > > > > > GPL2 projects included now in the form of automaticly built > external > > > > > > > > > > libraries... > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > > > > > > > > > > > > > >> Hi, > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your > down > > > > > > > > > > > > > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with > Freeswitch. > > > > The > > > > > > > > > > > > > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > > > > Freeswitch. > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Steve > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > > > > > > > > > > > > > >> > I was thinking if it could be possible to build a module like > the > > > > > > > > > > > > > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to > place > > > > > > > > > > > > > > > > > > > >> > calls throught GSM networks > > > > > > > > > > > > > > > > > > > >> > I can build an asterisk server for this, but i am working > > > > continously > > > > > > > > > > > > > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I > enjoy > > > > > > > > > > > > > > > > > > > >> > really work with freeswitch. I think that make this module > could be > > > > > > > > > > > > > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it > to > > > > > > > > > > > > > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > > > > > > > > > > > > > >> > Perhaps there is other module better than this made in > freeswitch > > > > and > > > > > > > > > > > > > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle > is > > > > that > > > > > > > > > > > > > > > > > > > >> > it is not needed to use any sound card or other device for this. > > > > Just > > > > > > > > > > > > > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > > > > > > > > > > > > > >> > The web page for the chan is this: > > > > > > > > > > > > > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > > > > > > > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > > > > > > > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > > > > many > > > > > > > > > > > > > > > > > > > >> > mistakes not knowing completely the posibilities with other > modules > > > > > > > > > > > > > > > > > > > >> > that can do this job. > > > > > > > > > > > > > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > > > > > > > > > > > > > >> > * > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Josue Diaz Cruz > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > C/ Balsicas 3 > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > www.coinfru.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > // > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > * > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > >> > > > > > > _________________________________________________________________________ > > > > > > > > > > > > > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > > > > > > > > > >> > consulting at freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://www.cluecon.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > > > > > > > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > >> > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > _________________________________________________________________________ > > > > > > > > > > > > > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > > > > > > > > > > > > > >> consulting at freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Official FreeSWITCH Sites > > > > > > > > > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://wiki.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://www.cluecon.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > > > > > > > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > >> UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > > > consulting at freeswitch.org > > > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > http://wiki.freeswitch.org > > > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/ca19122b/attachment-0001.html From jdiaz at coinfru.com Tue Jan 3 22:09:26 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Tue, 3 Jan 2012 20:09:26 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: <201201031318.10145.justlikeef@gmail.com> References: <201201031158.28269.justlikeef@gmail.com> <201201031318.10145.justlikeef@gmail.com> Message-ID: So at the end of the day we can not use this chan dongle if the author do not convert it to LGPL. And other option is to wait Giovanni build the mod_gsmopen to use it with huawei modems. Am i right? Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com _____ De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Rob Hutton Enviado el: Tuesday, January 03, 2012 19:18 Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] Posibility to build a module for chan dongle Thank you for the clarification Anthony!!! I just didn't want someone walking away with the absolute and finding out someone else that there are other opinions and getting pissed off because they got something else from what they were told here. What I was trying to say was that the leaders of any project have the absolute right to decide what they will and will not accept. There has been a great amount of work in making the licenses compatable with each other to address this exact situation. Many projects are multiple licensed at this point, and many projects that are MPL'd include GPL code as a dependency, although the opposite is largely not true, for many reasons. Modules under other licenses can be included and distributed without modifying either their license or the FreeSWITCH licensing as long as the terms of that license are adhered to. For this project, it needs to be an LGPL or other "free without restriction" license like OpenLDAP, etc. Not knowing the history, I took Steve's original comment to mean that this was a licensing incompatability. Sorry to ruffle feathers!!! On Tuesday 03 January 2012 12:43:43 Anthony Minessale wrote: > The reason MPL is incompatible with GPL is because the GPL is designed to > keep free software free by making it mandatory for its code and code > combined with it to fall under its license. > > MPL only requires changes to be shared with the author of the code they are > working with. MPL has no restriction on distribution or trying to keep the > resulting work free. > > We keep our code free because we want to and do not use the GPL to enforce > it. We intend for our software to be made into products and we put enough > work into the project that we feel those who benefit from our software will > come back and contribute to our community which is much stronger and able > to maintain the code than any private entity and anyone who can't see that > probably cannot be convinced otherwise. > > There is a severe lack of education about open source licenses. Its very > political and most of the people fighting about it don't even actually > understand the details. I'm not accusing anyone on this thread of not > understanding but it's important to realize that a large number of people > who even use the GPL just default to it assuming its the only choice for > open source and that it actually means open source. GPL tries very hard to > underscore that FREE means FREE as in liberty and not FREE as in "no cost". > > I choose not to license my software under GPL but that does not mean I do > not completely understand its intent or appreciate anything that has come > from its existence. > > Anyway, We can't use GPL code. We can't release any of FS as GPL or it > would break some of the other licenses, but you can ask the author of the > GPL code to make it LGPL which is compat since it allows linking without > contamination. > > You can also read the other code to see how it works and make your own > implementation. > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton wrote: > > > ** > > > > I'm not going to start a holy war on the list, nor spread fud, but: > > > > > > http://www.mozilla.org/MPL/MPL-1.1.html > > > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > > defined as "means a work which combines Covered Code or portions thereof > > with code not governed by the terms of this License", and other licenses > > are specifially allowed under these provisions under the MPL. > > > > > > The problem is with including MPL code in works licensed under GNU > > licence, as covered http://www.gnu.org/licenses/license-list.html, and as > > long as you are including MPL2 code and have any of the GNU2 series > > license, you are OK that way, also. > > > > > > From Wikipedia: > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > > have a provision that allows part of a program to offer the GNU GPL as an > > alternative choice, thereby allowing part of the program to have a > > GPL-compatible license. > > > > > > > > That doesn't mean that the project can't accept code, it means that that > > the project managers don't want to, for whatever reason, and that is > > completely up to them. But sending someone off with the misconception that > > it is not allowed by license is just going to lead to future problems for > > someone. > > > > > > Thanks, > > > > Rob > > > > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > > > Rob, > > > > > > > > > > no, you're wrong. > > > > > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > > > > > Please do not spread wrong info, that can lead to people wasting their > > time. > > > > > > > > > > -giovanni > > > > > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > > > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > > code is > > > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > > there are > > > > > > GPL2 projects included now in the form of automaticly built external > > > > > > libraries... > > > > > > > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > > > > > >> Hi, > > > > > > > > > > > >> > > > > > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your down > > > > > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. > > The > > > > > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > > Freeswitch. > > > > > > > > > > > >> > > > > > > > > > > > >> Steve > > > > > > > > > > > >> > > > > > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > > > > > >> > I was thinking if it could be possible to build a module like the > > > > > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > > > > > > > > > >> > calls throught GSM networks > > > > > > > > > > > >> > I can build an asterisk server for this, but i am working > > continously > > > > > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > > > > > > > > > >> > really work with freeswitch. I think that make this module could be > > > > > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > > > > > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > > > > > >> > Perhaps there is other module better than this made in freeswitch > > and > > > > > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle is > > that > > > > > > > > > > > >> > it is not needed to use any sound card or other device for this. > > Just > > > > > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > > > > > >> > The web page for the chan is this: > > > > > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > > many > > > > > > > > > > > >> > mistakes not knowing completely the posibilities with other modules > > > > > > > > > > > >> > that can do this job. > > > > > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > Josue Diaz Cruz > > > > > > > > > > > >> > > > > > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > > > > > >> > > > > > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > > > > > >> > > > > > > > > > > > > >> > C/ Balsicas 3 > > > > > > > > > > > >> > > > > > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > > > > > >> > > > > > > > > > > > > >> > www.coinfru.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > // > > > > > > > > > > > >> > > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > >> > > > _________________________________________________________________________ > > > > > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> > consulting at freeswitch.org > > > > > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > > > > > >> > http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > _________________________________________________________________________ > > > > > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> consulting at freeswitch.org > > > > > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> Official FreeSWITCH Sites > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> http://wiki.freeswitch.org > > > > > > > > > > > >> http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/228e0e38/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/228e0e38/attachment-0001.jpe From anthony.minessale at gmail.com Tue Jan 3 22:13:18 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jan 2012 13:13:18 -0600 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: <201201031158.28269.justlikeef@gmail.com> <201201031318.10145.justlikeef@gmail.com> Message-ID: Yes, those are the best 2 options. A third option is invite the author to make the FS module. We'd gladly supply assistance with questions etc. On Tue, Jan 3, 2012 at 1:09 PM, Josue Diaz Cruz wrote: > ** > So at the end of the day we can not use this chan dongle if the author do > not convert it to LGPL. And other option is to wait Giovanni build the > mod_gsmopen to use it with huawei modems. > > Am i right? > > * > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > jdiaz at coinfru.com > > > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.com > > > > > > * > > > ------------------------------ > *De:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *En nombre de *Rob Hutton > *Enviado el:* Tuesday, January 03, 2012 19:18 > *Para:* freeswitch-users at lists.freeswitch.org > *Asunto:* Re: [Freeswitch-users] Posibility to build a module for chan > dongle > > Thank you for the clarification Anthony!!! I just didn't want someone > walking away with the absolute and finding out someone else that there are > other opinions and getting pissed off because they got something else from > what they were told here. > > > What I was trying to say was that the leaders of any project have the > absolute right to decide what they will and will not accept. > > > There has been a great amount of work in making the licenses compatable > with each other to address this exact situation. Many projects are multiple > licensed at this point, and many projects that are MPL'd include GPL code > as a dependency, although the opposite is largely not true, for many > reasons. Modules under other licenses can be included and distributed > without modifying either their license or the FreeSWITCH licensing as long > as the terms of that license are adhered to. > > > For this project, it needs to be an LGPL or other "free without > restriction" license like OpenLDAP, etc. Not knowing the history, I took > Steve's original comment to mean that this was a licensing incompatability. > Sorry to ruffle feathers!!! > > > > On Tuesday 03 January 2012 12:43:43 Anthony Minessale wrote: > > > The reason MPL is incompatible with GPL is because the GPL is designed to > > > keep free software free by making it mandatory for its code and code > > > combined with it to fall under its license. > > > > > > MPL only requires changes to be shared with the author of the code they > are > > > working with. MPL has no restriction on distribution or trying to keep > the > > > resulting work free. > > > > > > We keep our code free because we want to and do not use the GPL to > enforce > > > it. We intend for our software to be made into products and we put enough > > > work into the project that we feel those who benefit from our software > will > > > come back and contribute to our community which is much stronger and able > > > to maintain the code than any private entity and anyone who can't see > that > > > probably cannot be convinced otherwise. > > > > > > There is a severe lack of education about open source licenses. Its very > > > political and most of the people fighting about it don't even actually > > > understand the details. I'm not accusing anyone on this thread of not > > > understanding but it's important to realize that a large number of people > > > who even use the GPL just default to it assuming its the only choice for > > > open source and that it actually means open source. GPL tries very hard > to > > > underscore that FREE means FREE as in liberty and not FREE as in "no > cost". > > > > > > I choose not to license my software under GPL but that does not mean I do > > > not completely understand its intent or appreciate anything that has come > > > from its existence. > > > > > > Anyway, We can't use GPL code. We can't release any of FS as GPL or it > > > would break some of the other licenses, but you can ask the author of the > > > GPL code to make it LGPL which is compat since it allows linking without > > > contamination. > > > > > > You can also read the other code to see how it works and make your own > > > implementation. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton > wrote: > > > > > > > ** > > > > > > > > I'm not going to start a holy war on the list, nor spread fud, but: > > > > > > > > > > > > http://www.mozilla.org/MPL/MPL-1.1.html > > > > > > > > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > > > > > > > > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > > > > defined as "means a work which combines Covered Code or portions > thereof > > > > with code not governed by the terms of this License", and other > licenses > > > > are specifially allowed under these provisions under the MPL. > > > > > > > > > > > > The problem is with including MPL code in works licensed under GNU > > > > licence, as covered http://www.gnu.org/licenses/license-list.html, > and as > > > > long as you are including MPL2 code and have any of the GNU2 series > > > > license, you are OK that way, also. > > > > > > > > > > > > From Wikipedia: > > > > > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > > > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > > > > have a provision that allows part of a program to offer the GNU GPL as > an > > > > alternative choice, thereby allowing part of the program to have a > > > > GPL-compatible license. > > > > > > > > > > > > > > > > That doesn't mean that the project can't accept code, it means that > that > > > > the project managers don't want to, for whatever reason, and that is > > > > completely up to them. But sending someone off with the misconception > that > > > > it is not allowed by license is just going to lead to future problems > for > > > > someone. > > > > > > > > > > > > Thanks, > > > > > > > > Rob > > > > > > > > > > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > > > > > > > Rob, > > > > > > > > > > > > > > > > > > no, you're wrong. > > > > > > > > > > > > > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > > > > > > > > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > > > > > > > > > > > > > Please do not spread wrong info, that can lead to people wasting > their > > > > time. > > > > > > > > > > > > > > > > > > -giovanni > > > > > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton > wrote: > > > > > > > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > > > > code is > > > > > > > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > > > > there are > > > > > > > > > > GPL2 projects included now in the form of automaticly built > external > > > > > > > > > > libraries... > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > > > > > > > > > > > > > >> Hi, > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your > down > > > > > > > > > > > > > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with > Freeswitch. > > > > The > > > > > > > > > > > > > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > > > > Freeswitch. > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Steve > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > > > > > > > > > > > > > >> > I was thinking if it could be possible to build a module like > the > > > > > > > > > > > > > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to > place > > > > > > > > > > > > > > > > > > > >> > calls throught GSM networks > > > > > > > > > > > > > > > > > > > >> > I can build an asterisk server for this, but i am working > > > > continously > > > > > > > > > > > > > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I > enjoy > > > > > > > > > > > > > > > > > > > >> > really work with freeswitch. I think that make this module > could be > > > > > > > > > > > > > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it > to > > > > > > > > > > > > > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > > > > > > > > > > > > > >> > Perhaps there is other module better than this made in > freeswitch > > > > and > > > > > > > > > > > > > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle > is > > > > that > > > > > > > > > > > > > > > > > > > >> > it is not needed to use any sound card or other device for this. > > > > Just > > > > > > > > > > > > > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > > > > > > > > > > > > > >> > The web page for the chan is this: > > > > > > > > > > > > > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > > > > > > > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > > > > > > > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > > > > many > > > > > > > > > > > > > > > > > > > >> > mistakes not knowing completely the posibilities with other > modules > > > > > > > > > > > > > > > > > > > >> > that can do this job. > > > > > > > > > > > > > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > > > > > > > > > > > > > >> > * > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Josue Diaz Cruz > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > C/ Balsicas 3 > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > www.coinfru.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > // > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > * > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > >> > > > > > > _________________________________________________________________________ > > > > > > > > > > > > > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > > > > > > > > > >> > consulting at freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://www.cluecon.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > > > > > > > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > >> > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > _________________________________________________________________________ > > > > > > > > > > > > > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > > > > > > > > > > > > > >> consulting at freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> Official FreeSWITCH Sites > > > > > > > > > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://wiki.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://www.cluecon.com > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > > > > > > > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > >> UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > > > consulting at freeswitch.org > > > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > http://wiki.freeswitch.org > > > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE: > > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/15c6cba3/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/15c6cba3/attachment-0001.jpe From jdiaz at coinfru.com Tue Jan 3 22:22:02 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Tue, 3 Jan 2012 20:22:02 +0100 Subject: [Freeswitch-users] Posibility to build a module for chan dongle In-Reply-To: References: <201201031158.28269.justlikeef@gmail.com><201201031318.10145.justlikeef@gmail.com> Message-ID: Ok i will apply for the last one option. It could the better to tell the author if he would like to make it for Freeswitch. Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com _____ De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Anthony Minessale Enviado el: Tuesday, January 03, 2012 20:13 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] Posibility to build a module for chan dongle Yes, those are the best 2 options. A third option is invite the author to make the FS module. We'd gladly supply assistance with questions etc. On Tue, Jan 3, 2012 at 1:09 PM, Josue Diaz Cruz wrote: So at the end of the day we can not use this chan dongle if the author do not convert it to LGPL. And other option is to wait Giovanni build the mod_gsmopen to use it with huawei modems. Am i right? Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com _____ De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Rob Hutton Enviado el: Tuesday, January 03, 2012 19:18 Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] Posibility to build a module for chan dongle Thank you for the clarification Anthony!!! I just didn't want someone walking away with the absolute and finding out someone else that there are other opinions and getting pissed off because they got something else from what they were told here. What I was trying to say was that the leaders of any project have the absolute right to decide what they will and will not accept. There has been a great amount of work in making the licenses compatable with each other to address this exact situation. Many projects are multiple licensed at this point, and many projects that are MPL'd include GPL code as a dependency, although the opposite is largely not true, for many reasons. Modules under other licenses can be included and distributed without modifying either their license or the FreeSWITCH licensing as long as the terms of that license are adhered to. For this project, it needs to be an LGPL or other "free without restriction" license like OpenLDAP, etc. Not knowing the history, I took Steve's original comment to mean that this was a licensing incompatability. Sorry to ruffle feathers!!! On Tuesday 03 January 2012 12:43:43 Anthony Minessale wrote: > The reason MPL is incompatible with GPL is because the GPL is designed to > keep free software free by making it mandatory for its code and code > combined with it to fall under its license. > > MPL only requires changes to be shared with the author of the code they are > working with. MPL has no restriction on distribution or trying to keep the > resulting work free. > > We keep our code free because we want to and do not use the GPL to enforce > it. We intend for our software to be made into products and we put enough > work into the project that we feel those who benefit from our software will > come back and contribute to our community which is much stronger and able > to maintain the code than any private entity and anyone who can't see that > probably cannot be convinced otherwise. > > There is a severe lack of education about open source licenses. Its very > political and most of the people fighting about it don't even actually > understand the details. I'm not accusing anyone on this thread of not > understanding but it's important to realize that a large number of people > who even use the GPL just default to it assuming its the only choice for > open source and that it actually means open source. GPL tries very hard to > underscore that FREE means FREE as in liberty and not FREE as in "no cost". > > I choose not to license my software under GPL but that does not mean I do > not completely understand its intent or appreciate anything that has come > from its existence. > > Anyway, We can't use GPL code. We can't release any of FS as GPL or it > would break some of the other licenses, but you can ask the author of the > GPL code to make it LGPL which is compat since it allows linking without > contamination. > > You can also read the other code to see how it works and make your own > implementation. > > > > > > > > > > > > > > On Tue, Jan 3, 2012 at 10:58 AM, Rob Hutton wrote: > > > ** > > > > I'm not going to start a holy war on the list, nor spread fud, but: > > > > > > http://www.mozilla.org/MPL/MPL-1.1.html > > > > http://mpl.mozilla.org/wp-content/uploads/2011/08/MPL-RC1-typography.html > > > > > > In section 3.3 of MPL 1.1 and section 3.7 of MPL 2, "Larger Works" are > > defined as "means a work which combines Covered Code or portions thereof > > with code not governed by the terms of this License", and other licenses > > are specifially allowed under these provisions under the MPL. > > > > > > The problem is with including MPL code in works licensed under GNU > > licence, as covered http://www.gnu.org/licenses/license-list.html, and as > > long as you are including MPL2 code and have any of the GNU2 series > > license, you are OK that way, also. > > > > > > From Wikipedia: > > > > The MPL is GPL-incompatible because the GPL module cannot be legally > > linked with an MPL module. However, versions of the MPL such as MPL 1.1 > > have a provision that allows part of a program to offer the GNU GPL as an > > alternative choice, thereby allowing part of the program to have a > > GPL-compatible license. > > > > > > > > That doesn't mean that the project can't accept code, it means that that > > the project managers don't want to, for whatever reason, and that is > > completely up to them. But sending someone off with the misconception that > > it is not allowed by license is just going to lead to future problems for > > someone. > > > > > > Thanks, > > > > Rob > > > > > > On Tuesday 03 January 2012 11:07:17 Giovanni Maruzzelli wrote: > > > > > Rob, > > > > > > > > > > no, you're wrong. > > > > > > > > > > You cannot include GPLd code in FreeSWITCH. > > > > > > > > > > That has been clarified multiple times in the mailing list too. > > > > > > > > > > Please do not spread wrong info, that can lead to people wasting their > > time. > > > > > > > > > > -giovanni > > > > > > > > > > On Tue, Jan 3, 2012 at 4:57 PM, Rob Hutton wrote: > > > > > > While MPL is not compatible with GPLd code, I don't think that GPLd > > code is > > > > > > compatible with MPL, as MPL is slightly more restrictive. In fact, > > there are > > > > > > GPL2 projects included now in the form of automaticly built external > > > > > > libraries... > > > > > > > > > > > > > > > > > > On Tuesday 03 January 2012 10:25:59 Steve Underwood wrote: > > > > > > > > > > > >> Hi, > > > > > > > > > > > >> > > > > > > > > > > > >> Whilst you can use the code from chan_dongle privately, for your down > > > > > > > > > > > >> purposes, with Freeswitch, it cannot be distributed with Freeswitch. > > The > > > > > > > > > > > >> licence for chan_dongle is GPL v2. This is incompatible with > > Freeswitch. > > > > > > > > > > > >> > > > > > > > > > > > >> Steve > > > > > > > > > > > >> > > > > > > > > > > > >> On 01/03/2012 11:03 PM, Josue Diaz Cruz wrote: > > > > > > > > > > > >> > I was thinking if it could be possible to build a module like the > > > > > > > > > > > >> > asterisk chan dongle. Uses the huawey USB GMS/UTMS modem to place > > > > > > > > > > > >> > calls throught GSM networks > > > > > > > > > > > >> > I can build an asterisk server for this, but i am working > > continously > > > > > > > > > > > >> > with freeswitch and i have not too much knowledge of c/c++. I enjoy > > > > > > > > > > > >> > really work with freeswitch. I think that make this module could be > > > > > > > > > > > >> > nice to use it in a low cost sip to GSM gateway. We can use it to > > > > > > > > > > > >> > develop pbx systems with gsm, or trunking. > > > > > > > > > > > >> > Perhaps there is other module better than this made in freeswitch > > and > > > > > > > > > > > >> > i am asking an stupid thing. What i like from this chan dongle is > > that > > > > > > > > > > > >> > it is not needed to use any sound card or other device for this. > > Just > > > > > > > > > > > >> > only as a module cause the huawey have the voice mode activable. > > > > > > > > > > > >> > The web page for the chan is this: > > > > > > > > > > > >> > http://wiki.e1550.mobi/doku.php?id=installation > > > > > > > > > > > >> > It is open source. so we can use the sources without any issue. > > > > > > > > > > > >> > Sorry me if the information is not really exaustive or if i maid > > many > > > > > > > > > > > >> > mistakes not knowing completely the posibilities with other modules > > > > > > > > > > > >> > that can do this job. > > > > > > > > > > > >> > By the way, sorry me for my english. It is not really good. > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > Josue Diaz Cruz > > > > > > > > > > > >> > > > > > > > > > > > > >> > Departamento Tecnico y Soporte > > > > > > > > > > > >> > > > > > > > > > > > > >> > /jdiaz at coinfru.com/ > > > > > > > > > > > >> > > > > > > > > > > > > >> > C/ Balsicas 3 > > > > > > > > > > > >> > > > > > > > > > > > > >> > Alquerias | 30580 | Murcia > > > > > > > > > > > >> > > > > > > > > > > > > >> > www.coinfru.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > // > > > > > > > > > > > >> > > > > > > > > > > > > >> > * > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > >> > > > _________________________________________________________________________ > > > > > > > > > > > >> > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> > consulting at freeswitch.org > > > > > > > > > > > >> > http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > > > > > > > > > > > > >> > Official FreeSWITCH Sites > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > http://wiki.freeswitch.org > > > > > > > > > > > >> > http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > > >> > FreeSWITCH-users mailing list > > > > > > > > > > > >> > FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> > http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > _________________________________________________________________________ > > > > > > > > > > > >> Professional FreeSWITCH Consulting Services: > > > > > > > > > > > >> consulting at freeswitch.org > > > > > > > > > > > >> http://www.freeswitchsolutions.com > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> > > > > > > > > > > > >> Official FreeSWITCH Sites > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> http://wiki.freeswitch.org > > > > > > > > > > > >> http://www.cluecon.com > > > > > > > > > > > >> > > > > > > > > > > > >> FreeSWITCH-users mailing list > > > > > > > > > > > >> FreeSWITCH-users at lists.freeswitch.org > > > > > > > > > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > >> UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > >> http://www.freeswitch.org > > > > > > > > > > > >> > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch 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Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/e28be13f/attachment-0003.jpe From msc at freeswitch.org Tue Jan 3 23:31:17 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 12:31:17 -0800 Subject: [Freeswitch-users] conf count In-Reply-To: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: Did you figure this one out yet? -MC On Sun, Dec 25, 2011 at 8:20 AM, Rodney wrote: > > I can select the conf-alone sound in caller controls and even set up a > bind digit for caller count once i am already in the conference using the > > Conference Announce Count Inline extension. > > but what i cant figure out is how to speak the caller count when entering > the conference room. can anyone help with this? currently with conf-alone > it only speaks the file when i am the only one left or there isnt anyone > else in the room when i get there. i would like it to tell me the total > callers of the room i am heading into. > > thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/79f5d523/attachment.html From msc at freeswitch.org Tue Jan 3 23:44:34 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 12:44:34 -0800 Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: <1324703154.9585.YahooMailClassic@web110808.mail.gq1.yahoo.com> References: <1324703154.9585.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: > but the station i need is this > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > and it is not there. I would appreciate your help > Bummer. ASF is a proprietary Microsoft technology, so I would contact the people using it and let them know that they are limiting their potential audience by using a locked-down container format. If you *really* want this then you can probably hire someone to do a workaround. My guess is that you'll need to pay a fair amount because the people with the requisite skills to do this will be in high demand. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/7ba532dc/attachment.html From justlikeef at gmail.com Tue Jan 3 23:55:34 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 3 Jan 2012 15:55:34 -0500 Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: References: <1324703154.9585.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: <201201031555.34895.justlikeef@gmail.com> If you can get permission to rebroadcast, you could set up a machine to transcode it into a format that is supportable... On Tuesday 03 January 2012 15:44:34 Michael Collins wrote: > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: > > > but the station i need is this > > > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > > > and it is not there. I would appreciate your help > > > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/4053ab3c/attachment.html From msc at freeswitch.org Wed Jan 4 00:33:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:33:23 -0800 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: References: Message-ID: I don't believe you can reliable put these tones in synch. I've never heard of any telephony system that does so, although I admit my experience is limited to legacy PBXes and FreeSWITCH. -MC On Fri, Dec 23, 2011 at 8:47 PM, wrote: > Hi, > > > I'm not sure that I understand what the issue is here. What two "tones" > > are > > not in sync? You said "dial tone" but I'm pretty sure that's not what you > > mean. Are you talking about the ringback tone that the caller hears? > > Exactly Michael, I was talking about the ringback tone. > I didn't know the correct expression and used a translation engine... > > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/c978f5d8/attachment.html From nasida at live.ru Wed Jan 4 00:40:37 2012 From: nasida at live.ru (Yuriy Nasida) Date: Wed, 4 Jan 2012 01:40:37 +0400 Subject: [Freeswitch-users] timeout for curl Message-ID: Hello FS users, I do some curl request from my dialplan. It looks like this:If cname provider doesn't answer for long time, I have issue because FS doesn't process a call.whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, FS must continue the logic of dialplan. I read about mod_cidlookup but it looks like that this module have same issuehttp://wiki.freeswitch.org/wiki/Mod_cidlookup Please advise Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/503b2448/attachment.html From msc at freeswitch.org Wed Jan 4 00:46:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:46:31 -0800 Subject: [Freeswitch-users] Silence ESL event In-Reply-To: <-3489375824995432417@unknownmsgid> References: <-3489375824995432417@unknownmsgid> Message-ID: You might be able to produce the same behavior with the "wait_for_silence" dp application. -MC On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar wrote: > Hi all, > > is there any way to receive a silence-detected event through ESL? > > The use case is to hangup the call in case nothing interesting is > going on on the channel. > > Sent from my mobile device > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/5f6c43a2/attachment-0001.html From msc at freeswitch.org Wed Jan 4 00:48:36 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:48:36 -0800 Subject: [Freeswitch-users] Continue after lua execute bridge In-Reply-To: References: Message-ID: No, you cannot do it this way. If you call the bridge from within your Lua script then the Lua script will be "active" for the duration of the bridge. If you transfer the call out to an XML dialplan and then perform the bridge then you'll be fine. -MC On Sat, Dec 24, 2011 at 7:01 AM, David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello Gang, > > I'm executing from a lua script: > > session:execute("bridge","sofia/gateway/" .. gw .. "/" .. out_number .. "") > > But I want my script to END if the bridge is succesfull, i.e.: Ringing, > Answer, Busy... > I've been testing, but it doesn't seem like the script is aware of the > result of the bridge until AFTER it has been release by the "bridge" > application... > > Is there ANY way to end my script in any of the scenarious mentioned > before? (Ringing, Answer, Busy) > > Thanks a lot > > oh, and MERRY CHRISTMAS or Happy Holidays, or whatever suits you! ;) > > David > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/5756358f/attachment.html From msc at freeswitch.org Wed Jan 4 00:51:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:51:49 -0800 Subject: [Freeswitch-users] Can not receive Events In-Reply-To: References: Message-ID: Try this same thing but use fs_cli. If it works with fs_cli then you know there's a problem with your terminal or something like that. If it doesn't work with fs_cli then you know you've got something really strange going on. Launch fs_cli. Type: /log 0 Type: /events plain all Then do your originate and you should see all sorts of events coming flying across the screen. -MC On Wed, Dec 28, 2011 at 5:19 AM, Arnuld Uttre (Phonologies) < arnuld at phonologies.com> wrote: > I am running freeswitch. I have connected to it suing telnet on event > socket. Through the same machine, using a different x-terminal, I have > subscribed to: > > event plain all > > Content-Type: command/reply > Reply-Text: +OK event listener enabled plain > > > I am able to make calls using "api originate" command and got the call > fine on my phone. I got +OK uuid-here too but I did not receive any > events. I event tried event plain CHANNEL_DESTROY from different terminal > but still no luck. I am usng freeswitch 1.0.7 on CentOS 5.5 with my own > application configured in autoload_configs/myapp.conf.xml > > >From where I should start solving the problem ? Here are the contents of > event_socket.conf.xml : > > > > > > > > > > > > > > > -- > Arnuld Uttre > Systems Software Engineer > > arnuld at Phonologies.COM > http://www.phonologies.com > > Phonologies (India) Private Limited > West Wing, Marri Deep, M. C. H. No. 12-5-4, > Lallaguda, Secunderabad 500017, INDIA. > Ph:+91-40-2701 8993 / 36 > Fax:+91-40-2701 8992 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/4587aed1/attachment.html From msc at freeswitch.org Wed Jan 4 00:54:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:54:45 -0800 Subject: [Freeswitch-users] Connecting to Event Socket In-Reply-To: References: Message-ID: fs_cli is a C program that uses ESL to abstract away a lot of the grunt work. If you need to listen for events then you should be using ESL, otherwise you're wasting time and energy reinventing the wheel, and unless you know more about the event socket than Anthony does then you're wheel will not be as good as his. :) -MC On Fri, Dec 30, 2011 at 3:19 AM, Arnuld Uttre (Phonologies) < arnuld at phonologies.com> wrote: > I mostly connect to event-socket using telnet and then subscribe to > receive events. I want to connect to event-socket using a simple TCP > client from within a C Program. Is it a goo idea or there is a better way > ? > > > > > -- > Arnuld Uttre > Systems Software Engineer > > arnuld at Phonologies.COM > http://www.phonologies.com > > Phonologies (India) Private Limited > West Wing, Marri Deep, M. C. H. No. 12-5-4, > Lallaguda, Secunderabad 500017, INDIA. > Ph:+91-40-2701 8993 / 36 > Fax:+91-40-2701 8992 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/43f5a8b3/attachment.html From msc at freeswitch.org Wed Jan 4 00:57:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:57:52 -0800 Subject: [Freeswitch-users] freeswitch b-leg data logging In-Reply-To: <1325381171.46859.YahooMailClassic@web110803.mail.gq1.yahoo.com> References: <1325381171.46859.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: Need more information. Pastebin your xml_cdr.conf.xml file. Also, go to fs_cli and do "reload mod_xml_cdr" and capture the output, also into the pastebin. There should be some information about why you're not getting b-legs. Lastly, you should probably pastebin the debug console output of a simple call that you know for a fact has a b-leg. -MC On Sat, Dec 31, 2011 at 5:26 PM, Sherif Omran wrote: > > > Hi > > I have a problem with freeswitch and don't get the b-leg information > although it is enabled in autoload_configs/xml_cdr.conf.xml file and > would appreciate if you could help. > > waiting your reply > > kind regards, > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/3fb0caac/attachment.html From msc at freeswitch.org Wed Jan 4 00:59:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 13:59:05 -0800 Subject: [Freeswitch-users] freeswitch as MGW In-Reply-To: References: Message-ID: Well, if you need one box for control and it doesn't need to do anything with the media then you might want to check out OpenSIPS and/or Kamailio for box1. Box2 would still be FreeSWITCH. -MC On Sun, Jan 1, 2012 at 8:53 AM, Juan Pablo L wrote: > Hello all, i was wondering if its possible the make a configuration in > such a way that you have one box, box1, with a freeswitch instance > to deal with all the sip related tasks and a next box, box2, that > deals with all the audio related tasks being controlled by box1 by > means of > MGCP, or any other protocol for that purpose, acting as a MGW (that > could also be used by other nodes in the network) ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/8f0ac653/attachment-0001.html From msc at freeswitch.org Wed Jan 4 01:02:03 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 14:02:03 -0800 Subject: [Freeswitch-users] Mod_SPANDSP In-Reply-To: References: Message-ID: You may want to check out the "execute_on_fax*" channel variables: http://wiki.freeswitch.org/wiki/Mod_spandsp#Execute_based_on_fax_session_outcome These allow for some elegant solutions. -MC On Mon, Jan 2, 2012 at 5:46 AM, Anto wrote: > hello > > I'm adapting the files are in > http://wiki.freeswitch.org/wiki/Mod_spandsp to make faxing my > business. I wanted to know whether to send the fax, an error occurs > (destination busy, etc.), this is still retrying successive times or > is there a way to handle this. I've been looking fax_result_code and > fax_result_text, to make a correct treatment but have not found > information on them. Does anyone have any examples or can guide me? > Thank you very much > > Best Regards > Anto > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/a7fb1263/attachment.html From msc at freeswitch.org Wed Jan 4 01:04:40 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 14:04:40 -0800 Subject: [Freeswitch-users] how do I play music on hold for A-leg while IVRing with B-leg In-Reply-To: <1325267973.42182.YahooMailNeo@web39704.mail.mud.yahoo.com> References: <1325267973.42182.YahooMailNeo@web39704.mail.mud.yahoo.com> Message-ID: Can you pastebin the dialplan and Lua script that you're using? The gang here will be happy to take a look. -MC On Fri, Dec 30, 2011 at 9:59 AM, Greg Buzzard wrote: > Hi All, new to FreeSwitch, great software! Read earlier posts, bought FS > book and am about half-way through it and still have a question. > > Problem context: I'm trying to do a form of call screening for home use > using a Lua script. I record a "self-intro" message from calling party > (A-leg), would then like to play hold music for A-leg while creating new > session and doing IVR dance with called party (B-leg). IVR objective is to > see if B-leg wishes to accept, add caller-ID to whitelist or blacklist, > and/or send to voicemail, etc. I have all of this working except the > ability to play hold music to A-leg before either bridging them, sending > them to voicemail or playing a message and dropping them. > > Specific problem: I don't have my head wrapped around the "clean" > approach to addressing this problem. I.e., seems like I may want to "park" > the A-leg (with music), while I do the IVR with B-leg. Then either: (1) > unpark A-leg and bridge to B-leg or (2) hang-up B-leg, unpark A-leg and > send to VM or (3) other embellishments where I add caller-ID (if it exists) > to either white or black lists and/or play recording to telemarketers to > leave me alone (with no VM), etc. If this is a good approach, getting a > few pointers on key commands/apps to use would be great. > > FWIW, my first simple-minded approach to transfer caller (A-leg) to > hold-music extension while connecting/IVRing with B-leg didn't work as > execution of the transfer was held until the IVRing was done). Similar to > what I saw in some earlier postings. If I get this figured out, I'm happy > to summarize and post useful snippets to the email archive, wiki page or > wherever, when done. > > -g > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/1dd8e476/attachment.html From devel at omninet.eu Wed Jan 4 01:07:46 2012 From: devel at omninet.eu (Anestis Mavro) Date: Wed, 4 Jan 2012 00:07:46 +0200 Subject: [Freeswitch-users] codec-ms for specific gateway Message-ID: <8D1B667979834E9C8350B83210BFFF47@omni1.local> Hi, I need to set a different packetization time for one specific gateway, keeping the default settings for other gateways. I have tried the parameter codec-ms (which is for sofia.conf.xml) in the settings for the specific gateway, but it doesn't work. Is there any other way to achieve this? Thank you Anestis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/ccbe6ac3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Anestis Mavrofillidis (omninet at omninet.gr).vcf Type: text/x-vcard Size: 244 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/ccbe6ac3/attachment.vcf From msc at freeswitch.org Wed Jan 4 01:20:47 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 14:20:47 -0800 Subject: [Freeswitch-users] Voicemail - see if user left a message? In-Reply-To: References: Message-ID: Avi, Are you looking at the XML CDR for the call leg that goes to voicemail? If so, check out the billsec value. The lower the value, the less likely they left a message. If it's zero then they definitely did not leave a message. I'd ask Tony for confirmation, but based on my experiments (with the "Local_Extension") I'd wager that the billsec variable only goes above zero if the voicemail app gets to the point where it actually records something. As a rule of thumb, I'd say anything less than four seconds should be considered "they did not leave a voicemail." Do some experimenting and let us know what you come up with. -MC On Mon, Jan 2, 2012 at 4:33 PM, Avi Marcus wrote: > Can the xml_cdrs tell me if the user actually left a voicemail, rather > than merely that VM picked up? I don't see where. > I know it's on ESL, and it hits the voicemail email code.. what's the > easiest way to integrate it with the CDRs? preferably with the xml_cdrs? > Maybe a patch to the xml_cdr mod? Or the easiest is to have mod_voicemail > set a channel variable once a voicemail gets marked as saved or the like? > > Is there currently a solution for this that I missed? > > Thanks, > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/e58a08c0/attachment.html From msc at freeswitch.org Wed Jan 4 01:30:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 14:30:31 -0800 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Can you expand upon this question a bit? I'm curious if there's a less hackish way of doing what you want to do. Under what circumstances do you need to add the leg variables? Also, can you give us the big picture? What's the problem you're solving? -MC On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese < brian.wiese.freeswitch at gmail.com> wrote: > Hi Everyone. > > I thought I read somewhere that this was possible, but I can't find it > now... > > I need a way to find-and-replace within a variable. So, for example, > I want to take variables that have values like this: > > 123abc123abc > abc123abc123abc > > ...and do a find/replace of the "abc" with "xyz" so the variables > would now return: > > 123xyz123xyz > xyz123xyz123xyz > > The use case I've run into is that I need to add leg variables to > group_call. In my case, group_call can return any number of members, > so I figured I would just replace the first "[" with > "[variable-I-want-to-set=...". > > Thanks for the help! > > ~Brian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/d19d2935/attachment-0001.html From notlikeme75 at yahoo.com Wed Jan 4 01:30:35 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 3 Jan 2012 14:30:35 -0800 (PST) Subject: [Freeswitch-users] conf count when entering room In-Reply-To: References: Message-ID: <1325629835.29228.YahooMailNeo@web65313.mail.ac2.yahoo.com> MC, I have not figured out how to get the conference count upon entering the room. I would imaging loading the?Conference Announce Count extension at the same time as dropping into the conference. I am unsure of how to do this so it gets passed the conference number i am entering. I would imagine this would be a good start: ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, January 3, 2012 4:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 20 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: conf count (Michael Collins) ? 2. Re: Radio stream binding ? (Michael Collins) ? 3. Re: Radio stream binding ? (Rob Hutton) ? 4. Re: int/ext dial tones not synchronous (Michael Collins) ? 5. timeout for curl (Yuriy Nasida) ? 6. Re: Silence ESL event (Michael Collins) Did you figure this one out yet? -MC On Sun, Dec 25, 2011 at 8:20 AM, Rodney wrote: > >I can select the conf-alone sound in caller controls and even set up a bind digit for caller count once i am already in the conference using the > >Conference Announce Count Inline? extension. > >but what i cant figure out is how to speak the caller count when entering the conference room.? can anyone help with this? currently with conf-alone it only speaks the file when i am the only one left or there isnt anyone else in the room when i get there. i would like it to tell me the total callers of the room i am heading into. > >thanks. > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: but the station i need is this > >mmsh://live.sis.gov.eg/live?MSWMExt=.asf > >and it is not there. I would appreciate your help > Bummer.? ASF is a proprietary Microsoft technology, so I would contact the people using it and let them know that they are limiting their potential audience by using a locked-down container format.? If you *really* want this then you can probably hire someone to do a workaround. My guess is that you'll need to pay a fair amount because the people with the requisite skills to do this will be in high demand. -MC If you can get permission to rebroadcast, you could set up a machine to transcode it into a format that is supportable... On Tuesday 03 January 2012 15:44:34 Michael Collins wrote: > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: > > > but the station i need is this > > > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > > > and it is not there. I would appreciate your help > > > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > I don't believe you can reliable put these tones in synch. I've never heard of any telephony system that does so, although I admit my experience is limited to legacy PBXes and FreeSWITCH. -MC On Fri, Dec 23, 2011 at 8:47 PM, wrote: Hi, > >> I'm not sure that I understand what the issue is here. What two "tones" >> are >> not in sync? You said "dial tone" but I'm pretty sure that's not what you >> mean. Are you talking about the ringback tone that the caller hears? > >Exactly Michael, I was talking about the ringback tone. >I didn't know the correct expression and used a translation engine... > >Georg > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Hello FS users, I do some curl request from my dialplan. It looks like this: If cname provider doesn't answer for long time, I have issue because FS doesn't process a call. whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, ?FS must continue the logic of dialplan. I read about ?mod_cidlookup but it looks like that this module have same issue http://wiki.freeswitch.org/wiki/Mod_cidlookup? Please advise Thanks. You might be able to produce the same behavior with the "wait_for_silence" dp application. -MC On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar wrote: Hi all, > >is there any way to receive a silence-detected event through ESL? > >The use case is to hangup the call in case nothing interesting is >going on on the channel. > >Sent from my mobile device > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/becfea44/attachment-0001.html From notlikeme75 at yahoo.com Wed Jan 4 01:32:11 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 3 Jan 2012 14:32:11 -0800 (PST) Subject: [Freeswitch-users] conf count when entering room In-Reply-To: <1325629835.29228.YahooMailNeo@web65313.mail.ac2.yahoo.com> References: <1325629835.29228.YahooMailNeo@web65313.mail.ac2.yahoo.com> Message-ID: <1325629931.86169.YahooMailNeo@web65311.mail.ac2.yahoo.com> sorry sent too soon. i want to bridge conf count extension and the 301 and then sched hangup bleg?? ________________________________ From: Rodney To: "freeswitch-users at lists.freeswitch.org" Sent: Tuesday, January 3, 2012 5:30 PM Subject: conf count when entering room MC, I have not figured out how to get the conference count upon entering the room. I would imaging loading the?Conference Announce Count extension at the same time as dropping into the conference. I am unsure of how to do this so it gets passed the conference number i am entering. I would imagine this would be a good start: ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, January 3, 2012 4:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 20 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: conf count (Michael Collins) ? 2. Re: Radio stream binding ? (Michael Collins) ? 3. Re: Radio stream binding ? (Rob Hutton) ? 4. Re: int/ext dial tones not synchronous (Michael Collins) ? 5. timeout for curl (Yuriy Nasida) ? 6. Re: Silence ESL event (Michael Collins) Did you figure this one out yet? -MC On Sun, Dec 25, 2011 at 8:20 AM, Rodney wrote: > >I can select the conf-alone sound in caller controls and even set up a bind digit for caller count once i am already in the conference using the > >Conference Announce Count Inline? extension. > >but what i cant figure out is how to speak the caller count when entering the conference room.? can anyone help with this? currently with conf-alone it only speaks the file when i am the only one left or there isnt anyone else in the room when i get there. i would like it to tell me the total callers of the room i am heading into. > >thanks. > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: but the station i need is this > >mmsh://live.sis.gov.eg/live?MSWMExt=.asf > >and it is not there. I would appreciate your help > Bummer.? ASF is a proprietary Microsoft technology, so I would contact the people using it and let them know that they are limiting their potential audience by using a locked-down container format.? If you *really* want this then you can probably hire someone to do a workaround. My guess is that you'll need to pay a fair amount because the people with the requisite skills to do this will be in high demand. -MC If you can get permission to rebroadcast, you could set up a machine to transcode it into a format that is supportable... On Tuesday 03 January 2012 15:44:34 Michael Collins wrote: > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: > > > but the station i need is this > > > > mmsh://live.sis.gov.eg/live?MSWMExt=.asf > > > > and it is not there. I would appreciate your help > > > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > I don't believe you can reliable put these tones in synch. I've never heard of any telephony system that does so, although I admit my experience is limited to legacy PBXes and FreeSWITCH. -MC On Fri, Dec 23, 2011 at 8:47 PM, wrote: Hi, > >> I'm not sure that I understand what the issue is here. What two "tones" >> are >> not in sync? You said "dial tone" but I'm pretty sure that's not what you >> mean. Are you talking about the ringback tone that the caller hears? > >Exactly Michael, I was talking about the ringback tone. >I didn't know the correct expression and used a translation engine... > >Georg > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Hello FS users, I do some curl request from my dialplan. It looks like this: If cname provider doesn't answer for long time, I have issue because FS doesn't process a call. whether there are any method to add some timeout for curl (3 sec for example) ? Thus, if cname provider doesn't answer for 3 sec, ?FS must continue the logic of dialplan. I read about ?mod_cidlookup but it looks like that this module have same issue http://wiki.freeswitch.org/wiki/Mod_cidlookup? Please advise Thanks. You might be able to produce the same behavior with the "wait_for_silence" dp application. -MC On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar wrote: Hi all, > >is there any way to receive a silence-detected event through ESL? > >The use case is to hangup the call in case nothing interesting is >going on on the channel. > >Sent from my mobile device > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/7847734d/attachment-0001.html From avi at avimarcus.net Wed Jan 4 01:45:09 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jan 2012 00:45:09 +0200 Subject: [Freeswitch-users] Voicemail - see if user left a message? In-Reply-To: References: Message-ID: With a patch still in http://jira.freeswitch.org/browse/FS-3797 ... mod_voicemail can save to the channel variables voicemail_message_len how long the actual VM is if one has been saved. Thanks Moc for holding my hand to write this tiny change. It has not been committed yet. -Avi Marcus On Tue, Jan 3, 2012 at 2:33 AM, Avi Marcus wrote: > > Can the xml_cdrs tell me if the user actually left a voicemail, rather than merely that VM picked up? I don't see where. > I know it's on ESL, and it hits the voicemail email code.. what's the easiest way to integrate it with the CDRs? preferably with the xml_cdrs? > Maybe a patch to the xml_cdr mod? Or the easiest is to have mod_voicemail set a channel variable once a voicemail gets marked as saved or the like? > > Is there currently a solution for this that I missed? > > Thanks, > -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/fc2ba268/attachment.html From brian.wiese.freeswitch at gmail.com Wed Jan 4 02:54:20 2012 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Tue, 3 Jan 2012 17:54:20 -0600 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Michael: I sure can! What I want to do is create template configurations that I can deploy to multiple servers. Part of the requirement of incoming calls is to configure which extension(s) ring immediately, 6-second delay, and 12-second delay. I have created groups for these extensions, and by using group_call I can get the full dial string for each group... perfect! Now, I just need a way to delay some of these extensions by 6 or 12 seconds. I ultimately want to inject leg variables into the dial string for each extension, so when the group_call is expanded each of them expand with the extra leg variable I define. Hope that helps... clear as mud? :) ~Brian On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins wrote: > Can you expand upon this question a bit? I'm curious if there's a less > hackish way of doing what you want to do. Under what circumstances do you > need to add the leg variables? Also, can you give us the big picture? What's > the problem you're solving? > > -MC > > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese > wrote: >> >> Hi Everyone. >> >> I thought I read somewhere that this was possible, but I can't find it >> now... >> >> I need a way to find-and-replace within a variable. ?So, for example, >> I want to take variables that have values like this: >> >> 123abc123abc >> abc123abc123abc >> >> ...and do a find/replace of the "abc" with "xyz" so the variables >> would now return: >> >> 123xyz123xyz >> xyz123xyz123xyz >> >> The use case I've run into is that I need to add leg variables to >> group_call. ?In my case, group_call can return any number of members, >> so I figured I would just replace the first "[" with >> "[variable-I-want-to-set=...". >> >> Thanks for the help! >> >> ~Brian >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed Jan 4 02:54:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 15:54:22 -0800 Subject: [Freeswitch-users] Voicemail - see if user left a message? In-Reply-To: References: Message-ID: So you didn't like my billsec trick, eh? :P -MC On Tue, Jan 3, 2012 at 2:45 PM, Avi Marcus wrote: > With a patch still in http://jira.freeswitch.org/browse/FS-3797 ... > mod_voicemail can save to the channel variables voicemail_message_len how > long the actual VM is if one has been saved. > Thanks Moc for holding my hand to write this tiny change. > > It has not been committed yet. > > -Avi Marcus > > > > On Tue, Jan 3, 2012 at 2:33 AM, Avi Marcus wrote: > > > > Can the xml_cdrs tell me if the user actually left a voicemail, rather > than merely that VM picked up? I don't see where. > > I know it's on ESL, and it hits the voicemail email code.. what's the > easiest way to integrate it with the CDRs? preferably with the xml_cdrs? > > Maybe a patch to the xml_cdr mod? Or the easiest is to have > mod_voicemail set a channel variable once a voicemail gets marked as saved > or the like? > > > > Is there currently a solution for this that I missed? > > > > Thanks, > > -Avi > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/0f688c87/attachment.html From avi at avimarcus.net Wed Jan 4 02:59:53 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jan 2012 01:59:53 +0200 Subject: [Freeswitch-users] Voicemail - see if user left a message? In-Reply-To: References: Message-ID: No, it's a hack, not a real method. And silly since the patch is ONE line in mod_voicemail. If the user discards their VM it will be long but no message. If that extension has a long VM message, then comparatively it's harder to see how long a VM is. There's no need for a hack here... -Avi On Wed, Jan 4, 2012 at 1:54 AM, Michael Collins wrote: > So you didn't like my billsec trick, eh? :P > -MC > > On Tue, Jan 3, 2012 at 2:45 PM, Avi Marcus wrote: > >> With a patch still in http://jira.freeswitch.org/browse/FS-3797 ... >> mod_voicemail can save to the channel variables voicemail_message_len how >> long the actual VM is if one has been saved. >> Thanks Moc for holding my hand to write this tiny change. >> >> It has not been committed yet. >> >> -Avi Marcus >> >> >> >> On Tue, Jan 3, 2012 at 2:33 AM, Avi Marcus wrote: >> > >> > Can the xml_cdrs tell me if the user actually left a voicemail, rather >> than merely that VM picked up? I don't see where. >> > I know it's on ESL, and it hits the voicemail email code.. what's the >> easiest way to integrate it with the CDRs? preferably with the xml_cdrs? >> > Maybe a patch to the xml_cdr mod? Or the easiest is to have >> mod_voicemail set a channel variable once a voicemail gets marked as saved >> or the like? >> > >> > Is there currently a solution for this that I missed? >> > >> > Thanks, >> > -Avi >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/e8482547/attachment.html From msc at freeswitch.org Wed Jan 4 03:10:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 16:10:33 -0800 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: This kinda sounds like a problem in need of mod_xml_curl. If that seems like too much of a hassle then I would fall back to a mod_lua or mod_perl script to do the regex stuff. How is your scripting fu? -MC On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese < brian.wiese.freeswitch at gmail.com> wrote: > Michael: > > I sure can! > > What I want to do is create template configurations that I can deploy > to multiple servers. Part of the requirement of incoming calls is to > configure which extension(s) ring immediately, 6-second delay, and > 12-second delay. I have created groups for these extensions, and by > using group_call I can get the full dial string for each group... > perfect! Now, I just need a way to delay some of these extensions by > 6 or 12 seconds. I ultimately want to inject leg variables into the > dial string for each extension, so when the group_call is expanded > each of them expand with the extra leg variable I define. > > Hope that helps... clear as mud? :) > > ~Brian > > On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins > wrote: > > Can you expand upon this question a bit? I'm curious if there's a less > > hackish way of doing what you want to do. Under what circumstances do you > > need to add the leg variables? Also, can you give us the big picture? > What's > > the problem you're solving? > > > > -MC > > > > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese > > wrote: > >> > >> Hi Everyone. > >> > >> I thought I read somewhere that this was possible, but I can't find it > >> now... > >> > >> I need a way to find-and-replace within a variable. So, for example, > >> I want to take variables that have values like this: > >> > >> 123abc123abc > >> abc123abc123abc > >> > >> ...and do a find/replace of the "abc" with "xyz" so the variables > >> would now return: > >> > >> 123xyz123xyz > >> xyz123xyz123xyz > >> > >> The use case I've run into is that I need to add leg variables to > >> group_call. In my case, group_call can return any number of members, > >> so I figured I would just replace the first "[" with > >> "[variable-I-want-to-set=...". > >> > >> Thanks for the help! > >> > >> ~Brian > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/bf75d17b/attachment-0001.html From brian.wiese.freeswitch at gmail.com Wed Jan 4 03:16:00 2012 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Tue, 3 Jan 2012 18:16:00 -0600 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Michael: I do think that mod_xml_curl is a little more than what I need here. I'll try a Lua script and see where I get. I'll try to call a Lua script and return the bridge string back to the dial plan. I'll let you know how my scripting fu works... ~Brian On Tue, Jan 3, 2012 at 6:10 PM, Michael Collins wrote: > This kinda sounds like a problem in need of mod_xml_curl. If that seems like > too much of a hassle then I would fall back to a mod_lua or mod_perl script > to do the regex stuff. How is your scripting fu? > > -MC > > > On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese > wrote: >> >> Michael: >> >> I sure can! >> >> What I want to do is create template configurations that I can deploy >> to multiple servers. ?Part of the requirement of incoming calls is to >> configure which extension(s) ring immediately, 6-second delay, and >> 12-second delay. ?I have created groups for these extensions, and by >> using group_call I can get the full dial string for each group... >> perfect! ?Now, I just need a way to delay some of these extensions by >> 6 or 12 seconds. ?I ultimately want to inject leg variables into the >> dial string for each extension, so when the group_call is expanded >> each of them expand with the extra leg variable I define. >> >> Hope that helps... ?clear as mud? ?:) >> >> ~Brian >> >> On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins >> wrote: >> > Can you expand upon this question a bit? I'm curious if there's a less >> > hackish way of doing what you want to do. Under what circumstances do >> > you >> > need to add the leg variables? Also, can you give us the big picture? >> > What's >> > the problem you're solving? >> > >> > -MC >> > >> > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese >> > wrote: >> >> >> >> Hi Everyone. >> >> >> >> I thought I read somewhere that this was possible, but I can't find it >> >> now... >> >> >> >> I need a way to find-and-replace within a variable. ?So, for example, >> >> I want to take variables that have values like this: >> >> >> >> 123abc123abc >> >> abc123abc123abc >> >> >> >> ...and do a find/replace of the "abc" with "xyz" so the variables >> >> would now return: >> >> >> >> 123xyz123xyz >> >> xyz123xyz123xyz >> >> >> >> The use case I've run into is that I need to add leg variables to >> >> group_call. ?In my case, group_call can return any number of members, >> >> so I figured I would just replace the first "[" with >> >> "[variable-I-want-to-set=...". >> >> >> >> Thanks for the help! >> >> >> >> ~Brian >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From cjbujold at accra.ca Wed Jan 4 03:20:11 2012 From: cjbujold at accra.ca (Charles Bujold) Date: Tue, 3 Jan 2012 20:20:11 -0400 Subject: [Freeswitch-users] error on git update Message-ID: <007e01ccca76$9f7af950$de70ebf0$@accra.ca> Tried updating Freeswitch (last update from GIT done Dec 1 2010) today and I'm getting the following error. Can somebody identify what needs to be done to get a proper update. Using Ubuntu 10.4 server. Never had a problem before! In file included from /usr/src/freeswitch.git/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch.git/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.git' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.git' make: *** [current] Error 2 Thanks CJB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/628acb24/attachment.html From msc at freeswitch.org Wed Jan 4 03:43:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 16:43:21 -0800 Subject: [Freeswitch-users] error on git update In-Reply-To: <007e01ccca76$9f7af950$de70ebf0$@accra.ca> References: <007e01ccca76$9f7af950$de70ebf0$@accra.ca> Message-ID: Try doing a ./bootstrap.sh and ./configure and then recompile. -MC On Tue, Jan 3, 2012 at 4:20 PM, Charles Bujold wrote: > Tried updating Freeswitch (last update from GIT done Dec 1 2010) today and > I?m getting the following error. Can somebody identify what needs to be > done to get a proper update. Using Ubuntu 10.4 server. Never had a > problem before!**** > > ** ** > > ** ** > > In file included from > /usr/src/freeswitch.git/libs/spandsp/src/spandsp.h:101,**** > > from ./src/include/private/switch_core_pvt.h:35,**** > > from src/switch_apr.c:37:**** > > /usr/src/freeswitch.git/libs/spandsp/src/spandsp/t4_tx.h:145: error: > expected declaration specifiers or ?...? before ?tz_t?**** > > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1**** > > make[2]: Leaving directory `/usr/src/freeswitch.git'**** > > make[1]: *** [all] Error 2**** > > make[1]: Leaving directory `/usr/src/freeswitch.git'**** > > make: *** [current] Error 2**** > > ** ** > > ** ** > > ** ** > > Thanks**** > > ** ** > > CJB**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/1f0158e3/attachment.html From notlikeme75 at yahoo.com Wed Jan 4 03:53:23 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 3 Jan 2012 16:53:23 -0800 (PST) Subject: [Freeswitch-users] mod limit for duplicate concurrent calls. In-Reply-To: References: Message-ID: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> thanks Vitaly, could someone tell me what date weekly call rupa talked about mod limit? i really would like to setup a method to hangup or transfer to a wav before hanging up anyone trying to call in with the same number ie. three way. ?if this indeed could work. where would i set this mod limit at? on the inbound extension or on the gateway. thanks. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, December 27, 2011 8:32 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 177 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: prevention of duplicate calls. (Vitaly Nikolaev) ? 2. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) ? 3. Re: sangoma A200 with mod_freetdm on windows?! (John) ? 4. Re: Call log - multiple entries CDR?? Billing? (curriegrad2004) Hello, You can try to use mod_limit http://wiki.freeswitch.org/wiki/Mod_limit hash by callerid+callid and set limit 1 call PS: i never used that this way but it might work On Tue, Dec 27, 2011 at 1:11 PM, Rodney wrote: is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up? and continuing the first call. > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > -- -- Vitaly Nikolaev Hi Moises! I allready sent one: http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 they haven't replied since 2 days :( Tamer Am 27.12.2011 20:17, schrieb Moises Silva: > On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi > wrote: > >? ? Hi people! >? ? I got my A200 board running with 1 FXS module on Linux along with >? ? mod_freetdm, but I am facing problems getting it to run on Windows. From >? ? Sangoma I followed the instructions to set up the board on Windows7 >? ? winpipe module, which works so far. > >? ? How do I get freeswitch with mod_freetdm to run on Windows that I can >? ? make use of the board (pbx) on a win machine? > > > You may want to send an email to Sangoma support. They are working > already in a wiki page for Windows setup, in the meantime they can help > you with instructions via email. > > *Moises Silva > **/Software Engineer, Development Manager/*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > ??? > > > t.? +1 800 388 2475 (N. America) > > t.? +1 905 474 1990 x128 > > f.? +1 905 474 9223 > >? > > ??? > > ** > > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter > `| > | YouTube > > >? ? ? ? ? ? ? ? VegaStream is now part of Sangoma! > > >? ? ? ? ? ? ? ? Ask us about both Gateway Appliances > and Internal > Gateways > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Tamer, This is linked from the Sangoma Support front page: http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-schedule John On 27/12/11 23:51, Tamer Higazi wrote: > Hi Moises! > I allready sent one: > > http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > > they haven't replied since 2 days :( > > > Tamer > > Am 27.12.2011 20:17, schrieb Moises Silva: >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi> >? wrote: >> >>? ? ? Hi people! >>? ? ? I got my A200 board running with 1 FXS module on Linux along with >>? ? ? mod_freetdm, but I am facing problems getting it to run on Windows. From >>? ? ? Sangoma I followed the instructions to set up the board on Windows7 >>? ? ? winpipe module, which works so far. >> >>? ? ? How do I get freeswitch with mod_freetdm to run on Windows that I can >>? ? ? make use of the board (pbx) on a win machine? >> >> >> You may want to send an email to Sangoma support. They are working >> already in a wiki page for Windows setup, in the meantime they can help >> you with instructions via email. >> >> *Moises Silva >> **/Software Engineer, Development Manager/*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> ??? >> >> >> t.? +1 800 388 2475 (N. America) >> >> t.? +1 905 474 1990 x128 >> >> f.? +1 905 474 9223 >> >> >> >> ??? >> >> ** >> >> >> Products >> ? | Solutions >> ? | Events >> ? | Contact >> ? | Wiki >> ? | Facebook >> ? | Twitter >> `| >> | YouTube >> >> >>? ? ? ? ? ? ? ? ? VegaStream is now part of Sangoma! >> >> >>? ? ? ? ? ? ? ? ? Ask us about both Gateway Appliances >> ? and Internal >> Gateways >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: ? ? to false ? ? All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran wrote: out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details > >Call logs are duplicated for 1 call? How can i prevent this? > > > > > >--- On Tue, 12/27/11, curriegrad2004 wrote: > > >>From: curriegrad2004 >>Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >>To: "FreeSWITCH Users Help" >>Date: Tuesday, December 27, 2011, 6:33 AM >> >> >>xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for >> >> >>On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: >> >>Hi, >>> >>>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >>> >>>thanks in advance >>> >>>regards, >>>Sherif Omran >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>> >> >>-----Inline Attachment Follows----- >> >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/9b4ba8fc/attachment-0001.html From sherifomran2000 at yahoo.com Wed Jan 4 04:02:08 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 3 Jan 2012 17:02:08 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! - Review In-Reply-To: Message-ID: <1325638928.76279.YahooMailClassic@web110805.mail.gq1.yahoo.com> Hi every body, I had a chance 3 days to install vBilling manually, which was not so trivial and would like to share? my experience with you. Installation script is clearly written, any developer with some linux experience can follow it and do the required changes. In fact, I did not use the installation script because I have custom modules enabled. Also, I use the freeswitch xml tree, which is different that what the script do. Additionally, I have BlueBox installed and they should work together. The installation script creates a custom freeswitch.xml file with the required modules. If these freeswitch modules were not previously installed, you have to install them manually. No need to reinstall freeswitch again using (make install) enable them in the freeswitch source and compile using make modulename-install Regarding the installation path: I used Centos 6 Server The software uses the /var/www/html/ folder as the base path. However, since i have bluebox installed, I tried to do the required changes in (freeswitch.xml) and install vBilling in /var/www/html/vBilling. However, after contacting Muhammed, he recommended to enabled mod-rewrite for the apache server. After checking out, I found that it was already enabled for centos using the .htaccess file. I could log to the front page but 404 error pops, if i enter the password leading to another path.? I checked the login php function and corrected the path. At least i could see that after login click, it was trying to call a page from the correct subfolder. However, I would not recommend to install it in a subfolder because it will not work properly. May be this needs some additional investement. When i placed it in the root web server folder, and made the required adjustments it worked fine. During the installation, I had to create a database using the given script, however I changed the username and password in the mysql tag. It returned that i could not login to the frontpage, since the password in encrypted. I had then to install phpmyadmin to revert the changes and change the password from the frontpage. It is now running fine. However, I still did not start playing with its reporting functions or any call log but it looks very promising. If i install some script, I usually check for spying functions such as sending back precious information without a permission to the author. However, the software is really clean. I did not find any spying functions or backholes. I checked the PHP files as well. Finally, i would like to thank Muhammed Bhatti for sharing this nice software with us. kind regards, Sherif Omran +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Sherif Omran Dr. sc. nat. (Title from University of Zurich) Signal and Image Processing, Acoustics, Artificial Intelligence Engineer and Neural Scientist. Design, Modeling and Simulation. Expert in Biomedical devices and Cochlear Implants. Telecommunication Consultant and ERICSSON Certified Engineer. Munich - Germany e-mail: sherif.omran at gmx.de +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ --- On Tue, 1/3/12, Muhammad Naseer Bhatti wrote: From: Muhammad Naseer Bhatti Subject: Re: [Freeswitch-users] vBilling Beta Program!! To: "FreeSWITCH Users Help" Date: Tuesday, January 3, 2012, 8:16 PM Sherif, have you installed manually? If so, you would have to enable mod_rewrite in your apache configuration. On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: Can you give more details how it does not work? I have the same situation. I can reach the frontpage and when i give the username i get 404. Is this the case you have? Do you have Centos or Redhat ? regards, Sherif --- On Tue, 1/3/12, Zenny wrote: From: Zenny Subject: Re: [Freeswitch-users] vBilling Beta Program!! To: "FreeSWITCH Users Help" Date: Tuesday, January 3, 2012, 11:18 AM looks promising, but the user login does not work. Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to all freeswitchers! On 1/3/12, nbhatti wrote: > Yes, it will support prepaid calling card and many more features soon. > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > wrote: >> It looks great! >> >> Will vBilling support batch user/prepaid calling card creation? >> >> ________________________________ >> If you reply to this email, your message will be added to the discussion >> below: >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >> To unsubscribe from vBilling Beta Program!!, click here. >> NAML > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/83781c6f/attachment.html From msc at freeswitch.org Wed Jan 4 04:04:29 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 17:04:29 -0800 Subject: [Freeswitch-users] timeout for curl In-Reply-To: References: Message-ID: I didn't see anything obvious in the mod_curl.c code nor in switch_curl.c which is not a good sign. However, you could probably use a mod_lua or mod_perl script to have more control over a call to the system curl. FWIW, I looked in libs/curl/include/curl/curl.h and there is a connect timeout hook but it does not look like switch_curl.c has an interface for it. A bounty, perhaps? -MC 2012/1/3 Yuriy Nasida > Hello FS users, > > I do some curl request from my dialplan. It looks like this: > > If cname provider doesn't answer for long time, I have issue because FS > doesn't process a call. > whether there are any method to add some timeout for curl (3 sec for > example) ? Thus, if cname provider doesn't answer for 3 sec, FS must > continue the logic of dialplan. > > I read about mod_cidlookup but it looks like that this module have same > issue > http://wiki.freeswitch.org/wiki/Mod_cidlookup > > Please advise > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/00534198/attachment.html From msc at freeswitch.org Wed Jan 4 04:06:27 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 17:06:27 -0800 Subject: [Freeswitch-users] mod limit for duplicate concurrent calls. In-Reply-To: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> References: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: On Tue, Jan 3, 2012 at 4:53 PM, Rodney wrote: > thanks Vitaly, > > could someone tell me what date weekly call rupa talked about mod limit? i > really would like to setup a method to hangup or transfer to a wav before > hanging up anyone trying to call in with the same number ie. three way. if > this indeed could work. where would i set this mod limit at? on the inbound > extension or on the gateway. thanks. > See link to audio files here: http://wiki.freeswitch.org/wiki/Mod_limit#Limit_Resource_Management -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/7f4d49e6/attachment-0001.html From sherifomran2000 at yahoo.com Wed Jan 4 04:06:37 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 3 Jan 2012 17:06:37 -0800 (PST) Subject: [Freeswitch-users] g722 In-Reply-To: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: <1325639197.1279.YahooMailClassic@web110816.mail.gq1.yahoo.com> Hell guys I noticed that module-g722 is missing. It pops out an error message during loading freeswitch. Does any body has? a clue where is the source code? best regards, Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/346e05a6/attachment.html From steveu at coppice.org Wed Jan 4 04:09:00 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 04 Jan 2012 09:09:00 +0800 Subject: [Freeswitch-users] g722 In-Reply-To: <1325639197.1279.YahooMailClassic@web110816.mail.gq1.yahoo.com> References: <1325639197.1279.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: <4F03A6AC.9010000@coppice.org> There is no G.722 module. The G.722 codec is provided by mod_spandsp. Steve On 01/04/2012 09:06 AM, Sherif Omran wrote: > Hell guys > > I noticed that module-g722 is missing. It pops out an error message > during loading freeswitch. Does any body has a clue where is the > source code? > > > best regards, > Sherif > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sherifomran2000 at yahoo.com Wed Jan 4 04:53:31 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 3 Jan 2012 17:53:31 -0800 (PST) Subject: [Freeswitch-users] g722 In-Reply-To: <4F03A6AC.9010000@coppice.org> Message-ID: <1325642011.60125.YahooMailClassic@web110808.mail.gq1.yahoo.com> I get the following warning 2012-01-04 01:52:14.363026 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_g722.so **/usr/local/freeswitch/mod/mod_g722.so: cannot open shared object file: No such file or directory** do you know how to avoid it? thanks regards, Sherif --- On Wed, 1/4/12, Steve Underwood wrote: From: Steve Underwood Subject: Re: [Freeswitch-users] g722 To: "FreeSWITCH Users Help" Date: Wednesday, January 4, 2012, 3:09 AM There is no G.722 module. The G.722 codec is provided by mod_spandsp. Steve On 01/04/2012 09:06 AM, Sherif Omran wrote: > Hell guys > > I noticed that module-g722 is missing. It pops out an error message > during loading freeswitch. Does any body has? a clue where is the > source code? > > > best regards, > Sherif > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/9cea4262/attachment.html From anthony.minessale at gmail.com Wed Jan 4 04:56:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jan 2012 19:56:01 -0600 Subject: [Freeswitch-users] error on git update In-Reply-To: <007e01ccca76$9f7af950$de70ebf0$@accra.ca> References: <007e01ccca76$9f7af950$de70ebf0$@accra.ca> Message-ID: make spandsp-reconf On Jan 3, 2012 6:22 PM, "Charles Bujold" wrote: > Tried updating Freeswitch (last update from GIT done Dec 1 2010) today and > I?m getting the following error. Can somebody identify what needs to be > done to get a proper update. Using Ubuntu 10.4 server. Never had a > problem before!**** > > ** ** > > ** ** > > In file included from > /usr/src/freeswitch.git/libs/spandsp/src/spandsp.h:101,**** > > from ./src/include/private/switch_core_pvt.h:35,**** > > from src/switch_apr.c:37:**** > > /usr/src/freeswitch.git/libs/spandsp/src/spandsp/t4_tx.h:145: error: > expected declaration specifiers or ?...? before ?tz_t?**** > > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1**** > > make[2]: Leaving directory `/usr/src/freeswitch.git'**** > > make[1]: *** [all] Error 2**** > > make[1]: Leaving directory `/usr/src/freeswitch.git'**** > > make: *** [current] Error 2**** > > ** ** > > ** ** > > ** ** > > Thanks**** > > ** ** > > CJB**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/6c35d843/attachment.html From sherifomran2000 at yahoo.com Wed Jan 4 05:00:59 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 3 Jan 2012 18:00:59 -0800 (PST) Subject: [Freeswitch-users] conference.conf missing? In-Reply-To: <1325639197.1279.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: <1325642459.39342.YahooMailClassic@web110811.mail.gq1.yahoo.com> Hello every body, I get the following error, any clue? 2012-01-04 01:52:14.234297 [ERR] mod_conference.c:7491 Open of conference.conf failed thanks regards, Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/18678abb/attachment.html From msc at freeswitch.org Wed Jan 4 07:12:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 20:12:26 -0800 Subject: [Freeswitch-users] g722 In-Reply-To: <1325642011.60125.YahooMailClassic@web110808.mail.gq1.yahoo.com> References: <4F03A6AC.9010000@coppice.org> <1325642011.60125.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: On Tue, Jan 3, 2012 at 5:53 PM, Sherif Omran wrote: > I get the following warning > > 2012-01-04 01:52:14.363026 [CRIT] switch_loadable_module.c:1281 Error > Loading module /usr/local/freeswitch/mod/mod_g722.so > **/usr/local/freeswitch/mod/mod_g722.so: cannot open shared object file: > No such file or directory** > > do you know how to avoid it? > Use latest git? There is no mod_g722 in the latest git. You may have updated from an older version and have an old modules.conf.xml file. I'd recommend installing a fresh, clean git version with the default configs and then re-integrate your customizations. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/c87bbcfe/attachment-0001.html From msc at freeswitch.org Wed Jan 4 07:14:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jan 2012 20:14:24 -0800 Subject: [Freeswitch-users] conference.conf missing? In-Reply-To: <1325642459.39342.YahooMailClassic@web110811.mail.gq1.yahoo.com> References: <1325639197.1279.YahooMailClassic@web110816.mail.gq1.yahoo.com> <1325642459.39342.YahooMailClassic@web110811.mail.gq1.yahoo.com> Message-ID: It means something happened to /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml, assuming that's your path. If you did not have a customized conference.conf.xml file then you can just copy the the file from your FS source directory under conf/autoload_configs/ -MC On Tue, Jan 3, 2012 at 6:00 PM, Sherif Omran wrote: > Hello every body, > > I get the following error, any clue? > > 2012-01-04 01:52:14.234297 [ERR] mod_conference.c:7491 Open of > conference.conf failed > > thanks > > regards, > Sherif Omran > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/3198b50d/attachment.html From curriegrad2004 at gmail.com Wed Jan 4 08:03:39 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 3 Jan 2012 21:03:39 -0800 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH Message-ID: One of the frequent complaints is that the default diaplan is huge... Well, this has been discussed about in one of the earlier conference calls last year and so far nobody has taken up any work on creating this configuration set, so I decided to take up the work on creating one. It's been done and I have it uploaded on my github. Testing and feedback is greatly appreciated: https://github.com/curriegrad2004/Freeswitch_slim_conf "A configuration set created for users new to FreeSWITCH. It doesn't expose what FreeSWITCH is truly capable of, but I hope new users won't get scared with this configuration set." as quoted from my description page on github. From sharad at coraltele.com Wed Jan 4 08:01:46 2012 From: sharad at coraltele.com (Sharad Garg) Date: Wed, 4 Jan 2012 10:31:46 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 7 References: Message-ID: <8D24ACE29A4C446B981F1BF4818F7923@sharad> All these are basically IP based PBX systems having PBX features. Regards ----- Original Message ----- From: To: Sent: Tuesday, January 03, 2012 7:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 7 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > > 1. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 2. Re: mod_sofia not binding to port 5060 (Brandon McGinty) > 3. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 4. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Giovanni Maruzzelli) > 5. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Kaushal Shriyan) > 6. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Zohair Raza) > 7. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > CallWeaver, and YATE. (Vitaly Nikolaev) > -------------------------------------------------------------------------------- > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nvitaly at gmail.com Wed Jan 4 08:17:01 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Wed, 4 Jan 2012 00:17:01 -0500 Subject: [Freeswitch-users] mod limit for duplicate concurrent calls. In-Reply-To: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> References: <1325638403.14665.YahooMailNeo@web65314.mail.ac2.yahoo.com> Message-ID: You need to check limit in dial plan before it hit conference, here is example: if same phone call 999 more then one time, it will get to limit_exceeded ext. On Jan 3, 2012, at 7:53 PM, Rodney wrote: > thanks Vitaly, > > could someone tell me what date weekly call rupa talked about mod limit? i really would like to setup a method to hangup or transfer to a wav before hanging up anyone trying to call in with the same number ie. three way. if this indeed could work. where would i set this mod limit at? on the inbound extension or on the gateway. thanks. > > > From: "freeswitch-users-request at lists.freeswitch.org" > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, December 27, 2011 8:32 PM > Subject: FreeSWITCH-users Digest, Vol 66, Issue 177 > > ----- Forwarded Message ----- > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: prevention of duplicate calls. (Vitaly Nikolaev) > 2. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) > 3. Re: sangoma A200 with mod_freetdm on windows?! (John) > 4. Re: Call log - multiple entries CDR?? Billing? (curriegrad2004) > Hello, > > You can try to use mod_limit > > http://wiki.freeswitch.org/wiki/Mod_limit > > hash by callerid+callid and set limit 1 call > > PS: i never used that this way but it might work > > > > > On Tue, Dec 27, 2011 at 1:11 PM, Rodney wrote: > is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up and continuing the first call. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > -- > Vitaly Nikolaev > Hi Moises! > I allready sent one: > > http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > > they haven't replied since 2 days :( > > > Tamer > > Am 27.12.2011 20:17, schrieb Moises Silva: > > On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi > > wrote: > > > > Hi people! > > I got my A200 board running with 1 FXS module on Linux along with > > mod_freetdm, but I am facing problems getting it to run on Windows. From > > Sangoma I followed the instructions to set up the board on Windows7 > > winpipe module, which works so far. > > > > How do I get freeswitch with mod_freetdm to run on Windows that I can > > make use of the board (pbx) on a win machine? > > > > > > You may want to send an email to Sangoma support. They are working > > already in a wiki page for Windows setup, in the meantime they can help > > you with instructions via email. > > > > *Moises Silva > > **/Software Engineer, Development Manager/*** > > > > msilva at sangoma.com > > > > Sangoma Technologies > > > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > > > > > > t. +1 800 388 2475 (N. America) > > > > t. +1 905 474 1990 x128 > > > > f. +1 905 474 9223 > > > > > > > > > > > > ** > > > > > > Products > > | Solutions > > | Events > > | Contact > > | Wiki > > | Facebook > > | Twitter > > `| > > | YouTube > > > > > > VegaStream is now part of Sangoma! > > > > > > Ask us about both Gateway Appliances > > and Internal > > Gateways > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > Tamer, > > This is linked from the Sangoma Support front page: > > http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-schedule > > John > > On 27/12/11 23:51, Tamer Higazi wrote: > > Hi Moises! > > I allready sent one: > > > > http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > > > > they haven't replied since 2 days :( > > > > > > Tamer > > > > Am 27.12.2011 20:17, schrieb Moises Silva: > >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi >> > wrote: > >> > >> Hi people! > >> I got my A200 board running with 1 FXS module on Linux along with > >> mod_freetdm, but I am facing problems getting it to run on Windows. From > >> Sangoma I followed the instructions to set up the board on Windows7 > >> winpipe module, which works so far. > >> > >> How do I get freeswitch with mod_freetdm to run on Windows that I can > >> make use of the board (pbx) on a win machine? > >> > >> > >> You may want to send an email to Sangoma support. They are working > >> already in a wiki page for Windows setup, in the meantime they can help > >> you with instructions via email. > >> > >> *Moises Silva > >> **/Software Engineer, Development Manager/*** > >> > >> msilva at sangoma.com > >> > >> Sangoma Technologies > >> > >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > >> > >> > >> > >> > >> t. +1 800 388 2475 (N. America) > >> > >> t. +1 905 474 1990 x128 > >> > >> f. +1 905 474 9223 > >> > >> > >> > >> > >> > >> ** > >> > >> > >> Products > >> | Solutions > >> | Events > >> | Contact > >> | Wiki > >> | Facebook > >> | Twitter > >> `| > >> | YouTube > >> > >> > >> VegaStream is now part of Sangoma! > >> > >> > >> Ask us about both Gateway Appliances > >> and Internal > >> Gateways > >> > >> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: > > > > If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: > > to false > > > All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration > > On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran wrote: > out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details > out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details > > > Call logs are duplicated for 1 call? How can i prevent this? > > > > > > --- On Tue, 12/27/11, curriegrad2004 wrote: > > From: curriegrad2004 > Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? > To: "FreeSWITCH Users Help" > Date: Tuesday, December 27, 2011, 6:33 AM > > xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for > > On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: > Hi, > > I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? > > thanks in advance > > regards, > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/6c949aa2/attachment-0001.html From notlikeme75 at yahoo.com Wed Jan 4 08:41:51 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 3 Jan 2012 21:41:51 -0800 (PST) Subject: [Freeswitch-users] mod limit for duplicate concurrent calls. In-Reply-To: References: Message-ID: <1325655711.39450.YahooMailNeo@web65302.mail.ac2.yahoo.com> vitaly, is it possible to put this limit before it even gets to my greeting/preamble? i don't even want them hitting the IVR. plus when does that count reset? i want the counter to reset once they hangup all current calls. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, January 4, 2012 12:17 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 29 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: conference.conf missing? (Michael Collins) ? 2. A New Config Set aimed for New People to??? FreeSWITCH ? ? ? (curriegrad2004) ? 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 7 (Sharad Garg) ? 4.?Re: mod limit for duplicate concurrent calls.??(Vitaly Nikolaev) It means something happened to /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml, assuming that's your path. If you did not have a customized conference.conf.xml file then you can just copy the the file from your FS source directory under conf/autoload_configs/ -MC On Tue, Jan 3, 2012 at 6:00 PM, Sherif Omran wrote: Hello every body, > >I get the following error, any clue? > >2012-01-04 01:52:14.234297 [ERR] mod_conference.c:7491 Open of conference.conf failed > >thanks > >regards, >Sherif Omran > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > One of the frequent complaints is that the default diaplan is huge... Well, this has been discussed about in one of the earlier conference calls last year and so far nobody has taken up any work on creating this configuration set, so I decided to take up the work on creating one. It's been done and I have it uploaded on my github. Testing and feedback is greatly appreciated: https://github.com/curriegrad2004/Freeswitch_slim_conf "A configuration set created for users new to FreeSWITCH. It doesn't expose what FreeSWITCH is truly capable of, but I hope new users won't get scared with this configuration set." as quoted from my description page on github. All these are basically IP based PBX systems having PBX features. Regards ----- Original Message ----- From: To: Sent: Tuesday, January 03, 2012 7:47 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 7 > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > -------------------------------------------------------------------------------- > Today's Topics: > >? 1. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Kaushal Shriyan) >? 2. Re: mod_sofia not binding to port 5060 (Brandon McGinty) >? 3. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Zohair Raza) >? 4. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Giovanni Maruzzelli) >? 5. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Kaushal Shriyan) >? 6. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Zohair Raza) >? 7. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, >? ? ? CallWeaver, and YATE. (Vitaly Nikolaev) > -------------------------------------------------------------------------------- > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > You need to check limit in dial plan before it hit conference, here is example: ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? if same phone call 999 more then one time, it will get to limit_exceeded ext. On Jan 3, 2012, at 7:53 PM, Rodney wrote: thanks Vitaly, > > >could someone tell me what date weekly call rupa talked about mod limit? i really would like to setup a method to hangup or transfer to a wav before hanging up anyone trying to call in with the same number ie. three way. ?if this indeed could work. where would i set this mod limit at? on the inbound extension or on the gateway. thanks. > > > > > >________________________________ > From: "freeswitch-users-request at lists.freeswitch.org" >To: freeswitch-users at lists.freeswitch.org >Sent: Tuesday, December 27, 2011 8:32 PM >Subject: FreeSWITCH-users Digest, Vol 66, Issue 177 > >----- Forwarded Message ----- > >Send FreeSWITCH-users mailing list submissions to >??? freeswitch-users at lists.freeswitch.org > >To subscribe or unsubscribe via the World Wide Web, visit >??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >or, via email, send a message with subject or body 'help' to >??? freeswitch-users-request at lists.freeswitch.org > >You can reach the person managing the list at >??? freeswitch-users-owner at lists.freeswitch.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of FreeSWITCH-users digest..." > >Today's Topics: > >? 1. Re: prevention of duplicate calls. (Vitaly Nikolaev) >? 2. Re: sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) >? 3. Re: sangoma A200 with mod_freetdm on windows?! (John) >? 4. Re: Call log - multiple entries CDR?? Billing? (curriegrad2004) > >Hello, > >You can try to use mod_limit > >http://wiki.freeswitch.org/wiki/Mod_limit > >hash by callerid+callid and set limit 1 call > >PS: i never used that this way but it might work > > > > > >On Tue, Dec 27, 2011 at 1:11 PM, Rodney wrote: > >is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up? and continuing the first call. >> >> >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > >-- >-- >Vitaly Nikolaev >Hi Moises! >I allready sent one: > >http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > >they haven't replied since 2 days :( > > >Tamer > >Am 27.12.2011 20:17, schrieb Moises Silva: >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi > > wrote: >> >>? ? Hi people! >>? ? I got my A200 board running with 1 FXS module on Linux along with >>? ? mod_freetdm, but I am facing problems getting it to run on Windows. From >>? ? Sangoma I followed the instructions to set up the board on Windows7 >>? ? winpipe module, which works so far. >> >>? ? How do I get freeswitch with mod_freetdm to run on Windows that I can >>? ? make use of the board (pbx) on a win machine? >> >> >> You may want to send an email to Sangoma support. They are working >> already in a wiki page for Windows setup, in the meantime they can help >> you with instructions via email. >> >> *Moises Silva >> **/Software Engineer, Development Manager/*** >> >> msilva at sangoma.com >> >> Sangoma Technologies >> >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> ??? >> >> >> t.? +1 800 388 2475 (N. America) >> >> t.? +1 905 474 1990 x128 >> >> f.? +1 905 474 9223 >> >>? >> >> ??? >> >> ** >> >> >> Products >> | Solutions >> | Events >> | Contact >> | Wiki >> | Facebook >> | Twitter >> `| >> | YouTube >> >> >>? ? ? ? ? ? ? ? VegaStream is now part of Sangoma! >> >> >>? ? ? ? ? ? ? ? Ask us about both Gateway Appliances >> and Internal >> Gateways >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > >Tamer, > >This is linked from the Sangoma Support front page: > >http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-schedule > >John > >On 27/12/11 23:51, Tamer Higazi wrote: >> Hi Moises! >> I allready sent one: >> >> http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 >> >> they haven't replied since 2 days :( >> >> >> Tamer >> >> Am 27.12.2011 20:17, schrieb Moises Silva: >>> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi>> >? wrote: >>> >>>? ? ? Hi people! >>>? ? ? I got my A200 board running with 1 FXS module on Linux along with >>>? ? ? mod_freetdm, but I am facing problems getting it to run on Windows. From >>>? ? ? Sangoma I followed the instructions to set up the board on Windows7 >>>? ? ? winpipe module, which works so far. >>> >>>? ? ? How do I get freeswitch with mod_freetdm to run on Windows that I can >>>? ? ? make use of the board (pbx) on a win machine? >>> >>> >>> You may want to send an email to Sangoma support. They are working >>> already in a wiki page for Windows setup, in the meantime they can help >>> you with instructions via email. >>> >>> *Moises Silva >>> **/Software Engineer, Development Manager/*** >>> >>> msilva at sangoma.com >>> >>> Sangoma Technologies >>> >>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>> >>> ??? >>> >>> >>> t.? +1 800 388 2475 (N. America) >>> >>> t.? +1 905 474 1990 x128 >>> >>> f.? +1 905 474 9223 >>> >>> >>> >>> ??? >>> >>> ** >>> >>> >>> Products >>> ? | Solutions >>> ? | Events >>> ? | Contact >>> ? | Wiki >>> ? | Facebook >>> ? | Twitter >>> `| >>> | YouTube >>> >>> >>>? ? ? ? ? ? ? ? ? VegaStream is now part of Sangoma! >>> >>> >>>? ? ? ? ? ? ? ? ? Ask us about both Gateway Appliances >>> ? and Internal >>> Gateways >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > >This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: > > > > > >If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: >? ? >to false >? ? > > >All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration > > >On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran wrote: > >out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >>out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details >> >>Call logs are duplicated for 1 call? How can i prevent this? >> >> >> >> >> >>--- On Tue, 12/27/11, curriegrad2004 wrote: >> >> >>>From: curriegrad2004 >>>Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >>>To: "FreeSWITCH Users Help" >>>Date: Tuesday, December 27, 2011, 6:33 AM >>> >>> >>>xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for >>> >>> >>>On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: >>> >>>Hi, >>>> >>>>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >>>> >>>>thanks in advance >>>> >>>>regards, >>>>Sherif Omran >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.com >>>> >>>>FreeSWITCH-users mailing list >>>>FreeSWITCH-users at lists.freeswitch.org >>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>http://www.freeswitch.org >>>> >>>> >>> >>>-----Inline Attachment Follows----- >>> >>> >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >>> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120103/9924b925/attachment-0001.html From hynek.cihlar at gmail.com Wed Jan 4 09:01:01 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 4 Jan 2012 07:01:01 +0100 Subject: [Freeswitch-users] Silence ESL event In-Reply-To: References: <-3489375824995432417@unknownmsgid> Message-ID: I'll give it a try, thanks! Hynek On Tue, Jan 3, 2012 at 10:46 PM, Michael Collins wrote: > You might be able to produce the same behavior with the "wait_for_silence" > dp application. > > -MC > > On Wed, Dec 28, 2011 at 3:20 AM, Hynek Cihlar wrote: > >> Hi all, >> >> is there any way to receive a silence-detected event through ESL? >> >> The use case is to hangup the call in case nothing interesting is >> going on on the channel. >> >> Sent from my mobile device >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/16f64eeb/attachment.html From gabe at gundy.org Wed Jan 4 09:12:45 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 3 Jan 2012 23:12:45 -0700 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH In-Reply-To: References: Message-ID: On Tue, Jan 3, 2012 at 10:03 PM, curriegrad2004 wrote: > It's been done and I have it uploaded on my github. Testing and > feedback is greatly appreciated: Love it. Gabe From nbhatti at gmail.com Wed Jan 4 13:46:10 2012 From: nbhatti at gmail.com (nbhatti) Date: Wed, 4 Jan 2012 02:46:10 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! - Review In-Reply-To: <1325638928.76279.YahooMailClassic@web110805.mail.gq1.yahoo.com> References: <1325549865328-7145235.post@n2.nabble.com> <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> <1325638928.76279.YahooMailClassic@web110805.mail.gq1.yahoo.com> Message-ID: Hello Sharif, Many thanks for the constructive comments for vBilling. This would really help us to improve and deliver more. As per your issues, vBilling is designed as a standalone application minimizing the footprint of FreeSWITCH and maximize the performance. We have not enabled any unnecessary modules which are not required. We already have included a complete management interface for FreeSWITCH. You can control almost every aspect of configuration without having a need of separate management interface. But If you really like to install it along with some other existence of any other application, you would have to make some changes in the following files accordingly. *application/config/config.php *Adjust *$config['base_url']* accordingly. You would also have to adjust * * in *freeswitch.xml *file as per your new path. Also make sure, /etc/hosts contains an entry for localhost pointing to 127.0.0.1 or vice versa. You would also have to manually install the database since it looks for a new install and configured the server accordingly. Don't also forget to configure odbc for the database you make. (/etc/odbc.ini) We are working on having vBilling exist with existing FS install base, but this will take some time to implement. As the core of vBilling is designed to serve dynamic configs from the DB using mod_xml_curl. Once again we are open for any (constructive) comments and contributions to improve vBilling. Thanks, Muhammad Naseer (Goni) On Wed, Jan 4, 2012 at 4:08 AM, sherif omran [via freeswitch-users] < ml-node+s2379917n7148890h48 at n2.nabble.com> wrote: > Hi every body, > > I had a chance 3 days to install vBilling manually, which was not so > trivial and would like to share my experience with you. > > Installation script is clearly written, any developer with some linux > experience can follow it and do the required changes. In fact, I did not > use the installation script because I have custom modules enabled. Also, I > use the freeswitch xml tree, which is different that what the script do. > Additionally, I have BlueBox installed and they should work together. > > The installation script creates a custom freeswitch.xml file with the > required modules. If these freeswitch modules were not previously > installed, you have to install them manually. No need to reinstall > freeswitch again using (make install) > > enable them in the freeswitch source and compile using > make modulename-install > > Regarding the installation path: > I used Centos 6 Server > > The software uses the /var/www/html/ folder as the base path. However, > since i have bluebox installed, I tried to do the required changes in > (freeswitch.xml) and install vBilling in /var/www/html/vBilling. However, > after contacting Muhammed, he recommended to enabled mod-rewrite for the > apache server. After checking out, I found that it was already enabled for > centos using the .htaccess file. I could log to the front page but 404 > error pops, if i enter the password leading to another path. I checked the > login php function and corrected the path. At least i could see that after > login click, it was trying to call a page from the correct subfolder. > However, I would not recommend to install it in a subfolder because it will > not work properly. May be this needs some additional investement. When i > placed it in the root web server folder, and made the required adjustments > it worked fine. > > During the installation, I had to create a database using the given > script, however I changed the username and password in the mysql tag. It > returned that i could not login to the frontpage, since the password in > encrypted. I had then to install phpmyadmin to revert the changes and > change the password from the frontpage. > > It is now running fine. However, I still did not start playing with its > reporting functions or any call log but it looks very promising. > > If i install some script, I usually check for spying functions such as > sending back precious information without a permission to the author. > However, the software is really clean. I did not find any spying functions > or backholes. I checked the PHP files as well. > > > Finally, i would like to thank Muhammed Bhatti for sharing this nice > software with us. > > kind regards, > Sherif Omran > > +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > Sherif Omran Dr. sc. nat. > (Title from University of Zurich) > Signal and Image Processing, Acoustics, Artificial Intelligence Engineer > and Neural Scientist. Design, Modeling and Simulation. Expert in Biomedical > devices and Cochlear Implants. Telecommunication Consultant and ERICSSON > Certified Engineer. > Munich - Germany > e-mail: [hidden email] > +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > > > --- On *Tue, 1/3/12, Muhammad Naseer Bhatti <[hidden email] > >* wrote: > > > From: Muhammad Naseer Bhatti <[hidden email] > > > > Subject: Re: [Freeswitch-users] vBilling Beta Program!! > To: "FreeSWITCH Users Help" <[hidden email] > > > > Date: Tuesday, January 3, 2012, 8:16 PM > > > Sherif, have you installed manually? If so, you would have to enable > mod_rewrite in your apache configuration. > > > On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran > > wrote: > > Can you give more details how it does not work? I have the same situation. > I can reach the frontpage and when i give the username i get 404. > > Is this the case you have? > > Do you have Centos or Redhat ? > > > regards, > Sherif > > > --- On *Tue, 1/3/12, Zenny > >* wrote: > > > From: Zenny > > > Subject: Re: [Freeswitch-users] vBilling Beta Program!! > To: "FreeSWITCH Users Help" > > > > Date: Tuesday, January 3, 2012, 11:18 AM > > > looks promising, but the user login does not work. > > Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to > all freeswitchers! > > On 1/3/12, nbhatti > > wrote: > > Yes, it will support prepaid calling card and many more features soon. > > > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > > > > wrote: > >> It looks great! > >> > >> Will vBilling support batch user/prepaid calling card creation? > >> > >> ________________________________ > >> If you reply to this email, your message will be added to the discussion > >> below: > >> > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html > >> To unsubscribe from vBilling Beta Program!!, click here. > >> NAML > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7148890.html > To unsubscribe from vBilling Beta Program!!, click here > . > NAML > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7149925.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/e53f9c75/attachment-0001.html From benkokakao at gmail.com Wed Jan 4 14:39:23 2012 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 4 Jan 2012 12:39:23 +0100 Subject: [Freeswitch-users] Debugging intermitent BLF problems In-Reply-To: References: Message-ID: On 2 January 2012 15:43, Yehavi Bourvine wrote: > It was rock-solid until mid-October (and that's why I have to stay at that > version in the meantime). > > I am willing to help debugging this, but need to know which debugs to add > which will be usefull... Since no one else answers: I guess the standard debug-log plus SIP-Debugging(NOTIFY-messages) should suffice. If it's not enough, you'll be told how to improve after you've filed the bug-report. Let us know the bug-id and i'll contribute if needed. Best regards Christian From sharad at coraltele.com Wed Jan 4 15:44:24 2012 From: sharad at coraltele.com (Sharad Garg) Date: Wed, 4 Jan 2012 18:14:24 +0530 Subject: [Freeswitch-users] Originate with dead destination References: Message-ID: <30934ADE01DC4CEE8B3AEB5DF8602C94@sharad> Hi friends, While using ORIGINATE API, if destination IP (local IP) is dead, FS marks this IP as dead with him till that IP alives. Is there any way to configure the FS not to mark as dead destination. Let it send SIP INVITE even if last SIP INVITE was not answered. Thanks in advance. Regards Sharad From miha at softnet.si Wed Jan 4 16:11:09 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 04 Jan 2012 14:11:09 +0100 Subject: [Freeswitch-users] ENUM Message-ID: <4F044FED.4060205@softnet.si> Hi, I need a pit of help for setting up ENUM. My outbount calls are going throught just fine. What I need is that, number get from enum are using for dialing in my dial plan. In enum.conf I have changed:(xxx.xxx.xxx.xxx is dns of my enum): References: <4F044FED.4060205@softnet.si> Message-ID: <4F0455C5.9020106@softnet.si> Hi, I found a problem with wireshark, but having problem with fixing it: \"Extension 018108500\" ;tag=rcKpcgQ244D6g it should be like this: ;tag=rcKpcgQ244D6g do you know why this happens, if I make a call without enum this thing (from) is ok. regads, Miha On 1/4/2012 2:11 PM, Miha Zoubek wrote: > Hi, > > I need a pit of help for setting up ENUM. > > My outbount calls are going throught just fine. What I need is that, > number get from enum are using for dialing in my dial plan. > > In enum.conf I have changed:(xxx.xxx.xxx.xxx is dns of my enum): > > > > > replace="sofia/gateway/sbc_trunk/$1"/ > > > In my dial plan I have set: > > > > I get this is wrong as my calls not working:) > > > Please help me out how to use number from enum in dialplan. > > Regards, > Miha > -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/fea64208/attachment.html From avi at avimarcus.net Wed Jan 4 16:45:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jan 2012 15:45:31 +0200 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH In-Reply-To: References: Message-ID: 1) Breaking up the default dialplan into the pieces make it easier to get a handle on. I wonder if you should just include all the others as a .noload in a folder named "advanced" or something. 2) I don't see any CDR modules in the autoload configs. cdr_csv for basic and/or xml_cdr to store the calls with all the variables. It's particularly helpful for when you don't know the variables that you can just go and see all that are there, even after the call. 3) I wrote up a little bit about the regex for the incoming extension (first actual dialplan they might see?).. https://github.com/avimar/Freeswitch_slim_conf/commit/91c49d7cf31f3c9a569b0258726d8b6511e468fb Not sure it's worth including, though. 4) Q: Why do your's and the default configs have: Wouldn't this work just as well without the added loopback channel? Why isn't this the standard? -Avi Marcus On Wed, Jan 4, 2012 at 8:12 AM, Gabriel Gunderson wrote: > On Tue, Jan 3, 2012 at 10:03 PM, curriegrad2004 > wrote: > > It's been done and I have it uploaded on my github. Testing and > > feedback is greatly appreciated: > > Love it. > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/b8235da6/attachment.html From nvitaly at gmail.com Wed Jan 4 16:49:27 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Wed, 4 Jan 2012 08:49:27 -0500 Subject: [Freeswitch-users] mod limit for duplicate concurrent calls. In-Reply-To: <1325655711.39450.YahooMailNeo@web65302.mail.ac2.yahoo.com> References: <1325655711.39450.YahooMailNeo@web65302.mail.ac2.yahoo.com> Message-ID: Rodney, That exactly how it works in code i sent yesterday. it limits concurrent calls from phone to destination. second call to same destination number will get recording that limit exceeded and hangup. If you have more questions it time to show your configs :) On Wed, Jan 4, 2012 at 12:41 AM, Rodney wrote: > vitaly, is it possible to put this limit before it even gets to my > greeting/preamble? i don't even want them hitting the IVR. plus when does > that count reset? i want the counter to reset once they hangup all current > calls. > > -- -- Vitaly Nikolaev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/11f82e28/attachment-0001.html From adrottenberg at gmail.com Wed Jan 4 17:51:47 2012 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Wed, 4 Jan 2012 09:51:47 -0500 Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: References: <1324703154.9585.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: I have actually started working on a mod_gstream to use the gstreamer library. Gstreamer has plugins that support playback of the Microsoft formats. I am currently tied down with other projects therefore I was unable to continue working on the module. Are there others that are interested in such a module. On Tue, Jan 3, 2012 at 3:44 PM, Michael Collins wrote: > > > On Fri, Dec 23, 2011 at 9:05 PM, Sherif Omran wrote: > >> but the station i need is this >> >> mmsh://live.sis.gov.eg/live?MSWMExt=.asf >> >> and it is not there. I would appreciate your help >> > Bummer. > > ASF is a proprietary Microsoft technology, so I would contact the people > using it and let them know that they are limiting their potential audience > by using a locked-down container format. > > If you *really* want this then you can probably hire someone to do a > workaround. My guess is that you'll need to pay a fair amount because the > people with the requisite skills to do this will be in high demand. > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/508b7917/attachment.html From steveu at coppice.org Wed Jan 4 17:54:27 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 04 Jan 2012 22:54:27 +0800 Subject: [Freeswitch-users] g722 In-Reply-To: References: <4F03A6AC.9010000@coppice.org> <1325642011.60125.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: <4F046823.5010509@coppice.org> On 01/04/2012 12:12 PM, Michael Collins wrote: > > > On Tue, Jan 3, 2012 at 5:53 PM, Sherif Omran > > wrote: > > I get the following warning > > 2012-01-04 01 :52:14.363026 [CRIT] > switch_loadable_module.c:1281 Error Loading module > /usr/local/freeswitch/mod/mod_g722.so > **/usr/local/freeswitch/mod/mod_g722.so: cannot open shared object > file: No such file or directory** > > do you know how to avoid it? > > Use latest git? There is no mod_g722 in the latest git. You may have > updated from an older version and have an old modules.conf.xml file. > I'd recommend installing a fresh, clean git version with the default > configs and then re-integrate your customizations. > This guy must be using something ancient. There hasn't been a mod_g722 for 3 or 4 years. Steve From curriegrad2004 at gmail.com Wed Jan 4 18:13:20 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 4 Jan 2012 07:13:20 -0800 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH In-Reply-To: References: Message-ID: I personally don't think that a new user should be exposed to CDRs just yet. They should have more time to know what FreeSWITCH is all about until they can move on. I might need somebody to remind me to convert all of the CRLF endings to LF when I get back from school On 2012-01-04 5:46 AM, "Avi Marcus" wrote: > 1) Breaking up the default dialplan into the pieces make it easier to get > a handle on. > I wonder if you should just include all the others as a .noload in a > folder named "advanced" or something. > > 2) I don't see any CDR modules in the autoload configs. cdr_csv for basic > and/or xml_cdr to store the calls with all the variables. It's particularly > helpful for when you don't know the variables that you can just go and see > all that are there, even after the call. > > 3) I wrote up a little bit about the regex for the incoming extension > (first actual dialplan they might see?).. > > https://github.com/avimar/Freeswitch_slim_conf/commit/91c49d7cf31f3c9a569b0258726d8b6511e468fb > Not sure it's worth including, though. > > 4) Q: Why do your's and the default configs have: > > > Wouldn't this work just as well without the added loopback channel? Why > isn't this the standard? > > > -Avi Marcus > > > On Wed, Jan 4, 2012 at 8:12 AM, Gabriel Gunderson wrote: > >> On Tue, Jan 3, 2012 at 10:03 PM, curriegrad2004 >> wrote: >> > It's been done and I have it uploaded on my github. Testing and >> > feedback is greatly appreciated: >> >> Love it. >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/8668e07e/attachment.html From avi at avimarcus.net Wed Jan 4 18:17:09 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jan 2012 17:17:09 +0200 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH In-Reply-To: References: Message-ID: It tickled me to see a log of every call in fusionpbx as soon as I hung up... CDRs are pretty important. I think. -Avi On Wed, Jan 4, 2012 at 5:13 PM, curriegrad2004 wrote: > I personally don't think that a new user should be exposed to CDRs just > yet. They should have more time to know what FreeSWITCH is all about until > they can move on. I might need somebody to remind me to convert all of the > CRLF endings to LF when I get back from school > On 2012-01-04 5:46 AM, "Avi Marcus" wrote: > >> 1) Breaking up the default dialplan into the pieces make it easier to get >> a handle on. >> I wonder if you should just include all the others as a .noload in a >> folder named "advanced" or something. >> >> 2) I don't see any CDR modules in the autoload configs. cdr_csv for basic >> and/or xml_cdr to store the calls with all the variables. It's particularly >> helpful for when you don't know the variables that you can just go and see >> all that are there, even after the call. >> >> 3) I wrote up a little bit about the regex for the incoming extension >> (first actual dialplan they might see?).. >> >> https://github.com/avimar/Freeswitch_slim_conf/commit/91c49d7cf31f3c9a569b0258726d8b6511e468fb >> Not sure it's worth including, though. >> >> 4) Q: Why do your's and the default configs have: >> >> >> Wouldn't this work just as well without the added loopback channel? Why >> isn't this the standard? >> >> >> -Avi Marcus >> >> >> On Wed, Jan 4, 2012 at 8:12 AM, Gabriel Gunderson wrote: >> >>> On Tue, Jan 3, 2012 at 10:03 PM, curriegrad2004 >>> wrote: >>> > It's been done and I have it uploaded on my github. Testing and >>> > feedback is greatly appreciated: >>> >>> Love it. >>> >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/c72b0b0b/attachment-0001.html From lists.freeswitch.org at sjau.ch Wed Jan 4 17:19:51 2012 From: lists.freeswitch.org at sjau.ch (lists.freeswitch.org at sjau.ch) Date: Wed, 04 Jan 2012 15:19:51 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN Message-ID: <4F046007.1080601@sjau.ch> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there Soon the Raspberry Pis will be available and I'm planning to buy a bunch of them. I think with a 4GB SD Card, ethernet port and 2x USB port it should be fine. However in one office that I plan on setting up FS is still ISDN being used. There are no plans on changing to VoIP however having an internal FS system would be nice. My question is, how can I hookup the Raspberry Pi to the ISDN lines. I guess I'd need some USB<->ISDN Adapter. The only thing I could find this far were USB ISDN modems. But modems are for data transmission and not voice - at least that's what I think. So, any of you have suggestions how I can hook up the Raspberry Pi to ISDN? Thx Stephan -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk8EYAEACgkQ9KcSxJrwrxgIjgCfaMa4sYLN1TD17+YMXd7WY0YW /CUAnAgPR6v0Dr2RhM7D9aWhO5770wST =m666 -----END PGP SIGNATURE----- From anita.hall at simmortel.com Wed Jan 4 08:51:49 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 4 Jan 2012 11:21:49 +0530 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: References: Message-ID: Hi We have purchased around 20 licenses of G729 from FS and plans to buy 100 more. So, yes we are aware of official licenses from FS and the patent issues. Thanks, Anita. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/39bfce07/attachment.html From msc at freeswitch.org Wed Jan 4 20:06:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 4 Jan 2012 09:06:33 -0800 Subject: [Freeswitch-users] Originate with dead destination In-Reply-To: <30934ADE01DC4CEE8B3AEB5DF8602C94@sharad> References: <30934ADE01DC4CEE8B3AEB5DF8602C94@sharad> Message-ID: How does it mark it "dead"? Are you talking about a gateway in FreeSWITCH? -MC On Wed, Jan 4, 2012 at 4:44 AM, Sharad Garg wrote: > Hi friends, > > While using ORIGINATE API, if destination IP (local IP) is dead, FS marks > this IP as dead with him till that IP alives. > > Is there any way to configure the FS not to mark as dead destination. Let > it > send SIP INVITE even if last SIP INVITE was not answered. > > Thanks in advance. > > Regards > Sharad > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/a05eda75/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Jan 4 20:23:57 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 04 Jan 2012 18:23:57 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F046007.1080601@sjau.ch> References: <4F046007.1080601@sjau.ch> Message-ID: <4F048B2D.4020802@puzzled.xs4all.nl> On 04-01-12 15:19, lists.freeswitch.org at sjau.ch wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi there > > Soon the Raspberry Pis will be available and I'm planning to buy a > bunch of them. I think with a 4GB SD Card, ethernet port and 2x USB > port it should be fine. > > However in one office that I plan on setting up FS is still ISDN being > used. There are no plans on changing to VoIP however having an > internal FS system would be nice. > > My question is, how can I hookup the Raspberry Pi to the ISDN lines. I > guess I'd need some USB<->ISDN Adapter. The only thing I could find > this far were USB ISDN modems. But modems are for data transmission > and not voice - at least that's what I think. > > So, any of you have suggestions how I can hook up the Raspberry Pi to How about a beroNet gateway? http://www.beronet.com/product/berofix-gateways/ You did not mention if the office uses BRI or PRI lines. There used to be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal Adapters (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem available anymore. Afaik FreeSWITCH does not support them (a HFC BRI/PRI module would be cool) but they should work with the latest dahdi+zaphfc patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking of it maybe you could make lcr talk to FreeSWITCH but I never investigated that. Regards, Patrick From lists.freeswitch.org at sjau.ch Wed Jan 4 20:36:18 2012 From: lists.freeswitch.org at sjau.ch (lists.freeswitch.org at sjau.ch) Date: Wed, 04 Jan 2012 18:36:18 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F048B2D.4020802@puzzled.xs4all.nl> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> Message-ID: <4F048E12.1050408@sjau.ch> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I think it's BRI. Right now we can only have 4 incoming/outgoing calls. So I think it's 2 BRI Lines with 2 channels each. But how to find out for sure? Well, I'd prefer fully going the VoIP route but the others are sceptical. As for the rest, I'm completely lost. What's HFC? From what I read on the FS wiki there are ISDN modules: http://wiki.freeswitch.org/wiki/FreeTDM#ISDN_Modules . Would those help somehow? Thanks for the reply Stephan On 01/04/2012 06:23 PM, Patrick Lists wrote: > You did not mention if the office uses BRI or PRI lines. There used > to be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal > Adapters (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem > available anymore. Afaik FreeSWITCH does not support them (a HFC > BRI/PRI module would be cool) but they should work with the latest > dahdi+zaphfc patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking > of it maybe you could make lcr talk to FreeSWITCH but I never > investigated that. > > Regards, Patrick -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk8EjhEACgkQ9KcSxJrwrxgNDwCcCkNRNbHDusPWYkXY7tkLpsqe 9acAnRTrpjepL1IoNmO/T76DqeVlGa6h =vvrx -----END PGP SIGNATURE----- From curriegrad2004 at gmail.com Wed Jan 4 20:37:10 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 4 Jan 2012 09:37:10 -0800 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F048B2D.4020802@puzzled.xs4all.nl> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> Message-ID: A better choice would be to get a T1-E1 interface card over USB. What if we can use the GPIO pins? On 2012-01-04 9:24 AM, "Patrick Lists" wrote: > On 04-01-12 15:19, lists.freeswitch.org at sjau.ch wrote: > > -----BEGIN PGP SIGNED MESSAGE----- > > Hash: SHA1 > > > > Hi there > > > > Soon the Raspberry Pis will be available and I'm planning to buy a > > bunch of them. I think with a 4GB SD Card, ethernet port and 2x USB > > port it should be fine. > > > > However in one office that I plan on setting up FS is still ISDN being > > used. There are no plans on changing to VoIP however having an > > internal FS system would be nice. > > > > My question is, how can I hookup the Raspberry Pi to the ISDN lines. I > > guess I'd need some USB<->ISDN Adapter. The only thing I could find > > this far were USB ISDN modems. But modems are for data transmission > > and not voice - at least that's what I think. > > > > So, any of you have suggestions how I can hook up the Raspberry Pi to > > How about a beroNet gateway? > http://www.beronet.com/product/berofix-gateways/ > > You did not mention if the office uses BRI or PRI lines. There used to > be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal Adapters > (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem available > anymore. Afaik FreeSWITCH does not support them (a HFC BRI/PRI module > would be cool) but they should work with the latest dahdi+zaphfc > patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking of it maybe you > could make lcr talk to FreeSWITCH but I never investigated that. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/3d7b9c6d/attachment.html From curriegrad2004 at gmail.com Wed Jan 4 20:38:57 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 4 Jan 2012 09:38:57 -0800 Subject: [Freeswitch-users] g729 sound format for Intel IPP codecs In-Reply-To: References: Message-ID: Try mod_native_file. It does what you want it to do. On 2012-01-04 9:03 AM, "Anita Hall" wrote: > Hi > > We have purchased around 20 licenses of G729 from FS and plans to buy 100 > more. So, yes we are aware of official licenses from FS and the patent > issues. > > Thanks, > Anita. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/e7ccb742/attachment.html From gmaruzz at gmail.com Wed Jan 4 20:53:58 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 4 Jan 2012 18:53:58 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> Message-ID: They got BRI, almost for sure. Try to look into Fritz products, or other German products, they're very strong on BRI. Maybe an embedded BRI<->SIP gateway will fit nicely your bill. -giovanni On Wed, Jan 4, 2012 at 6:37 PM, curriegrad2004 wrote: > A better choice would be to get a T1-E1 interface card over USB. What if we > can use the GPIO pins? > > On 2012-01-04 9:24 AM, "Patrick Lists" > wrote: >> >> On 04-01-12 15:19, lists.freeswitch.org at sjau.ch wrote: >> > -----BEGIN PGP SIGNED MESSAGE----- >> > Hash: SHA1 >> > >> > Hi there >> > >> > Soon the Raspberry Pis will be available and I'm planning to buy a >> > bunch of them. I think with a 4GB SD Card, ethernet port and 2x USB >> > port it should be fine. >> > >> > However in one office that I plan on setting up FS is still ISDN being >> > used. There are no plans on changing to VoIP however having an >> > internal FS system would be nice. >> > >> > My question is, how can I hookup the Raspberry Pi to the ISDN lines. I >> > guess I'd need some USB<->ISDN Adapter. The only thing I could find >> > this far were USB ISDN modems. But modems are for data transmission >> > and not voice - at least that's what I think. >> > >> > So, any of you have suggestions how I can hook up the Raspberry Pi to >> >> How about a beroNet gateway? >> http://www.beronet.com/product/berofix-gateways/ >> >> You did not mention if the office uses BRI or PRI lines. There used to >> be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal Adapters >> (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem available >> anymore. Afaik FreeSWITCH does not support them (a HFC BRI/PRI module >> would be cool) but they should work with the latest dahdi+zaphfc >> patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking of it maybe you >> could make lcr talk to FreeSWITCH but I never investigated that. >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From yehavi.bourvine at gmail.com Wed Jan 4 20:55:27 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 4 Jan 2012 19:55:27 +0200 Subject: [Freeswitch-users] Debugging intermitent BLF problems In-Reply-To: References: Message-ID: Hi, It is bugID 3794. Anthony did some changes which I'll test tomorrow. He also suggested to watch the sip_subscriptions table and have a TCPdump of the phone's signalling. Regards, __Yehavi: 2012/1/4 Christian Benke > On 2 January 2012 15:43, Yehavi Bourvine > wrote: > > It was rock-solid until mid-October (and that's why I have to stay at > that > > version in the meantime). > > > > I am willing to help debugging this, but need to know which debugs to add > > which will be usefull... > > > Since no one else answers: I guess the standard debug-log plus > SIP-Debugging(NOTIFY-messages) should suffice. If it's not enough, > you'll be told how to improve after you've filed the bug-report. Let > us know the bug-id and i'll contribute if needed. > > Best regards > Christian > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/1fe25d07/attachment.html From lists.freeswitch.org at sjau.ch Wed Jan 4 21:10:36 2012 From: lists.freeswitch.org at sjau.ch (lists.freeswitch.org at sjau.ch) Date: Wed, 04 Jan 2012 19:10:36 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F048B2D.4020802@puzzled.xs4all.nl> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> Message-ID: <4F04961C.2080205@sjau.ch> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Addon: How would I connect the berofix gateway to the raspberry pi board? Btw, I just search a bit for usb isdn 128 TAs and I came accross a Minivigor 128 ( http://www.draytek.de/produkte/archiv/minivigor128.html ). Could that work? I could try to obtain such a device and give SSH access to interested parties. Stephan On 01/04/2012 06:23 PM, Patrick Lists wrote: > How about a beroNet gateway? > http://www.beronet.com/product/berofix-gateways/ > > You did not mention if the office uses BRI or PRI lines. There used > to be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal > Adapters (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem > available anymore. Afaik FreeSWITCH does not support them (a HFC > BRI/PRI module would be cool) but they should work with the latest > dahdi+zaphfc patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking > of it maybe you could make lcr talk to FreeSWITCH but I never > investigated that. > > Regards, Patrick -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk8ElhsACgkQ9KcSxJrwrxj1AgCgw0QxWqGoDlrM0KpRewQvNINJ ohsAn0IgWmDOrbBh3oGoace9WME/lZzt =bnGq -----END PGP SIGNATURE----- From georg at riseup.net Wed Jan 4 21:15:53 2012 From: georg at riseup.net (georg at riseup.net) Date: Wed, 4 Jan 2012 19:15:53 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F04961C.2080205@sjau.ch> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> <4F04961C.2080205@sjau.ch> Message-ID: <5ce1889de8e0f393d6b786dd0aee0174.squirrel@fulvetta.riseup.net> He Stephan, As far as I know, these devices are normally connected via ethernet. The gateways of Patton Inalp (aka "Smartnode") are also great. Georg > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Addon: How would I connect the berofix gateway to the raspberry pi board? > > Btw, I just search a bit for usb isdn 128 TAs and I came accross a > Minivigor 128 ( > http://www.draytek.de/produkte/archiv/minivigor128.html ). Could that > work? > > I could try to obtain such a device and give SSH access to interested > parties. > > Stephan > > On 01/04/2012 06:23 PM, Patrick Lists wrote: >> How about a beroNet gateway? >> http://www.beronet.com/product/berofix-gateways/ >> >> You did not mention if the office uses BRI or PRI lines. There used >> to be cheap USB HFC chipset based ISDN BRI (not PRI) Terminal >> Adapters (e.g. Billion tinyUSB ISDN TA 128k) but they don't seem >> available anymore. Afaik FreeSWITCH does not support them (a HFC >> BRI/PRI module would be cool) but they should work with the latest >> dahdi+zaphfc patch/libpri/asterisk or mISDN/lcr/asterisk. Thinking >> of it maybe you could make lcr talk to FreeSWITCH but I never >> investigated that. >> >> Regards, Patrick > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.11 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk8ElhsACgkQ9KcSxJrwrxj1AgCgw0QxWqGoDlrM0KpRewQvNINJ > ohsAn0IgWmDOrbBh3oGoace9WME/lZzt > =bnGq > -----END PGP SIGNATURE----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lists.freeswitch.org at sjau.ch Wed Jan 4 21:28:59 2012 From: lists.freeswitch.org at sjau.ch (lists.freeswitch.org at sjau.ch) Date: Wed, 04 Jan 2012 19:28:59 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <5ce1889de8e0f393d6b786dd0aee0174.squirrel@fulvetta.riseup.net> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> <4F04961C.2080205@sjau.ch> <5ce1889de8e0f393d6b786dd0aee0174.squirrel@fulvetta.riseup.net> Message-ID: <4F049A6B.8040500@sjau.ch> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Georg I see.... well, they kinda defeat the purpose of a Raspberry Pi. Instead of buying a Beronet box I could just make a full PC instead. The beauty of the Raspberry Pi lies in it's size, pricetag and power consumption. Thanks for your explanation. Stephan On 01/04/2012 07:15 PM, georg at riseup.net wrote: > He Stephan, > > As far as I know, these devices are normally connected via > ethernet. The gateways of Patton Inalp (aka "Smartnode") are also > great. > > Georg -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.11 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk8EmmoACgkQ9KcSxJrwrxgERgCZAa/KE7XHszqAbOepIGnB38KI yaQAn0DVXwSo4OXVlzehr+CDpEZ2UYa2 =dwxa -----END PGP SIGNATURE----- From freeswitch-list at puzzled.xs4all.nl Wed Jan 4 21:56:24 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 04 Jan 2012 19:56:24 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F048E12.1050408@sjau.ch> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> <4F048E12.1050408@sjau.ch> Message-ID: <4F04A0D8.9070404@puzzled.xs4all.nl> Please do not top post. On 04-01-12 18:36, lists.freeswitch.org at sjau.ch wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > I think it's BRI. Right now we can only have 4 incoming/outgoing > calls. So I think it's 2 BRI Lines with 2 channels each. > > But how to find out for sure? Check the invoice or the actual phone setup? > Well, I'd prefer fully going the VoIP route but the others are sceptical. > > As for the rest, I'm completely lost. What's HFC? From what I read on HFC is an ISDN chipset made by Cologne http://www.colognechip.com > the FS wiki there are ISDN modules: > http://wiki.freeswitch.org/wiki/FreeTDM#ISDN_Modules . Would those > help somehow? Sangoma ISDN module is for Sangoma products which it clearly mentions on the wiki ("only supported with Sangoma cards"). If you have no experience with/knowledge of integrating ISDN into FreeSWITCH and/or Asterisk then you may be better off with something like a beroNet gateway or hire a Consultant. Regards, Patrick From freeswitch-list at puzzled.xs4all.nl Wed Jan 4 21:57:45 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 04 Jan 2012 19:57:45 +0100 Subject: [Freeswitch-users] Raspberry Pi & ISDN In-Reply-To: <4F04961C.2080205@sjau.ch> References: <4F046007.1080601@sjau.ch> <4F048B2D.4020802@puzzled.xs4all.nl> <4F04961C.2080205@sjau.ch> Message-ID: <4F04A129.4030602@puzzled.xs4all.nl> Please do not top post. On 04-01-12 19:10, lists.freeswitch.org at sjau.ch wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Addon: How would I connect the berofix gateway to the raspberry pi board? You would use a LAN connection. > Btw, I just search a bit for usb isdn 128 TAs and I came accross a > Minivigor 128 ( > http://www.draytek.de/produkte/archiv/minivigor128.html ). Could that > work? If it has a HFC chipset then yes. If not then no. Call Draytek and find out. > I could try to obtain such a device and give SSH access to interested > parties. If you want to hire a Consultant then contact the FreeSWITCH developer team (I have no affiliation). Email consulting at freeswitch.org or call +1-918-420-9266 or tollfree +1-877-742-2583. Regards. Patrick From nbhatti at gmail.com Wed Jan 4 22:14:11 2012 From: nbhatti at gmail.com (nbhatti) Date: Wed, 4 Jan 2012 11:14:11 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! - Review In-Reply-To: <1325638928.76279.YahooMailClassic@web110805.mail.gq1.yahoo.com> References: <1325549865328-7145235.post@n2.nabble.com> <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> <1325638928.76279.YahooMailClassic@web110805.mail.gq1.yahoo.com> Message-ID: And you would also have to change .htaccess On Wed, Jan 4, 2012 at 4:08 AM, sherif omran [via freeswitch-users] < ml-node+s2379917n7148890h48 at n2.nabble.com> wrote: > Hi every body, > > I had a chance 3 days to install vBilling manually, which was not so > trivial and would like to share my experience with you. > > Installation script is clearly written, any developer with some linux > experience can follow it and do the required changes. In fact, I did not > use the installation script because I have custom modules enabled. Also, I > use the freeswitch xml tree, which is different that what the script do. > Additionally, I have BlueBox installed and they should work together. > > The installation script creates a custom freeswitch.xml file with the > required modules. If these freeswitch modules were not previously > installed, you have to install them manually. No need to reinstall > freeswitch again using (make install) > > enable them in the freeswitch source and compile using > make modulename-install > > Regarding the installation path: > I used Centos 6 Server > > The software uses the /var/www/html/ folder as the base path. However, > since i have bluebox installed, I tried to do the required changes in > (freeswitch.xml) and install vBilling in /var/www/html/vBilling. However, > after contacting Muhammed, he recommended to enabled mod-rewrite for the > apache server. After checking out, I found that it was already enabled for > centos using the .htaccess file. I could log to the front page but 404 > error pops, if i enter the password leading to another path. I checked the > login php function and corrected the path. At least i could see that after > login click, it was trying to call a page from the correct subfolder. > However, I would not recommend to install it in a subfolder because it will > not work properly. May be this needs some additional investement. When i > placed it in the root web server folder, and made the required adjustments > it worked fine. > > During the installation, I had to create a database using the given > script, however I changed the username and password in the mysql tag. It > returned that i could not login to the frontpage, since the password in > encrypted. I had then to install phpmyadmin to revert the changes and > change the password from the frontpage. > > It is now running fine. However, I still did not start playing with its > reporting functions or any call log but it looks very promising. > > If i install some script, I usually check for spying functions such as > sending back precious information without a permission to the author. > However, the software is really clean. I did not find any spying functions > or backholes. I checked the PHP files as well. > > > Finally, i would like to thank Muhammed Bhatti for sharing this nice > software with us. > > kind regards, > Sherif Omran > > +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > Sherif Omran Dr. sc. nat. > (Title from University of Zurich) > Signal and Image Processing, Acoustics, Artificial Intelligence Engineer > and Neural Scientist. Design, Modeling and Simulation. Expert in Biomedical > devices and Cochlear Implants. Telecommunication Consultant and ERICSSON > Certified Engineer. > Munich - Germany > e-mail: [hidden email] > +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > > > --- On *Tue, 1/3/12, Muhammad Naseer Bhatti <[hidden email] > >* wrote: > > > From: Muhammad Naseer Bhatti <[hidden email] > > > > Subject: Re: [Freeswitch-users] vBilling Beta Program!! > To: "FreeSWITCH Users Help" <[hidden email] > > > > Date: Tuesday, January 3, 2012, 8:16 PM > > > Sherif, have you installed manually? If so, you would have to enable > mod_rewrite in your apache configuration. > > > On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran > > wrote: > > Can you give more details how it does not work? I have the same situation. > I can reach the frontpage and when i give the username i get 404. > > Is this the case you have? > > Do you have Centos or Redhat ? > > > regards, > Sherif > > > --- On *Tue, 1/3/12, Zenny > >* wrote: > > > From: Zenny > > > Subject: Re: [Freeswitch-users] vBilling Beta Program!! > To: "FreeSWITCH Users Help" > > > > Date: Tuesday, January 3, 2012, 11:18 AM > > > looks promising, but the user login does not work. > > Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to > all freeswitchers! > > On 1/3/12, nbhatti > > wrote: > > Yes, it will support prepaid calling card and many more features soon. > > > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > > > > wrote: > >> It looks great! > >> > >> Will vBilling support batch user/prepaid calling card creation? > >> > >> ________________________________ > >> If you reply to this email, your message will be added to the discussion > >> below: > >> > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html > >> To unsubscribe from vBilling Beta Program!!, click here. > >> NAML > > > > > > -- > > View this message in context: > > > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7148890.html > To unsubscribe from vBilling Beta Program!!, click here > . > NAML > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7151552.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/d8b1b609/attachment-0001.html From peter at uringme.com Wed Jan 4 22:52:50 2012 From: peter at uringme.com (peter at uringme.com) Date: Wed, 4 Jan 2012 11:52:50 -0800 (PST) Subject: [Freeswitch-users] X-Lite and Video Message-ID: <1325706770.19930.YahooMailClassic@web2808.biz.mail.ne1.yahoo.com> (Apologies to the list owner for originally sending this to the wrong address) I have two laptops running X-Lite 4.? I have them registered to a FreeSwitch server (latest git) as extensions 7777 and 7778.? I have a dialplan for each (quick and dirty) that just bridges them when one is dialed from the other: ? ????????? ????????? ????????? ??? (and vice-versa for 7778). I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem. I'm trying to get video going.? Both X-Lites have H.263 and H.263-1998 enabled in their settings.? Freeswitch has the following in vars.xml: ? When I try to make a video call from one extension to the other, the calling extension seems to think it's in video, but the called extension doesn't. INVITE from freeswitch console: ?? ------------------------------------------------------------------------ ?? INVITE sip:7778 at test SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport ?? Max-Forwards: 70 ?? Contact: ?? To: ?? From: "Peter Test";tag=a7031d83 ?? Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ. ?? CSeq: 1 INVITE ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ?? Content-Type: application/sdp ?? Supported: replaces ?? User-Agent: X-Lite 4 release 4.1 stamp 63214 ?? Content-Length: 681 ?? v=0 ?? o=- 12970176286658638 1 IN IP4 192.168.1.7 ?? s=CounterPath X-Lite 4.1 ?? c=IN IP4 192.168.1.7 ?? t=0 0 ?? a=ice-ufrag:20fef6 ?? a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb ?? m=audio 58632 RTP/AVP 107 0 8 101 ?? a=rtpmap:107 BV32/16000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-15 ?? a=sendrecv ?? a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host ?? a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host ?? m=video 50994 RTP/AVP 34 115 ?? a=rtpmap:34 H263/90000 ?? a=fmtp:34 QCIF=2;CIF=2;VGA=2 ?? a=rtpmap:115 H263-1998/90000 ?? a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1 ?? a=sendrecv ?? a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host ?? a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host I do see freeswitch seeing the audio and video codecs: 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4683 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:2800 Set Codec sofia/external/7777 at test PCMU/8000 20 ms 160 samples 64000 bits 2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777 at test) State NEW 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34] However, when freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE: ?? INVITE sip:7778 at 68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0 ?? Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 ?? Max-Forwards: 69 ?? From: "Peter Test" ;tag=ZeZUav5XXat5e ?? To: ?? Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc ?? CSeq: 22515274 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, hold, refer ?? Content-Type: application/sdp ?? Content-Disposition: session ?? Content-Length: 207 ?? X-FS-Support: update_display ?? Remote-Party-ID: "Peter Test" ;party=calling;screen=yes;privacy=off ?? v=0 ?? o=FreeSWITCH 1325690959 1325690960 IN IP4 204.13.175.89 ?? s=FreeSWITCH ?? c=IN IP4 204.13.175.89 ?? t=0 0 ?? m=audio 11718 RTP/AVP 0 8 3 101 13 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=ptime:20 So, am I missing something?? Do I need to use something other than "bridge" in the dialplan, or do I need to add some variables to be able to pass on the video?? All I'm trying to do is make a video call between two X-Lites that are locally SIP registered to freeswitch.? Because I want to record the video at some point in the future, I don't want to divert the media -- I want it streaming/passing through freeswitch. When the call is connected, the caller shows a "Waiting for video", but the called doesn't show this.? When I try to start the video, it says "Failed to Start Video". -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/373c2c20/attachment.html From brandon.mcginty at gmail.com Thu Jan 5 01:07:32 2012 From: brandon.mcginty at gmail.com (Brandon McGinty) Date: Wed, 04 Jan 2012 17:07:32 -0500 Subject: [Freeswitch-users] Flite speed and pitch increase after update. Message-ID: <4F04CDA4.3070707@gmail.com> Hi everyone. We upgraded Freeswitch on the 30th of December, using the git branch revision from the 28th of December. However, while the conference features improved, the flite package has had its pitch and rate increased for all tts speech. We are using the mod_tts module in our conferences, and it now sounds extrordinarily sped up and pitched up, similar to what one would consider a chipmunk sound. I've gone through the mod_tts sources, the flite sources, the modules directory, the autoload_modules directory, the configs for the conference and flite, and I can't find anywhere how to change the pitch and rate of the synthesizer. I've tried tts-rate and tts-pitch, etc, to no affect. I've also tried google, and noone else seems to be dealing with this strangeness. We are all using soft-phones or pots to sip devices, so the speech output is quite important. Any help you all can provide would be truly be appreciated. Brandon McGinty-Carroll From avi at avimarcus.net Thu Jan 5 01:24:22 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 5 Jan 2012 00:24:22 +0200 Subject: [Freeswitch-users] Flite speed and pitch increase after update. In-Reply-To: <4F04CDA4.3070707@gmail.com> References: <4F04CDA4.3070707@gmail.com> Message-ID: Perhaps related: while in the old version, imo, "Kal" sounded the best.. on the box I just built kal sounds like a chipmunk and the rest sound normal ("rms" sounded decent). -Avi On Thu, Jan 5, 2012 at 12:07 AM, Brandon McGinty wrote: > Hi everyone. > We upgraded Freeswitch on the 30th of December, using the git branch > revision from the 28th of December. However, while the conference > features improved, the flite package has had its pitch and rate > increased for all tts speech. We are using the mod_tts module in our > conferences, and it now sounds extrordinarily sped up and pitched up, > similar to what one would consider a chipmunk sound. > I've gone through the mod_tts sources, the flite sources, the modules > directory, the autoload_modules directory, the configs for the > conference and flite, and I can't find anywhere how to change the pitch > and rate of the synthesizer. I've tried tts-rate and tts-pitch, etc, to > no affect. I've also tried google, and noone else seems to be dealing > with this strangeness. > We are all using soft-phones or pots to sip devices, so the speech > output is quite important. > Any help you all can provide would be truly be appreciated. > > Brandon McGinty-Carroll > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/62897827/attachment.html From bdfoster at endigotech.com Thu Jan 5 01:24:36 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 4 Jan 2012 17:24:36 -0500 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 7 In-Reply-To: <8D24ACE29A4C446B981F1BF4818F7923@sharad> References: <8D24ACE29A4C446B981F1BF4818F7923@sharad> Message-ID: When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." -Brian On Jan 4, 2012 12:05 AM, "Sharad Garg" wrote: > > All these are basically IP based PBX systems having PBX features. > > Regards > > ----- Original Message ----- > From: > To: > Sent: Tuesday, January 03, 2012 7:47 PM > Subject: FreeSWITCH-users Digest, Vol 67, Issue 7 > > > > Send FreeSWITCH-users mailing list submissions to > > freeswitch-users at lists.freeswitch.org > > > > To subscribe or unsubscribe via the World Wide Web, visit > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > or, via email, send a message with subject or body 'help' to > > freeswitch-users-request at lists.freeswitch.org > > > > You can reach the person managing the list at > > freeswitch-users-owner at lists.freeswitch.org > > > > When replying, please edit your Subject line so it is more specific > > than "Re: Contents of FreeSWITCH-users digest..." > > > > > -------------------------------------------------------------------------------- > > > > Today's Topics: > > > > 1. Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Kaushal Shriyan) > > 2. Re: mod_sofia not binding to port 5060 (Brandon McGinty) > > 3. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Zohair Raza) > > 4. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Giovanni Maruzzelli) > > 5. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Kaushal Shriyan) > > 6. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Zohair Raza) > > 7. Re: Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, > > CallWeaver, and YATE. (Vitaly Nikolaev) > > > > > -------------------------------------------------------------------------------- > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/3934dfbe/attachment-0001.html From darknesslabs at gmail.com Thu Jan 5 02:05:06 2012 From: darknesslabs at gmail.com (Karol) Date: Wed, 4 Jan 2012 18:05:06 -0500 Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: <4F025951.2040009@googlemail.com> References: <4F025951.2040009@googlemail.com> Message-ID: Thanks all. I just wanted to see, if anyone else is watching logs. On Jan 2, 2012 8:32 PM, "Tamer Higazi" wrote: > Or better! > Make all clients behind static ip addresses available, and limit the > registration with those ip-addresses. > > > > Tamer > > Am 03.01.2012 00:26, schrieb Brian Foster: > > This is just part of having a switch open to the world. We get these > > registration attempts on a daily, almost hourly basis. It is your > > responsibility to make sure your system is secured. Please look at > > fail2ban and iptables as ways to protect your system. If you don't, it > > could cost you big money. > > > > On Jan 2, 2012 4:59 PM, "Karol" > > wrote: > > > > Anyone else see this?? > > > > http://pastebin.freeswitch.com/18078 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/2082f21d/attachment.html From darknesslabs at gmail.com Thu Jan 5 02:07:56 2012 From: darknesslabs at gmail.com (Karol) Date: Wed, 4 Jan 2012 18:07:56 -0500 Subject: [Freeswitch-users] Torrents for the weekly conference Message-ID: Has everyone given up on this method? I have 3 torrents frozen, in their tracks. - Karol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/c57ce107/attachment.html From jeff at jefflenk.com Thu Jan 5 02:27:44 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 4 Jan 2012 15:27:44 -0800 (PST) Subject: [Freeswitch-users] Flite speed and pitch increase after update. In-Reply-To: <4F04CDA4.3070707@gmail.com> References: <4F04CDA4.3070707@gmail.com> Message-ID: <1325719664163-7152479.post@n2.nabble.com> Please open a Jira and document the problem so it doesnt get forgotten. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Flite-speed-and-pitch-increase-after-update-tp7152216p7152479.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Thu Jan 5 03:38:39 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 4 Jan 2012 18:38:39 -0600 Subject: [Freeswitch-users] Flite speed and pitch increase after update. In-Reply-To: <1325719664163-7152479.post@n2.nabble.com> References: <4F04CDA4.3070707@gmail.com> <1325719664163-7152479.post@n2.nabble.com> Message-ID: confirmed this is the kal voice specific, slt and rms both work This is an update to the actual flite lib, we will have to report it upstream. On Wed, Jan 4, 2012 at 5:27 PM, Jeff Lenk wrote: > Please open a Jira and document the problem so it doesnt get forgotten. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Flite-speed-and-pitch-increase-after-update-tp7152216p7152479.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120104/fff8820c/attachment.html From olimonkey at gmail.com Thu Jan 5 10:06:26 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Thu, 5 Jan 2012 15:06:26 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: *bump* So I think maybe the way I'm doing the originate is the problem? In my call string I'm creating a connection directly from the CISCO (192.168.x.x) to the managed application, which may be why it starts playing straight away? Maybe I should be originating a call first and then only once I know the other side has picked up will I bridge the call to the IVR managed application. Problem is I dunno how to tell whether the other person has picked up (or even if the cisco is going to tell me) and I don't know how to do things to a call once it has been established. I'm currently reading the Dialplan wiki page, hoping to get something out of it there. Cheers Oliver On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: > I've been battling while creating an IVR using FreeSWITCH mod_managed > and connecting through a CISCO 2811. Most things now work quite well, > but I am having a few issues with the way the system answers calls (or > doesn't answer calls...). > > I have FreeSWITCH running as a windows service on Windows Server 2008, > which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > which is then connected to a POTS phone line. > > > Take the following scenario: > > 1. Managed .NET application creates a call string and uses ESL to talk > to freeswitch and originate a call: > > string callstring = > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x > '&managed(ivrAppName)'"; > eslConnection.API("originate", callstring); > > where 192.168.x.x is the CISCO IP. > > 2. The CISCO sees that the phone number (1091234567) starts with a 1 > so it uses FXO port 1 and strips the 1 and uses the remaining phone > number (091234567) to make the call. > > 3. My phone rings, I pick up and I can hear my IVR playing. > > > > These are my current problems: > > - IVR starts playing before I even pick up the phone. This means that > if the system calls a mobile phone and the person doesn't pick up, the > IVR will start playing and eventually the mobile phone will divert to > voice mail. Obviously I then get a missed call and an sms saying I > have a new voice mail, which is annoying. Instead I would like it to > KNOW that no one has picked up, but I don't know how to do this. > Somehow the CISCO needs to be able to tell FreeSWITCH that the call > has not yet been answered. For some reason however as soon as the > CISCO starts calling FreeSWITCH thinks the call is already connected. > It doesn't know that the CISCO is actually still ringing. Maybe I'm > doing originate the wrong way or something ... > > - The phone only rings for about 10 seconds before hanging up. I've > tried "call_timeout", "bridge_answer_timeout". I've also tried setting > CISCO "ring number". Nothing works, my phone still only rings for > about 10 seconds. I don't know if this is a FreeSWITCH issue or a > CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > starts playing even if no one answers the phone. > > > > > > CISCO Config for relevant FXO port: > > voice service voip > ?allow-connections h323 to h323 > ?allow-connections h323 to sip > ?allow-connections sip to h323 > ?allow-connections sip to sip > ?no supplementary-service h450.2 > ?no supplementary-service h450.3 > ?supplementary-service h450.12 > ?no supplementary-service sip moved-temporarily > ?no supplementary-service sip refer > ?fax protocol cisco > ?sip > ?registrar server expires max 3600 min 3600 > ?no update-callerid > ?no call service stop > > voice-port 0/3/2 > ?output attenuation -3 > ?no comfort-noise > ?cptone AU > ?impedance complex1 > ?caller-id enable > ! > dial-peer voice 100 pots > ?preference 1 > ?destination-pattern 1T > ?port 0/3/2 > ! > > > > Many Thanks, > > Oliver From olimonkey at gmail.com Thu Jan 5 10:17:56 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Thu, 5 Jan 2012 15:17:56 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: Also, maybe I should be doing something like this: sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' instead of: sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' but, I don't really have the CISCO configured as a gateway, nor do I know how really...probably not on the right track there. On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: > *bump* > > > So I think maybe the way I'm doing the originate is the problem? In my > call string I'm creating a connection directly from the CISCO > (192.168.x.x) to the managed application, which may be why it starts > playing straight away? > > Maybe I should be originating a call first and then only once I know > the other side has picked up will I bridge the call to the IVR managed > application. > > Problem is I dunno how to tell whether the other person has picked up > (or even if the cisco is going to tell me) and I don't know how to do > things to a call once it has been established. > > > I'm currently reading the Dialplan wiki page, hoping to get something > out of it there. > > > Cheers > > Oliver > > > On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> and connecting through a CISCO 2811. Most things now work quite well, >> but I am having a few issues with the way the system answers calls (or >> doesn't answer calls...). >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> which is then connected to a POTS phone line. >> >> >> Take the following scenario: >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> to freeswitch and originate a call: >> >> string callstring = >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >> '&managed(ivrAppName)'"; >> eslConnection.API("originate", callstring); >> >> where 192.168.x.x is the CISCO IP. >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> number (091234567) to make the call. >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> These are my current problems: >> >> - IVR starts playing before I even pick up the phone. This means that >> if the system calls a mobile phone and the person doesn't pick up, the >> IVR will start playing and eventually the mobile phone will divert to >> voice mail. Obviously I then get a missed call and an sms saying I >> have a new voice mail, which is annoying. Instead I would like it to >> KNOW that no one has picked up, but I don't know how to do this. >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> has not yet been answered. For some reason however as soon as the >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> doing originate the wrong way or something ... >> >> - The phone only rings for about 10 seconds before hanging up. I've >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> CISCO "ring number". Nothing works, my phone still only rings for >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> starts playing even if no one answers the phone. >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> voice service voip >> ?allow-connections h323 to h323 >> ?allow-connections h323 to sip >> ?allow-connections sip to h323 >> ?allow-connections sip to sip >> ?no supplementary-service h450.2 >> ?no supplementary-service h450.3 >> ?supplementary-service h450.12 >> ?no supplementary-service sip moved-temporarily >> ?no supplementary-service sip refer >> ?fax protocol cisco >> ?sip >> ?registrar server expires max 3600 min 3600 >> ?no update-callerid >> ?no call service stop >> >> voice-port 0/3/2 >> ?output attenuation -3 >> ?no comfort-noise >> ?cptone AU >> ?impedance complex1 >> ?caller-id enable >> ! >> dial-peer voice 100 pots >> ?preference 1 >> ?destination-pattern 1T >> ?port 0/3/2 >> ! >> >> >> >> Many Thanks, >> >> Oliver From miha at softnet.si Thu Jan 5 10:19:22 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 05 Jan 2012 08:19:22 +0100 Subject: [Freeswitch-users] Password in dialplan Message-ID: <4F054EFA.2000004@softnet.si> Hi, in dial plan I have this line (for radius): Deafult password is set to 1234. As I do not wont default password for radius, I would just like to have here password which is set for user in directory. How to set in dialplan that the password will be taken from user/dir? Like this? Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/187f8ede/attachment.html From sharad at coraltele.com Thu Jan 5 07:53:06 2012 From: sharad at coraltele.com (Sharad Garg) Date: Thu, 5 Jan 2012 10:23:06 +0530 Subject: [Freeswitch-users] Originate with dead destination References: <30934ADE01DC4CEE8B3AEB5DF8602C94@sharad> Message-ID: <728FEDA19D644DE08E9CD193A94515CD@sharad> Hi Thanks a lot for your kind reply... I am sorry not to explain my query in detail.. I am using sip_profile 'external' for making point to point SIP call (without registration). Everything is working perfectly except one point. Point is.....if detination host (the url used in originate API), is switched-off, Freeswitch does not get any response from destination host. In this case, Freeswitch sends 4-5 SIP INVITE & revert back with error message. Now when I use the same originate API to the same url, Freeswitch does not generate the SIP Invite to this url. In this case, Freeswitch diretly reverts back with error without making the SIP INVITE to the defined url. I had waited for 10-15 minutes but every time FS reverted back with error message without generating INVITE. When the defined host is switched-on, there is some broadcast "netbios" message on network. I think from this broadcast message, FS comes to know that the host has become live now. So if I make originate to the same url, this time FS generates the INVITE. I might be wrong but this is what I analyzed. If I am wrong, plz correct me. So from this, I understood that if a destination host is switched-off, FS marks that host as `dead' with him till he gets some broadcast message from that host. So my query is...is there any way to configure FS so that it will generate the SIP invite on every ORIGINATE API even if host is not live. Please note that in my testing the host is a PC on which a softphone EYEBEAM 1.5 is installed on 5060 & it is configured without Registration. Means the url in my ORIGINATE is for a unregistered end point. Hope it is clear. Best Regards Sharad ----- Original Message ----- From: Michael Collins To: FreeSWITCH Users Help Sent: Wednesday, January 04, 2012 10:36 PM Subject: Re: [Freeswitch-users] Originate with dead destination How does it mark it "dead"? Are you talking about a gateway in FreeSWITCH? -MC On Wed, Jan 4, 2012 at 4:44 AM, Sharad Garg wrote: Hi friends, While using ORIGINATE API, if destination IP (local IP) is dead, FS marks this IP as dead with him till that IP alives. Is there any way to configure the FS not to mark as dead destination. Let it send SIP INVITE even if last SIP INVITE was not answered. Thanks in advance. Regards Sharad _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/b3f911a0/attachment.html From peter.olsson at visionutveckling.se Thu Jan 5 10:36:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Jan 2012 07:36:20 +0000 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F1502B9B8@cantor.std.visionutv.se> I think you're doing it right. It seems to me that the Cisco responds with 200 OK, and lets FS believe the call was answered, even though it was not really answered. You will need to check the settings in the Cisco. Also trace the SIP packets and log during a call, and pastebin it, and someone here can take a look on it. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Oliver Schenk Skickat: den 5 januari 2012 08:06 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR *bump* So I think maybe the way I'm doing the originate is the problem? In my call string I'm creating a connection directly from the CISCO (192.168.x.x) to the managed application, which may be why it starts playing straight away? Maybe I should be originating a call first and then only once I know the other side has picked up will I bridge the call to the IVR managed application. Problem is I dunno how to tell whether the other person has picked up (or even if the cisco is going to tell me) and I don't know how to do things to a call once it has been established. I'm currently reading the Dialplan wiki page, hoping to get something out of it there. Cheers Oliver On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: > I've been battling while creating an IVR using FreeSWITCH mod_managed > and connecting through a CISCO 2811. Most things now work quite well, > but I am having a few issues with the way the system answers calls (or > doesn't answer calls...). > > I have FreeSWITCH running as a windows service on Windows Server 2008, > which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > which is then connected to a POTS phone line. > > > Take the following scenario: > > 1. Managed .NET application creates a call string and uses ESL to talk > to freeswitch and originate a call: > > string callstring = > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sof > ia/internal/1091234567 at 192.168.x.x > '&managed(ivrAppName)'"; > eslConnection.API("originate", callstring); > > where 192.168.x.x is the CISCO IP. > > 2. The CISCO sees that the phone number (1091234567) starts with a 1 > so it uses FXO port 1 and strips the 1 and uses the remaining phone > number (091234567) to make the call. > > 3. My phone rings, I pick up and I can hear my IVR playing. > > > > These are my current problems: > > - IVR starts playing before I even pick up the phone. This means that > if the system calls a mobile phone and the person doesn't pick up, the > IVR will start playing and eventually the mobile phone will divert to > voice mail. Obviously I then get a missed call and an sms saying I > have a new voice mail, which is annoying. Instead I would like it to > KNOW that no one has picked up, but I don't know how to do this. > Somehow the CISCO needs to be able to tell FreeSWITCH that the call > has not yet been answered. For some reason however as soon as the > CISCO starts calling FreeSWITCH thinks the call is already connected. > It doesn't know that the CISCO is actually still ringing. Maybe I'm > doing originate the wrong way or something ... > > - The phone only rings for about 10 seconds before hanging up. I've > tried "call_timeout", "bridge_answer_timeout". I've also tried setting > CISCO "ring number". Nothing works, my phone still only rings for > about 10 seconds. I don't know if this is a FreeSWITCH issue or a > CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > starts playing even if no one answers the phone. > > > > > > CISCO Config for relevant FXO port: > > voice service voip > ?allow-connections h323 to h323 > ?allow-connections h323 to sip > ?allow-connections sip to h323 > ?allow-connections sip to sip > ?no supplementary-service h450.2 > ?no supplementary-service h450.3 > ?supplementary-service h450.12 > ?no supplementary-service sip moved-temporarily > ?no supplementary-service sip refer > ?fax protocol cisco > ?sip > ?registrar server expires max 3600 min 3600 > ?no update-callerid > ?no call service stop > > voice-port 0/3/2 > ?output attenuation -3 > ?no comfort-noise > ?cptone AU > ?impedance complex1 > ?caller-id enable > ! > dial-peer voice 100 pots > ?preference 1 > ?destination-pattern 1T > ?port 0/3/2 > ! > > > > Many Thanks, > > Oliver _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f054ba032765570613771! From olimonkey at gmail.com Thu Jan 5 10:50:35 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Thu, 5 Jan 2012 15:50:35 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: <1FFF97C269757C458224B7C895F35F1502B9B8@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1502B9B8@cantor.std.visionutv.se> Message-ID: I have another theory... In the managed application that gets bridged to the outgoing cisco call, l have: context.Session.Answer(); immediately. Should I be waiting for some kind of event first before I call "Answer()" to make sure it doesn't execute my code too quickly? On Thu, Jan 5, 2012 at 3:36 PM, Peter Olsson wrote: > I think you're doing it right. It seems to me that the Cisco responds with 200 OK, and lets FS believe the call was answered, even though it was not really answered. You will need to check the settings in the Cisco. Also trace the SIP packets and log during a call, and pastebin it, and someone here can take a look on it. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Oliver Schenk > Skickat: den 5 januari 2012 08:06 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > *bump* > > > So I think maybe the way I'm doing the originate is the problem? In my call string I'm creating a connection directly from the CISCO > (192.168.x.x) to the managed application, which may be why it starts playing straight away? > > Maybe I should be originating a call first and then only once I know the other side has picked up will I bridge the call to the IVR managed application. > > Problem is I dunno how to tell whether the other person has picked up (or even if the cisco is going to tell me) and I don't know how to do things to a call once it has been established. > > > I'm currently reading the Dialplan wiki page, hoping to get something out of it there. > > > Cheers > > Oliver > > > On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> and connecting through a CISCO 2811. Most things now work quite well, >> but I am having a few issues with the way the system answers calls (or >> doesn't answer calls...). >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> which is then connected to a POTS phone line. >> >> >> Take the following scenario: >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> to freeswitch and originate a call: >> >> string callstring = >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sof >> ia/internal/1091234567 at 192.168.x.x >> '&managed(ivrAppName)'"; >> eslConnection.API("originate", callstring); >> >> where 192.168.x.x is the CISCO IP. >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> number (091234567) to make the call. >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> These are my current problems: >> >> - IVR starts playing before I even pick up the phone. This means that >> if the system calls a mobile phone and the person doesn't pick up, the >> IVR will start playing and eventually the mobile phone will divert to >> voice mail. Obviously I then get a missed call and an sms saying I >> have a new voice mail, which is annoying. Instead I would like it to >> KNOW that no one has picked up, but I don't know how to do this. >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> has not yet been answered. For some reason however as soon as the >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> doing originate the wrong way or something ... >> >> - The phone only rings for about 10 seconds before hanging up. I've >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> CISCO "ring number". Nothing works, my phone still only rings for >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> starts playing even if no one answers the phone. >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> voice service voip >> ?allow-connections h323 to h323 >> ?allow-connections h323 to sip >> ?allow-connections sip to h323 >> ?allow-connections sip to sip >> ?no supplementary-service h450.2 >> ?no supplementary-service h450.3 >> ?supplementary-service h450.12 >> ?no supplementary-service sip moved-temporarily >> ?no supplementary-service sip refer >> ?fax protocol cisco >> ?sip >> ?registrar server expires max 3600 min 3600 >> ?no update-callerid >> ?no call service stop >> >> voice-port 0/3/2 >> ?output attenuation -3 >> ?no comfort-noise >> ?cptone AU >> ?impedance complex1 >> ?caller-id enable >> ! >> dial-peer voice 100 pots >> ?preference 1 >> ?destination-pattern 1T >> ?port 0/3/2 >> ! >> >> >> >> Many Thanks, >> >> Oliver > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f054ba032765570613771! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brandon.mcginty at gmail.com Thu Jan 5 12:00:51 2012 From: brandon.mcginty at gmail.com (Brandon McGinty) Date: Thu, 05 Jan 2012 04:00:51 -0500 Subject: [Freeswitch-users] Flite speed and pitch increase after update. In-Reply-To: References: <4F04CDA4.3070707@gmail.com> <1325719664163-7152479.post@n2.nabble.com> Message-ID: <4F0566C3.5080704@gmail.com> Thank you all. I shall report tomorrow, and see what I can do about creating a quick patch and sending it here. Greatly appreciate the quick responses. Have a good one. Brandon McGinty-Carroll On 1/4/2012 7:38 PM, Anthony Minessale wrote: > confirmed this is the kal voice specific, slt and rms both work > > This is an update to the actual flite lib, we will have to report it > upstream. > > > On Wed, Jan 4, 2012 at 5:27 PM, Jeff Lenk > wrote: > > Please open a Jira and document the problem so it doesnt get forgotten. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Flite-speed-and-pitch-increase-after-update-tp7152216p7152479.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Thu Jan 5 12:35:59 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Jan 2012 09:35:59 +0000 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1502B9B8@cantor.std.visionutv.se> Message-ID: <1FFF97C269757C458224B7C895F35F1502BA8E@cantor.std.visionutv.se> No, I don't think that the call should execute the application until the call has been answered. However, there should be no need for your application to answer it, since it will be answered already. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Oliver Schenk Skickat: den 5 januari 2012 08:51 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR I have another theory... In the managed application that gets bridged to the outgoing cisco call, l have: context.Session.Answer(); immediately. Should I be waiting for some kind of event first before I call "Answer()" to make sure it doesn't execute my code too quickly? On Thu, Jan 5, 2012 at 3:36 PM, Peter Olsson wrote: > I think you're doing it right. It seems to me that the Cisco responds with 200 OK, and lets FS believe the call was answered, even though it was not really answered. You will need to check the settings in the Cisco. Also trace the SIP packets and log during a call, and pastebin it, and someone here can take a look on it. > > /Peter > > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Oliver > Schenk > Skickat: den 5 januari 2012 08:06 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > *bump* > > > So I think maybe the way I'm doing the originate is the problem? In my > call string I'm creating a connection directly from the CISCO > (192.168.x.x) to the managed application, which may be why it starts playing straight away? > > Maybe I should be originating a call first and then only once I know the other side has picked up will I bridge the call to the IVR managed application. > > Problem is I dunno how to tell whether the other person has picked up (or even if the cisco is going to tell me) and I don't know how to do things to a call once it has been established. > > > I'm currently reading the Dialplan wiki page, hoping to get something out of it there. > > > Cheers > > Oliver > > > On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> and connecting through a CISCO 2811. Most things now work quite well, >> but I am having a few issues with the way the system answers calls >> (or doesn't answer calls...). >> >> I have FreeSWITCH running as a windows service on Windows Server >> 2008, which is connected via LAN to a CISCO 2811 with a 4 port FXO >> card, which is then connected to a POTS phone line. >> >> >> Take the following scenario: >> >> 1. Managed .NET application creates a call string and uses ESL to >> talk to freeswitch and originate a call: >> >> string callstring = >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}so >> f >> ia/internal/1091234567 at 192.168.x.x >> '&managed(ivrAppName)'"; >> eslConnection.API("originate", callstring); >> >> where 192.168.x.x is the CISCO IP. >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> number (091234567) to make the call. >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> These are my current problems: >> >> - IVR starts playing before I even pick up the phone. This means that >> if the system calls a mobile phone and the person doesn't pick up, >> the IVR will start playing and eventually the mobile phone will >> divert to voice mail. Obviously I then get a missed call and an sms >> saying I have a new voice mail, which is annoying. Instead I would >> like it to KNOW that no one has picked up, but I don't know how to do this. >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> has not yet been answered. For some reason however as soon as the >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> doing originate the wrong way or something ... >> >> - The phone only rings for about 10 seconds before hanging up. I've >> tried "call_timeout", "bridge_answer_timeout". I've also tried >> setting CISCO "ring number". Nothing works, my phone still only rings >> for about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> starts playing even if no one answers the phone. >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> voice service voip >> ?allow-connections h323 to h323 >> ?allow-connections h323 to sip >> ?allow-connections sip to h323 >> ?allow-connections sip to sip >> ?no supplementary-service h450.2 >> ?no supplementary-service h450.3 >> ?supplementary-service h450.12 >> ?no supplementary-service sip moved-temporarily >> ?no supplementary-service sip refer >> ?fax protocol cisco >> ?sip >> ?registrar server expires max 3600 min 3600 >> ?no update-callerid >> ?no call service stop >> >> voice-port 0/3/2 >> ?output attenuation -3 >> ?no comfort-noise >> ?cptone AU >> ?impedance complex1 >> ?caller-id enable >> ! >> dial-peer voice 100 pots >> ?preference 1 >> ?destination-pattern 1T >> ?port 0/3/2 >> ! >> >> >> >> Many Thanks, >> >> Oliver > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f05562f32761050517028! From Prometheus001 at gmx.net Thu Jan 5 13:59:54 2012 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 05 Jan 2012 11:59:54 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable Message-ID: <4F0582AA.9060708@gmx.net> Hello, I have a strange phenomen: When a target UA is busy, it returns "486 Busy" to Freeswitch. But Freeswitch then returns "480 Temporarily Unavailable" to the called party. Where does this come from and how can I change this behaviour? See (anonymized) SIP trace with ngrep: UA to Freeswitch: ======================== U 2012/01/04 13:59:44.928775 :5060 -> :5080 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP :5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. From: "026xxxxxxxx" >;tag=py094Kv7vr03a. To: ;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. CSeq: 22504928 INVITE. User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). Content-Length: 0. Freeswitch to Caller: ======================== U 2012/01/04 13:59:44.930387 :5060 -> :5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP :5060;branch=z9hG4bK-4896-2830DFA. From: ;tag=13517-HB-08a98588-2622da197. To: ;user=phone>;tag=XQtc5US24QgDa. Call-ID: 13517-SG-08a98587-0a352e121 at sip.provider.de. CSeq: 134781549 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 13-52-28 -0600. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. P-Asserted-Identity: "069xxxxxxxx" >. Best regards Peter From michal.zubac at comgate.cz Thu Jan 5 15:20:03 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Thu, 05 Jan 2012 13:20:03 +0100 Subject: [Freeswitch-users] bridging 2 SIP destinations from PBX behind NAT Message-ID: <4F059573.5030006@comgate.cz> Hi. I encountered problem with bridging two SIP calls. My FreeSwitch is behind NAT, I'm not sure if endpoint switches are, but I think it doesn't matter now. Situation looks like this: SIP1 <--> NAT <--> PBX <--> NAT <--> SIP2 SIP1 is caller switch, SIP2 is destination switch so call is initiated like this: SIP1->PBX->SIP2 I tested both endpoints separately using ISDN endpoint and they worked. Like this ISDN1->PBX->SIP2 & SIP1->PBX->ISDN2 But when I connected SIP1 & SIP2 together, call is estabilished, but no audio was going through our firewall. Both audio streams from SIP1 and SIP2 were filtered out by our firewall, because Freeswitch (PBX) wasn't sending any initial packets to SIP endpoints, so no NAT holes were created. It look strange to me. I expected FreeSwitch to at least send some "empty" RTP packets to SIP endpoints as soon as call estabilishment is confirmed on SIP channel. But Freeswitch doesn't do it and only sends ringing indication. I had problems in scenario, where SIP2 endpoint doesn't ring and immediately answers the call. Call was estabilished, but audio was stuck in our firewall, because Freeswitch haven't initiated any RTP communication yet and was probably waiting for something. I managed to solve this problem by adding between my and commands in dialplan. Now it works, but I'm feeling kind of uncertain with this solution, because I don't really understand why. What was Freeswitch waiting for, before I added instant_ringback? Why can't FreeSwitch prepare NAT holes (as described before) as soon as SDP are interchanged? Thanks for clarification. Regards -- Michal Zubac ComGate Interactive s.r.o. Prague Marina Office Center Jankovcova 1596/14a 17000 Praha 7, Czech Republic From frank at telonium.com Thu Jan 5 16:21:51 2012 From: frank at telonium.com (Frank Park) Date: Thu, 5 Jan 2012 08:21:51 -0500 Subject: [Freeswitch-users] How to troubleshoot "503 Maximum Calls In Progress"? In-Reply-To: <7454A296C7EDE34EA57199FAA401E2F1220CB5293C@VMBX113.ihostexchange.net> References: <7454A296C7EDE34EA57199FAA401E2F1220CB5293C@VMBX113.ihostexchange.net> Message-ID: Chris, Was this issue ever resolved? Anything you learned after this email was sent? Thank you, Frank On Mon, May 23, 2011 at 6:49 PM, Chris Fowler wrote: > Hi,**** > > ** ** > > On our Production FreeSWITCH box: **** > > ** ** > > FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500)**** > > ** ** > > UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, > 751 microseconds**** > > 12519 session(s) since startup**** > > 2 session(s) 0/30**** > > 1000 session(s) max**** > > min idle cpu 0.00/100.00**** > > ** ** > > At 14:30:37 a reloadxml command was issued.**** > > At 14:30:43 the box started rejecting all calls with 503 error. **** > > At 14:38 it started working again. **** > > ** ** > > ** ** > > Looking at the code I see three conditions can trigger this behavior:**** > > ** ** > > if (sess_count >= sess_max || !sofia_test_pflag(profile, > PFLAG_RUNNING) || !switch_core_ready()) {**** > > nua_respond(nh, 503, "Maximum Calls In Progress", > SIPTAG_RETRY_AFTER_STR("300"), TAG_END());**** > > ** ** > > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n");**** > > ** ** > > goto done;**** > > }**** > > ** ** > > Logs show it wasn?t lack of available sessions. How can > sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and > busy the system?**** > > ** ** > > Thoughts? Thx, Chris.**** > > ** ** > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/e0da7406/attachment-0001.html From kochanowski.wojtek at gmail.com Thu Jan 5 16:06:49 2012 From: kochanowski.wojtek at gmail.com (Wojtek Kochanowski) Date: Thu, 5 Jan 2012 14:06:49 +0100 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: References: <14AA2B32-ACE0-429F-AB60-2E9CAF98BD96@freeswitch.org> Message-ID: Hi! I've had similar problem after separate contexts from default to external and internal. What is your tonegroup in autoload_configs/freetdm.conf.xml ? Make sure value of this parameter exist in tones.conf. Greet, Wojtek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/897f2124/attachment.html From msc at freeswitch.org Thu Jan 5 18:49:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Jan 2012 07:49:55 -0800 Subject: [Freeswitch-users] Password in dialplan In-Reply-To: <4F054EFA.2000004@softnet.si> References: <4F054EFA.2000004@softnet.si> Message-ID: How is this user being authenticated? -MC On Wed, Jan 4, 2012 at 11:19 PM, Miha Zoubek wrote: > Hi, > > in dial plan I have this line (for radius): > > data="PASSWD=$${default_password}"/> > > Deafult password is set to 1234. As I do not wont default password for > radius, I would just like to have here password which is set for user in > directory. > > > > How to set in dialplan that the password will be taken from user/dir? > > Like > this? > > Thanks! > > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/206739ac/attachment.html From gustavomarsico at gmail.com Thu Jan 5 19:51:59 2012 From: gustavomarsico at gmail.com (=?iso-8859-1?Q?Gustavo_M=E1rsico?=) Date: Thu, 5 Jan 2012 13:51:59 -0300 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: I think I've a similar problem related to callcenter app. When I made an originate like this: originate loopback/2500/default/XML &bridge(user/2001) 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false Dialplan: loopback/2500-b Action set(ignore_early_media=true) Dialplan: loopback/2500-b Action callcenter(click2call) 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE EXECUTE loopback/2500-b set(open=true) 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] EXECUTE loopback/2500-b set(ignore_early_media=true) 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY EXECUTE loopback/2500-b callcenter(click2call) 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > Also, maybe I should be doing something like this: > > sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' > > instead of: > > sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' > > > but, I don't really have the CISCO configured as a gateway, nor do I > know how really...probably not on the right track there. > > > > > On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >> *bump* >> >> >> So I think maybe the way I'm doing the originate is the problem? In my >> call string I'm creating a connection directly from the CISCO >> (192.168.x.x) to the managed application, which may be why it starts >> playing straight away? >> >> Maybe I should be originating a call first and then only once I know >> the other side has picked up will I bridge the call to the IVR managed >> application. >> >> Problem is I dunno how to tell whether the other person has picked up >> (or even if the cisco is going to tell me) and I don't know how to do >> things to a call once it has been established. >> >> >> I'm currently reading the Dialplan wiki page, hoping to get something >> out of it there. >> >> >> Cheers >> >> Oliver >> >> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>> and connecting through a CISCO 2811. Most things now work quite well, >>> but I am having a few issues with the way the system answers calls (or >>> doesn't answer calls...). >>> >>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>> which is then connected to a POTS phone line. >>> >>> >>> Take the following scenario: >>> >>> 1. Managed .NET application creates a call string and uses ESL to talk >>> to freeswitch and originate a call: >>> >>> string callstring = >>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>> '&managed(ivrAppName)'"; >>> eslConnection.API("originate", callstring); >>> >>> where 192.168.x.x is the CISCO IP. >>> >>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>> number (091234567) to make the call. >>> >>> 3. My phone rings, I pick up and I can hear my IVR playing. >>> >>> >>> >>> These are my current problems: >>> >>> - IVR starts playing before I even pick up the phone. This means that >>> if the system calls a mobile phone and the person doesn't pick up, the >>> IVR will start playing and eventually the mobile phone will divert to >>> voice mail. Obviously I then get a missed call and an sms saying I >>> have a new voice mail, which is annoying. Instead I would like it to >>> KNOW that no one has picked up, but I don't know how to do this. >>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>> has not yet been answered. For some reason however as soon as the >>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>> doing originate the wrong way or something ... >>> >>> - The phone only rings for about 10 seconds before hanging up. I've >>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>> CISCO "ring number". Nothing works, my phone still only rings for >>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>> starts playing even if no one answers the phone. >>> >>> >>> >>> >>> >>> CISCO Config for relevant FXO port: >>> >>> voice service voip >>> allow-connections h323 to h323 >>> allow-connections h323 to sip >>> allow-connections sip to h323 >>> allow-connections sip to sip >>> no supplementary-service h450.2 >>> no supplementary-service h450.3 >>> supplementary-service h450.12 >>> no supplementary-service sip moved-temporarily >>> no supplementary-service sip refer >>> fax protocol cisco >>> sip >>> registrar server expires max 3600 min 3600 >>> no update-callerid >>> no call service stop >>> >>> voice-port 0/3/2 >>> output attenuation -3 >>> no comfort-noise >>> cptone AU >>> impedance complex1 >>> caller-id enable >>> ! >>> dial-peer voice 100 pots >>> preference 1 >>> destination-pattern 1T >>> port 0/3/2 >>> ! >>> >>> >>> >>> Many Thanks, >>> >>> Oliver > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter at uringme.com Thu Jan 5 21:20:02 2012 From: peter at uringme.com (peter at uringme.com) Date: Thu, 5 Jan 2012 10:20:02 -0800 (PST) Subject: [Freeswitch-users] X-Lite and Video In-Reply-To: <1325706770.19930.YahooMailClassic@web2808.biz.mail.ne1.yahoo.com> Message-ID: <1325787602.37602.YahooMailClassic@web2804.biz.mail.ne1.yahoo.com> For anyone who may have the same issue in the future, I did finally get it to work. I had to not only add the codecs to global_codec_prefs, but add it to outbound_codec_prefs as well (then I restarted FS -- don't know if a reloadxml would have worked or not).? Here's mine now: ? ? That got me a video call between registered extensions.? However, the video was blank even though I had full audio.? tcpdump showed that video data was being sent and received, but both video screens said: "Waiting for video".? I tried a couple things like firewalls and NAT ICE/STUN on my own side, then finally tried removing H.263-1998 from both X-Lites to force H.263, and that worked. I had two-way video at that point.? My X-Lite 4 generic doesn't have H.264 support, so I haven't tested that. --- On Wed, 1/4/12, peter at uringme.com wrote: From: peter at uringme.com Subject: [Freeswitch-users] X-Lite and Video To: freeswitch-users at lists.freeswitch.org Date: Wednesday, January 4, 2012, 2:52 PM (Apologies to the list owner for originally sending this to the wrong address) I have two laptops running X-Lite 4.? I have them registered to a FreeSwitch server (latest git) as extensions 7777 and 7778.? I have a dialplan for each (quick and dirty) that just bridges them when one is dialed from the other: ? ????????? ????????? ????????? ??? (and vice-versa for 7778). I can dial between them just fine for audio calls -- bidirectional audio, etc, no problem. I'm trying to get video going.? Both X-Lites have H.263 and H.263-1998 enabled in their settings.? Freeswitch has the following in vars.xml: ? When I try to make a video call from one extension to the other, the calling extension seems to think it's in video, but the called extension doesn't. INVITE from freeswitch console: ?? ------------------------------------------------------------------------ ?? INVITE sip:7778 at test SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.1.7:50350;branch=z9hG4bK-d8754z-9ea3618224c5cf23-1---d8754z-;rport ?? Max-Forwards: 70 ?? Contact: ?? To: ?? From: "Peter Test";tag=a7031d83 ?? Call-ID: ZDUzZGE1YjUyOTQ2ZGNmZTY0Yjc5ODA5NTE4NDAzMGQ. ?? CSeq: 1 INVITE ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ?? Content-Type: application/sdp ?? Supported: replaces ?? User-Agent: X-Lite 4 release 4.1 stamp 63214 ?? Content-Length: 681 ?? v=0 ?? o=- 12970176286658638 1 IN IP4 192.168.1.7 ?? s=CounterPath X-Lite 4.1 ?? c=IN IP4 192.168.1.7 ?? t=0 0 ?? a=ice-ufrag:20fef6 ?? a=ice-pwd:0a03863684bc5f16a9c862dcdccdd8eb ?? m=audio 58632 RTP/AVP 107 0 8 101 ?? a=rtpmap:107 BV32/16000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-15 ?? a=sendrecv ?? a=candidate:1 1 UDP 659136 192.168.1.7 58632 typ host ?? a=candidate:1 2 UDP 659134 192.168.1.7 58633 typ host ?? m=video 50994 RTP/AVP 34 115 ?? a=rtpmap:34 H263/90000 ?? a=fmtp:34 QCIF=2;CIF=2;VGA=2 ?? a=rtpmap:115 H263-1998/90000 ?? a=fmtp:115 QCIF=2;CIF=2;VGA=2;I=1;J=1;T=1 ?? a=sendrecv ?? a=candidate:1 1 UDP 659136 192.168.1.7 50994 typ host ?? a=candidate:1 2 UDP 659134 192.168.1.7 50995 typ host I do see freeswitch seeing the audio and video codecs: 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4683 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:2800 Set Codec sofia/external/7777 at test PCMU/8000 20 ms 160 samples 64000 bits 2012-01-04 13:44:37.188657 [DEBUG] switch_core_state_machine.c:343 (sofia/external/7777 at test) State NEW 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4797 Set 2833 dtmf send/recv payload to 101 2012-01-04 13:44:37.188657 [DEBUG] sofia_glue.c:4856 Video Codec Compare [H263:34]/[H263:34] However, when freeswitch starts the bridge and calls the far-end party, it doesn't send along video information in the INVITE: ?? INVITE sip:7778 at 68.202.69.172:32834;transport=udp;rinstance=677b87c43ee7970a SIP/2.0 ?? Via: SIP/2.0/UDP 204.13.175.89:5080;rport;branch=z9hG4bK8jm56tcmZ6p6j2012-01-04 ?? Max-Forwards: 69 ?? From: "Peter Test" ;tag=ZeZUav5XXat5e ?? To: ?? Call-ID: fe64b2f5-b1a6-122f-a187-00144f49eecc ?? CSeq: 22515274 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY ?? Supported: timer, precondition, path, replaces ?? Allow-Events: talk, hold, refer ?? Content-Type: application/sdp ?? Content-Disposition: session ?? Content-Length: 207 ?? X-FS-Support: update_display ?? Remote-Party-ID: "Peter Test" ;party=calling;screen=yes;privacy=off ?? v=0 ?? o=FreeSWITCH 1325690959 1325690960 IN IP4 204.13.175.89 ?? s=FreeSWITCH ?? c=IN IP4 204.13.175.89 ?? t=0 0 ?? m=audio 11718 RTP/AVP 0 8 3 101 13 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=ptime:20 So, am I missing something?? Do I need to use something other than "bridge" in the dialplan, or do I need to add some variables to be able to pass on the video?? All I'm trying to do is make a video call between two X-Lites that are locally SIP registered to freeswitch.? Because I want to record the video at some point in the future, I don't want to divert the media -- I want it streaming/passing through freeswitch. When the call is connected, the caller shows a "Waiting for video", but the called doesn't show this.? When I try to start the video, it says "Failed to Start Video". -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/90421941/attachment-0001.html From govoiper at gmail.com Thu Jan 5 22:18:39 2012 From: govoiper at gmail.com (Sammy Govind) Date: Fri, 6 Jan 2012 00:18:39 +0500 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F0582AA.9060708@gmx.net> References: <4F0582AA.9060708@gmx.net> Message-ID: Hi, Thats really magical !! I wonder how would the console logs look like for this,. 1- Please do share the logs in pastebin. 2- take sip trace of the same call as well. 3- identify and paste the relevant dialplan code. As a new follower of FS this is really alarming for me to know how a 486 coming from carrier is relayed back to my sip client!! Regards, Sammy On Thu, Jan 5, 2012 at 3:59 PM, Peter P GMX wrote: > Hello, > > I have a strange phenomen: > > When a target UA is busy, it returns "486 Busy" to Freeswitch. But > Freeswitch then returns "480 Temporarily Unavailable" to the called party. > Where does this come from and how can I change this behaviour? > > See (anonymized) SIP trace with ngrep: > > UA to Freeswitch: > ======================== > U 2012/01/04 13:59:44.928775 :5060 -> :5080 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP > :5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. > From: "026xxxxxxxx" >;tag=py094Kv7vr03a. > To: > ;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. > Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. > CSeq: 22504928 INVITE. > User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). > Content-Length: 0. > > Freeswitch to Caller: > ======================== > U 2012/01/04 13:59:44.930387 :5060 -> :5060 > SIP/2.0 480 Temporarily Unavailable. > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-4896-2830DFA. > From: > ;user=phone>;tag=13517-HB-08a98588-2622da197. > To: ;user=phone>;tag=XQtc5US24QgDa. > Call-ID: 13517-SG-08a98587-0a352e121 at sip.provider.de. > CSeq: 134781549 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 > 13-52-28 -0600. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: precondition, path, replaces. > Allow-Events: talk, hold, refer. > Content-Length: 0. > P-Asserted-Identity: "069xxxxxxxx" >. > > Best regards > Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/41cfbf0d/attachment.html From Hector.Geraldino at ip-soft.net Thu Jan 5 22:31:21 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 5 Jan 2012 14:31:21 -0500 Subject: [Freeswitch-users] help with wait_for_silence Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507651@NY1-EXMB-01.ip-soft.net> Hi, I'm trying to use the wait_for_silence application to stop recording audio from a channel, and I need to use it not from a FS dialplan but from an ESL application. So far I can execute the wait_for_silence application using 'execute wait_for_silence arguments' with no issues. I can even see the result of the execution of this command on the fs_cli console: 2012-01-05 19:04:40.266001 [DEBUG] switch_ivr.c:576 sofia/internal/5512 at 192.168.8.11 Command Execute wait_for_silence(200 15 15 5000) EXECUTE sofia/internal/5512 at 192.168.8.11 wait_for_silence(200 15 15 5000) The problem is that I don't know if this is working or not, as it is not raising any event that can be received by the application. I don't see any event related to this application (or to detect silence using any other mechanism) on the wiki Events List page [http://wiki.freeswitch.org/wiki/Event_list], so I'm not sure it is usable in an socket outbound mode. Resuming: is it possible to detect the silence on a channel and then perform an action in an ESL application, using wait_for_silence app or any other mechanism? Thanks, Hector -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/9ad0649f/attachment.html From freeswitch at peely.com Thu Jan 5 22:53:36 2012 From: freeswitch at peely.com (peely) Date: Thu, 5 Jan 2012 11:53:36 -0800 (PST) Subject: [Freeswitch-users] SIP attacks from 188.161.101.73 In-Reply-To: References: <4F025951.2040009@googlemail.com> Message-ID: <1325793216400-7155883.post@n2.nabble.com> We force all users to register against our DNS name, then use IP Tables to reject REGISTER methods coming against our IP address: iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "REGISTER sip:a.c.d.e SIP" --algo bm iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "REGISTER sip:a.c.d.e:5060 SIP" --algo bm Where a.b.c.d is your public IP address. This will stop responses to initial SIPVicious explorations also, which means you don't get on their list in the first place and save a whole lot of useless UDP blasts which come from their brute force attacks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-attacks-from-188-161-101-73-tp7144973p7155883.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Jan 5 23:04:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Jan 2012 12:04:38 -0800 Subject: [Freeswitch-users] help with wait_for_silence In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225507651@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0225507651@NY1-EXMB-01.ip-soft.net> Message-ID: What are you using to do the recording of the audio? The record app has built-in silence detection. If you are using record_session then I can see your dilemma. It's almost as if you'd need "execute_on_silence" as opposed to wait_for_silence. Anyone out there have any suggestions for Hector on how to do this? -MC On Thu, Jan 5, 2012 at 11:31 AM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > Hi,**** > > ** ** > > I?m trying to use the wait_for_silence application to stop recording audio > from a channel, and I need to use it not from a FS dialplan but from an ESL > application. **** > > ** ** > > So far I can execute the wait_for_silence application using ?execute > wait_for_silence *arguments*? with no issues. I can even see the result > of the execution of this command on the fs_cli console:**** > > ** ** > > 2012-01-05 19:04:40.266001 [DEBUG] switch_ivr.c:576 sofia/internal/ > 5512 at 192.168.8.11 Command Execute wait_for_silence(200 15 15 5000)**** > > EXECUTE sofia/internal/5512 at 192.168.8.11 wait_for_silence(200 15 15 5000)* > *** > > ** ** > > The problem is that I don?t know if this is working or not, as it is not > raising any event that can be received by the application. I don?t see any > event related to this application (or to detect silence using any other > mechanism) on the wiki Events List page [ > http://wiki.freeswitch.org/wiki/Event_list], so I?m not sure it is usable > in an socket outbound mode. **** > > ** ** > > Resuming: is it possible to detect the silence on a channel and then > perform an action in an ESL application, using wait_for_silence app or any > other mechanism? **** > > ** ** > > Thanks,**** > > Hector**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/2c08f80d/attachment.html From Hector.Geraldino at ip-soft.net Thu Jan 5 23:33:39 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 5 Jan 2012 15:33:39 -0500 Subject: [Freeswitch-users] help with wait_for_silence In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD0225507651@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022550765C@NY1-EXMB-01.ip-soft.net> Great, that was exactly what I was looking for. I was using uuid_record, not the 'record' app. I replaced it with 'execute record params' and now I receive a RECORD_STOP event. I think I can move forward with this small change. Thanks a lot Michael! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, January 05, 2012 3:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] help with wait_for_silence What are you using to do the recording of the audio? The record app has built-in silence detection. If you are using record_session then I can see your dilemma. It's almost as if you'd need "execute_on_silence" as opposed to wait_for_silence. Anyone out there have any suggestions for Hector on how to do this? -MC On Thu, Jan 5, 2012 at 11:31 AM, Hector Geraldino > wrote: Hi, I'm trying to use the wait_for_silence application to stop recording audio from a channel, and I need to use it not from a FS dialplan but from an ESL application. So far I can execute the wait_for_silence application using 'execute wait_for_silence arguments' with no issues. I can even see the result of the execution of this command on the fs_cli console: 2012-01-05 19:04:40.266001 [DEBUG] switch_ivr.c:576 sofia/internal/5512 at 192.168.8.11 Command Execute wait_for_silence(200 15 15 5000) EXECUTE sofia/internal/5512 at 192.168.8.11 wait_for_silence(200 15 15 5000) The problem is that I don't know if this is working or not, as it is not raising any event that can be received by the application. I don't see any event related to this application (or to detect silence using any other mechanism) on the wiki Events List page [http://wiki.freeswitch.org/wiki/Event_list], so I'm not sure it is usable in an socket outbound mode. Resuming: is it possible to detect the silence on a channel and then perform an action in an ESL application, using wait_for_silence app or any other mechanism? Thanks, Hector _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/ec41898f/attachment-0001.html From brian.wiese.freeswitch at gmail.com Fri Jan 6 00:19:03 2012 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Thu, 5 Jan 2012 15:19:03 -0600 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Michael: Here's what I came up with, and it works really well! I can embed this in a dialplan or use it at the CLI: lua ~stream:write(tostring(string.gsub("string_to_search", "string_to_find", "replacement_string"))) It's awesome! Thanks for your help! ~Brian On Tue, Jan 3, 2012 at 6:16 PM, Brian Wiese wrote: > Michael: > > I do think that mod_xml_curl is a little more than what I need here. > I'll try a Lua script and see where I get. ?I'll try to call a Lua > script and return the bridge string back to the dial plan. > > I'll let you know how my scripting fu works... > > ~Brian > > On Tue, Jan 3, 2012 at 6:10 PM, Michael Collins wrote: >> This kinda sounds like a problem in need of mod_xml_curl. If that seems like >> too much of a hassle then I would fall back to a mod_lua or mod_perl script >> to do the regex stuff. How is your scripting fu? >> >> -MC >> >> >> On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese >> wrote: >>> >>> Michael: >>> >>> I sure can! >>> >>> What I want to do is create template configurations that I can deploy >>> to multiple servers. ?Part of the requirement of incoming calls is to >>> configure which extension(s) ring immediately, 6-second delay, and >>> 12-second delay. ?I have created groups for these extensions, and by >>> using group_call I can get the full dial string for each group... >>> perfect! ?Now, I just need a way to delay some of these extensions by >>> 6 or 12 seconds. ?I ultimately want to inject leg variables into the >>> dial string for each extension, so when the group_call is expanded >>> each of them expand with the extra leg variable I define. >>> >>> Hope that helps... ?clear as mud? ?:) >>> >>> ~Brian >>> >>> On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins >>> wrote: >>> > Can you expand upon this question a bit? I'm curious if there's a less >>> > hackish way of doing what you want to do. Under what circumstances do >>> > you >>> > need to add the leg variables? Also, can you give us the big picture? >>> > What's >>> > the problem you're solving? >>> > >>> > -MC >>> > >>> > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese >>> > wrote: >>> >> >>> >> Hi Everyone. >>> >> >>> >> I thought I read somewhere that this was possible, but I can't find it >>> >> now... >>> >> >>> >> I need a way to find-and-replace within a variable. ?So, for example, >>> >> I want to take variables that have values like this: >>> >> >>> >> 123abc123abc >>> >> abc123abc123abc >>> >> >>> >> ...and do a find/replace of the "abc" with "xyz" so the variables >>> >> would now return: >>> >> >>> >> 123xyz123xyz >>> >> xyz123xyz123xyz >>> >> >>> >> The use case I've run into is that I need to add leg variables to >>> >> group_call. ?In my case, group_call can return any number of members, >>> >> so I figured I would just replace the first "[" with >>> >> "[variable-I-want-to-set=...". >>> >> >>> >> Thanks for the help! >>> >> >>> >> ~Brian >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From gb10hkzo-freeswitch at yahoo.co.uk Fri Jan 6 00:38:33 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 5 Jan 2012 21:38:33 +0000 (GMT) Subject: [Freeswitch-users] Encrypted clients to unencrypted destination ... possible ? Message-ID: <1325799513.45414.YahooMailNeo@web29401.mail.ird.yahoo.com> Hello List, To save me much head-banging and time-wasting, could someone please tell me whether it is possible to configure Freeswitch so that an endpoint can register over TLS, but can route out to and receive calls from the big bad unencrypted outside world ? I've got a basic DDI config at the moment that I can receive calls on fine when endpoint registers as unencrypted. ?But the moment I turn on TLS, the DDI no longer rings. I've followed the instructions here?http://wiki.freeswitch.org/wiki/SIP_TLS to enable TLS (only enabling TLS on the internal Sofia obviously). ?By the way, it would be great if someone could update those instructions with more detail as to how to get things working with a commercial certificate, as I had no luck making it work, so there's obviously a trick that's missing, and had to resort to a self signed one. ? Thanks Bob From sherifomran2000 at yahoo.com Fri Jan 6 01:37:50 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 5 Jan 2012 14:37:50 -0800 (PST) Subject: [Freeswitch-users] bridging 2 SIP destinations from PBX behind NAT In-Reply-To: <4F059573.5030006@comgate.cz> Message-ID: <1325803070.75371.YahooMailClassic@web110806.mail.gq1.yahoo.com> Try using stun server ex stun.ekiga.org best regards --- On Thu, 1/5/12, Michal Zub?? wrote: From: Michal Zub?? Subject: [Freeswitch-users] bridging 2 SIP destinations from PBX behind NAT To: "FreeSWITCH Users Help" Date: Thursday, January 5, 2012, 2:20 PM Hi. I encountered problem with bridging two SIP calls. My FreeSwitch is behind NAT, I'm not sure if endpoint switches are, but I think it doesn't matter now. Situation looks like this: ???SIP1 <--> NAT <--> PBX <--> NAT <--> SIP2 SIP1 is caller switch, SIP2 is destination switch so call is initiated like this: ???SIP1->PBX->SIP2 I tested both endpoints separately using ISDN endpoint and they worked. Like this ???ISDN1->PBX->SIP2 & ???SIP1->PBX->ISDN2 But when I connected SIP1 & SIP2 together, call is estabilished, but no audio was going through our firewall. Both audio streams from SIP1 and SIP2 were filtered out by our firewall, because Freeswitch (PBX) wasn't sending any initial packets to SIP endpoints, so no NAT holes were created. It look strange to me. I expected FreeSwitch to at least send some "empty" RTP packets to SIP endpoints as soon as call estabilishment is confirmed on SIP channel. But Freeswitch doesn't do it and only sends ringing indication. I had problems in scenario, where SIP2 endpoint doesn't ring and immediately answers the call. Call was estabilished, but audio was stuck in our firewall, because Freeswitch haven't initiated any RTP communication yet and was probably waiting for something. I managed to solve this problem by adding between my and commands in dialplan. Now it works, but I'm feeling kind of uncertain with this solution, because I don't really understand why. What was Freeswitch waiting for, before I added instant_ringback? Why can't FreeSwitch prepare NAT holes (as described before) as soon as SDP are interchanged? Thanks for clarification. Regards -- Michal Zubac ComGate Interactive s.r.o. Prague Marina Office Center Jankovcova 1596/14a 17000 Praha 7, Czech Republic _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/06e878ce/attachment.html From lists at telefaks.de Fri Jan 6 02:21:50 2012 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 06 Jan 2012 00:21:50 +0100 Subject: [Freeswitch-users] Encrypted clients to unencrypted destination ... possible ? In-Reply-To: <1325799513.45414.YahooMailNeo@web29401.mail.ird.yahoo.com> References: <1325799513.45414.YahooMailNeo@web29401.mail.ird.yahoo.com> Message-ID: <4F06308E.6090908@telefaks.de> Hello Bob, sure, this does work. I have a couple of phones working with TLS and SRTP and they all communicate to the outside world through Freeswitch via SIP and RTP. You have to ensure though that Freeswitch is not is proxy or bypass media mode. Best regards Peter Am 05.01.2012 22:38, schrieb Bob Smith: > Hello List, > > To save me much head-banging and time-wasting, could someone please tell me whether it is possible to configure Freeswitch so that an endpoint can register over TLS, but can route out to and receive calls from the big bad unencrypted outside world ? > > I've got a basic DDI config at the moment that I can receive calls on fine when endpoint registers as unencrypted. But the moment I turn on TLS, the DDI no longer rings. > > I've followed the instructions here http://wiki.freeswitch.org/wiki/SIP_TLS to enable TLS (only enabling TLS on the internal Sofia obviously). By the way, it would be great if someone could update those instructions with more detail as to how to get things working with a commercial certificate, as I had no luck making it work, so there's obviously a trick that's missing, and had to resort to a self signed one. > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From lists at telefaks.de Fri Jan 6 02:25:23 2012 From: lists at telefaks.de (Peter Steinbach) Date: Fri, 06 Jan 2012 00:25:23 +0100 Subject: [Freeswitch-users] Session timeout Message-ID: <4F063163.50005@telefaks.de> I have enabled session timers in my sofia internal profile the following way: However I do not see any Re-Invite or Update packets sent from Freeswitch to my User Agents. What am I doing wrong? Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de From gb10hkzo-freeswitch at yahoo.co.uk Fri Jan 6 02:29:47 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 5 Jan 2012 23:29:47 +0000 (GMT) Subject: [Freeswitch-users] Encrypted clients to unencrypted destination ... possible ? Message-ID: <1325806187.90487.YahooMailNeo@web29406.mail.ird.yahoo.com> Hello Peter, I will have another go at this over the next couple of days and let you know ! Bob >Hello Bob, > >sure, this does work. I have a couple of phones working with TLS and >SRTP and they all communicate to the outside world through Freeswitch >via SIP and RTP. You have to ensure though that Freeswitch is not is >proxy or bypass media mode. > >Best regards >Peter From msc at freeswitch.org Fri Jan 6 02:36:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Jan 2012 15:36:52 -0800 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Nicely done! Be sure to wikify this one. If you have any questions about updating the wiki just email me off list. -MC On Thu, Jan 5, 2012 at 1:19 PM, Brian Wiese < brian.wiese.freeswitch at gmail.com> wrote: > Michael: > > Here's what I came up with, and it works really well! I can embed > this in a dialplan or use it at the CLI: > > lua ~stream:write(tostring(string.gsub("string_to_search", > "string_to_find", "replacement_string"))) > > It's awesome! > > Thanks for your help! > > ~Brian > > On Tue, Jan 3, 2012 at 6:16 PM, Brian Wiese > wrote: > > Michael: > > > > I do think that mod_xml_curl is a little more than what I need here. > > I'll try a Lua script and see where I get. I'll try to call a Lua > > script and return the bridge string back to the dial plan. > > > > I'll let you know how my scripting fu works... > > > > ~Brian > > > > On Tue, Jan 3, 2012 at 6:10 PM, Michael Collins > wrote: > >> This kinda sounds like a problem in need of mod_xml_curl. If that seems > like > >> too much of a hassle then I would fall back to a mod_lua or mod_perl > script > >> to do the regex stuff. How is your scripting fu? > >> > >> -MC > >> > >> > >> On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese > >> wrote: > >>> > >>> Michael: > >>> > >>> I sure can! > >>> > >>> What I want to do is create template configurations that I can deploy > >>> to multiple servers. Part of the requirement of incoming calls is to > >>> configure which extension(s) ring immediately, 6-second delay, and > >>> 12-second delay. I have created groups for these extensions, and by > >>> using group_call I can get the full dial string for each group... > >>> perfect! Now, I just need a way to delay some of these extensions by > >>> 6 or 12 seconds. I ultimately want to inject leg variables into the > >>> dial string for each extension, so when the group_call is expanded > >>> each of them expand with the extra leg variable I define. > >>> > >>> Hope that helps... clear as mud? :) > >>> > >>> ~Brian > >>> > >>> On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins > >>> wrote: > >>> > Can you expand upon this question a bit? I'm curious if there's a > less > >>> > hackish way of doing what you want to do. Under what circumstances do > >>> > you > >>> > need to add the leg variables? Also, can you give us the big picture? > >>> > What's > >>> > the problem you're solving? > >>> > > >>> > -MC > >>> > > >>> > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese > >>> > wrote: > >>> >> > >>> >> Hi Everyone. > >>> >> > >>> >> I thought I read somewhere that this was possible, but I can't find > it > >>> >> now... > >>> >> > >>> >> I need a way to find-and-replace within a variable. So, for > example, > >>> >> I want to take variables that have values like this: > >>> >> > >>> >> 123abc123abc > >>> >> abc123abc123abc > >>> >> > >>> >> ...and do a find/replace of the "abc" with "xyz" so the variables > >>> >> would now return: > >>> >> > >>> >> 123xyz123xyz > >>> >> xyz123xyz123xyz > >>> >> > >>> >> The use case I've run into is that I need to add leg variables to > >>> >> group_call. In my case, group_call can return any number of > members, > >>> >> so I figured I would just replace the first "[" with > >>> >> "[variable-I-want-to-set=...". > >>> >> > >>> >> Thanks for the help! > >>> >> > >>> >> ~Brian > >>> >> > >>> >> > >>> >> > _________________________________________________________________________ > >>> >> Professional FreeSWITCH Consulting Services: > >>> >> consulting at freeswitch.org > >>> >> http://www.freeswitchsolutions.com > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> Official FreeSWITCH Sites > >>> >> http://www.freeswitch.org > >>> >> http://wiki.freeswitch.org > >>> >> http://www.cluecon.com > >>> >> > >>> >> FreeSWITCH-users mailing list > >>> >> FreeSWITCH-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> > > >>> > > >>> > > >>> > > >>> > > _________________________________________________________________________ > >>> > Professional FreeSWITCH Consulting Services: > >>> > consulting at freeswitch.org > >>> > http://www.freeswitchsolutions.com > >>> > > >>> > > >>> > > >>> > > >>> > Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://wiki.freeswitch.org > >>> > http://www.cluecon.com > >>> > > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at lists.freeswitch.org > >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> > http://www.freeswitch.org > >>> > > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/b15cda44/attachment.html From olimonkey at gmail.com Fri Jan 6 03:01:08 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 08:01:08 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: Thanks for the help so far. Here is a pastebin of FreeSWITCH output: http://pastebin.com/i6Qgc7ws Notice how the "has been answered" log message comes immediately (within a few milliseconds) after the call was originated. I think this would suggest that the CISCO is immediately sending a 200 OK, as you suggested. I also turned on CISCO debugging, but I'm just trying to figure out how to get the information regarding SIP messages back to Freeswitch. I'll run the test again and see if I can get some useful CISCO debug. Which "debug ccsip" commands are relevant to what I want for the CISCO SIP debugging? Thanks! 2012/1/6 Gustavo M?rsico : > I think I've a similar problem related to callcenter app. When I made an originate like this: > > originate loopback/2500/default/XML &bridge(user/2001) > > 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know > When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. > In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. > > originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) > > Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false > Dialplan: loopback/2500-b Action set(ignore_early_media=true) > Dialplan: loopback/2500-b Action callcenter(click2call) > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE > EXECUTE loopback/2500-b set(open=true) > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] > EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) > EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) > EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) > EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] > EXECUTE loopback/2500-b set(ignore_early_media=true) > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL > 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY > EXECUTE loopback/2500-b callcenter(click2call) > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL > 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE > 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" > > > > > On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > >> Also, maybe I should be doing something like this: >> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >> >> instead of: >> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >> >> >> but, I don't really have the CISCO configured as a gateway, nor do I >> know how really...probably not on the right track there. >> >> >> >> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>> *bump* >>> >>> >>> So I think maybe the way I'm doing the originate is the problem? In my >>> call string I'm creating a connection directly from the CISCO >>> (192.168.x.x) to the managed application, which may be why it starts >>> playing straight away? >>> >>> Maybe I should be originating a call first and then only once I know >>> the other side has picked up will I bridge the call to the IVR managed >>> application. >>> >>> Problem is I dunno how to tell whether the other person has picked up >>> (or even if the cisco is going to tell me) and I don't know how to do >>> things to a call once it has been established. >>> >>> >>> I'm currently reading the Dialplan wiki page, hoping to get something >>> out of it there. >>> >>> >>> Cheers >>> >>> Oliver >>> >>> >>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>> and connecting through a CISCO 2811. Most things now work quite well, >>>> but I am having a few issues with the way the system answers calls (or >>>> doesn't answer calls...). >>>> >>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>> which is then connected to a POTS phone line. >>>> >>>> >>>> Take the following scenario: >>>> >>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>> to freeswitch and originate a call: >>>> >>>> string callstring = >>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>> '&managed(ivrAppName)'"; >>>> eslConnection.API("originate", callstring); >>>> >>>> where 192.168.x.x is the CISCO IP. >>>> >>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>> number (091234567) to make the call. >>>> >>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>> >>>> >>>> >>>> These are my current problems: >>>> >>>> - IVR starts playing before I even pick up the phone. This means that >>>> if the system calls a mobile phone and the person doesn't pick up, the >>>> IVR will start playing and eventually the mobile phone will divert to >>>> voice mail. Obviously I then get a missed call and an sms saying I >>>> have a new voice mail, which is annoying. Instead I would like it to >>>> KNOW that no one has picked up, but I don't know how to do this. >>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>> has not yet been answered. For some reason however as soon as the >>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>> doing originate the wrong way or something ... >>>> >>>> - The phone only rings for about 10 seconds before hanging up. I've >>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>> CISCO "ring number". Nothing works, my phone still only rings for >>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>> starts playing even if no one answers the phone. >>>> >>>> >>>> >>>> >>>> >>>> CISCO Config for relevant FXO port: >>>> >>>> voice service voip >>>> ?allow-connections h323 to h323 >>>> ?allow-connections h323 to sip >>>> ?allow-connections sip to h323 >>>> ?allow-connections sip to sip >>>> ?no supplementary-service h450.2 >>>> ?no supplementary-service h450.3 >>>> ?supplementary-service h450.12 >>>> ?no supplementary-service sip moved-temporarily >>>> ?no supplementary-service sip refer >>>> ?fax protocol cisco >>>> ?sip >>>> ?registrar server expires max 3600 min 3600 >>>> ?no update-callerid >>>> ?no call service stop >>>> >>>> voice-port 0/3/2 >>>> ?output attenuation -3 >>>> ?no comfort-noise >>>> ?cptone AU >>>> ?impedance complex1 >>>> ?caller-id enable >>>> ! >>>> dial-peer voice 100 pots >>>> ?preference 1 >>>> ?destination-pattern 1T >>>> ?port 0/3/2 >>>> ! >>>> >>>> >>>> >>>> Many Thanks, >>>> >>>> Oliver >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From olimonkey at gmail.com Fri Jan 6 03:04:15 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 08:04:15 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: By the way: I tried {ignore_early_media=true} as well, but as I think we determined, my problem is probably with the CISCO telling FS that the call has been answered when really it hasn't yet. On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: > Thanks for the help so far. > > > Here is a pastebin of FreeSWITCH output: > http://pastebin.com/i6Qgc7ws > > Notice how the "has been answered" log message comes immediately > (within a few milliseconds) after the call was originated. I think > this would suggest that the CISCO is immediately sending a 200 OK, as > you suggested. I also turned on CISCO debugging, but I'm just trying > to figure out how to get the information regarding SIP messages back > to Freeswitch. I'll run the test again and see if I can get some > useful CISCO debug. > > Which "debug ccsip" commands are relevant to what I want for the CISCO > SIP debugging? > > > Thanks! > > > > > 2012/1/6 Gustavo M?rsico : >> I think I've a similar problem related to callcenter app. When I made an originate like this: >> >> originate loopback/2500/default/XML &bridge(user/2001) >> >> 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know >> When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. >> In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. >> >> originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) >> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >> Dialplan: loopback/2500-b Action callcenter(click2call) >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE >> EXECUTE loopback/2500-b set(open=true) >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >> EXECUTE loopback/2500-b set(ignore_early_media=true) >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY >> EXECUTE loopback/2500-b callcenter(click2call) >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" >> >> >> >> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >> >>> Also, maybe I should be doing something like this: >>> >>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>> >>> instead of: >>> >>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>> >>> >>> but, I don't really have the CISCO configured as a gateway, nor do I >>> know how really...probably not on the right track there. >>> >>> >>> >>> >>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>> *bump* >>>> >>>> >>>> So I think maybe the way I'm doing the originate is the problem? In my >>>> call string I'm creating a connection directly from the CISCO >>>> (192.168.x.x) to the managed application, which may be why it starts >>>> playing straight away? >>>> >>>> Maybe I should be originating a call first and then only once I know >>>> the other side has picked up will I bridge the call to the IVR managed >>>> application. >>>> >>>> Problem is I dunno how to tell whether the other person has picked up >>>> (or even if the cisco is going to tell me) and I don't know how to do >>>> things to a call once it has been established. >>>> >>>> >>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>> out of it there. >>>> >>>> >>>> Cheers >>>> >>>> Oliver >>>> >>>> >>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>> but I am having a few issues with the way the system answers calls (or >>>>> doesn't answer calls...). >>>>> >>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>> which is then connected to a POTS phone line. >>>>> >>>>> >>>>> Take the following scenario: >>>>> >>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>> to freeswitch and originate a call: >>>>> >>>>> string callstring = >>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>> '&managed(ivrAppName)'"; >>>>> eslConnection.API("originate", callstring); >>>>> >>>>> where 192.168.x.x is the CISCO IP. >>>>> >>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>> number (091234567) to make the call. >>>>> >>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>> >>>>> >>>>> >>>>> These are my current problems: >>>>> >>>>> - IVR starts playing before I even pick up the phone. This means that >>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>> IVR will start playing and eventually the mobile phone will divert to >>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>> has not yet been answered. For some reason however as soon as the >>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>> doing originate the wrong way or something ... >>>>> >>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>> starts playing even if no one answers the phone. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> CISCO Config for relevant FXO port: >>>>> >>>>> voice service voip >>>>> ?allow-connections h323 to h323 >>>>> ?allow-connections h323 to sip >>>>> ?allow-connections sip to h323 >>>>> ?allow-connections sip to sip >>>>> ?no supplementary-service h450.2 >>>>> ?no supplementary-service h450.3 >>>>> ?supplementary-service h450.12 >>>>> ?no supplementary-service sip moved-temporarily >>>>> ?no supplementary-service sip refer >>>>> ?fax protocol cisco >>>>> ?sip >>>>> ?registrar server expires max 3600 min 3600 >>>>> ?no update-callerid >>>>> ?no call service stop >>>>> >>>>> voice-port 0/3/2 >>>>> ?output attenuation -3 >>>>> ?no comfort-noise >>>>> ?cptone AU >>>>> ?impedance complex1 >>>>> ?caller-id enable >>>>> ! >>>>> dial-peer voice 100 pots >>>>> ?preference 1 >>>>> ?destination-pattern 1T >>>>> ?port 0/3/2 >>>>> ! >>>>> >>>>> >>>>> >>>>> Many Thanks, >>>>> >>>>> Oliver >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From olimonkey at gmail.com Fri Jan 6 03:25:08 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 08:25:08 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: I just noticed something else, if I don't pick up the phone at all. The IVR just keeps playing until the menu timeout kicks in. So here is a CISCO SIP log: http://pastebin.com/Y9sYkuxi The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. I hope the CISCO log is readable, it's a bit long because I just did "debug ccsip all". In this test I didn't bother picking up the phone at all, but I can see that FS answered anyway and the IVR kept playing until it timed out. I'm not an expert, but here is what I picked out of it: At 00:08:10 we get a Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" the further down at the same timestamp we get Sent: "SIP/2.0 100 Trying" At 00:08:13 we get a Sent: "SIP/2.0 183 Session Progress" At 00:18:13 we get a Sent: "SIP/2.0 200 OK" Then at the same timestamp we get: Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" Once the IVR times out at 00:09:16 we get Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" And then the reply right after Sent: "SIP/2.0 200 OK" So I think you were right, the CISCO is sending back an "OK" 3 seconds after the "INVITE" is received. The part that is beyond my field of expertise so far is WHY? Thanks, Oliver On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: > By the way: > > I tried {ignore_early_media=true} as well, but as I think we > determined, my problem is probably with the CISCO telling FS that the > call has been answered when really it hasn't yet. > > > > On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >> Thanks for the help so far. >> >> >> Here is a pastebin of FreeSWITCH output: >> http://pastebin.com/i6Qgc7ws >> >> Notice how the "has been answered" log message comes immediately >> (within a few milliseconds) after the call was originated. I think >> this would suggest that the CISCO is immediately sending a 200 OK, as >> you suggested. I also turned on CISCO debugging, but I'm just trying >> to figure out how to get the information regarding SIP messages back >> to Freeswitch. I'll run the test again and see if I can get some >> useful CISCO debug. >> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >> SIP debugging? >> >> >> Thanks! >> >> >> >> >> 2012/1/6 Gustavo M?rsico : >>> I think I've a similar problem related to callcenter app. When I made an originate like this: >>> >>> originate loopback/2500/default/XML &bridge(user/2001) >>> >>> 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know >>> When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. >>> In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. >>> >>> originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) >>> >>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false >>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>> Dialplan: loopback/2500-b Action callcenter(click2call) >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE >>> EXECUTE loopback/2500-b set(open=true) >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] >>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b >>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY >>> EXECUTE loopback/2500-b callcenter(click2call) >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE >>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE >>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" >>> >>> >>> >>> >>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>> >>>> Also, maybe I should be doing something like this: >>>> >>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>> >>>> instead of: >>>> >>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>> >>>> >>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>> know how really...probably not on the right track there. >>>> >>>> >>>> >>>> >>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>> *bump* >>>>> >>>>> >>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>> call string I'm creating a connection directly from the CISCO >>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>> playing straight away? >>>>> >>>>> Maybe I should be originating a call first and then only once I know >>>>> the other side has picked up will I bridge the call to the IVR managed >>>>> application. >>>>> >>>>> Problem is I dunno how to tell whether the other person has picked up >>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>> things to a call once it has been established. >>>>> >>>>> >>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>> out of it there. >>>>> >>>>> >>>>> Cheers >>>>> >>>>> Oliver >>>>> >>>>> >>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>>> but I am having a few issues with the way the system answers calls (or >>>>>> doesn't answer calls...). >>>>>> >>>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>>> which is then connected to a POTS phone line. >>>>>> >>>>>> >>>>>> Take the following scenario: >>>>>> >>>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>>> to freeswitch and originate a call: >>>>>> >>>>>> string callstring = >>>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>>> '&managed(ivrAppName)'"; >>>>>> eslConnection.API("originate", callstring); >>>>>> >>>>>> where 192.168.x.x is the CISCO IP. >>>>>> >>>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>>> number (091234567) to make the call. >>>>>> >>>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>>> >>>>>> >>>>>> >>>>>> These are my current problems: >>>>>> >>>>>> - IVR starts playing before I even pick up the phone. This means that >>>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>>> IVR will start playing and eventually the mobile phone will divert to >>>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>>> has not yet been answered. For some reason however as soon as the >>>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>>> doing originate the wrong way or something ... >>>>>> >>>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>>> starts playing even if no one answers the phone. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> CISCO Config for relevant FXO port: >>>>>> >>>>>> voice service voip >>>>>> ?allow-connections h323 to h323 >>>>>> ?allow-connections h323 to sip >>>>>> ?allow-connections sip to h323 >>>>>> ?allow-connections sip to sip >>>>>> ?no supplementary-service h450.2 >>>>>> ?no supplementary-service h450.3 >>>>>> ?supplementary-service h450.12 >>>>>> ?no supplementary-service sip moved-temporarily >>>>>> ?no supplementary-service sip refer >>>>>> ?fax protocol cisco >>>>>> ?sip >>>>>> ?registrar server expires max 3600 min 3600 >>>>>> ?no update-callerid >>>>>> ?no call service stop >>>>>> >>>>>> voice-port 0/3/2 >>>>>> ?output attenuation -3 >>>>>> ?no comfort-noise >>>>>> ?cptone AU >>>>>> ?impedance complex1 >>>>>> ?caller-id enable >>>>>> ! >>>>>> dial-peer voice 100 pots >>>>>> ?preference 1 >>>>>> ?destination-pattern 1T >>>>>> ?port 0/3/2 >>>>>> ! >>>>>> >>>>>> >>>>>> >>>>>> Many Thanks, >>>>>> >>>>>> Oliver >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org From olimonkey at gmail.com Fri Jan 6 04:13:12 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 09:13:12 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: Shouldn't there be a 180 RINGING somewhere in there? On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: > I just noticed something else, if I don't pick up the phone at all. > The IVR just keeps playing until the menu timeout kicks in. > > So here is a CISCO SIP log: > http://pastebin.com/Y9sYkuxi > > The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. > I hope the CISCO log is readable, it's a bit long because I just did > "debug ccsip all". > > > > In this test I didn't bother picking up the phone at all, but I can > see that FS answered anyway and the IVR kept playing until it timed > out. > I'm not an expert, but here is what I picked out of it: > > At 00:08:10 we get a > Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" > > the further down at the same timestamp we get > Sent: "SIP/2.0 100 Trying" > > At 00:08:13 we get a > Sent: "SIP/2.0 183 Session Progress" > > At 00:18:13 we get a > Sent: "SIP/2.0 200 OK" > > Then at the same timestamp we get: > Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > > > Once the IVR times out at 00:09:16 we get > Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > And then the reply right after > Sent: "SIP/2.0 200 OK" > > > > So I think you were right, the CISCO is sending back an "OK" 3 seconds > after the "INVITE" is received. > > > > The part that is beyond my field of expertise so far is WHY? > > > > Thanks, > > > Oliver > > > > On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >> By the way: >> >> I tried {ignore_early_media=true} as well, but as I think we >> determined, my problem is probably with the CISCO telling FS that the >> call has been answered when really it hasn't yet. >> >> >> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>> Thanks for the help so far. >>> >>> >>> Here is a pastebin of FreeSWITCH output: >>> http://pastebin.com/i6Qgc7ws >>> >>> Notice how the "has been answered" log message comes immediately >>> (within a few milliseconds) after the call was originated. I think >>> this would suggest that the CISCO is immediately sending a 200 OK, as >>> you suggested. I also turned on CISCO debugging, but I'm just trying >>> to figure out how to get the information regarding SIP messages back >>> to Freeswitch. I'll run the test again and see if I can get some >>> useful CISCO debug. >>> >>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>> SIP debugging? >>> >>> >>> Thanks! >>> >>> >>> >>> >>> 2012/1/6 Gustavo M?rsico : >>>> I think I've a similar problem related to callcenter app. When I made an originate like this: >>>> >>>> originate loopback/2500/default/XML &bridge(user/2001) >>>> >>>> 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know >>>> When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. >>>> In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. >>>> >>>> originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) >>>> >>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false >>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE >>>> EXECUTE loopback/2500-b set(open=true) >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] >>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b >>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY >>>> EXECUTE loopback/2500-b callcenter(click2call) >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE >>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE >>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" >>>> >>>> >>>> >>>> >>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>> >>>>> Also, maybe I should be doing something like this: >>>>> >>>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>>> >>>>> instead of: >>>>> >>>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>>> >>>>> >>>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>>> know how really...probably not on the right track there. >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>>> *bump* >>>>>> >>>>>> >>>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>>> call string I'm creating a connection directly from the CISCO >>>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>>> playing straight away? >>>>>> >>>>>> Maybe I should be originating a call first and then only once I know >>>>>> the other side has picked up will I bridge the call to the IVR managed >>>>>> application. >>>>>> >>>>>> Problem is I dunno how to tell whether the other person has picked up >>>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>>> things to a call once it has been established. >>>>>> >>>>>> >>>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>>> out of it there. >>>>>> >>>>>> >>>>>> Cheers >>>>>> >>>>>> Oliver >>>>>> >>>>>> >>>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>>>> but I am having a few issues with the way the system answers calls (or >>>>>>> doesn't answer calls...). >>>>>>> >>>>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>>>> which is then connected to a POTS phone line. >>>>>>> >>>>>>> >>>>>>> Take the following scenario: >>>>>>> >>>>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>>>> to freeswitch and originate a call: >>>>>>> >>>>>>> string callstring = >>>>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>>>> '&managed(ivrAppName)'"; >>>>>>> eslConnection.API("originate", callstring); >>>>>>> >>>>>>> where 192.168.x.x is the CISCO IP. >>>>>>> >>>>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>>>> number (091234567) to make the call. >>>>>>> >>>>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>>>> >>>>>>> >>>>>>> >>>>>>> These are my current problems: >>>>>>> >>>>>>> - IVR starts playing before I even pick up the phone. This means that >>>>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>>>> IVR will start playing and eventually the mobile phone will divert to >>>>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>>>> has not yet been answered. For some reason however as soon as the >>>>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>>>> doing originate the wrong way or something ... >>>>>>> >>>>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>>>> starts playing even if no one answers the phone. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> CISCO Config for relevant FXO port: >>>>>>> >>>>>>> voice service voip >>>>>>> ?allow-connections h323 to h323 >>>>>>> ?allow-connections h323 to sip >>>>>>> ?allow-connections sip to h323 >>>>>>> ?allow-connections sip to sip >>>>>>> ?no supplementary-service h450.2 >>>>>>> ?no supplementary-service h450.3 >>>>>>> ?supplementary-service h450.12 >>>>>>> ?no supplementary-service sip moved-temporarily >>>>>>> ?no supplementary-service sip refer >>>>>>> ?fax protocol cisco >>>>>>> ?sip >>>>>>> ?registrar server expires max 3600 min 3600 >>>>>>> ?no update-callerid >>>>>>> ?no call service stop >>>>>>> >>>>>>> voice-port 0/3/2 >>>>>>> ?output attenuation -3 >>>>>>> ?no comfort-noise >>>>>>> ?cptone AU >>>>>>> ?impedance complex1 >>>>>>> ?caller-id enable >>>>>>> ! >>>>>>> dial-peer voice 100 pots >>>>>>> ?preference 1 >>>>>>> ?destination-pattern 1T >>>>>>> ?port 0/3/2 >>>>>>> ! >>>>>>> >>>>>>> >>>>>>> >>>>>>> Many Thanks, >>>>>>> >>>>>>> Oliver >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org From brian at freeswitch.org Fri Jan 6 04:20:28 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 5 Jan 2012 19:20:28 -0600 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: Message-ID: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 is usually RINGING (generate ringback locally) while a 183 has media... aka early media and usually provides ringback inband. /b On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: > Shouldn't there be a 180 RINGING somewhere in there? > > > > > On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >> I just noticed something else, if I don't pick up the phone at all. >> The IVR just keeps playing until the menu timeout kicks in. >> >> So here is a CISCO SIP log: >> http://pastebin.com/Y9sYkuxi >> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >> I hope the CISCO log is readable, it's a bit long because I just did >> "debug ccsip all". >> >> >> >> In this test I didn't bother picking up the phone at all, but I can >> see that FS answered anyway and the IVR kept playing until it timed >> out. >> I'm not an expert, but here is what I picked out of it: >> >> At 00:08:10 we get a >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >> >> the further down at the same timestamp we get >> Sent: "SIP/2.0 100 Trying" >> >> At 00:08:13 we get a >> Sent: "SIP/2.0 183 Session Progress" >> >> At 00:18:13 we get a >> Sent: "SIP/2.0 200 OK" >> >> Then at the same timestamp we get: >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> >> Once the IVR times out at 00:09:16 we get >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> And then the reply right after >> Sent: "SIP/2.0 200 OK" >> >> >> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >> after the "INVITE" is received. >> >> >> >> The part that is beyond my field of expertise so far is WHY? >> >> >> >> Thanks, >> >> >> Oliver >> >> >> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>> By the way: >>> >>> I tried {ignore_early_media=true} as well, but as I think we >>> determined, my problem is probably with the CISCO telling FS that the >>> call has been answered when really it hasn't yet. >>> >>> >>> >>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>>> Thanks for the help so far. >>>> >>>> >>>> Here is a pastebin of FreeSWITCH output: >>>> http://pastebin.com/i6Qgc7ws >>>> >>>> Notice how the "has been answered" log message comes immediately >>>> (within a few milliseconds) after the call was originated. I think >>>> this would suggest that the CISCO is immediately sending a 200 OK, as >>>> you suggested. I also turned on CISCO debugging, but I'm just trying >>>> to figure out how to get the information regarding SIP messages back >>>> to Freeswitch. I'll run the test again and see if I can get some >>>> useful CISCO debug. >>>> >>>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>>> SIP debugging? >>>> >>>> >>>> Thanks! >>>> >>>> >>>> >>>> >>>> 2012/1/6 Gustavo M?rsico : >>>>> I think I've a similar problem related to callcenter app. When I made an originate like this: >>>>> >>>>> originate loopback/2500/default/XML &bridge(user/2001) >>>>> >>>>> 2500 is an extension that leads to a callcenter application. In this case, we dial first to the queue and when an agent answered we call to the customer. As far as I know >>>>> When the A-leg reaches to the queue, without selecting an agent, the call is automatically sent to the B-leg. As far as I see, there is a pre-answer method that fs needs to send the media to A-leg. >>>>> In order to try to avoid this, I tried using ignore_early_media=true as part of the originate in A-leg and/or B-leg, with no luck. >>>>> >>>>> originate {ignore_early_media=true}loopback/2500/default/XML &bridge({ignore_early_media=true}user/2001) >>>>> >>>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] destination_number(2500) =~ /^(2500)$/ break=on-false >>>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal loopback/2500-b [BREAK] >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 (loopback/2500-b) State ROUTING going to sleep >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 (loopback/2500-b) Running State Change CS_EXECUTE >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 (loopback/2500-b) State EXECUTE >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b CHANNEL EXECUTE >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 loopback/2500-b Standard EXECUTE >>>>> EXECUTE loopback/2500-b set(open=true) >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [open]=[true] >>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET [ignore_early_media]=[true] >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application callcenter Requires media! pre_answering channel loopback/2500-b >>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/2500-a! >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) Callstate Change RINGING -> EARLY >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer loopback/2500-b! >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) Callstate Change RINGING -> EARLY >>>>> EXECUTE loopback/2500-b callcenter(click2call) >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) Callstate Change EARLY -> ACTIVE >>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel [loopback/2500-a] has been answered >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal loopback/2500-b [BREAK] >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b CHANNEL KILL >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [loopback/2500-a] >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) Callstate Change EARLY -> ACTIVE >>>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a Flipping CID from "" <0000000000> to "Outbound Call" >>>>> >>>>> >>>>> >>>>> >>>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>>> >>>>>> Also, maybe I should be doing something like this: >>>>>> >>>>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>>>> >>>>>> instead of: >>>>>> >>>>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>>>> >>>>>> >>>>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>>>> know how really...probably not on the right track there. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>>>> *bump* >>>>>>> >>>>>>> >>>>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>>>> call string I'm creating a connection directly from the CISCO >>>>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>>>> playing straight away? >>>>>>> >>>>>>> Maybe I should be originating a call first and then only once I know >>>>>>> the other side has picked up will I bridge the call to the IVR managed >>>>>>> application. >>>>>>> >>>>>>> Problem is I dunno how to tell whether the other person has picked up >>>>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>>>> things to a call once it has been established. >>>>>>> >>>>>>> >>>>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>>>> out of it there. >>>>>>> >>>>>>> >>>>>>> Cheers >>>>>>> >>>>>>> Oliver >>>>>>> >>>>>>> >>>>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>>>>> but I am having a few issues with the way the system answers calls (or >>>>>>>> doesn't answer calls...). >>>>>>>> >>>>>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>>>>> which is then connected to a POTS phone line. >>>>>>>> >>>>>>>> >>>>>>>> Take the following scenario: >>>>>>>> >>>>>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>>>>> to freeswitch and originate a call: >>>>>>>> >>>>>>>> string callstring = >>>>>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>>>>> '&managed(ivrAppName)'"; >>>>>>>> eslConnection.API("originate", callstring); >>>>>>>> >>>>>>>> where 192.168.x.x is the CISCO IP. >>>>>>>> >>>>>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>>>>> number (091234567) to make the call. >>>>>>>> >>>>>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> These are my current problems: >>>>>>>> >>>>>>>> - IVR starts playing before I even pick up the phone. This means that >>>>>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>>>>> IVR will start playing and eventually the mobile phone will divert to >>>>>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>>>>> has not yet been answered. For some reason however as soon as the >>>>>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>>>>> doing originate the wrong way or something ... >>>>>>>> >>>>>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>>>>> starts playing even if no one answers the phone. >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> CISCO Config for relevant FXO port: >>>>>>>> >>>>>>>> voice service voip >>>>>>>> allow-connections h323 to h323 >>>>>>>> allow-connections h323 to sip >>>>>>>> allow-connections sip to h323 >>>>>>>> allow-connections sip to sip >>>>>>>> no supplementary-service h450.2 >>>>>>>> no supplementary-service h450.3 >>>>>>>> supplementary-service h450.12 >>>>>>>> no supplementary-service sip moved-temporarily >>>>>>>> no supplementary-service sip refer >>>>>>>> fax protocol cisco >>>>>>>> sip >>>>>>>> registrar server expires max 3600 min 3600 >>>>>>>> no update-callerid >>>>>>>> no call service stop >>>>>>>> >>>>>>>> voice-port 0/3/2 >>>>>>>> output attenuation -3 >>>>>>>> no comfort-noise >>>>>>>> cptone AU >>>>>>>> impedance complex1 >>>>>>>> caller-id enable >>>>>>>> ! >>>>>>>> dial-peer voice 100 pots >>>>>>>> preference 1 >>>>>>>> destination-pattern 1T >>>>>>>> port 0/3/2 >>>>>>>> ! >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> Many Thanks, >>>>>>>> >>>>>>>> Oliver >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120105/7912ec97/attachment-0001.html From brian.wiese.freeswitch at gmail.com Fri Jan 6 04:32:18 2012 From: brian.wiese.freeswitch at gmail.com (Brian Wiese) Date: Thu, 5 Jan 2012 19:32:18 -0600 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Wikified: http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Search_and_replace :) On Thu, Jan 5, 2012 at 5:36 PM, Michael Collins wrote: > Nicely done! Be sure to wikify this one. If you have any questions about > updating the wiki just email me off list. > > -MC > > > On Thu, Jan 5, 2012 at 1:19 PM, Brian Wiese > wrote: >> >> Michael: >> >> Here's what I came up with, and it works really well! ?I can embed >> this in a dialplan or use it at the CLI: >> >> lua ~stream:write(tostring(string.gsub("string_to_search", >> "string_to_find", "replacement_string"))) >> >> It's awesome! >> >> Thanks for your help! >> >> ~Brian >> >> On Tue, Jan 3, 2012 at 6:16 PM, Brian Wiese >> wrote: >> > Michael: >> > >> > I do think that mod_xml_curl is a little more than what I need here. >> > I'll try a Lua script and see where I get. ?I'll try to call a Lua >> > script and return the bridge string back to the dial plan. >> > >> > I'll let you know how my scripting fu works... >> > >> > ~Brian >> > >> > On Tue, Jan 3, 2012 at 6:10 PM, Michael Collins >> > wrote: >> >> This kinda sounds like a problem in need of mod_xml_curl. If that seems >> >> like >> >> too much of a hassle then I would fall back to a mod_lua or mod_perl >> >> script >> >> to do the regex stuff. How is your scripting fu? >> >> >> >> -MC >> >> >> >> >> >> On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese >> >> wrote: >> >>> >> >>> Michael: >> >>> >> >>> I sure can! >> >>> >> >>> What I want to do is create template configurations that I can deploy >> >>> to multiple servers. ?Part of the requirement of incoming calls is to >> >>> configure which extension(s) ring immediately, 6-second delay, and >> >>> 12-second delay. ?I have created groups for these extensions, and by >> >>> using group_call I can get the full dial string for each group... >> >>> perfect! ?Now, I just need a way to delay some of these extensions by >> >>> 6 or 12 seconds. ?I ultimately want to inject leg variables into the >> >>> dial string for each extension, so when the group_call is expanded >> >>> each of them expand with the extra leg variable I define. >> >>> >> >>> Hope that helps... ?clear as mud? ?:) >> >>> >> >>> ~Brian >> >>> >> >>> On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins >> >>> wrote: >> >>> > Can you expand upon this question a bit? I'm curious if there's a >> >>> > less >> >>> > hackish way of doing what you want to do. Under what circumstances >> >>> > do >> >>> > you >> >>> > need to add the leg variables? Also, can you give us the big >> >>> > picture? >> >>> > What's >> >>> > the problem you're solving? >> >>> > >> >>> > -MC >> >>> > >> >>> > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese >> >>> > wrote: >> >>> >> >> >>> >> Hi Everyone. >> >>> >> >> >>> >> I thought I read somewhere that this was possible, but I can't find >> >>> >> it >> >>> >> now... >> >>> >> >> >>> >> I need a way to find-and-replace within a variable. ?So, for >> >>> >> example, >> >>> >> I want to take variables that have values like this: >> >>> >> >> >>> >> 123abc123abc >> >>> >> abc123abc123abc >> >>> >> >> >>> >> ...and do a find/replace of the "abc" with "xyz" so the variables >> >>> >> would now return: >> >>> >> >> >>> >> 123xyz123xyz >> >>> >> xyz123xyz123xyz >> >>> >> >> >>> >> The use case I've run into is that I need to add leg variables to >> >>> >> group_call. ?In my case, group_call can return any number of >> >>> >> members, >> >>> >> so I figured I would just replace the first "[" with >> >>> >> "[variable-I-want-to-set=...". >> >>> >> >> >>> >> Thanks for the help! >> >>> >> >> >>> >> ~Brian >> >>> >> >> >>> >> >> >>> >> >> >>> >> _________________________________________________________________________ >> >>> >> Professional FreeSWITCH Consulting Services: >> >>> >> consulting at freeswitch.org >> >>> >> http://www.freeswitchsolutions.com >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> Official FreeSWITCH Sites >> >>> >> http://www.freeswitch.org >> >>> >> http://wiki.freeswitch.org >> >>> >> http://www.cluecon.com >> >>> >> >> >>> >> FreeSWITCH-users mailing list >> >>> >> FreeSWITCH-users at lists.freeswitch.org >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >> >>> >> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> >> http://www.freeswitch.org >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > _________________________________________________________________________ >> >>> > Professional FreeSWITCH Consulting Services: >> >>> > consulting at freeswitch.org >> >>> > http://www.freeswitchsolutions.com >> >>> > >> >>> > >> >>> > >> >>> > >> >>> > Official FreeSWITCH Sites >> >>> > http://www.freeswitch.org >> >>> > http://wiki.freeswitch.org >> >>> > http://www.cluecon.com >> >>> > >> >>> > FreeSWITCH-users mailing list >> >>> > FreeSWITCH-users at lists.freeswitch.org >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> > >> >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> > http://www.freeswitch.org >> >>> > >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From valery.kalinin at gmail.com Fri Jan 6 07:56:00 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Fri, 6 Jan 2012 10:56:00 +0600 Subject: [Freeswitch-users] Why database is locked? Message-ID: Hi all! Periodically database is locked and all calls stopped. Log: 2012-01-06 08:32:52.492796 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] update sip_dialogs set presence_id='2002 at 192.168.205.1',presence_data='' where uuid='f506627c-ea28-481c-ba72-beaeebc16a6b'; 2012-01-06 08:33:12.486159 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2012-01-06 08:33:22.591484 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] insert into sip_dialogs (call_id,uuid,sip_to_user,sip_to_host,sip_to_tag,sip_from_user,sip_from_host,sip_from_tag,contact_user,contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,call_info,rcd) values('810ae706-b2b1-122f-aa88-6cf049ef3354','b925d777-a70c-4d71-a07b-bce5a804f48a','2002','192.168.205.1','e21ByaF046ppQ','2003','192.168.205.1','1436972916','2003','192.168.205.217','early','outbound','Yealink SIP-T22P 7.60.14.5','internal','srv-pok-phone','< sip:2003 at 192.168.205.217:5062>','2003 at 192.168.205.1','','',1325817146) 2012-01-06 08:33:41.089426 [ERR] ftmod_zt.c:1280 [s1c16][1:16] Dropping event 8 to be able to write data 2012-01-06 08:33:52.667878 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] delete from sip_dialogs where uuid='8ed4b985-1f1f-4901-938d-56619a42eb1c' 2012-01-06 08:34:02.189552 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC abort frame received (ZT_EVENT_ABORT) 2012-01-06 08:34:22.607797 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] delete from sip_authentication where expires > 0 and expires <= 1325817161 and hostname='srv-pok-phone.lpurs.argos-group.ru' 2012-01-06 08:34:52.460796 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('9643f3a4-acc0-41cf-8a08-ca9dd791dbbb', 1325817222, 'internal', 'srv-pok-phone', 0) 2012-01-06 08:42:54.233881 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('4611d24f-cfa0-47d4-a8d3-5fdce08ed85a', 1325817734, 'internal', 'srv-pok-phone', 0) 2012-01-06 08:42:54.884449 [CRIT] ftdm_io.c:5467 [s1c9][1:9] Forcing hangup since the user did not confirmed our hangup after 3000ms 2012-01-06 08:43:06.435420 [ERR] ftmod_libpri.c:132 XXX Progress message requested but no information is provided 2012-01-06 08:43:06.813352 [ERR] ftmod_libpri.c:132 Received unsolicited status: Mandatory information element is missing 2012-01-06 08:43:24.297760 [ERR] switch_core_sqldb.c:481 SQL ERR [database is locked] delete from sip_dialogs where uuid='563a394c-5c82-4381-868d-35e344f3ccd9' Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/d99a9cab/attachment.html From bdfoster at endigotech.com Fri Jan 6 08:18:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 6 Jan 2012 00:18:58 -0500 Subject: [Freeswitch-users] Session timeout In-Reply-To: <4F063163.50005@telefaks.de> References: <4F063163.50005@telefaks.de> Message-ID: Shot in the dark here... have you reloaded your internal profile? On Jan 5, 2012 6:27 PM, "Peter Steinbach" wrote: > I have enabled session timers in my sofia internal profile the following > way: > > > > However I do not see any Re-Invite or Update packets sent from > Freeswitch to my User Agents. > > What am I doing wrong? > > Peter > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/e30e79fe/attachment.html From olimonkey at gmail.com Fri Jan 6 08:55:51 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 13:55:51 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> Message-ID: I've tried looking at disable-early-media configuration command, but that didn't work and I doubt that has anything to do with the CISCO sending a 200 OK right after a 183 SESSION PROGRESS. On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: > Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 > is usually RINGING (generate ringback locally) while a 183 has media... aka > early media and usually provides ringback inband. > > /b > > On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: > > Shouldn't there be a ?180 RINGING ?somewhere in there? > > > > > On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: > > I just noticed something else, if I don't pick up the phone at all. > > The IVR just keeps playing until the menu timeout kicks in. > > > So here is a CISCO SIP log: > > http://pastebin.com/Y9sYkuxi > > > The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. > > I hope the CISCO log is readable, it's a bit long because I just did > > "debug ccsip all". > > > > > In this test I didn't bother picking up the phone at all, but I can > > see that FS answered anyway and the IVR kept playing until it timed > > out. > > I'm not an expert, but here is what I picked out of it: > > > At 00:08:10 we get a > > Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" > > > the further down at the same timestamp we get > > Sent: "SIP/2.0 100 Trying" > > > At 00:08:13 we get a > > Sent: "SIP/2.0 183 Session Progress" > > > At 00:18:13 we get a > > Sent: "SIP/2.0 200 OK" > > > Then at the same timestamp we get: > > Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > > > > Once the IVR times out at 00:09:16 we get > > Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > > And then the reply right after > > Sent: "SIP/2.0 200 OK" > > > > > So I think you were right, the CISCO is sending back an "OK" 3 seconds > > after the "INVITE" is received. > > > > > The part that is beyond my field of expertise so far is WHY? > > > > > Thanks, > > > > Oliver > > > > > On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: > > By the way: > > > I tried {ignore_early_media=true} as well, but as I think we > > determined, my problem is probably with the CISCO telling FS that the > > call has been answered when really it hasn't yet. > > > > > On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: > > Thanks for the help so far. > > > > Here is a pastebin of FreeSWITCH output: > > http://pastebin.com/i6Qgc7ws > > > Notice how the "has been answered" log message comes immediately > > (within a few milliseconds) after the call was originated. I think > > this would suggest that the CISCO is immediately sending a 200 OK, as > > you suggested. I also turned on CISCO debugging, but I'm just trying > > to figure out how to get the information regarding SIP messages back > > to Freeswitch. I'll run the test again and see if I can get some > > useful CISCO debug. > > > Which "debug ccsip" commands are relevant to what I want for the CISCO > > SIP debugging? > > > > Thanks! > > > > > > 2012/1/6 Gustavo M?rsico : > > I think I've a similar problem related to callcenter app. When I made an > originate like this: > > > originate loopback/2500/default/XML &bridge(user/2001) > > > 2500 is an extension that leads to a callcenter application. In this case, > we dial first to the queue and when an agent answered we call to the > customer. As far as I know > > When the A-leg reaches to the queue, without selecting an agent, the call is > automatically sent to the B-leg. As far as I see, there is a pre-answer > method that fs needs to send the media to A-leg. > > In order to try to avoid this, I tried using ignore_early_media=true as part > of the originate in A-leg and/or B-leg, with no luck. > > > originate {ignore_early_media=true}loopback/2500/default/XML > &bridge({ignore_early_media=true}user/2001) > > > Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] > destination_number(2500) =~ /^(2500)$/ break=on-false > > Dialplan: loopback/2500-b Action set(ignore_early_media=true) > > Dialplan: loopback/2500-b Action callcenter(click2call) > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 > (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 > (loopback/2500-b) State ROUTING going to sleep > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 > (loopback/2500-b) Running State Change CS_EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 > (loopback/2500-b) State EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b > CHANNEL EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 > loopback/2500-b Standard EXECUTE > > EXECUTE loopback/2500-b set(open=true) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [open]=[true] > > EXECUTE loopback/2500-b > hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) > > EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) > > EXECUTE loopback/2500-b > hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) > > EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] > > EXECUTE loopback/2500-b set(ignore_early_media=true) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [ignore_early_media]=[true] > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application > callcenter Requires media! pre_answering channel loopback/2500-b > > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer > loopback/2500-a! > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) > Callstate Change RINGING -> EARLY > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer > loopback/2500-b! > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) > Callstate Change RINGING -> EARLY > > EXECUTE loopback/2500-b callcenter(click2call) > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) > Callstate Change EARLY -> ACTIVE > > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel > [loopback/2500-a] has been answered > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate > Resulted in Success: [loopback/2500-a] > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) > Callstate Change EARLY -> ACTIVE > > 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a > Flipping CID from "" <0000000000> to "Outbound Call" > > > > > > On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > > > Also, maybe I should be doing something like this: > > > sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' > > > instead of: > > > sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' > > > > but, I don't really have the CISCO configured as a gateway, nor do I > > know how really...probably not on the right track there. > > > > > > On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: > > *bump* > > > > So I think maybe the way I'm doing the originate is the problem? In my > > call string I'm creating a connection directly from the CISCO > > (192.168.x.x) to the managed application, which may be why it starts > > playing straight away? > > > Maybe I should be originating a call first and then only once I know > > the other side has picked up will I bridge the call to the IVR managed > > application. > > > Problem is I dunno how to tell whether the other person has picked up > > (or even if the cisco is going to tell me) and I don't know how to do > > things to a call once it has been established. > > > > I'm currently reading the Dialplan wiki page, hoping to get something > > out of it there. > > > > Cheers > > > Oliver > > > > On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: > > I've been battling while creating an IVR using FreeSWITCH mod_managed > > and connecting through a CISCO 2811. Most things now work quite well, > > but I am having a few issues with the way the system answers calls (or > > doesn't answer calls...). > > > I have FreeSWITCH running as a windows service on Windows Server 2008, > > which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > > which is then connected to a POTS phone line. > > > > Take the following scenario: > > > 1. Managed .NET application creates a call string and uses ESL to talk > > to freeswitch and originate a call: > > > string callstring = > > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x > > '&managed(ivrAppName)'"; > > eslConnection.API("originate", callstring); > > > where 192.168.x.x is the CISCO IP. > > > 2. The CISCO sees that the phone number (1091234567) starts with a 1 > > so it uses FXO port 1 and strips the 1 and uses the remaining phone > > number (091234567) to make the call. > > > 3. My phone rings, I pick up and I can hear my IVR playing. > > > > > These are my current problems: > > > - IVR starts playing before I even pick up the phone. This means that > > if the system calls a mobile phone and the person doesn't pick up, the > > IVR will start playing and eventually the mobile phone will divert to > > voice mail. Obviously I then get a missed call and an sms saying I > > have a new voice mail, which is annoying. Instead I would like it to > > KNOW that no one has picked up, but I don't know how to do this. > > Somehow the CISCO needs to be able to tell FreeSWITCH that the call > > has not yet been answered. For some reason however as soon as the > > CISCO starts calling FreeSWITCH thinks the call is already connected. > > It doesn't know that the CISCO is actually still ringing. Maybe I'm > > doing originate the wrong way or something ... > > > - The phone only rings for about 10 seconds before hanging up. I've > > tried "call_timeout", "bridge_answer_timeout". I've also tried setting > > CISCO "ring number". Nothing works, my phone still only rings for > > about 10 seconds. I don't know if this is a FreeSWITCH issue or a > > CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > > starts playing even if no one answers the phone. > > > > > > > CISCO Config for relevant FXO port: > > > voice service voip > > ?allow-connections h323 to h323 > > ?allow-connections h323 to sip > > ?allow-connections sip to h323 > > ?allow-connections sip to sip > > ?no supplementary-service h450.2 > > ?no supplementary-service h450.3 > > ?supplementary-service h450.12 > > ?no supplementary-service sip moved-temporarily > > ?no supplementary-service sip refer > > ?fax protocol cisco > > ?sip > > ?registrar server expires max 3600 min 3600 > > ?no update-callerid > > ?no call service stop > > > voice-port 0/3/2 > > ?output attenuation -3 > > ?no comfort-noise > > ?cptone AU > > ?impedance complex1 > > ?caller-id enable > > ! > > dial-peer voice 100 pots > > ?preference 1 > > ?destination-pattern 1T > > ?port 0/3/2 > > ! > > > > > Many Thanks, > > > Oliver > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: ? +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peter.olsson at visionutveckling.se Fri Jan 6 11:20:49 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 6 Jan 2012 08:20:49 +0000 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org>, Message-ID: <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> If it sends 200 OK right after 183, this IS the problem. 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. Seems to me there is something broken in the Cisco. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] Skickat: den 6 januari 2012 06:55 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR I've tried looking at disable-early-media configuration command, but that didn't work and I doubt that has anything to do with the CISCO sending a 200 OK right after a 183 SESSION PROGRESS. On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: > Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 > is usually RINGING (generate ringback locally) while a 183 has media... aka > early media and usually provides ringback inband. > > /b > > On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: > > Shouldn't there be a 180 RINGING somewhere in there? > > > > > On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: > > I just noticed something else, if I don't pick up the phone at all. > > The IVR just keeps playing until the menu timeout kicks in. > > > So here is a CISCO SIP log: > > http://pastebin.com/Y9sYkuxi > > > The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. > > I hope the CISCO log is readable, it's a bit long because I just did > > "debug ccsip all". > > > > > In this test I didn't bother picking up the phone at all, but I can > > see that FS answered anyway and the IVR kept playing until it timed > > out. > > I'm not an expert, but here is what I picked out of it: > > > At 00:08:10 we get a > > Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" > > > the further down at the same timestamp we get > > Sent: "SIP/2.0 100 Trying" > > > At 00:08:13 we get a > > Sent: "SIP/2.0 183 Session Progress" > > > At 00:18:13 we get a > > Sent: "SIP/2.0 200 OK" > > > Then at the same timestamp we get: > > Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > > > > Once the IVR times out at 00:09:16 we get > > Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > > > And then the reply right after > > Sent: "SIP/2.0 200 OK" > > > > > So I think you were right, the CISCO is sending back an "OK" 3 seconds > > after the "INVITE" is received. > > > > > The part that is beyond my field of expertise so far is WHY? > > > > > Thanks, > > > > Oliver > > > > > On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: > > By the way: > > > I tried {ignore_early_media=true} as well, but as I think we > > determined, my problem is probably with the CISCO telling FS that the > > call has been answered when really it hasn't yet. > > > > > On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: > > Thanks for the help so far. > > > > Here is a pastebin of FreeSWITCH output: > > http://pastebin.com/i6Qgc7ws > > > Notice how the "has been answered" log message comes immediately > > (within a few milliseconds) after the call was originated. I think > > this would suggest that the CISCO is immediately sending a 200 OK, as > > you suggested. I also turned on CISCO debugging, but I'm just trying > > to figure out how to get the information regarding SIP messages back > > to Freeswitch. I'll run the test again and see if I can get some > > useful CISCO debug. > > > Which "debug ccsip" commands are relevant to what I want for the CISCO > > SIP debugging? > > > > Thanks! > > > > > > 2012/1/6 Gustavo M?rsico : > > I think I've a similar problem related to callcenter app. When I made an > originate like this: > > > originate loopback/2500/default/XML &bridge(user/2001) > > > 2500 is an extension that leads to a callcenter application. In this case, > we dial first to the queue and when an agent answered we call to the > customer. As far as I know > > When the A-leg reaches to the queue, without selecting an agent, the call is > automatically sent to the B-leg. As far as I see, there is a pre-answer > method that fs needs to send the media to A-leg. > > In order to try to avoid this, I tried using ignore_early_media=true as part > of the originate in A-leg and/or B-leg, with no luck. > > > originate {ignore_early_media=true}loopback/2500/default/XML > &bridge({ignore_early_media=true}user/2001) > > > Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] > destination_number(2500) =~ /^(2500)$/ break=on-false > > Dialplan: loopback/2500-b Action set(ignore_early_media=true) > > Dialplan: loopback/2500-b Action callcenter(click2call) > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 > (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 > (loopback/2500-b) State ROUTING going to sleep > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 > (loopback/2500-b) Running State Change CS_EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 > (loopback/2500-b) State EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b > CHANNEL EXECUTE > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 > loopback/2500-b Standard EXECUTE > > EXECUTE loopback/2500-b set(open=true) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [open]=[true] > > EXECUTE loopback/2500-b > hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) > > EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) > > EXECUTE loopback/2500-b > hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) > > EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] > > EXECUTE loopback/2500-b set(ignore_early_media=true) > > 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET > [ignore_early_media]=[true] > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application > callcenter Requires media! pre_answering channel loopback/2500-b > > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer > loopback/2500-a! > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) > Callstate Change RINGING -> EARLY > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer > loopback/2500-b! > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) > Callstate Change RINGING -> EARLY > > EXECUTE loopback/2500-b callcenter(click2call) > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) > Callstate Change EARLY -> ACTIVE > > 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel > [loopback/2500-a] has been answered > > 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal > loopback/2500-b [BREAK] > > 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > CHANNEL KILL > > 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate > Resulted in Success: [loopback/2500-a] > > 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) > Callstate Change EARLY -> ACTIVE > > 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a > Flipping CID from "" <0000000000> to "Outbound Call" > > > > > > On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > > > Also, maybe I should be doing something like this: > > > sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' > > > instead of: > > > sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' > > > > but, I don't really have the CISCO configured as a gateway, nor do I > > know how really...probably not on the right track there. > > > > > > On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: > > *bump* > > > > So I think maybe the way I'm doing the originate is the problem? In my > > call string I'm creating a connection directly from the CISCO > > (192.168.x.x) to the managed application, which may be why it starts > > playing straight away? > > > Maybe I should be originating a call first and then only once I know > > the other side has picked up will I bridge the call to the IVR managed > > application. > > > Problem is I dunno how to tell whether the other person has picked up > > (or even if the cisco is going to tell me) and I don't know how to do > > things to a call once it has been established. > > > > I'm currently reading the Dialplan wiki page, hoping to get something > > out of it there. > > > > Cheers > > > Oliver > > > > On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: > > I've been battling while creating an IVR using FreeSWITCH mod_managed > > and connecting through a CISCO 2811. Most things now work quite well, > > but I am having a few issues with the way the system answers calls (or > > doesn't answer calls...). > > > I have FreeSWITCH running as a windows service on Windows Server 2008, > > which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > > which is then connected to a POTS phone line. > > > > Take the following scenario: > > > 1. Managed .NET application creates a call string and uses ESL to talk > > to freeswitch and originate a call: > > > string callstring = > > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x > > '&managed(ivrAppName)'"; > > eslConnection.API("originate", callstring); > > > where 192.168.x.x is the CISCO IP. > > > 2. The CISCO sees that the phone number (1091234567) starts with a 1 > > so it uses FXO port 1 and strips the 1 and uses the remaining phone > > number (091234567) to make the call. > > > 3. My phone rings, I pick up and I can hear my IVR playing. > > > > > These are my current problems: > > > - IVR starts playing before I even pick up the phone. This means that > > if the system calls a mobile phone and the person doesn't pick up, the > > IVR will start playing and eventually the mobile phone will divert to > > voice mail. Obviously I then get a missed call and an sms saying I > > have a new voice mail, which is annoying. Instead I would like it to > > KNOW that no one has picked up, but I don't know how to do this. > > Somehow the CISCO needs to be able to tell FreeSWITCH that the call > > has not yet been answered. For some reason however as soon as the > > CISCO starts calling FreeSWITCH thinks the call is already connected. > > It doesn't know that the CISCO is actually still ringing. Maybe I'm > > doing originate the wrong way or something ... > > > - The phone only rings for about 10 seconds before hanging up. I've > > tried "call_timeout", "bridge_answer_timeout". I've also tried setting > > CISCO "ring number". Nothing works, my phone still only rings for > > about 10 seconds. I don't know if this is a FreeSWITCH issue or a > > CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > > starts playing even if no one answers the phone. > > > > > > > CISCO Config for relevant FXO port: > > > voice service voip > > allow-connections h323 to h323 > > allow-connections h323 to sip > > allow-connections sip to h323 > > allow-connections sip to sip > > no supplementary-service h450.2 > > no supplementary-service h450.3 > > supplementary-service h450.12 > > no supplementary-service sip moved-temporarily > > no supplementary-service sip refer > > fax protocol cisco > > sip > > registrar server expires max 3600 min 3600 > > no update-callerid > > no call service stop > > > voice-port 0/3/2 > > output attenuation -3 > > no comfort-noise > > cptone AU > > impedance complex1 > > caller-id enable > > ! > > dial-peer voice 100 pots > > preference 1 > > destination-pattern 1T > > port 0/3/2 > > ! > > > > > Many Thanks, > > > Oliver > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f068c9f32761270174137! From miha at softnet.si Fri Jan 6 11:40:52 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 06 Jan 2012 09:40:52 +0100 Subject: [Freeswitch-users] Password in dialplan In-Reply-To: References: <4F054EFA.2000004@softnet.si> Message-ID: <4F06B394.6000808@softnet.si> Hi Michael, this I have in my dialplan: in user/dir: user id="013108500"> When user make call we authenticate it with freeradius. I would like that, the same password for user login is also for radius authentication, but I do not know how to link this with dial plan (that the dial plan will parse password from user dir for auth_function. Thanks! Regards, Miha On 1/5/2012 4:49 PM, Michael Collins wrote: > How is this user being authenticated? > -MC > > On Wed, Jan 4, 2012 at 11:19 PM, Miha Zoubek > wrote: > > Hi, > > in dial plan I have this line (for radius): > > data="PASSWD=$${default_password}"/> > > Deafult password is set to 1234. As I do not wont default password > for radius, I would just like to have here password which is set > for user in directory. > > > > How to set in dialplan that the password will be taken from user/dir? > > data="PASSWD=$${password}"/> Like this? > > Thanks! > > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/0379162e/attachment-0001.html From freeswitch at peely.com Fri Jan 6 13:01:28 2012 From: freeswitch at peely.com (peely) Date: Fri, 6 Jan 2012 02:01:28 -0800 (PST) Subject: [Freeswitch-users] Session timeout In-Reply-To: <4F063163.50005@telefaks.de> References: <4F063163.50005@telefaks.de> Message-ID: <1325844088052-7157839.post@n2.nabble.com> Did the other end declare support for timer i.e.: UAC: Supported: Timer UAS: Require: Timer, Session Expires: yyy ? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Session-timeout-tp7156616p7157839.html Sent from the freeswitch-users mailing list archive at Nabble.com. From engineerzuhairraza at gmail.com Fri Jan 6 13:21:11 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Fri, 6 Jan 2012 14:21:11 +0400 Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: References: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: Hi, username and password not working for me Username : demouser password : P at ssw0rd Are they changed ? Regards, Zohair Raza On Tue, Jan 3, 2012 at 10:16 PM, Muhammad Naseer Bhatti wrote: > > Sherif, have you installed manually? If so, you would have to enable > mod_rewrite in your apache configuration. > > > > On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: > >> Can you give more details how it does not work? I have the same situation. >> I can reach the frontpage and when i give the username i get 404. >> >> Is this the case you have? >> >> Do you have Centos or Redhat ? >> >> >> regards, >> Sherif >> >> >> --- On *Tue, 1/3/12, Zenny * wrote: >> >> >> From: Zenny >> Subject: Re: [Freeswitch-users] vBilling Beta Program!! >> To: "FreeSWITCH Users Help" >> Date: Tuesday, January 3, 2012, 11:18 AM >> >> >> looks promising, but the user login does not work. >> >> Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to >> all freeswitchers! >> >> On 1/3/12, nbhatti > >> wrote: >> > Yes, it will support prepaid calling card and many more features soon. >> > >> > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] >> > > >> wrote: >> >> It looks great! >> >> >> >> Will vBilling support batch user/prepaid calling card creation? >> >> >> >> ________________________________ >> >> If you reply to this email, your message will be added to the >> discussion >> >> below: >> >> >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >> >> To unsubscribe from vBilling Beta Program!!, click here. >> >> NAML >> > >> > >> > -- >> > View this message in context: >> > >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/256bef25/attachment.html From manjiri05_deshpande at yahoo.co.in Fri Jan 6 13:50:48 2012 From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande) Date: Fri, 6 Jan 2012 16:20:48 +0530 (IST) Subject: [Freeswitch-users] ring tone is heard for 2 sec before busy tone on Lync client Message-ID: <1325847048.31905.YahooMailNeo@web95905.mail.in.yahoo.com> Hi, I am using freeswitch 107 on windows platform I am working for a product where lync client can call a phone registred with Lync server. When Phone is set to DND mode and lync client calls that phone,I could hear ring tone for abt 2 sec and then busy tone comes. The behvaiour should be ,only busy tone should be heard. I removed "ringback" and "ignore_early_media" from dialplan. Please find dial plan , freeswitch logs and Wireshark capture attached with this mail. From wireshark it is clear that no 180 or 183 messages are flowing.But still from FS logs it could be figure out that,FS is entering into "RINGING"? mode. Thanks, Manjiri Deshpande -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/1be074e2/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: For_FS.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/1be074e2/attachment-0001.txt -------------- next part -------------- A non-text attachment was scrubbed... Name: ring_back.pcap Type: application/octet-stream Size: 37712 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/1be074e2/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: [dialplan]{from_ocs}.xml Type: application/octet-stream Size: 3277 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/1be074e2/attachment-0003.obj From olimonkey at gmail.com Fri Jan 6 14:04:14 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Fri, 6 Jan 2012 19:04:14 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> Message-ID: Because I'm using an FXO card with voice, I added something to my CISCO conf. Many others had the same thing: voice-port 0/3/0 ... supervisory disconnect dualtone mid-call supervisory answer dualtone <---- ADDED THIS ONE ... Once I added this, the FS output now just showed the following while the phone was ringing: 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel sofia/internal/109212xxxx at 192.168.x.x [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer sofia/internal/109212xxxx at 192.168.x.x! Where as previous it would show the above and also show the following: 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" <0000000000> to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" <1092122856> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel [sofia/internal/109212xxxx at 192.168.x.x] has been answered BUT, the IVR still started playing even before I pick up the phone. Hmmmm.....so why is FS still starting the managed application when the call has not been answered yet. Are we all sure that the managed application should not be executed until the call "has been answered" shows up in the log file? Will have to keep testing on monday as I don't have access to the CISCO from where i am now. I'll have to see whether the CISCO changes had any impact on the times at which the SIP messages are sent back and forth. Especially the 200 OK message. Thanks again for help, maybe getting somewhere now...... Oliver On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson wrote: > If it sends 200 OK right after 183, this IS the problem. > > 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. > > Seems to me there is something broken in the Cisco. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] > Skickat: den 6 januari 2012 06:55 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > I've tried looking at disable-early-media configuration command, but > that didn't work and I doubt that has anything to do with the CISCO > sending a 200 OK right after a 183 SESSION PROGRESS. > > > > > On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >> is usually RINGING (generate ringback locally) while a 183 has media... aka >> early media and usually provides ringback inband. >> >> /b >> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >> >> Shouldn't there be a ?180 RINGING ?somewhere in there? >> >> >> >> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >> >> I just noticed something else, if I don't pick up the phone at all. >> >> The IVR just keeps playing until the menu timeout kicks in. >> >> >> So here is a CISCO SIP log: >> >> http://pastebin.com/Y9sYkuxi >> >> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >> >> I hope the CISCO log is readable, it's a bit long because I just did >> >> "debug ccsip all". >> >> >> >> >> In this test I didn't bother picking up the phone at all, but I can >> >> see that FS answered anyway and the IVR kept playing until it timed >> >> out. >> >> I'm not an expert, but here is what I picked out of it: >> >> >> At 00:08:10 we get a >> >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >> >> >> the further down at the same timestamp we get >> >> Sent: "SIP/2.0 100 Trying" >> >> >> At 00:08:13 we get a >> >> Sent: "SIP/2.0 183 Session Progress" >> >> >> At 00:18:13 we get a >> >> Sent: "SIP/2.0 200 OK" >> >> >> Then at the same timestamp we get: >> >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> >> >> Once the IVR times out at 00:09:16 we get >> >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> And then the reply right after >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >> >> after the "INVITE" is received. >> >> >> >> >> The part that is beyond my field of expertise so far is WHY? >> >> >> >> >> Thanks, >> >> >> >> Oliver >> >> >> >> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >> >> By the way: >> >> >> I tried {ignore_early_media=true} as well, but as I think we >> >> determined, my problem is probably with the CISCO telling FS that the >> >> call has been answered when really it hasn't yet. >> >> >> >> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >> >> Thanks for the help so far. >> >> >> >> Here is a pastebin of FreeSWITCH output: >> >> http://pastebin.com/i6Qgc7ws >> >> >> Notice how the "has been answered" log message comes immediately >> >> (within a few milliseconds) after the call was originated. I think >> >> this would suggest that the CISCO is immediately sending a 200 OK, as >> >> you suggested. I also turned on CISCO debugging, but I'm just trying >> >> to figure out how to get the information regarding SIP messages back >> >> to Freeswitch. I'll run the test again and see if I can get some >> >> useful CISCO debug. >> >> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >> >> SIP debugging? >> >> >> >> Thanks! >> >> >> >> >> >> 2012/1/6 Gustavo M?rsico : >> >> I think I've a similar problem related to callcenter app. When I made an >> originate like this: >> >> >> originate loopback/2500/default/XML &bridge(user/2001) >> >> >> 2500 is an extension that leads to a callcenter application. In this case, >> we dial first to the queue and when an agent answered we call to the >> customer. As far as I know >> >> When the A-leg reaches to the queue, without selecting an agent, the call is >> automatically sent to the B-leg. As far as I see, there is a pre-answer >> method that fs needs to send the media to A-leg. >> >> In order to try to avoid this, I tried using ignore_early_media=true as part >> of the originate in A-leg and/or B-leg, with no luck. >> >> >> originate {ignore_early_media=true}loopback/2500/default/XML >> &bridge({ignore_early_media=true}user/2001) >> >> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >> destination_number(2500) =~ /^(2500)$/ break=on-false >> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >> >> Dialplan: loopback/2500-b Action callcenter(click2call) >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >> (loopback/2500-b) State ROUTING going to sleep >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >> (loopback/2500-b) Running State Change CS_EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >> (loopback/2500-b) State EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >> CHANNEL EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >> loopback/2500-b Standard EXECUTE >> >> EXECUTE loopback/2500-b set(open=true) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [open]=[true] >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [ignore_early_media]=[true] >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >> callcenter Requires media! pre_answering channel loopback/2500-b >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >> loopback/2500-a! >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >> Callstate Change RINGING -> EARLY >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >> loopback/2500-b! >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >> Callstate Change RINGING -> EARLY >> >> EXECUTE loopback/2500-b callcenter(click2call) >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >> Callstate Change EARLY -> ACTIVE >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >> [loopback/2500-a] has been answered >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >> Resulted in Success: [loopback/2500-a] >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >> Callstate Change EARLY -> ACTIVE >> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >> Flipping CID from "" <0000000000> to "Outbound Call" >> >> >> >> >> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >> >> >> Also, maybe I should be doing something like this: >> >> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >> >> >> instead of: >> >> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >> >> >> >> but, I don't really have the CISCO configured as a gateway, nor do I >> >> know how really...probably not on the right track there. >> >> >> >> >> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >> >> *bump* >> >> >> >> So I think maybe the way I'm doing the originate is the problem? In my >> >> call string I'm creating a connection directly from the CISCO >> >> (192.168.x.x) to the managed application, which may be why it starts >> >> playing straight away? >> >> >> Maybe I should be originating a call first and then only once I know >> >> the other side has picked up will I bridge the call to the IVR managed >> >> application. >> >> >> Problem is I dunno how to tell whether the other person has picked up >> >> (or even if the cisco is going to tell me) and I don't know how to do >> >> things to a call once it has been established. >> >> >> >> I'm currently reading the Dialplan wiki page, hoping to get something >> >> out of it there. >> >> >> >> Cheers >> >> >> Oliver >> >> >> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> >> and connecting through a CISCO 2811. Most things now work quite well, >> >> but I am having a few issues with the way the system answers calls (or >> >> doesn't answer calls...). >> >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> >> which is then connected to a POTS phone line. >> >> >> >> Take the following scenario: >> >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> >> to freeswitch and originate a call: >> >> >> string callstring = >> >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >> >> '&managed(ivrAppName)'"; >> >> eslConnection.API("originate", callstring); >> >> >> where 192.168.x.x is the CISCO IP. >> >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> >> number (091234567) to make the call. >> >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> >> These are my current problems: >> >> >> - IVR starts playing before I even pick up the phone. This means that >> >> if the system calls a mobile phone and the person doesn't pick up, the >> >> IVR will start playing and eventually the mobile phone will divert to >> >> voice mail. Obviously I then get a missed call and an sms saying I >> >> have a new voice mail, which is annoying. Instead I would like it to >> >> KNOW that no one has picked up, but I don't know how to do this. >> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> >> has not yet been answered. For some reason however as soon as the >> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> >> doing originate the wrong way or something ... >> >> >> - The phone only rings for about 10 seconds before hanging up. I've >> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> >> CISCO "ring number". Nothing works, my phone still only rings for >> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> >> starts playing even if no one answers the phone. >> >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> >> voice service voip >> >> ?allow-connections h323 to h323 >> >> ?allow-connections h323 to sip >> >> ?allow-connections sip to h323 >> >> ?allow-connections sip to sip >> >> ?no supplementary-service h450.2 >> >> ?no supplementary-service h450.3 >> >> ?supplementary-service h450.12 >> >> ?no supplementary-service sip moved-temporarily >> >> ?no supplementary-service sip refer >> >> ?fax protocol cisco >> >> ?sip >> >> ?registrar server expires max 3600 min 3600 >> >> ?no update-callerid >> >> ?no call service stop >> >> >> voice-port 0/3/2 >> >> ?output attenuation -3 >> >> ?no comfort-noise >> >> ?cptone AU >> >> ?impedance complex1 >> >> ?caller-id enable >> >> ! >> >> dial-peer voice 100 pots >> >> ?preference 1 >> >> ?destination-pattern 1T >> >> ?port 0/3/2 >> >> ! >> >> >> >> >> Many Thanks, >> >> >> Oliver >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: ? +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f068c9f32761270174137! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nbhatti at gmail.com Fri Jan 6 14:15:21 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 6 Jan 2012 14:15:21 +0300 Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: References: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: No, this is not the correct username/password. You may login with login: admin and password: vBilling On Fri, Jan 6, 2012 at 1:21 PM, Zohair Raza wrote: > Hi, > username and password not working for me > > Username : demouser > password : P at ssw0rd > > Are they changed ? > > Regards, > Zohair Raza > > > > > On Tue, Jan 3, 2012 at 10:16 PM, Muhammad Naseer Bhatti > wrote: > >> >> Sherif, have you installed manually? If so, you would have to enable >> mod_rewrite in your apache configuration. >> >> >> >> On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: >> >>> Can you give more details how it does not work? I have the same >>> situation. >>> I can reach the frontpage and when i give the username i get 404. >>> >>> Is this the case you have? >>> >>> Do you have Centos or Redhat ? >>> >>> >>> regards, >>> Sherif >>> >>> >>> --- On *Tue, 1/3/12, Zenny * wrote: >>> >>> >>> From: Zenny >>> Subject: Re: [Freeswitch-users] vBilling Beta Program!! >>> To: "FreeSWITCH Users Help" >>> Date: Tuesday, January 3, 2012, 11:18 AM >>> >>> >>> looks promising, but the user login does not work. >>> >>> Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to >>> all freeswitchers! >>> >>> On 1/3/12, nbhatti > >>> wrote: >>> > Yes, it will support prepaid calling card and many more features soon. >>> > >>> > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] >>> > > >>> wrote: >>> >> It looks great! >>> >> >>> >> Will vBilling support batch user/prepaid calling card creation? >>> >> >>> >> ________________________________ >>> >> If you reply to this email, your message will be added to the >>> discussion >>> >> below: >>> >> >>> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >>> >> To unsubscribe from vBilling Beta Program!!, click here. >>> >> NAML >>> > >>> > >>> > -- >>> > View this message in context: >>> > >>> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html >>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/7c1467cd/attachment-0001.html From engineerzuhairraza at gmail.com Fri Jan 6 14:18:22 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Fri, 6 Jan 2012 15:18:22 +0400 Subject: [Freeswitch-users] vBilling Beta Program!! In-Reply-To: References: <1325614119.30774.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: thanks Regards, Zohair Raza On Fri, Jan 6, 2012 at 3:15 PM, Muhammad Naseer Bhatti wrote: > > No, this is not the correct username/password. You may login with login: > admin and password: vBilling > > > On Fri, Jan 6, 2012 at 1:21 PM, Zohair Raza wrote: > >> Hi, >> username and password not working for me >> >> Username : demouser >> password : P at ssw0rd >> >> Are they changed ? >> >> Regards, >> Zohair Raza >> >> >> >> >> On Tue, Jan 3, 2012 at 10:16 PM, Muhammad Naseer Bhatti < >> nbhatti at gmail.com> wrote: >> >>> >>> Sherif, have you installed manually? If so, you would have to enable >>> mod_rewrite in your apache configuration. >>> >>> >>> >>> On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: >>> >>>> Can you give more details how it does not work? I have the same >>>> situation. >>>> I can reach the frontpage and when i give the username i get 404. >>>> >>>> Is this the case you have? >>>> >>>> Do you have Centos or Redhat ? >>>> >>>> >>>> regards, >>>> Sherif >>>> >>>> >>>> --- On *Tue, 1/3/12, Zenny * wrote: >>>> >>>> >>>> From: Zenny >>>> Subject: Re: [Freeswitch-users] vBilling Beta Program!! >>>> To: "FreeSWITCH Users Help" >>>> Date: Tuesday, January 3, 2012, 11:18 AM >>>> >>>> >>>> looks promising, but the user login does not work. >>>> >>>> Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to >>>> all freeswitchers! >>>> >>>> On 1/3/12, nbhatti > >>>> wrote: >>>> > Yes, it will support prepaid calling card and many more features soon. >>>> > >>>> > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] >>>> > > >>>> wrote: >>>> >> It looks great! >>>> >> >>>> >> Will vBilling support batch user/prepaid calling card creation? >>>> >> >>>> >> ________________________________ >>>> >> If you reply to this email, your message will be added to the >>>> discussion >>>> >> below: >>>> >> >>>> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >>>> >> To unsubscribe from vBilling Beta Program!!, click here. >>>> >> NAML >>>> > >>>> > >>>> > -- >>>> > View this message in context: >>>> > >>>> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html >>>> > Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/a853b65d/attachment.html From juanito1982 at gmail.com Fri Jan 6 15:46:23 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Jan 2012 13:46:23 +0100 Subject: [Freeswitch-users] LUA getVariable Message-ID: Hello, I have one LUA script that make calls to one custom FS app. That app sets up some session channel variables and makes an outgoing call: if(caller_channel = switch_core_session_get_channel(session)){ switch_channel_set_variable(caller_channel, "var", "val") } If the caller hangs up the call, the app returns and I try to get the val of var as: session:getVariable("var") but a nil value is returned. Do you know why? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/6f97928f/attachment.html From amit.nakum2009 at gmail.com Fri Jan 6 13:28:00 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Fri, 6 Jan 2012 15:58:00 +0530 Subject: [Freeswitch-users] in ivr menu Message-ID: Hi, I want to play wav files from the different path. As now it is playing the wav files from: "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/" path. Please tell me how to play wav files from other path in ivr menu using greet-short parameter . Thanks in advance.... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/1c2ed9c1/attachment-0001.html From bharatlalcheta at gmail.com Fri Jan 6 16:18:44 2012 From: bharatlalcheta at gmail.com (Bharat Lalcheta) Date: Fri, 6 Jan 2012 18:48:44 +0530 Subject: [Freeswitch-users] Codec Preferance Message-ID: Hiii, I am new to freeswitch. Prior to freeswitch i was using asterisk. I have 200 extensions working in my office and want to move all to freeswitch from asterisk. In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch. Follwing is one example.... ----------------------------------------------------------------- ---------------------------------------------------------- Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having. I want to use different codecs for different extensions. Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ? If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ? Please guide me and provide solution for the same Thanks in advance Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/b67f853d/attachment-0001.html From bharatlalcheta at gmail.com Fri Jan 6 16:24:12 2012 From: bharatlalcheta at gmail.com (Bharat Lalcheta) Date: Fri, 6 Jan 2012 18:54:12 +0530 Subject: [Freeswitch-users] Codec selection Message-ID: Hiii, I am new to freeswitch. Prior to freeswitch i was using asterisk. I have 200 extensions working in my office and want to move all to freeswitch from asterisk. In asterisk, i can give codec selection and preferance in sip.conf to all extensions. In the same way i created 200 extensions under internal profile in freeswitch. Follwing is one example.... ----------------------------------------------------------------- ---------------------------------------------------------- Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone other than defined in 590.xml. It is seding codes which is mentioned in my conf/sip_profiles/internal.xml and codec negotiation done on whatever codec my sip phone having. I want to use different codecs for different extensions. Is it common behaviour of Freeswitch ? Should i override codec prerfrance in my extension list from my internal profile or not ? If no, then is it that i have to create 200 profiles in freeswitch to solve this problem ? Please guide me and provide solution for the same Thanks in advance -- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/5e2a49f0/attachment.html From engineerzuhairraza at gmail.com Fri Jan 6 16:31:23 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Fri, 6 Jan 2012 17:31:23 +0400 Subject: [Freeswitch-users] Codec Preferance In-Reply-To: References: Message-ID: Hi, You can set codec from dialplan. See http://wiki.freeswitch.org/wiki/Codec_negotiation make sure to set inbound-codec-negotiation from generous to greedy in your sip profile Regards, Zohair Raza On Fri, Jan 6, 2012 at 5:18 PM, Bharat Lalcheta wrote: > Hiii, > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > I have 200 extensions working in my office and want to move all to > freeswitch from asterisk. > > In asterisk, i can give codec selection and preferance in sip.conf to all > extensions. In the same way i created 200 extensions under internal profile > in freeswitch. > > Follwing is one example.... > ----------------------------------------------------------------- > > > > > > > > > > > > > > > > > > ---------------------------------------------------------- > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone > other than defined in 590.xml. It is seding codes which is mentioned in my > conf/sip_profiles/internal.xml and codec negotiation done on whatever codec > my sip phone having. > > I want to use different codecs for different extensions. > > Is it common behaviour of Freeswitch ? Should i override codec prerfrance > in my extension list from my internal profile or not ? > > If no, then is it that i have to create 200 profiles in freeswitch to > solve this problem ? > > Please guide me and provide solution for the same > > > Thanks in advance > > Bharat Lalcheta > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/e55e88ef/attachment.html From freeswitch at peely.com Fri Jan 6 16:59:32 2012 From: freeswitch at peely.com (peely) Date: Fri, 6 Jan 2012 05:59:32 -0800 (PST) Subject: [Freeswitch-users] Help on RTMP In-Reply-To: References: <9CD21E5D-581E-45AB-854E-2CD973A0CCF0@freeswitch.org> Message-ID: <1325858372244-7158479.post@n2.nabble.com> What are you passing in as the user? I found it needs to be user at ip_address. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-on-RTMP-tp7134331p7158479.html Sent from the freeswitch-users mailing list archive at Nabble.com. From curriegrad2004 at gmail.com Fri Jan 6 18:11:52 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 6 Jan 2012 07:11:52 -0800 Subject: [Freeswitch-users] Codec Preferance In-Reply-To: References: Message-ID: I highly would recommend that you change the name of those codecs to something else because you might be making matters worse later down the road On 2012-01-06 5:21 AM, "Bharat Lalcheta" wrote: > Hiii, > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > I have 200 extensions working in my office and want to move all to > freeswitch from asterisk. > > In asterisk, i can give codec selection and preferance in sip.conf to all > extensions. In the same way i created 200 extensions under internal profile > in freeswitch. > > Follwing is one example.... > ----------------------------------------------------------------- > > > > > > > > > > > > > > > > > > ---------------------------------------------------------- > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone > other than defined in 590.xml. It is seding codes which is mentioned in my > conf/sip_profiles/internal.xml and codec negotiation done on whatever codec > my sip phone having. > > I want to use different codecs for different extensions. > > Is it common behaviour of Freeswitch ? Should i override codec prerfrance > in my extension list from my internal profile or not ? > > If no, then is it that i have to create 200 profiles in freeswitch to > solve this problem ? > > Please guide me and provide solution for the same > > > Thanks in advance > > Bharat Lalcheta > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/a42316c0/attachment.html From peter.olsson at visionutveckling.se Fri Jan 6 18:21:00 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 6 Jan 2012 15:21:00 +0000 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Are you still using ignore_early_media=true - this must be set for this to work correctly. You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" Park application is only executed after the call was answered. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] Skickat: den 6 januari 2012 12:04 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR Because I'm using an FXO card with voice, I added something to my CISCO conf. Many others had the same thing: voice-port 0/3/0 ... supervisory disconnect dualtone mid-call supervisory answer dualtone <---- ADDED THIS ONE ... Once I added this, the FS output now just showed the following while the phone was ringing: 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel sofia/internal/109212xxxx at 192.168.x.x [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer sofia/internal/109212xxxx at 192.168.x.x! Where as previous it would show the above and also show the following: 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" <0000000000> to "Outbound Call" <109212xxxx> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" <1092122856> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel [sofia/internal/109212xxxx at 192.168.x.x] has been answered BUT, the IVR still started playing even before I pick up the phone. Hmmmm.....so why is FS still starting the managed application when the call has not been answered yet. Are we all sure that the managed application should not be executed until the call "has been answered" shows up in the log file? Will have to keep testing on monday as I don't have access to the CISCO from where i am now. I'll have to see whether the CISCO changes had any impact on the times at which the SIP messages are sent back and forth. Especially the 200 OK message. Thanks again for help, maybe getting somewhere now...... Oliver On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson wrote: > If it sends 200 OK right after 183, this IS the problem. > > 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. > > Seems to me there is something broken in the Cisco. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] > Skickat: den 6 januari 2012 06:55 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > I've tried looking at disable-early-media configuration command, but > that didn't work and I doubt that has anything to do with the CISCO > sending a 200 OK right after a 183 SESSION PROGRESS. > > > > > On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >> is usually RINGING (generate ringback locally) while a 183 has media... aka >> early media and usually provides ringback inband. >> >> /b >> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >> >> Shouldn't there be a 180 RINGING somewhere in there? >> >> >> >> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >> >> I just noticed something else, if I don't pick up the phone at all. >> >> The IVR just keeps playing until the menu timeout kicks in. >> >> >> So here is a CISCO SIP log: >> >> http://pastebin.com/Y9sYkuxi >> >> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >> >> I hope the CISCO log is readable, it's a bit long because I just did >> >> "debug ccsip all". >> >> >> >> >> In this test I didn't bother picking up the phone at all, but I can >> >> see that FS answered anyway and the IVR kept playing until it timed >> >> out. >> >> I'm not an expert, but here is what I picked out of it: >> >> >> At 00:08:10 we get a >> >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >> >> >> the further down at the same timestamp we get >> >> Sent: "SIP/2.0 100 Trying" >> >> >> At 00:08:13 we get a >> >> Sent: "SIP/2.0 183 Session Progress" >> >> >> At 00:18:13 we get a >> >> Sent: "SIP/2.0 200 OK" >> >> >> Then at the same timestamp we get: >> >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> >> >> Once the IVR times out at 00:09:16 we get >> >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> And then the reply right after >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >> >> after the "INVITE" is received. >> >> >> >> >> The part that is beyond my field of expertise so far is WHY? >> >> >> >> >> Thanks, >> >> >> >> Oliver >> >> >> >> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >> >> By the way: >> >> >> I tried {ignore_early_media=true} as well, but as I think we >> >> determined, my problem is probably with the CISCO telling FS that the >> >> call has been answered when really it hasn't yet. >> >> >> >> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >> >> Thanks for the help so far. >> >> >> >> Here is a pastebin of FreeSWITCH output: >> >> http://pastebin.com/i6Qgc7ws >> >> >> Notice how the "has been answered" log message comes immediately >> >> (within a few milliseconds) after the call was originated. I think >> >> this would suggest that the CISCO is immediately sending a 200 OK, as >> >> you suggested. I also turned on CISCO debugging, but I'm just trying >> >> to figure out how to get the information regarding SIP messages back >> >> to Freeswitch. I'll run the test again and see if I can get some >> >> useful CISCO debug. >> >> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >> >> SIP debugging? >> >> >> >> Thanks! >> >> >> >> >> >> 2012/1/6 Gustavo M?rsico : >> >> I think I've a similar problem related to callcenter app. When I made an >> originate like this: >> >> >> originate loopback/2500/default/XML &bridge(user/2001) >> >> >> 2500 is an extension that leads to a callcenter application. In this case, >> we dial first to the queue and when an agent answered we call to the >> customer. As far as I know >> >> When the A-leg reaches to the queue, without selecting an agent, the call is >> automatically sent to the B-leg. As far as I see, there is a pre-answer >> method that fs needs to send the media to A-leg. >> >> In order to try to avoid this, I tried using ignore_early_media=true as part >> of the originate in A-leg and/or B-leg, with no luck. >> >> >> originate {ignore_early_media=true}loopback/2500/default/XML >> &bridge({ignore_early_media=true}user/2001) >> >> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >> destination_number(2500) =~ /^(2500)$/ break=on-false >> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >> >> Dialplan: loopback/2500-b Action callcenter(click2call) >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >> (loopback/2500-b) State ROUTING going to sleep >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >> (loopback/2500-b) Running State Change CS_EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >> (loopback/2500-b) State EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >> CHANNEL EXECUTE >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >> loopback/2500-b Standard EXECUTE >> >> EXECUTE loopback/2500-b set(open=true) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [open]=[true] >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >> [ignore_early_media]=[true] >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >> callcenter Requires media! pre_answering channel loopback/2500-b >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >> loopback/2500-a! >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >> Callstate Change RINGING -> EARLY >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >> loopback/2500-b! >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >> Callstate Change RINGING -> EARLY >> >> EXECUTE loopback/2500-b callcenter(click2call) >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >> Callstate Change EARLY -> ACTIVE >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >> [loopback/2500-a] has been answered >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >> loopback/2500-b [BREAK] >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> CHANNEL KILL >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >> Resulted in Success: [loopback/2500-a] >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >> Callstate Change EARLY -> ACTIVE >> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >> Flipping CID from "" <0000000000> to "Outbound Call" >> >> >> >> >> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >> >> >> Also, maybe I should be doing something like this: >> >> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >> >> >> instead of: >> >> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >> >> >> >> but, I don't really have the CISCO configured as a gateway, nor do I >> >> know how really...probably not on the right track there. >> >> >> >> >> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >> >> *bump* >> >> >> >> So I think maybe the way I'm doing the originate is the problem? In my >> >> call string I'm creating a connection directly from the CISCO >> >> (192.168.x.x) to the managed application, which may be why it starts >> >> playing straight away? >> >> >> Maybe I should be originating a call first and then only once I know >> >> the other side has picked up will I bridge the call to the IVR managed >> >> application. >> >> >> Problem is I dunno how to tell whether the other person has picked up >> >> (or even if the cisco is going to tell me) and I don't know how to do >> >> things to a call once it has been established. >> >> >> >> I'm currently reading the Dialplan wiki page, hoping to get something >> >> out of it there. >> >> >> >> Cheers >> >> >> Oliver >> >> >> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> >> and connecting through a CISCO 2811. Most things now work quite well, >> >> but I am having a few issues with the way the system answers calls (or >> >> doesn't answer calls...). >> >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> >> which is then connected to a POTS phone line. >> >> >> >> Take the following scenario: >> >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> >> to freeswitch and originate a call: >> >> >> string callstring = >> >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >> >> '&managed(ivrAppName)'"; >> >> eslConnection.API("originate", callstring); >> >> >> where 192.168.x.x is the CISCO IP. >> >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> >> number (091234567) to make the call. >> >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> >> These are my current problems: >> >> >> - IVR starts playing before I even pick up the phone. This means that >> >> if the system calls a mobile phone and the person doesn't pick up, the >> >> IVR will start playing and eventually the mobile phone will divert to >> >> voice mail. Obviously I then get a missed call and an sms saying I >> >> have a new voice mail, which is annoying. Instead I would like it to >> >> KNOW that no one has picked up, but I don't know how to do this. >> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> >> has not yet been answered. For some reason however as soon as the >> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> >> doing originate the wrong way or something ... >> >> >> - The phone only rings for about 10 seconds before hanging up. I've >> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> >> CISCO "ring number". Nothing works, my phone still only rings for >> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> >> starts playing even if no one answers the phone. >> >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> >> voice service voip >> >> allow-connections h323 to h323 >> >> allow-connections h323 to sip >> >> allow-connections sip to h323 >> >> allow-connections sip to sip >> >> no supplementary-service h450.2 >> >> no supplementary-service h450.3 >> >> supplementary-service h450.12 >> >> no supplementary-service sip moved-temporarily >> >> no supplementary-service sip refer >> >> fax protocol cisco >> >> sip >> >> registrar server expires max 3600 min 3600 >> >> no update-callerid >> >> no call service stop >> >> >> voice-port 0/3/2 >> >> output attenuation -3 >> >> no comfort-noise >> >> cptone AU >> >> impedance complex1 >> >> caller-id enable >> >> ! >> >> dial-peer voice 100 pots >> >> preference 1 >> >> destination-pattern 1T >> >> port 0/3/2 >> >> ! >> >> >> >> >> Many Thanks, >> >> >> Oliver >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f06d49b32762089563979! From kris at kriskinc.com Fri Jan 6 18:24:01 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 6 Jan 2012 10:24:01 -0500 Subject: [Freeswitch-users] Codec Preferance In-Reply-To: References: Message-ID: inbound_codec_prefs outbound_codec_prefs These are Sofia profile configuration options. They are NOT valid options for the directory UNLESS you're setting them as variable to use in your dialplan later. If you want to control codes on a per-user basis you have to set late negotiation and use the dialplan. On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta wrote: > Hiii, > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > I have 200 extensions working in my office and want to move all to > freeswitch from asterisk. > > In asterisk, i can give codec selection and preferance?in sip.conf to all > extensions. In the same way i created 200 extensions under internal profile > in freeswitch. > > Follwing is one example.... > ----------------------------------------------------------------- > > ? > ??? > ????? > ????? > ????? > ????? > ????? > ??? > ??? > ????? > ????? > ????? > ????? > ??? > ? > > ---------------------------------------------------------- > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip phone > other than defined in 590.xml. It is seding codes which is mentioned in my > conf/sip_profiles/internal.xml and codec negotiation done on whatever codec > my sip phone having. > > I want to?use different codecs?for different extensions. > > Is it common behaviour of Freeswitch ? Should i override codec prerfrance in > my extension list from my internal profile or not ? > > If no, then is it that i have to create 200 profiles in freeswitch to solve > this problem ? > > Please guide me and provide solution for the same > > > Thanks in advance > > Bharat Lalcheta > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From curriegrad2004 at gmail.com Fri Jan 6 18:37:06 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 6 Jan 2012 07:37:06 -0800 Subject: [Freeswitch-users] Codec Preferance In-Reply-To: References: Message-ID: In theory he could use those variable names but it would involve some extra work ;) On 2012-01-06 7:24 AM, "Kristian Kielhofner" wrote: > inbound_codec_prefs > outbound_codec_prefs > > These are Sofia profile configuration options. They are NOT valid > options for the directory UNLESS you're setting them as variable to > use in your dialplan later. > > If you want to control codes on a per-user basis you have to set late > negotiation and use the dialplan. > > On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta > wrote: > > Hiii, > > > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > > > I have 200 extensions working in my office and want to move all to > > freeswitch from asterisk. > > > > In asterisk, i can give codec selection and preferance in sip.conf to all > > extensions. In the same way i created 200 extensions under internal > profile > > in freeswitch. > > > > Follwing is one example.... > > ----------------------------------------------------------------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ---------------------------------------------------------- > > > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip > phone > > other than defined in 590.xml. It is seding codes which is mentioned in > my > > conf/sip_profiles/internal.xml and codec negotiation done on whatever > codec > > my sip phone having. > > > > I want to use different codecs for different extensions. > > > > Is it common behaviour of Freeswitch ? Should i override codec > prerfrance in > > my extension list from my internal profile or not ? > > > > If no, then is it that i have to create 200 profiles in freeswitch to > solve > > this problem ? > > > > Please guide me and provide solution for the same > > > > > > Thanks in advance > > > > Bharat Lalcheta > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/9b3ad20d/attachment.html From bharatlalcheta at gmail.com Fri Jan 6 18:40:01 2012 From: bharatlalcheta at gmail.com (Bharat Lalcheta) Date: Fri, 6 Jan 2012 21:10:01 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 In-Reply-To: References: Message-ID: can you please explain in details what you want to tell ? > ---------- Forwarded message ---------- > From: curriegrad2004 > To: FreeSWITCH Users Help > Cc: > Date: Fri, 6 Jan 2012 07:11:52 -0800 > Subject: Re: [Freeswitch-users] Codec Preferance > > I highly would recommend that you change the name of those codecs to > something else because you might be making matters worse later down the > road > On 2012-01-06 5:21 AM, "Bharat Lalcheta" wrote: > >> Hiii, >> >> I am new to freeswitch. Prior to freeswitch i was using asterisk. >> >> I have 200 extensions working in my office and want to move all to >> freeswitch from asterisk. >> >> In asterisk, i can give codec selection and preferance in sip.conf to all >> extensions. In the same way i created 200 extensions under internal profile >> in freeswitch. >> >> Follwing is one example.... >> ----------------------------------------------------------------- >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> ---------------------------------------------------------- >> >> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone other than defined in 590.xml. It is seding codes which is mentioned >> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec my sip phone having. >> >> I want to use different codecs for different extensions. >> >> Is it common behaviour of Freeswitch ? Should i override codec prerfrance >> in my extension list from my internal profile or not ? >> >> If no, then is it that i have to create 200 profiles in freeswitch to >> solve this problem ? >> >> Please guide me and provide solution for the same >> >> >> Thanks in advance >> >> Bharat Lalcheta >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > ---------- Forwarded message ---------- > From: Peter Olsson > To: FreeSWITCH Users Help > Cc: > Date: Fri, 6 Jan 2012 15:21:00 +0000 > Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > Are you still using ignore_early_media=true - this must be set for this to > work correctly. > > You will see a EXECUTE log line when FS executes the application, with > ignore_early_media enabled it shouldn't execute until the call has been > answered. I just tried it myself, and it works as expected. > > Example "originate {ignore_early_media=true}sofia/internal/number at host&park()" > > Park application is only executed after the call was answered. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ > olimonkey at gmail.com] > Skickat: den 6 januari 2012 12:04 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > Because I'm using an FXO card with voice, I added something to my > CISCO conf. Many others had the same thing: > > > voice-port 0/3/0 > ... > supervisory disconnect dualtone mid-call > supervisory answer dualtone <---- ADDED THIS ONE > ... > > > > Once I added this, the FS output now just showed the following while > the phone was ringing: > > 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel > sofia/internal/109212xxxx at 192.168.x.x > [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] > 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 > sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound > Call" <109212xxxx> > 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer > sofia/internal/109212xxxx at 192.168.x.x! > > > Where as previous it would show the above and also show the following: > > 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 > sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" > <0000000000> to "Outbound Call" <109212xxxx> > 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 > sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" > <1092122856> > 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel > [sofia/internal/109212xxxx at 192.168.x.x] has been answered > > > > BUT, the IVR still started playing even before I pick up the phone. > Hmmmm.....so why is FS still starting the managed application when the > call has not been answered yet. Are we all sure that the managed > application should not be executed until the call "has been answered" > shows up in the log file? > > > Will have to keep testing on monday as I don't have access to the > CISCO from where i am now. I'll have to see whether the CISCO changes > had any impact on the times at which the SIP messages are sent back > and forth. Especially the 200 OK message. > > > Thanks again for help, maybe getting somewhere now...... > > Oliver > > > > > On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson > wrote: > > If it sends 200 OK right after 183, this IS the problem. > > > > 200 OK means that the call was answered, it should not be sent until the > call was actually picked up in the remote end. When 200 OK arrives to FS it > will execute your app, and you will start playing the files. > > > > Seems to me there is something broken in the Cisco. > > > > /Peter > > > > ________________________________________ > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ > olimonkey at gmail.com] > > Skickat: den 6 januari 2012 06:55 > > Till: FreeSWITCH Users Help > > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > > > I've tried looking at disable-early-media configuration command, but > > that didn't work and I doubt that has anything to do with the CISCO > > sending a 200 OK right after a 183 SESSION PROGRESS. > > > > > > > > > > On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: > >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. > 180 > >> is usually RINGING (generate ringback locally) while a 183 has media... > aka > >> early media and usually provides ringback inband. > >> > >> /b > >> > >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: > >> > >> Shouldn't there be a 180 RINGING somewhere in there? > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk > wrote: > >> > >> I just noticed something else, if I don't pick up the phone at all. > >> > >> The IVR just keeps playing until the menu timeout kicks in. > >> > >> > >> So here is a CISCO SIP log: > >> > >> http://pastebin.com/Y9sYkuxi > >> > >> > >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. > >> > >> I hope the CISCO log is readable, it's a bit long because I just did > >> > >> "debug ccsip all". > >> > >> > >> > >> > >> In this test I didn't bother picking up the phone at all, but I can > >> > >> see that FS answered anyway and the IVR kept playing until it timed > >> > >> out. > >> > >> I'm not an expert, but here is what I picked out of it: > >> > >> > >> At 00:08:10 we get a > >> > >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" > >> > >> > >> the further down at the same timestamp we get > >> > >> Sent: "SIP/2.0 100 Trying" > >> > >> > >> At 00:08:13 we get a > >> > >> Sent: "SIP/2.0 183 Session Progress" > >> > >> > >> At 00:18:13 we get a > >> > >> Sent: "SIP/2.0 200 OK" > >> > >> > >> Then at the same timestamp we get: > >> > >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > >> > >> > >> > >> > >> Once the IVR times out at 00:09:16 we get > >> > >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" > >> > >> > >> And then the reply right after > >> > >> Sent: "SIP/2.0 200 OK" > >> > >> > >> > >> > >> So I think you were right, the CISCO is sending back an "OK" 3 seconds > >> > >> after the "INVITE" is received. > >> > >> > >> > >> > >> The part that is beyond my field of expertise so far is WHY? > >> > >> > >> > >> > >> Thanks, > >> > >> > >> > >> Oliver > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk > wrote: > >> > >> By the way: > >> > >> > >> I tried {ignore_early_media=true} as well, but as I think we > >> > >> determined, my problem is probably with the CISCO telling FS that the > >> > >> call has been answered when really it hasn't yet. > >> > >> > >> > >> > >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk > wrote: > >> > >> Thanks for the help so far. > >> > >> > >> > >> Here is a pastebin of FreeSWITCH output: > >> > >> http://pastebin.com/i6Qgc7ws > >> > >> > >> Notice how the "has been answered" log message comes immediately > >> > >> (within a few milliseconds) after the call was originated. I think > >> > >> this would suggest that the CISCO is immediately sending a 200 OK, as > >> > >> you suggested. I also turned on CISCO debugging, but I'm just trying > >> > >> to figure out how to get the information regarding SIP messages back > >> > >> to Freeswitch. I'll run the test again and see if I can get some > >> > >> useful CISCO debug. > >> > >> > >> Which "debug ccsip" commands are relevant to what I want for the CISCO > >> > >> SIP debugging? > >> > >> > >> > >> Thanks! > >> > >> > >> > >> > >> > >> 2012/1/6 Gustavo M?rsico : > >> > >> I think I've a similar problem related to callcenter app. When I made an > >> originate like this: > >> > >> > >> originate loopback/2500/default/XML &bridge(user/2001) > >> > >> > >> 2500 is an extension that leads to a callcenter application. In this > case, > >> we dial first to the queue and when an agent answered we call to the > >> customer. As far as I know > >> > >> When the A-leg reaches to the queue, without selecting an agent, the > call is > >> automatically sent to the B-leg. As far as I see, there is a pre-answer > >> method that fs needs to send the media to A-leg. > >> > >> In order to try to avoid this, I tried using ignore_early_media=true as > part > >> of the originate in A-leg and/or B-leg, with no luck. > >> > >> > >> originate {ignore_early_media=true}loopback/2500/default/XML > >> &bridge({ignore_early_media=true}user/2001) > >> > >> > >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] > >> destination_number(2500) =~ /^(2500)$/ break=on-false > >> > >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) > >> > >> Dialplan: loopback/2500-b Action callcenter(click2call) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 > >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 > >> (loopback/2500-b) State ROUTING going to sleep > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 > >> (loopback/2500-b) Running State Change CS_EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 > >> (loopback/2500-b) State EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b > >> CHANNEL EXECUTE > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 > >> loopback/2500-b Standard EXECUTE > >> > >> EXECUTE loopback/2500-b set(open=true) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [open]=[true] > >> > >> EXECUTE loopback/2500-b > >> > hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) > >> > >> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) > >> > >> EXECUTE loopback/2500-b > >> > hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) > >> > >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 > -0300) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] > >> > >> EXECUTE loopback/2500-b set(ignore_early_media=true) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b > SET > >> [ignore_early_media]=[true] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 > Application > >> callcenter Requires media! pre_answering channel loopback/2500-b > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer > >> loopback/2500-a! > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 > (loopback/2500-a) > >> Callstate Change RINGING -> EARLY > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 > Pre-Answer > >> loopback/2500-b! > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 > (loopback/2500-b) > >> Callstate Change RINGING -> EARLY > >> > >> EXECUTE loopback/2500-b callcenter(click2call) > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 > (loopback/2500-a) > >> Callstate Change EARLY -> ACTIVE > >> > >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel > >> [loopback/2500-a] has been answered > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send > signal > >> loopback/2500-b [BREAK] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b > >> CHANNEL KILL > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 > Originate > >> Resulted in Success: [loopback/2500-a] > >> > >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 > (loopback/2500-b) > >> Callstate Change EARLY -> ACTIVE > >> > >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a > >> Flipping CID from "" <0000000000> to "Outbound Call" > >> > >> > >> > >> > >> > >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: > >> > >> > >> Also, maybe I should be doing something like this: > >> > >> > >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' > >> > >> > >> instead of: > >> > >> > >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' > >> > >> > >> > >> but, I don't really have the CISCO configured as a gateway, nor do I > >> > >> know how really...probably not on the right track there. > >> > >> > >> > >> > >> > >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk > wrote: > >> > >> *bump* > >> > >> > >> > >> So I think maybe the way I'm doing the originate is the problem? In my > >> > >> call string I'm creating a connection directly from the CISCO > >> > >> (192.168.x.x) to the managed application, which may be why it starts > >> > >> playing straight away? > >> > >> > >> Maybe I should be originating a call first and then only once I know > >> > >> the other side has picked up will I bridge the call to the IVR managed > >> > >> application. > >> > >> > >> Problem is I dunno how to tell whether the other person has picked up > >> > >> (or even if the cisco is going to tell me) and I don't know how to do > >> > >> things to a call once it has been established. > >> > >> > >> > >> I'm currently reading the Dialplan wiki page, hoping to get something > >> > >> out of it there. > >> > >> > >> > >> Cheers > >> > >> > >> Oliver > >> > >> > >> > >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk > wrote: > >> > >> I've been battling while creating an IVR using FreeSWITCH mod_managed > >> > >> and connecting through a CISCO 2811. Most things now work quite well, > >> > >> but I am having a few issues with the way the system answers calls (or > >> > >> doesn't answer calls...). > >> > >> > >> I have FreeSWITCH running as a windows service on Windows Server 2008, > >> > >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, > >> > >> which is then connected to a POTS phone line. > >> > >> > >> > >> Take the following scenario: > >> > >> > >> 1. Managed .NET application creates a call string and uses ESL to talk > >> > >> to freeswitch and originate a call: > >> > >> > >> string callstring = > >> > >> > "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x > >> > >> '&managed(ivrAppName)'"; > >> > >> eslConnection.API("originate", callstring); > >> > >> > >> where 192.168.x.x is the CISCO IP. > >> > >> > >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 > >> > >> so it uses FXO port 1 and strips the 1 and uses the remaining phone > >> > >> number (091234567) to make the call. > >> > >> > >> 3. My phone rings, I pick up and I can hear my IVR playing. > >> > >> > >> > >> > >> These are my current problems: > >> > >> > >> - IVR starts playing before I even pick up the phone. This means that > >> > >> if the system calls a mobile phone and the person doesn't pick up, the > >> > >> IVR will start playing and eventually the mobile phone will divert to > >> > >> voice mail. Obviously I then get a missed call and an sms saying I > >> > >> have a new voice mail, which is annoying. Instead I would like it to > >> > >> KNOW that no one has picked up, but I don't know how to do this. > >> > >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call > >> > >> has not yet been answered. For some reason however as soon as the > >> > >> CISCO starts calling FreeSWITCH thinks the call is already connected. > >> > >> It doesn't know that the CISCO is actually still ringing. Maybe I'm > >> > >> doing originate the wrong way or something ... > >> > >> > >> - The phone only rings for about 10 seconds before hanging up. I've > >> > >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting > >> > >> CISCO "ring number". Nothing works, my phone still only rings for > >> > >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a > >> > >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just > >> > >> starts playing even if no one answers the phone. > >> > >> > >> > >> > >> > >> > >> CISCO Config for relevant FXO port: > >> > >> > >> voice service voip > >> > >> allow-connections h323 to h323 > >> > >> allow-connections h323 to sip > >> > >> allow-connections sip to h323 > >> > >> allow-connections sip to sip > >> > >> no supplementary-service h450.2 > >> > >> no supplementary-service h450.3 > >> > >> supplementary-service h450.12 > >> > >> no supplementary-service sip moved-temporarily > >> > >> no supplementary-service sip refer > >> > >> fax protocol cisco > >> > >> sip > >> > >> registrar server expires max 3600 min 3600 > >> > >> no update-callerid > >> > >> no call service stop > >> > >> > >> voice-port 0/3/2 > >> > >> output attenuation -3 > >> > >> no comfort-noise > >> > >> cptone AU > >> > >> impedance complex1 > >> > >> caller-id enable > >> > >> ! > >> > >> dial-peer voice 100 pots > >> > >> preference 1 > >> > >> destination-pattern 1T > >> > >> port 0/3/2 > >> > >> ! > >> > >> > >> > >> > >> Many Thanks, > >> > >> > >> Oliver > >> > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> > >> consulting at freeswitch.org > >> > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> > >> http://www.freeswitch.org > >> > >> http://wiki.freeswitch.org > >> > >> http://www.cluecon.com > >> > >> > >> FreeSWITCH-users mailing list > >> > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> > >> Professional FreeSWITCH Consulting Services: > >> > >> consulting at freeswitch.org > >> > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> > >> http://www.freeswitch.org > >> > >> http://wiki.freeswitch.org > >> > >> http://www.cluecon.com > >> > >> > >> FreeSWITCH-users mailing list > >> > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> -- > >> Brian West > >> FreeSWITCH Solutions, LLC > >> Phone: +1 (918) 420-9266 > >> Fax: +1 (918) 420-9267 > >> brian at freeswitch.org > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f06d49b32762089563979! > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/205878fe/attachment-0001.html From bharatlalcheta at gmail.com Fri Jan 6 18:46:11 2012 From: bharatlalcheta at gmail.com (Bharat Lalcheta) Date: Fri, 6 Jan 2012 21:16:11 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 52 In-Reply-To: References: Message-ID: Can you please provide a simple example to use different codec in both leg using dialplan Regards, Bharat Lalcheta On Fri, Jan 6, 2012 at 9:10 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Codec Preferance (Kristian Kielhofner) > 2. Re: Codec Preferance (curriegrad2004) > 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta) > > > ---------- Forwarded message ---------- > From: Kristian Kielhofner > To: FreeSWITCH Users Help > Cc: > Date: Fri, 6 Jan 2012 10:24:01 -0500 > Subject: Re: [Freeswitch-users] Codec Preferance > inbound_codec_prefs > outbound_codec_prefs > > These are Sofia profile configuration options. They are NOT valid > options for the directory UNLESS you're setting them as variable to > use in your dialplan later. > > If you want to control codes on a per-user basis you have to set late > negotiation and use the dialplan. > > On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta > wrote: > > Hiii, > > > > I am new to freeswitch. Prior to freeswitch i was using asterisk. > > > > I have 200 extensions working in my office and want to move all to > > freeswitch from asterisk. > > > > In asterisk, i can give codec selection and preferance in sip.conf to all > > extensions. In the same way i created 200 extensions under internal > profile > > in freeswitch. > > > > Follwing is one example.... > > ----------------------------------------------------------------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ---------------------------------------------------------- > > > > Now when ever i called to 590 freeswitch sends all codecs to 590 sip > phone > > other than defined in 590.xml. It is seding codes which is mentioned in > my > > conf/sip_profiles/internal.xml and codec negotiation done on whatever > codec > > my sip phone having. > > > > I want to use different codecs for different extensions. > > > > Is it common behaviour of Freeswitch ? Should i override codec > prerfrance in > > my extension list from my internal profile or not ? > > > > If no, then is it that i have to create 200 profiles in freeswitch to > solve > > this problem ? > > > > Please guide me and provide solution for the same > > > > > > Thanks in advance > > > > Bharat Lalcheta > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > > > > ---------- Forwarded message ---------- > From: curriegrad2004 > To: FreeSWITCH Users Help > Cc: > Date: Fri, 6 Jan 2012 07:37:06 -0800 > Subject: Re: [Freeswitch-users] Codec Preferance > > In theory he could use those variable names but it would involve some > extra work ;) > On 2012-01-06 7:24 AM, "Kristian Kielhofner" wrote: > >> inbound_codec_prefs >> outbound_codec_prefs >> >> These are Sofia profile configuration options. They are NOT valid >> options for the directory UNLESS you're setting them as variable to >> use in your dialplan later. >> >> If you want to control codes on a per-user basis you have to set late >> negotiation and use the dialplan. >> >> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >> wrote: >> > Hiii, >> > >> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >> > >> > I have 200 extensions working in my office and want to move all to >> > freeswitch from asterisk. >> > >> > In asterisk, i can give codec selection and preferance in sip.conf to >> all >> > extensions. In the same way i created 200 extensions under internal >> profile >> > in freeswitch. >> > >> > Follwing is one example.... >> > ----------------------------------------------------------------- >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > ---------------------------------------------------------- >> > >> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone >> > other than defined in 590.xml. It is seding codes which is mentioned in >> my >> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec >> > my sip phone having. >> > >> > I want to use different codecs for different extensions. >> > >> > Is it common behaviour of Freeswitch ? Should i override codec >> prerfrance in >> > my extension list from my internal profile or not ? >> > >> > If no, then is it that i have to create 200 profiles in freeswitch to >> solve >> > this problem ? >> > >> > Please guide me and provide solution for the same >> > >> > >> > Thanks in advance >> > >> > Bharat Lalcheta >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Bharat Lalcheta > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Fri, 6 Jan 2012 21:10:01 +0530 > Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 > can you please explain in details what you want to tell ? > > > > >> ---------- Forwarded message ---------- >> From: curriegrad2004 >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 6 Jan 2012 07:11:52 -0800 >> Subject: Re: [Freeswitch-users] Codec Preferance >> >> I highly would recommend that you change the name of those codecs to >> something else because you might be making matters worse later down the >> road >> On 2012-01-06 5:21 AM, "Bharat Lalcheta" >> wrote: >> >>> Hiii, >>> >>> I am new to freeswitch. Prior to freeswitch i was using asterisk. >>> >>> I have 200 extensions working in my office and want to move all to >>> freeswitch from asterisk. >>> >>> In asterisk, i can give codec selection and preferance in sip.conf to >>> all extensions. In the same way i created 200 extensions under internal >>> profile in freeswitch. >>> >>> Follwing is one example.... >>> ----------------------------------------------------------------- >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ---------------------------------------------------------- >>> >>> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>> phone other than defined in 590.xml. It is seding codes which is mentioned >>> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >>> codec my sip phone having. >>> >>> I want to use different codecs for different extensions. >>> >>> Is it common behaviour of Freeswitch ? Should i override codec >>> prerfrance in my extension list from my internal profile or not ? >>> >>> If no, then is it that i have to create 200 profiles in freeswitch to >>> solve this problem ? >>> >>> Please guide me and provide solution for the same >>> >>> >>> Thanks in advance >>> >>> Bharat Lalcheta >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> ---------- Forwarded message ---------- >> From: Peter Olsson >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 6 Jan 2012 15:21:00 +0000 >> Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> Are you still using ignore_early_media=true - this must be set for this >> to work correctly. >> >> You will see a EXECUTE log line when FS executes the application, with >> ignore_early_media enabled it shouldn't execute until the call has been >> answered. I just tried it myself, and it works as expected. >> >> Example "originate {ignore_early_media=true}sofia/internal/number at host&park()" >> >> Park application is only executed after the call was answered. >> >> /Peter >> >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ >> olimonkey at gmail.com] >> Skickat: den 6 januari 2012 12:04 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> >> Because I'm using an FXO card with voice, I added something to my >> CISCO conf. Many others had the same thing: >> >> >> voice-port 0/3/0 >> ... >> supervisory disconnect dualtone mid-call >> supervisory answer dualtone <---- ADDED THIS ONE >> ... >> >> >> >> Once I added this, the FS output now just showed the following while >> the phone was ringing: >> >> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >> sofia/internal/109212xxxx at 192.168.x.x >> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >> Call" <109212xxxx> >> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >> sofia/internal/109212xxxx at 192.168.x.x! >> >> >> Where as previous it would show the above and also show the following: >> >> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >> <0000000000> to "Outbound Call" <109212xxxx> >> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >> <1092122856> >> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >> >> >> >> BUT, the IVR still started playing even before I pick up the phone. >> Hmmmm.....so why is FS still starting the managed application when the >> call has not been answered yet. Are we all sure that the managed >> application should not be executed until the call "has been answered" >> shows up in the log file? >> >> >> Will have to keep testing on monday as I don't have access to the >> CISCO from where i am now. I'll have to see whether the CISCO changes >> had any impact on the times at which the SIP messages are sent back >> and forth. Especially the 200 OK message. >> >> >> Thanks again for help, maybe getting somewhere now...... >> >> Oliver >> >> >> >> >> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >> wrote: >> > If it sends 200 OK right after 183, this IS the problem. >> > >> > 200 OK means that the call was answered, it should not be sent until >> the call was actually picked up in the remote end. When 200 OK arrives to >> FS it will execute your app, and you will start playing the files. >> > >> > Seems to me there is something broken in the Cisco. >> > >> > /Peter >> > >> > ________________________________________ >> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ >> olimonkey at gmail.com] >> > Skickat: den 6 januari 2012 06:55 >> > Till: FreeSWITCH Users Help >> > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> > >> > I've tried looking at disable-early-media configuration command, but >> > that didn't work and I doubt that has anything to do with the CISCO >> > sending a 200 OK right after a 183 SESSION PROGRESS. >> > >> > >> > >> > >> > On Fri, Jan 6, 2012 at 9:20 AM, Brian West >> wrote: >> >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. >> 180 >> >> is usually RINGING (generate ringback locally) while a 183 has >> media... aka >> >> early media and usually provides ringback inband. >> >> >> >> /b >> >> >> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >> >> >> >> Shouldn't there be a 180 RINGING somewhere in there? >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk >> wrote: >> >> >> >> I just noticed something else, if I don't pick up the phone at all. >> >> >> >> The IVR just keeps playing until the menu timeout kicks in. >> >> >> >> >> >> So here is a CISCO SIP log: >> >> >> >> http://pastebin.com/Y9sYkuxi >> >> >> >> >> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >> >> >> >> I hope the CISCO log is readable, it's a bit long because I just did >> >> >> >> "debug ccsip all". >> >> >> >> >> >> >> >> >> >> In this test I didn't bother picking up the phone at all, but I can >> >> >> >> see that FS answered anyway and the IVR kept playing until it timed >> >> >> >> out. >> >> >> >> I'm not an expert, but here is what I picked out of it: >> >> >> >> >> >> At 00:08:10 we get a >> >> >> >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >> >> >> >> >> >> the further down at the same timestamp we get >> >> >> >> Sent: "SIP/2.0 100 Trying" >> >> >> >> >> >> At 00:08:13 we get a >> >> >> >> Sent: "SIP/2.0 183 Session Progress" >> >> >> >> >> >> At 00:18:13 we get a >> >> >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> >> Then at the same timestamp we get: >> >> >> >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> >> >> >> >> >> >> >> Once the IVR times out at 00:09:16 we get >> >> >> >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >> >> >> >> >> >> And then the reply right after >> >> >> >> Sent: "SIP/2.0 200 OK" >> >> >> >> >> >> >> >> >> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >> >> >> >> after the "INVITE" is received. >> >> >> >> >> >> >> >> >> >> The part that is beyond my field of expertise so far is WHY? >> >> >> >> >> >> >> >> >> >> Thanks, >> >> >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk >> wrote: >> >> >> >> By the way: >> >> >> >> >> >> I tried {ignore_early_media=true} as well, but as I think we >> >> >> >> determined, my problem is probably with the CISCO telling FS that the >> >> >> >> call has been answered when really it hasn't yet. >> >> >> >> >> >> >> >> >> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk >> wrote: >> >> >> >> Thanks for the help so far. >> >> >> >> >> >> >> >> Here is a pastebin of FreeSWITCH output: >> >> >> >> http://pastebin.com/i6Qgc7ws >> >> >> >> >> >> Notice how the "has been answered" log message comes immediately >> >> >> >> (within a few milliseconds) after the call was originated. I think >> >> >> >> this would suggest that the CISCO is immediately sending a 200 OK, as >> >> >> >> you suggested. I also turned on CISCO debugging, but I'm just trying >> >> >> >> to figure out how to get the information regarding SIP messages back >> >> >> >> to Freeswitch. I'll run the test again and see if I can get some >> >> >> >> useful CISCO debug. >> >> >> >> >> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >> >> >> >> SIP debugging? >> >> >> >> >> >> >> >> Thanks! >> >> >> >> >> >> >> >> >> >> >> >> 2012/1/6 Gustavo M?rsico : >> >> >> >> I think I've a similar problem related to callcenter app. When I made >> an >> >> originate like this: >> >> >> >> >> >> originate loopback/2500/default/XML &bridge(user/2001) >> >> >> >> >> >> 2500 is an extension that leads to a callcenter application. In this >> case, >> >> we dial first to the queue and when an agent answered we call to the >> >> customer. As far as I know >> >> >> >> When the A-leg reaches to the queue, without selecting an agent, the >> call is >> >> automatically sent to the B-leg. As far as I see, there is a pre-answer >> >> method that fs needs to send the media to A-leg. >> >> >> >> In order to try to avoid this, I tried using ignore_early_media=true >> as part >> >> of the originate in A-leg and/or B-leg, with no luck. >> >> >> >> >> >> originate {ignore_early_media=true}loopback/2500/default/XML >> >> &bridge({ignore_early_media=true}user/2001) >> >> >> >> >> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >> >> destination_number(2500) =~ /^(2500)$/ break=on-false >> >> >> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >> >> >> >> Dialplan: loopback/2500-b Action callcenter(click2call) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >> >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >> >> (loopback/2500-b) State ROUTING going to sleep >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >> >> (loopback/2500-b) Running State Change CS_EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >> >> (loopback/2500-b) State EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >> >> CHANNEL EXECUTE >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >> >> loopback/2500-b Standard EXECUTE >> >> >> >> EXECUTE loopback/2500-b set(open=true) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [open]=[true] >> >> >> >> EXECUTE loopback/2500-b >> >> >> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> >> >> EXECUTE loopback/2500-b >> hash(insert/10.8.0.70-last_dial/0000000000/2500) >> >> >> >> EXECUTE loopback/2500-b >> >> >> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >> >> >> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 >> -0300) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >> >> >> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b >> SET >> >> [ignore_early_media]=[true] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 >> Application >> >> callcenter Requires media! pre_answering channel loopback/2500-b >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >> >> loopback/2500-a! >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >> (loopback/2500-a) >> >> Callstate Change RINGING -> EARLY >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 >> Pre-Answer >> >> loopback/2500-b! >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >> (loopback/2500-b) >> >> Callstate Change RINGING -> EARLY >> >> >> >> EXECUTE loopback/2500-b callcenter(click2call) >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >> (loopback/2500-a) >> >> Callstate Change EARLY -> ACTIVE >> >> >> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >> >> [loopback/2500-a] has been answered >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >> signal >> >> loopback/2500-b [BREAK] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >> >> CHANNEL KILL >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 >> Originate >> >> Resulted in Success: [loopback/2500-a] >> >> >> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >> (loopback/2500-b) >> >> Callstate Change EARLY -> ACTIVE >> >> >> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 >> loopback/2500-a >> >> Flipping CID from "" <0000000000> to "Outbound Call" >> >> >> >> >> >> >> >> >> >> >> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >> >> >> >> >> >> Also, maybe I should be doing something like this: >> >> >> >> >> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >> >> >> >> >> >> instead of: >> >> >> >> >> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >> >> >> >> >> >> >> >> but, I don't really have the CISCO configured as a gateway, nor do I >> >> >> >> know how really...probably not on the right track there. >> >> >> >> >> >> >> >> >> >> >> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk >> wrote: >> >> >> >> *bump* >> >> >> >> >> >> >> >> So I think maybe the way I'm doing the originate is the problem? In my >> >> >> >> call string I'm creating a connection directly from the CISCO >> >> >> >> (192.168.x.x) to the managed application, which may be why it starts >> >> >> >> playing straight away? >> >> >> >> >> >> Maybe I should be originating a call first and then only once I know >> >> >> >> the other side has picked up will I bridge the call to the IVR managed >> >> >> >> application. >> >> >> >> >> >> Problem is I dunno how to tell whether the other person has picked up >> >> >> >> (or even if the cisco is going to tell me) and I don't know how to do >> >> >> >> things to a call once it has been established. >> >> >> >> >> >> >> >> I'm currently reading the Dialplan wiki page, hoping to get something >> >> >> >> out of it there. >> >> >> >> >> >> >> >> Cheers >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk >> wrote: >> >> >> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >> >> >> >> and connecting through a CISCO 2811. Most things now work quite well, >> >> >> >> but I am having a few issues with the way the system answers calls (or >> >> >> >> doesn't answer calls...). >> >> >> >> >> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >> >> >> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >> >> >> >> which is then connected to a POTS phone line. >> >> >> >> >> >> >> >> Take the following scenario: >> >> >> >> >> >> 1. Managed .NET application creates a call string and uses ESL to talk >> >> >> >> to freeswitch and originate a call: >> >> >> >> >> >> string callstring = >> >> >> >> >> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >> >> >> >> '&managed(ivrAppName)'"; >> >> >> >> eslConnection.API("originate", callstring); >> >> >> >> >> >> where 192.168.x.x is the CISCO IP. >> >> >> >> >> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >> >> >> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >> >> >> >> number (091234567) to make the call. >> >> >> >> >> >> 3. My phone rings, I pick up and I can hear my IVR playing. >> >> >> >> >> >> >> >> >> >> These are my current problems: >> >> >> >> >> >> - IVR starts playing before I even pick up the phone. This means that >> >> >> >> if the system calls a mobile phone and the person doesn't pick up, the >> >> >> >> IVR will start playing and eventually the mobile phone will divert to >> >> >> >> voice mail. Obviously I then get a missed call and an sms saying I >> >> >> >> have a new voice mail, which is annoying. Instead I would like it to >> >> >> >> KNOW that no one has picked up, but I don't know how to do this. >> >> >> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >> >> >> >> has not yet been answered. For some reason however as soon as the >> >> >> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >> >> >> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >> >> >> >> doing originate the wrong way or something ... >> >> >> >> >> >> - The phone only rings for about 10 seconds before hanging up. I've >> >> >> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >> >> >> >> CISCO "ring number". Nothing works, my phone still only rings for >> >> >> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >> >> >> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >> >> >> >> starts playing even if no one answers the phone. >> >> >> >> >> >> >> >> >> >> >> >> >> >> CISCO Config for relevant FXO port: >> >> >> >> >> >> voice service voip >> >> >> >> allow-connections h323 to h323 >> >> >> >> allow-connections h323 to sip >> >> >> >> allow-connections sip to h323 >> >> >> >> allow-connections sip to sip >> >> >> >> no supplementary-service h450.2 >> >> >> >> no supplementary-service h450.3 >> >> >> >> supplementary-service h450.12 >> >> >> >> no supplementary-service sip moved-temporarily >> >> >> >> no supplementary-service sip refer >> >> >> >> fax protocol cisco >> >> >> >> sip >> >> >> >> registrar server expires max 3600 min 3600 >> >> >> >> no update-callerid >> >> >> >> no call service stop >> >> >> >> >> >> voice-port 0/3/2 >> >> >> >> output attenuation -3 >> >> >> >> no comfort-noise >> >> >> >> cptone AU >> >> >> >> impedance complex1 >> >> >> >> caller-id enable >> >> >> >> ! >> >> >> >> dial-peer voice 100 pots >> >> >> >> preference 1 >> >> >> >> destination-pattern 1T >> >> >> >> port 0/3/2 >> >> >> >> ! >> >> >> >> >> >> >> >> >> >> Many Thanks, >> >> >> >> >> >> Oliver >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> consulting at freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> consulting at freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> Brian West >> >> FreeSWITCH Solutions, LLC >> >> Phone: +1 (918) 420-9266 >> >> Fax: +1 (918) 420-9267 >> >> brian at freeswitch.org >> >> http://www.freeswitch.org >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f06d49b32762089563979! >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Bharat Lalcheta > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/16a821f3/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 6 18:50:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Jan 2012 09:50:03 -0600 Subject: [Freeswitch-users] Why database is locked? In-Reply-To: References: Message-ID: Are you running the latest version of FS? Also are you doing anything externally to mess with the sql db files? Another possibility is a slow filesystem, we used to have people get this problem on suse linux. You could try running your db dir in a ramdisk. On Thu, Jan 5, 2012 at 10:56 PM, Valery Kalinin wrote: > Hi all! > > Periodically database is locked and all calls stopped. > > Log: > 2012-01-06 08:32:52.492796 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > update sip_dialogs set presence_id='2002 at 192.168.205.1',presence_data='' > where uuid='f506627c-ea28-481c-ba72-beaeebc16a6b'; > 2012-01-06 08:33:12.486159 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC abort > frame received (ZT_EVENT_ABORT) > 2012-01-06 08:33:22.591484 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > insert into sip_dialogs > (call_id,uuid,sip_to_user,sip_to_host,sip_to_tag,sip_from_user,sip_from_host,sip_from_tag,contact_user,contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,call_info,rcd) > values('810ae706-b2b1-122f-aa88-6cf049ef3354','b925d777-a70c-4d71-a07b-bce5a804f48a','2002','192.168.205.1','e21ByaF046ppQ','2003','192.168.205.1','1436972916','2003','192.168.205.217','early','outbound','Yealink > SIP-T22P 7.60.14.5','internal','srv-pok-phone','< > sip:2003 at 192.168.205.217:5062>','2003 at 192.168.205.1','','',1325817146) > 2012-01-06 08:33:41.089426 [ERR] ftmod_zt.c:1280 [s1c16][1:16] Dropping > event 8 to be able to write data > 2012-01-06 08:33:52.667878 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > delete from sip_dialogs where uuid='8ed4b985-1f1f-4901-938d-56619a42eb1c' > 2012-01-06 08:34:02.189552 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC abort > frame received (ZT_EVENT_ABORT) > 2012-01-06 08:34:22.607797 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > delete from sip_authentication where expires > 0 and expires <= 1325817161 > and hostname='srv-pok-phone.lpurs.argos-group.ru' > 2012-01-06 08:34:52.460796 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > insert into sip_authentication (nonce,expires,profile_name,hostname, > last_nc) values('9643f3a4-acc0-41cf-8a08-ca9dd791dbbb', 1325817222, > 'internal', 'srv-pok-phone', 0) > 2012-01-06 08:42:54.233881 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > insert into sip_authentication (nonce,expires,profile_name,hostname, > last_nc) values('4611d24f-cfa0-47d4-a8d3-5fdce08ed85a', 1325817734, > 'internal', 'srv-pok-phone', 0) > 2012-01-06 08:42:54.884449 [CRIT] ftdm_io.c:5467 [s1c9][1:9] Forcing > hangup since the user did not confirmed our hangup after 3000ms > 2012-01-06 08:43:06.435420 [ERR] ftmod_libpri.c:132 XXX Progress message > requested but no information is provided > 2012-01-06 08:43:06.813352 [ERR] ftmod_libpri.c:132 Received unsolicited > status: Mandatory information element is missing > 2012-01-06 08:43:24.297760 [ERR] switch_core_sqldb.c:481 SQL ERR > [database is locked] > delete from sip_dialogs where uuid='563a394c-5c82-4381-868d-35e344f3ccd9' > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/a2a22480/attachment.html From bharatlalcheta at gmail.com Fri Jan 6 19:13:52 2012 From: bharatlalcheta at gmail.com (Bharat Lalcheta) Date: Fri, 6 Jan 2012 21:43:52 +0530 Subject: [Freeswitch-users] Codec Preferance Message-ID: > Can you please provide a simple example to use different codec in both leg > using dialplan > > Regards, > > Bharat Lalcheta > > On Fri, Jan 6, 2012 at 9:10 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> Send FreeSWITCH-users mailing list submissions to >> freeswitch-users at lists.freeswitch.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> or, via email, send a message with subject or body 'help' to >> freeswitch-users-request at lists.freeswitch.org >> >> You can reach the person managing the list at >> freeswitch-users-owner at lists.freeswitch.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of FreeSWITCH-users digest..." >> >> Today's Topics: >> >> 1. Re: Codec Preferance (Kristian Kielhofner) >> 2. Re: Codec Preferance (curriegrad2004) >> 3. Re: FreeSWITCH-users Digest, Vol 67, Issue 51 (Bharat Lalcheta) >> >> >> ---------- Forwarded message ---------- >> From: Kristian Kielhofner >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 6 Jan 2012 10:24:01 -0500 >> Subject: Re: [Freeswitch-users] Codec Preferance >> inbound_codec_prefs >> outbound_codec_prefs >> >> These are Sofia profile configuration options. They are NOT valid >> options for the directory UNLESS you're setting them as variable to >> use in your dialplan later. >> >> If you want to control codes on a per-user basis you have to set late >> negotiation and use the dialplan. >> >> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >> wrote: >> > Hiii, >> > >> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >> > >> > I have 200 extensions working in my office and want to move all to >> > freeswitch from asterisk. >> > >> > In asterisk, i can give codec selection and preferance in sip.conf to >> all >> > extensions. In the same way i created 200 extensions under internal >> profile >> > in freeswitch. >> > >> > Follwing is one example.... >> > ----------------------------------------------------------------- >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > ---------------------------------------------------------- >> > >> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >> phone >> > other than defined in 590.xml. It is seding codes which is mentioned in >> my >> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >> codec >> > my sip phone having. >> > >> > I want to use different codecs for different extensions. >> > >> > Is it common behaviour of Freeswitch ? Should i override codec >> prerfrance in >> > my extension list from my internal profile or not ? >> > >> > If no, then is it that i have to create 200 profiles in freeswitch to >> solve >> > this problem ? >> > >> > Please guide me and provide solution for the same >> > >> > >> > Thanks in advance >> > >> > Bharat Lalcheta >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Kristian Kielhofner >> >> >> >> >> ---------- Forwarded message ---------- >> From: curriegrad2004 >> To: FreeSWITCH Users Help >> Cc: >> Date: Fri, 6 Jan 2012 07:37:06 -0800 >> Subject: Re: [Freeswitch-users] Codec Preferance >> >> In theory he could use those variable names but it would involve some >> extra work ;) >> On 2012-01-06 7:24 AM, "Kristian Kielhofner" wrote: >> >>> inbound_codec_prefs >>> outbound_codec_prefs >>> >>> These are Sofia profile configuration options. They are NOT valid >>> options for the directory UNLESS you're setting them as variable to >>> use in your dialplan later. >>> >>> If you want to control codes on a per-user basis you have to set late >>> negotiation and use the dialplan. >>> >>> On Fri, Jan 6, 2012 at 8:18 AM, Bharat Lalcheta >>> wrote: >>> > Hiii, >>> > >>> > I am new to freeswitch. Prior to freeswitch i was using asterisk. >>> > >>> > I have 200 extensions working in my office and want to move all to >>> > freeswitch from asterisk. >>> > >>> > In asterisk, i can give codec selection and preferance in sip.conf to >>> all >>> > extensions. In the same way i created 200 extensions under internal >>> profile >>> > in freeswitch. >>> > >>> > Follwing is one example.... >>> > ----------------------------------------------------------------- >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > ---------------------------------------------------------- >>> > >>> > Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>> phone >>> > other than defined in 590.xml. It is seding codes which is mentioned >>> in my >>> > conf/sip_profiles/internal.xml and codec negotiation done on whatever >>> codec >>> > my sip phone having. >>> > >>> > I want to use different codecs for different extensions. >>> > >>> > Is it common behaviour of Freeswitch ? Should i override codec >>> prerfrance in >>> > my extension list from my internal profile or not ? >>> > >>> > If no, then is it that i have to create 200 profiles in freeswitch to >>> solve >>> > this problem ? >>> > >>> > Please guide me and provide solution for the same >>> > >>> > >>> > Thanks in advance >>> > >>> > Bharat Lalcheta >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> >>> -- >>> Kristian Kielhofner >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> ---------- Forwarded message ---------- >> From: Bharat Lalcheta >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Fri, 6 Jan 2012 21:10:01 +0530 >> Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 51 >> can you please explain in details what you want to tell ? >> >> >> >> >>> ---------- Forwarded message ---------- >>> From: curriegrad2004 >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Fri, 6 Jan 2012 07:11:52 -0800 >>> Subject: Re: [Freeswitch-users] Codec Preferance >>> >>> I highly would recommend that you change the name of those codecs to >>> something else because you might be making matters worse later down the >>> road >>> On 2012-01-06 5:21 AM, "Bharat Lalcheta" >>> wrote: >>> >>>> Hiii, >>>> >>>> I am new to freeswitch. Prior to freeswitch i was using asterisk. >>>> >>>> I have 200 extensions working in my office and want to move all to >>>> freeswitch from asterisk. >>>> >>>> In asterisk, i can give codec selection and preferance in sip.conf to >>>> all extensions. In the same way i created 200 extensions under internal >>>> profile in freeswitch. >>>> >>>> Follwing is one example.... >>>> ----------------------------------------------------------------- >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ---------------------------------------------------------- >>>> >>>> Now when ever i called to 590 freeswitch sends all codecs to 590 sip >>>> phone other than defined in 590.xml. It is seding codes which is mentioned >>>> in my conf/sip_profiles/internal.xml and codec negotiation done on whatever >>>> codec my sip phone having. >>>> >>>> I want to use different codecs for different extensions. >>>> >>>> Is it common behaviour of Freeswitch ? Should i override codec >>>> prerfrance in my extension list from my internal profile or not ? >>>> >>>> If no, then is it that i have to create 200 profiles in freeswitch to >>>> solve this problem ? >>>> >>>> Please guide me and provide solution for the same >>>> >>>> >>>> Thanks in advance >>>> >>>> Bharat Lalcheta >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> ---------- Forwarded message ---------- >>> From: Peter Olsson >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Fri, 6 Jan 2012 15:21:00 +0000 >>> Subject: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> Are you still using ignore_early_media=true - this must be set for this >>> to work correctly. >>> >>> You will see a EXECUTE log line when FS executes the application, with >>> ignore_early_media enabled it shouldn't execute until the call has been >>> answered. I just tried it myself, and it works as expected. >>> >>> Example "originate {ignore_early_media=true}sofia/internal/number at host&park()" >>> >>> Park application is only executed after the call was answered. >>> >>> /Peter >>> >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>> freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ >>> olimonkey at gmail.com] >>> Skickat: den 6 januari 2012 12:04 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> >>> Because I'm using an FXO card with voice, I added something to my >>> CISCO conf. Many others had the same thing: >>> >>> >>> voice-port 0/3/0 >>> ... >>> supervisory disconnect dualtone mid-call >>> supervisory answer dualtone <---- ADDED THIS ONE >>> ... >>> >>> >>> >>> Once I added this, the FS output now just showed the following while >>> the phone was ringing: >>> >>> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >>> sofia/internal/109212xxxx at 192.168.x.x >>> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >>> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >>> Call" <109212xxxx> >>> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >>> sofia/internal/109212xxxx at 192.168.x.x! >>> >>> >>> Where as previous it would show the above and also show the following: >>> >>> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >>> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >>> <0000000000> to "Outbound Call" <109212xxxx> >>> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >>> <1092122856> >>> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >>> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >>> >>> >>> >>> BUT, the IVR still started playing even before I pick up the phone. >>> Hmmmm.....so why is FS still starting the managed application when the >>> call has not been answered yet. Are we all sure that the managed >>> application should not be executed until the call "has been answered" >>> shows up in the log file? >>> >>> >>> Will have to keep testing on monday as I don't have access to the >>> CISCO from where i am now. I'll have to see whether the CISCO changes >>> had any impact on the times at which the SIP messages are sent back >>> and forth. Especially the 200 OK message. >>> >>> >>> Thanks again for help, maybe getting somewhere now...... >>> >>> Oliver >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >>> wrote: >>> > If it sends 200 OK right after 183, this IS the problem. >>> > >>> > 200 OK means that the call was answered, it should not be sent until >>> the call was actually picked up in the remote end. When 200 OK arrives to >>> FS it will execute your app, and you will start playing the files. >>> > >>> > Seems to me there is something broken in the Cisco. >>> > >>> > /Peter >>> > >>> > ________________________________________ >>> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ >>> freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [ >>> olimonkey at gmail.com] >>> > Skickat: den 6 januari 2012 06:55 >>> > Till: FreeSWITCH Users Help >>> > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> > >>> > I've tried looking at disable-early-media configuration command, but >>> > that didn't work and I doubt that has anything to do with the CISCO >>> > sending a 200 OK right after a 183 SESSION PROGRESS. >>> > >>> > >>> > >>> > >>> > On Fri, Jan 6, 2012 at 9:20 AM, Brian West >>> wrote: >>> >> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some >>> devices.. 180 >>> >> is usually RINGING (generate ringback locally) while a 183 has >>> media... aka >>> >> early media and usually provides ringback inband. >>> >> >>> >> /b >>> >> >>> >> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>> >> >>> >> Shouldn't there be a 180 RINGING somewhere in there? >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk >>> wrote: >>> >> >>> >> I just noticed something else, if I don't pick up the phone at all. >>> >> >>> >> The IVR just keeps playing until the menu timeout kicks in. >>> >> >>> >> >>> >> So here is a CISCO SIP log: >>> >> >>> >> http://pastebin.com/Y9sYkuxi >>> >> >>> >> >>> >> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>> >> >>> >> I hope the CISCO log is readable, it's a bit long because I just did >>> >> >>> >> "debug ccsip all". >>> >> >>> >> >>> >> >>> >> >>> >> In this test I didn't bother picking up the phone at all, but I can >>> >> >>> >> see that FS answered anyway and the IVR kept playing until it timed >>> >> >>> >> out. >>> >> >>> >> I'm not an expert, but here is what I picked out of it: >>> >> >>> >> >>> >> At 00:08:10 we get a >>> >> >>> >> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>> >> >>> >> >>> >> the further down at the same timestamp we get >>> >> >>> >> Sent: "SIP/2.0 100 Trying" >>> >> >>> >> >>> >> At 00:08:13 we get a >>> >> >>> >> Sent: "SIP/2.0 183 Session Progress" >>> >> >>> >> >>> >> At 00:18:13 we get a >>> >> >>> >> Sent: "SIP/2.0 200 OK" >>> >> >>> >> >>> >> Then at the same timestamp we get: >>> >> >>> >> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>> >> >>> >> >>> >> >>> >> >>> >> Once the IVR times out at 00:09:16 we get >>> >> >>> >> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>> >> >>> >> >>> >> And then the reply right after >>> >> >>> >> Sent: "SIP/2.0 200 OK" >>> >> >>> >> >>> >> >>> >> >>> >> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>> >> >>> >> after the "INVITE" is received. >>> >> >>> >> >>> >> >>> >> >>> >> The part that is beyond my field of expertise so far is WHY? >>> >> >>> >> >>> >> >>> >> >>> >> Thanks, >>> >> >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk >>> wrote: >>> >> >>> >> By the way: >>> >> >>> >> >>> >> I tried {ignore_early_media=true} as well, but as I think we >>> >> >>> >> determined, my problem is probably with the CISCO telling FS that the >>> >> >>> >> call has been answered when really it hasn't yet. >>> >> >>> >> >>> >> >>> >> >>> >> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk >>> wrote: >>> >> >>> >> Thanks for the help so far. >>> >> >>> >> >>> >> >>> >> Here is a pastebin of FreeSWITCH output: >>> >> >>> >> http://pastebin.com/i6Qgc7ws >>> >> >>> >> >>> >> Notice how the "has been answered" log message comes immediately >>> >> >>> >> (within a few milliseconds) after the call was originated. I think >>> >> >>> >> this would suggest that the CISCO is immediately sending a 200 OK, as >>> >> >>> >> you suggested. I also turned on CISCO debugging, but I'm just trying >>> >> >>> >> to figure out how to get the information regarding SIP messages back >>> >> >>> >> to Freeswitch. I'll run the test again and see if I can get some >>> >> >>> >> useful CISCO debug. >>> >> >>> >> >>> >> Which "debug ccsip" commands are relevant to what I want for the CISCO >>> >> >>> >> SIP debugging? >>> >> >>> >> >>> >> >>> >> Thanks! >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> 2012/1/6 Gustavo M?rsico : >>> >> >>> >> I think I've a similar problem related to callcenter app. When I made >>> an >>> >> originate like this: >>> >> >>> >> >>> >> originate loopback/2500/default/XML &bridge(user/2001) >>> >> >>> >> >>> >> 2500 is an extension that leads to a callcenter application. In this >>> case, >>> >> we dial first to the queue and when an agent answered we call to the >>> >> customer. As far as I know >>> >> >>> >> When the A-leg reaches to the queue, without selecting an agent, the >>> call is >>> >> automatically sent to the B-leg. As far as I see, there is a >>> pre-answer >>> >> method that fs needs to send the media to A-leg. >>> >> >>> >> In order to try to avoid this, I tried using ignore_early_media=true >>> as part >>> >> of the originate in A-leg and/or B-leg, with no luck. >>> >> >>> >> >>> >> originate {ignore_early_media=true}loopback/2500/default/XML >>> >> &bridge({ignore_early_media=true}user/2001) >>> >> >>> >> >>> >> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>> >> destination_number(2500) =~ /^(2500)$/ break=on-false >>> >> >>> >> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>> >> >>> >> Dialplan: loopback/2500-b Action callcenter(click2call) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>> >> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>> >> (loopback/2500-b) State ROUTING going to sleep >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>> >> (loopback/2500-b) Running State Change CS_EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>> >> (loopback/2500-b) State EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>> >> CHANNEL EXECUTE >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>> >> loopback/2500-b Standard EXECUTE >>> >> >>> >> EXECUTE loopback/2500-b set(open=true) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [open]=[true] >>> >> >>> >> EXECUTE loopback/2500-b >>> >> >>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >> >>> >> EXECUTE loopback/2500-b >>> hash(insert/10.8.0.70-last_dial/0000000000/2500) >>> >> >>> >> EXECUTE loopback/2500-b >>> >> >>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >> >>> >> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 >>> -0300) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>> >> >>> >> EXECUTE loopback/2500-b set(ignore_early_media=true) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 >>> loopback/2500-b SET >>> >> [ignore_early_media]=[true] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 >>> Application >>> >> callcenter Requires media! pre_answering channel loopback/2500-b >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>> >> loopback/2500-a! >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >>> (loopback/2500-a) >>> >> Callstate Change RINGING -> EARLY >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 >>> Pre-Answer >>> >> loopback/2500-b! >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 >>> (loopback/2500-b) >>> >> Callstate Change RINGING -> EARLY >>> >> >>> >> EXECUTE loopback/2500-b callcenter(click2call) >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >>> (loopback/2500-a) >>> >> Callstate Change EARLY -> ACTIVE >>> >> >>> >> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>> >> [loopback/2500-a] has been answered >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send >>> signal >>> >> loopback/2500-b [BREAK] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> >> CHANNEL KILL >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 >>> Originate >>> >> Resulted in Success: [loopback/2500-a] >>> >> >>> >> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 >>> (loopback/2500-b) >>> >> Callstate Change EARLY -> ACTIVE >>> >> >>> >> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 >>> loopback/2500-a >>> >> Flipping CID from "" <0000000000> to "Outbound Call" >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>> >> >>> >> >>> >> Also, maybe I should be doing something like this: >>> >> >>> >> >>> >> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>> >> >>> >> >>> >> instead of: >>> >> >>> >> >>> >> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>> >> >>> >> >>> >> >>> >> but, I don't really have the CISCO configured as a gateway, nor do I >>> >> >>> >> know how really...probably not on the right track there. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk >>> wrote: >>> >> >>> >> *bump* >>> >> >>> >> >>> >> >>> >> So I think maybe the way I'm doing the originate is the problem? In my >>> >> >>> >> call string I'm creating a connection directly from the CISCO >>> >> >>> >> (192.168.x.x) to the managed application, which may be why it starts >>> >> >>> >> playing straight away? >>> >> >>> >> >>> >> Maybe I should be originating a call first and then only once I know >>> >> >>> >> the other side has picked up will I bridge the call to the IVR managed >>> >> >>> >> application. >>> >> >>> >> >>> >> Problem is I dunno how to tell whether the other person has picked up >>> >> >>> >> (or even if the cisco is going to tell me) and I don't know how to do >>> >> >>> >> things to a call once it has been established. >>> >> >>> >> >>> >> >>> >> I'm currently reading the Dialplan wiki page, hoping to get something >>> >> >>> >> out of it there. >>> >> >>> >> >>> >> >>> >> Cheers >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> >> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk >>> wrote: >>> >> >>> >> I've been battling while creating an IVR using FreeSWITCH mod_managed >>> >> >>> >> and connecting through a CISCO 2811. Most things now work quite well, >>> >> >>> >> but I am having a few issues with the way the system answers calls (or >>> >> >>> >> doesn't answer calls...). >>> >> >>> >> >>> >> I have FreeSWITCH running as a windows service on Windows Server 2008, >>> >> >>> >> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>> >> >>> >> which is then connected to a POTS phone line. >>> >> >>> >> >>> >> >>> >> Take the following scenario: >>> >> >>> >> >>> >> 1. Managed .NET application creates a call string and uses ESL to talk >>> >> >>> >> to freeswitch and originate a call: >>> >> >>> >> >>> >> string callstring = >>> >> >>> >> >>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>> >> >>> >> '&managed(ivrAppName)'"; >>> >> >>> >> eslConnection.API("originate", callstring); >>> >> >>> >> >>> >> where 192.168.x.x is the CISCO IP. >>> >> >>> >> >>> >> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>> >> >>> >> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>> >> >>> >> number (091234567) to make the call. >>> >> >>> >> >>> >> 3. My phone rings, I pick up and I can hear my IVR playing. >>> >> >>> >> >>> >> >>> >> >>> >> These are my current problems: >>> >> >>> >> >>> >> - IVR starts playing before I even pick up the phone. This means that >>> >> >>> >> if the system calls a mobile phone and the person doesn't pick up, the >>> >> >>> >> IVR will start playing and eventually the mobile phone will divert to >>> >> >>> >> voice mail. Obviously I then get a missed call and an sms saying I >>> >> >>> >> have a new voice mail, which is annoying. Instead I would like it to >>> >> >>> >> KNOW that no one has picked up, but I don't know how to do this. >>> >> >>> >> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>> >> >>> >> has not yet been answered. For some reason however as soon as the >>> >> >>> >> CISCO starts calling FreeSWITCH thinks the call is already connected. >>> >> >>> >> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>> >> >>> >> doing originate the wrong way or something ... >>> >> >>> >> >>> >> - The phone only rings for about 10 seconds before hanging up. I've >>> >> >>> >> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>> >> >>> >> CISCO "ring number". Nothing works, my phone still only rings for >>> >> >>> >> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>> >> >>> >> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>> >> >>> >> starts playing even if no one answers the phone. >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> CISCO Config for relevant FXO port: >>> >> >>> >> >>> >> voice service voip >>> >> >>> >> allow-connections h323 to h323 >>> >> >>> >> allow-connections h323 to sip >>> >> >>> >> allow-connections sip to h323 >>> >> >>> >> allow-connections sip to sip >>> >> >>> >> no supplementary-service h450.2 >>> >> >>> >> no supplementary-service h450.3 >>> >> >>> >> supplementary-service h450.12 >>> >> >>> >> no supplementary-service sip moved-temporarily >>> >> >>> >> no supplementary-service sip refer >>> >> >>> >> fax protocol cisco >>> >> >>> >> sip >>> >> >>> >> registrar server expires max 3600 min 3600 >>> >> >>> >> no update-callerid >>> >> >>> >> no call service stop >>> >> >>> >> >>> >> voice-port 0/3/2 >>> >> >>> >> output attenuation -3 >>> >> >>> >> no comfort-noise >>> >> >>> >> cptone AU >>> >> >>> >> impedance complex1 >>> >> >>> >> caller-id enable >>> >> >>> >> ! >>> >> >>> >> dial-peer voice 100 pots >>> >> >>> >> preference 1 >>> >> >>> >> destination-pattern 1T >>> >> >>> >> port 0/3/2 >>> >> >>> >> ! >>> >> >>> >> >>> >> >>> >> >>> >> Many Thanks, >>> >> >>> >> >>> >> Oliver >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> >>> >> Professional FreeSWITCH Consulting Services: >>> >> >>> >> consulting at freeswitch.org >>> >> >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> http://wiki.freeswitch.org >>> >> >>> >> http://www.cluecon.com >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> >>> >> Professional FreeSWITCH Consulting Services: >>> >> >>> >> consulting at freeswitch.org >>> >> >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> http://wiki.freeswitch.org >>> >> >>> >> http://www.cluecon.com >>> >> >>> >> >>> >> FreeSWITCH-users mailing list >>> >> >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> -- >>> >> Brian West >>> >> FreeSWITCH Solutions, LLC >>> >> Phone: +1 (918) 420-9266 >>> >> Fax: +1 (918) 420-9267 >>> >> brian at freeswitch.org >>> >> http://www.freeswitch.org >>> >> >>> >> >>> >> >>> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> >> >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4f06d49b32762089563979! >>> >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Bharat Lalcheta >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Bharat Lalcheta > -- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/3ab0406d/attachment-0001.html From curriegrad2004 at gmail.com Fri Jan 6 20:37:36 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 6 Jan 2012 09:37:36 -0800 Subject: [Freeswitch-users] Why database is locked? In-Reply-To: References: Message-ID: Running the DB out of cifs can also cause the same symptoms of a locked database On 2012-01-06 7:50 AM, "Anthony Minessale" wrote: > Are you running the latest version of FS? > Also are you doing anything externally to mess with the sql db files? > Another possibility is a slow filesystem, we used to have people get this > problem on suse linux. > You could try running your db dir in a ramdisk. > > > On Thu, Jan 5, 2012 at 10:56 PM, Valery Kalinin wrote: > >> Hi all! >> >> Periodically database is locked and all calls stopped. >> >> Log: >> 2012-01-06 08:32:52.492796 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> update sip_dialogs set presence_id='2002 at 192.168.205.1',presence_data='' >> where uuid='f506627c-ea28-481c-ba72-beaeebc16a6b'; >> 2012-01-06 08:33:12.486159 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC >> abort frame received (ZT_EVENT_ABORT) >> 2012-01-06 08:33:22.591484 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> insert into sip_dialogs >> (call_id,uuid,sip_to_user,sip_to_host,sip_to_tag,sip_from_user,sip_from_host,sip_from_tag,contact_user,contact_host,state,direction,user_agent,profile_name,hostname,contact,presence_id,presence_data,call_info,rcd) >> values('810ae706-b2b1-122f-aa88-6cf049ef3354','b925d777-a70c-4d71-a07b-bce5a804f48a','2002','192.168.205.1','e21ByaF046ppQ','2003','192.168.205.1','1436972916','2003','192.168.205.217','early','outbound','Yealink >> SIP-T22P 7.60.14.5','internal','srv-pok-phone','< >> sip:2003 at 192.168.205.217:5062>','2003 at 192.168.205.1','','',1325817146) >> 2012-01-06 08:33:41.089426 [ERR] ftmod_zt.c:1280 [s1c16][1:16] Dropping >> event 8 to be able to write data >> 2012-01-06 08:33:52.667878 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> delete from sip_dialogs where uuid='8ed4b985-1f1f-4901-938d-56619a42eb1c' >> 2012-01-06 08:34:02.189552 [ERR] ftmod_zt.c:1104 [s1c16][1:16] HDLC >> abort frame received (ZT_EVENT_ABORT) >> 2012-01-06 08:34:22.607797 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> delete from sip_authentication where expires > 0 and expires <= >> 1325817161 and hostname='srv-pok-phone.lpurs.argos-group.ru' >> 2012-01-06 08:34:52.460796 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> insert into sip_authentication (nonce,expires,profile_name,hostname, >> last_nc) values('9643f3a4-acc0-41cf-8a08-ca9dd791dbbb', 1325817222, >> 'internal', 'srv-pok-phone', 0) >> 2012-01-06 08:42:54.233881 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> insert into sip_authentication (nonce,expires,profile_name,hostname, >> last_nc) values('4611d24f-cfa0-47d4-a8d3-5fdce08ed85a', 1325817734, >> 'internal', 'srv-pok-phone', 0) >> 2012-01-06 08:42:54.884449 [CRIT] ftdm_io.c:5467 [s1c9][1:9] Forcing >> hangup since the user did not confirmed our hangup after 3000ms >> 2012-01-06 08:43:06.435420 [ERR] ftmod_libpri.c:132 XXX Progress message >> requested but no information is provided >> 2012-01-06 08:43:06.813352 [ERR] ftmod_libpri.c:132 Received unsolicited >> status: Mandatory information element is missing >> 2012-01-06 08:43:24.297760 [ERR] switch_core_sqldb.c:481 SQL ERR >> [database is locked] >> delete from sip_dialogs where uuid='563a394c-5c82-4381-868d-35e344f3ccd9' >> >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/46016a05/attachment.html From curriegrad2004 at gmail.com Fri Jan 6 20:43:11 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 6 Jan 2012 09:43:11 -0800 Subject: [Freeswitch-users] A New Config Set aimed for New People to FreeSWITCH In-Reply-To: References: Message-ID: The original github repository has been moved to FreeSWITCH-sample-conf report under my account. That repo's now public and again, feedback is greatly appreciated. On 2012-01-04 7:17 AM, "Avi Marcus" wrote: > It tickled me to see a log of every call in fusionpbx as soon as I hung > up... CDRs are pretty important. I think. > > -Avi > > > On Wed, Jan 4, 2012 at 5:13 PM, curriegrad2004 wrote: > >> I personally don't think that a new user should be exposed to CDRs just >> yet. They should have more time to know what FreeSWITCH is all about until >> they can move on. I might need somebody to remind me to convert all of the >> CRLF endings to LF when I get back from school >> On 2012-01-04 5:46 AM, "Avi Marcus" wrote: >> >>> 1) Breaking up the default dialplan into the pieces make it easier to >>> get a handle on. >>> I wonder if you should just include all the others as a .noload in a >>> folder named "advanced" or something. >>> >>> 2) I don't see any CDR modules in the autoload configs. cdr_csv for >>> basic and/or xml_cdr to store the calls with all the variables. It's >>> particularly helpful for when you don't know the variables that you can >>> just go and see all that are there, even after the call. >>> >>> 3) I wrote up a little bit about the regex for the incoming extension >>> (first actual dialplan they might see?).. >>> >>> https://github.com/avimar/Freeswitch_slim_conf/commit/91c49d7cf31f3c9a569b0258726d8b6511e468fb >>> Not sure it's worth including, though. >>> >>> 4) Q: Why do your's and the default configs have: >>> >>> >>> Wouldn't this work just as well without the added loopback channel? Why >>> isn't this the standard? >>> >>> >>> -Avi Marcus >>> >>> >>> On Wed, Jan 4, 2012 at 8:12 AM, Gabriel Gunderson wrote: >>> >>>> On Tue, Jan 3, 2012 at 10:03 PM, curriegrad2004 >>>> wrote: >>>> > It's been done and I have it uploaded on my github. Testing and >>>> > feedback is greatly appreciated: >>>> >>>> Love it. >>>> >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/03964d4b/attachment-0001.html From chad at apartmentlines.com Fri Jan 6 21:04:11 2012 From: chad at apartmentlines.com (Chad Phillips) Date: Fri, 6 Jan 2012 10:04:11 -0800 Subject: [Freeswitch-users] LUA getVariable In-Reply-To: References: Message-ID: this might do the trick: http://wiki.freeswitch.org/wiki/Variable_session_in_hangup_hook Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, January 6, 2012 at 4:46 AM, Juan Antonio Iba?ez Santorum wrote: > Hello, > > I have one LUA script that make calls to one custom FS app. That app sets up some session channel variables and makes an outgoing call: > > > if(caller_channel = switch_core_session_get_channel(session)){ > > switch_channel_set_variable(caller_channel, "var", "val") > > > > } > > > > > > > If the caller hangs up the call, the app returns and I try to get the val of var as: > > > session:getVariable("var") > > but a nil value is returned. Do you know why? > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/a7178252/attachment.html From freeswitchlist at gmail.com Fri Jan 6 18:42:01 2012 From: freeswitchlist at gmail.com (bob smith) Date: Fri, 6 Jan 2012 10:42:01 -0500 Subject: [Freeswitch-users] dialer Message-ID: Hi guys, we are new to this list. Can someone make a recommendation for a good dialer available for freeswitch. RS -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/6794ce5c/attachment.html From basit.engg at gmail.com Fri Jan 6 22:10:21 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Sat, 7 Jan 2012 00:10:21 +0500 Subject: [Freeswitch-users] dialer In-Reply-To: References: Message-ID: here u go. http://www.newfies-dialer.org/tag/freeswitch/ On Fri, Jan 6, 2012 at 8:42 PM, bob smith wrote: > Hi guys, we are new to this list. > > Can someone make a recommendation for a good dialer available for > freeswitch. > > RS > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/328edd1f/attachment.html From juanito1982 at gmail.com Fri Jan 6 23:40:43 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Fri, 6 Jan 2012 21:40:43 +0100 Subject: [Freeswitch-users] Bridge answered Message-ID: Hello, Is there any way to get if an originate-bridge has been answered and duration doing the originate-bridge from an already answered channel? Could be considered hangup cause NORMAL_CLEARING=ANSWER? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/34b441db/attachment.html From justlikeef at gmail.com Fri Jan 6 23:49:44 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Fri, 6 Jan 2012 15:49:44 -0500 Subject: [Freeswitch-users] Getting 405 Method Not Allowed from Grandstream presence registration Message-ID: <201201061549.44963.justlikeef@gmail.com> recv 598 bytes from udp/[10.0.1.67]:5060 at 20:48:20.589315: ------------------------------------------------------------------------ SUBSCRIBE sip:102 at usgagasrv1.pts-hvac.com SIP/2.0 Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK1880398731;rport From: ;tag=1864547776 To: Call-ID: 1978847509-5060-4 at BA.A.B.GH CSeq: 30500 SUBSCRIBE Contact: X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2100 1.0.1.108 Expires: 3630 Supported: replaces, path, timer Event: reg Accept: application/reginfo+xml Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 ------------------------------------------------------------------------ send 462 bytes to udp/[10.0.1.67]:5060 at 20:48:20.590001: ------------------------------------------------------------------------ SIP/2.0 405 Method Not Allowed Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK1880398731;rport=5060 From: ;tag=1864547776 To: ;tag=XDmee5t4Nj3ae Call-ID: 1978847509-5060-4 at BA.A.B.GH CSeq: 30500 SUBSCRIBE User-Agent: Configured by 2600hz! Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/022d7155/attachment.html From kris at kriskinc.com Sat Jan 7 02:51:07 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 6 Jan 2012 18:51:07 -0500 Subject: [Freeswitch-users] Getting 405 Method Not Allowed from Grandstream presence registration In-Reply-To: <201201061549.44963.justlikeef@gmail.com> References: <201201061549.44963.justlikeef@gmail.com> Message-ID: Check this: http://wiki.freeswitch.org/wiki/Shared_Line_Appearance#FreeSWITCH_Configuration Pay special attention to for your internal profile. On Fri, Jan 6, 2012 at 3:49 PM, Rob Hutton wrote: > recv 598 bytes from udp/[10.0.1.67]:5060 at 20:48:20.589315: > > ------------------------------------------------------------------------ > > SUBSCRIBE sip:102 at usgagasrv1.pts-hvac.com SIP/2.0 > > Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK1880398731;rport > > From: ;tag=1864547776 > > To: > > Call-ID: 1978847509-5060-4 at BA.A.B.GH > > CSeq: 30500 SUBSCRIBE > > Contact: > > X-Grandstream-PBX: true > > Max-Forwards: 70 > > User-Agent: Grandstream GXP2100 1.0.1.108 > > Expires: 3630 > > Supported: replaces, path, timer > > Event: reg > > Accept: application/reginfo+xml > > Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, > UPDATE, MESSAGE > > Content-Length: 0 > > ------------------------------------------------------------------------ > > send 462 bytes to udp/[10.0.1.67]:5060 at 20:48:20.590001: > > ------------------------------------------------------------------------ > > SIP/2.0 405 Method Not Allowed > > Via: SIP/2.0/UDP 10.0.1.67:5060;branch=z9hG4bK1880398731;rport=5060 > > From: ;tag=1864547776 > > To: ;tag=XDmee5t4Nj3ae > > Call-ID: 1978847509-5060-4 at BA.A.B.GH > > CSeq: 30500 SUBSCRIBE > > User-Agent: Configured by 2600hz! > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY > > Supported: precondition, path, replaces > > Content-Length: 0 > > ------------------------------------------------------------------------ > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From engelster at gmail.com Sat Jan 7 01:45:45 2012 From: engelster at gmail.com (Der Engel) Date: Fri, 6 Jan 2012 17:45:45 -0500 Subject: [Freeswitch-users] Javascript vs Lua Message-ID: Hello, Just begun to play with FreeSWITCH and was wondering what scripting language to use to extend its functionality, by reading the wiki I understand that the recommended one is Lua but I don't know the language yet, I do know Javascript, so my question is, will FreeSWTCH support Javascript as a first class citizen in the future (by looking at the source the spidermonkey lib hasn't been update in years) or should I just get started in learning Lua for future support/conformance sake? Thanks, Der From olimonkey at gmail.com Sat Jan 7 04:28:56 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Sat, 7 Jan 2012 09:28:56 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Message-ID: No, at the time I was no longer using "ignore_early_media= true", because initially it didn't work. I was actually thinking of putting this property back in again, thanks a lot! I'll try it first thing Monday. With regard to EXECUTE log line, either I don't have the right level of logging turned on or something else is going on, because I've never seen any such log entries before with my managed application; yet my IVR app definitely does get executed. I don't think I have the DEBUG logging output level turned on so that could explain it... Thanks all, Oliver On Fri, Jan 6, 2012 at 11:21 PM, Peter Olsson wrote: > Are you still using ignore_early_media=true - this must be set for this to work correctly. > > You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. > > Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" > > Park application is only executed after the call was answered. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] > Skickat: den 6 januari 2012 12:04 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR > > Because I'm using an FXO card with voice, I added something to my > CISCO conf. Many others had the same thing: > > > voice-port 0/3/0 > ? ... > ? supervisory disconnect dualtone mid-call > ? supervisory answer dualtone ? ?<---- ADDED THIS ONE > ? ... > > > > Once I added this, the FS output now just showed the following while > the phone was ringing: > > 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel > sofia/internal/109212xxxx at 192.168.x.x > [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] > 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 > sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound > Call" <109212xxxx> > 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer > sofia/internal/109212xxxx at 192.168.x.x! > > > Where as previous it would show the above and also show the following: > > 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 > sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" > <0000000000> to "Outbound Call" <109212xxxx> > 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 > sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" > <1092122856> > 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel > [sofia/internal/109212xxxx at 192.168.x.x] has been answered > > > > BUT, the IVR still started playing even before I pick up the phone. > Hmmmm.....so why is FS still starting the managed application when the > call has not been answered yet. Are we all sure that the managed > application should not be executed until the call "has been answered" > shows up in the log file? > > > Will have to keep testing on monday as I don't have access to the > CISCO from where i am now. I'll have to see whether the CISCO changes > had any impact on the times at which the SIP messages are sent back > and forth. Especially the 200 OK message. > > > Thanks again for help, maybe getting somewhere now...... > > Oliver > > > > > On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson > wrote: >> If it sends 200 OK right after 183, this IS the problem. >> >> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. >> >> Seems to me there is something broken in the Cisco. >> >> /Peter >> >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >> Skickat: den 6 januari 2012 06:55 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> >> I've tried looking at disable-early-media configuration command, but >> that didn't work and I doubt that has anything to do with the CISCO >> sending a 200 OK right after a 183 SESSION PROGRESS. >> >> >> >> >> On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >>> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >>> is usually RINGING (generate ringback locally) while a 183 has media... aka >>> early media and usually provides ringback inband. >>> >>> /b >>> >>> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>> >>> Shouldn't there be a ?180 RINGING ?somewhere in there? >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >>> >>> I just noticed something else, if I don't pick up the phone at all. >>> >>> The IVR just keeps playing until the menu timeout kicks in. >>> >>> >>> So here is a CISCO SIP log: >>> >>> http://pastebin.com/Y9sYkuxi >>> >>> >>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>> >>> I hope the CISCO log is readable, it's a bit long because I just did >>> >>> "debug ccsip all". >>> >>> >>> >>> >>> In this test I didn't bother picking up the phone at all, but I can >>> >>> see that FS answered anyway and the IVR kept playing until it timed >>> >>> out. >>> >>> I'm not an expert, but here is what I picked out of it: >>> >>> >>> At 00:08:10 we get a >>> >>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>> >>> >>> the further down at the same timestamp we get >>> >>> Sent: "SIP/2.0 100 Trying" >>> >>> >>> At 00:08:13 we get a >>> >>> Sent: "SIP/2.0 183 Session Progress" >>> >>> >>> At 00:18:13 we get a >>> >>> Sent: "SIP/2.0 200 OK" >>> >>> >>> Then at the same timestamp we get: >>> >>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>> >>> >>> >>> >>> Once the IVR times out at 00:09:16 we get >>> >>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>> >>> >>> And then the reply right after >>> >>> Sent: "SIP/2.0 200 OK" >>> >>> >>> >>> >>> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>> >>> after the "INVITE" is received. >>> >>> >>> >>> >>> The part that is beyond my field of expertise so far is WHY? >>> >>> >>> >>> >>> Thanks, >>> >>> >>> >>> Oliver >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>> >>> By the way: >>> >>> >>> I tried {ignore_early_media=true} as well, but as I think we >>> >>> determined, my problem is probably with the CISCO telling FS that the >>> >>> call has been answered when really it hasn't yet. >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>> >>> Thanks for the help so far. >>> >>> >>> >>> Here is a pastebin of FreeSWITCH output: >>> >>> http://pastebin.com/i6Qgc7ws >>> >>> >>> Notice how the "has been answered" log message comes immediately >>> >>> (within a few milliseconds) after the call was originated. I think >>> >>> this would suggest that the CISCO is immediately sending a 200 OK, as >>> >>> you suggested. I also turned on CISCO debugging, but I'm just trying >>> >>> to figure out how to get the information regarding SIP messages back >>> >>> to Freeswitch. I'll run the test again and see if I can get some >>> >>> useful CISCO debug. >>> >>> >>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>> >>> SIP debugging? >>> >>> >>> >>> Thanks! >>> >>> >>> >>> >>> >>> 2012/1/6 Gustavo M?rsico : >>> >>> I think I've a similar problem related to callcenter app. When I made an >>> originate like this: >>> >>> >>> originate loopback/2500/default/XML &bridge(user/2001) >>> >>> >>> 2500 is an extension that leads to a callcenter application. In this case, >>> we dial first to the queue and when an agent answered we call to the >>> customer. As far as I know >>> >>> When the A-leg reaches to the queue, without selecting an agent, the call is >>> automatically sent to the B-leg. As far as I see, there is a pre-answer >>> method that fs needs to send the media to A-leg. >>> >>> In order to try to avoid this, I tried using ignore_early_media=true as part >>> of the originate in A-leg and/or B-leg, with no luck. >>> >>> >>> originate {ignore_early_media=true}loopback/2500/default/XML >>> &bridge({ignore_early_media=true}user/2001) >>> >>> >>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>> destination_number(2500) =~ /^(2500)$/ break=on-false >>> >>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>> >>> Dialplan: loopback/2500-b Action callcenter(click2call) >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >>> loopback/2500-b [BREAK] >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> CHANNEL KILL >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>> (loopback/2500-b) State ROUTING going to sleep >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>> (loopback/2500-b) Running State Change CS_EXECUTE >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>> (loopback/2500-b) State EXECUTE >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>> CHANNEL EXECUTE >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>> loopback/2500-b Standard EXECUTE >>> >>> EXECUTE loopback/2500-b set(open=true) >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>> [open]=[true] >>> >>> EXECUTE loopback/2500-b >>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>> >>> EXECUTE loopback/2500-b >>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>> >>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>> >>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>> [ignore_early_media]=[true] >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >>> callcenter Requires media! pre_answering channel loopback/2500-b >>> >>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>> loopback/2500-a! >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >>> Callstate Change RINGING -> EARLY >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>> loopback/2500-b [BREAK] >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> CHANNEL KILL >>> >>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >>> loopback/2500-b! >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >>> Callstate Change RINGING -> EARLY >>> >>> EXECUTE loopback/2500-b callcenter(click2call) >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >>> Callstate Change EARLY -> ACTIVE >>> >>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>> [loopback/2500-a] has been answered >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>> loopback/2500-b [BREAK] >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>> CHANNEL KILL >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >>> Resulted in Success: [loopback/2500-a] >>> >>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >>> Callstate Change EARLY -> ACTIVE >>> >>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >>> Flipping CID from "" <0000000000> to "Outbound Call" >>> >>> >>> >>> >>> >>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>> >>> >>> Also, maybe I should be doing something like this: >>> >>> >>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>> >>> >>> instead of: >>> >>> >>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>> >>> >>> >>> but, I don't really have the CISCO configured as a gateway, nor do I >>> >>> know how really...probably not on the right track there. >>> >>> >>> >>> >>> >>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>> >>> *bump* >>> >>> >>> >>> So I think maybe the way I'm doing the originate is the problem? In my >>> >>> call string I'm creating a connection directly from the CISCO >>> >>> (192.168.x.x) to the managed application, which may be why it starts >>> >>> playing straight away? >>> >>> >>> Maybe I should be originating a call first and then only once I know >>> >>> the other side has picked up will I bridge the call to the IVR managed >>> >>> application. >>> >>> >>> Problem is I dunno how to tell whether the other person has picked up >>> >>> (or even if the cisco is going to tell me) and I don't know how to do >>> >>> things to a call once it has been established. >>> >>> >>> >>> I'm currently reading the Dialplan wiki page, hoping to get something >>> >>> out of it there. >>> >>> >>> >>> Cheers >>> >>> >>> Oliver >>> >>> >>> >>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>> >>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>> >>> and connecting through a CISCO 2811. Most things now work quite well, >>> >>> but I am having a few issues with the way the system answers calls (or >>> >>> doesn't answer calls...). >>> >>> >>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>> >>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>> >>> which is then connected to a POTS phone line. >>> >>> >>> >>> Take the following scenario: >>> >>> >>> 1. Managed .NET application creates a call string and uses ESL to talk >>> >>> to freeswitch and originate a call: >>> >>> >>> string callstring = >>> >>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>> >>> '&managed(ivrAppName)'"; >>> >>> eslConnection.API("originate", callstring); >>> >>> >>> where 192.168.x.x is the CISCO IP. >>> >>> >>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>> >>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>> >>> number (091234567) to make the call. >>> >>> >>> 3. My phone rings, I pick up and I can hear my IVR playing. >>> >>> >>> >>> >>> These are my current problems: >>> >>> >>> - IVR starts playing before I even pick up the phone. This means that >>> >>> if the system calls a mobile phone and the person doesn't pick up, the >>> >>> IVR will start playing and eventually the mobile phone will divert to >>> >>> voice mail. Obviously I then get a missed call and an sms saying I >>> >>> have a new voice mail, which is annoying. Instead I would like it to >>> >>> KNOW that no one has picked up, but I don't know how to do this. >>> >>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>> >>> has not yet been answered. For some reason however as soon as the >>> >>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>> >>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>> >>> doing originate the wrong way or something ... >>> >>> >>> - The phone only rings for about 10 seconds before hanging up. I've >>> >>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>> >>> CISCO "ring number". Nothing works, my phone still only rings for >>> >>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>> >>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>> >>> starts playing even if no one answers the phone. >>> >>> >>> >>> >>> >>> >>> CISCO Config for relevant FXO port: >>> >>> >>> voice service voip >>> >>> ?allow-connections h323 to h323 >>> >>> ?allow-connections h323 to sip >>> >>> ?allow-connections sip to h323 >>> >>> ?allow-connections sip to sip >>> >>> ?no supplementary-service h450.2 >>> >>> ?no supplementary-service h450.3 >>> >>> ?supplementary-service h450.12 >>> >>> ?no supplementary-service sip moved-temporarily >>> >>> ?no supplementary-service sip refer >>> >>> ?fax protocol cisco >>> >>> ?sip >>> >>> ?registrar server expires max 3600 min 3600 >>> >>> ?no update-callerid >>> >>> ?no call service stop >>> >>> >>> voice-port 0/3/2 >>> >>> ?output attenuation -3 >>> >>> ?no comfort-noise >>> >>> ?cptone AU >>> >>> ?impedance complex1 >>> >>> ?caller-id enable >>> >>> ! >>> >>> dial-peer voice 100 pots >>> >>> ?preference 1 >>> >>> ?destination-pattern 1T >>> >>> ?port 0/3/2 >>> >>> ! >>> >>> >>> >>> >>> Many Thanks, >>> >>> >>> Oliver >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> -- >>> Brian West >>> FreeSWITCH Solutions, LLC >>> Phone: +1 (918) 420-9266 >>> Fax: ? +1 (918) 420-9267 >>> brian at freeswitch.org >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4f06d49b32762089563979! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From th982a at googlemail.com Sat Jan 7 04:51:06 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Sat, 07 Jan 2012 02:51:06 +0100 Subject: [Freeswitch-users] mod_sofia not binding to port 5060 In-Reply-To: <4F030729.1070208@gmail.com> References: <4F01E360.9010007@gmail.com> <4F025B10.4030705@googlemail.com> <4F030729.1070208@gmail.com> Message-ID: <4F07A50A.3000608@googlemail.com> Hi Brandon! I thought you had taken allready a look in the init script?! Also, look as what user freeswitch is being started. Don't know, perhaps that might be the reason either. Tamer Am 03.01.2012 14:48, schrieb Brandon McGinty: > Thanks for these. However, I checked all three of these resources. > For some reason, this issue is only occuring when the init.d script for > debian, is used to start freeswitch; from the command line, the issue > goes away completely. > All that is running is the -nc switch, so I'm not sure what could be > causing the issue. > Thanks again. > > Brandon McGinty-Carroll > > > On 1/2/2012 8:34 PM, Tamer Higazi wrote: >> Look at: >> >> conf/autoload_configs/switch.conf.xml >> >> read this: >> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files >> >> >> look at "sip-port" >> >> >> I am tired, I am going to bed. >> >> >> Tamer >> >> >> Am 02.01.2012 18:03, schrieb Brandon McGinty: >>> Good afternoon. >>> We've got a freeswitch machine, no external SIP gateways for outside >>> calls, eight extensions, two conferences, and a completely public IP for >>> the machine in question (no NAT weirdness), running on Debian 6 (weezy/sid). >>> I woke up this morning, to find that sip_external_port 5060 was not >>> listening, though it was enabled in config/vars.xml. >>> The internal port, 5080, is listening, and I have folks confirming that >>> they can speak using that port. >>> I've set netcat to listen via 5080, so I know it isn't firewalled, or >>> not at least ina conventional manner. >>> Any help you all can provide, things I can check, would be greatly >>> appreciated. >>> >>> Sincerely, >>> Brandon McGinty-Carroll >>> From notlikeme75 at yahoo.com Sat Jan 7 06:45:15 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 6 Jan 2012 19:45:15 -0800 (PST) Subject: [Freeswitch-users] 407 undefined? sip port 5060 Message-ID: <1325907915.75948.YahooMailNeo@web65315.mail.ac2.yahoo.com> I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response:? 2012-01-06 21:52:46.0 000000000?? 614??????? ? ???6174530953?(????????? DID-IN:RNK) (H1 usinteractive-nextdi)?? 407?????? undefined? I have the following gateway setup in?C:\Program Files\FreeSWITCH\conf\sip_profiles\external ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/3db214ea/attachment.html From valery.kalinin at gmail.com Sat Jan 7 07:00:16 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Sat, 7 Jan 2012 10:00:16 +0600 Subject: [Freeswitch-users] Why database is locked? Message-ID: > Are you running the latest version of FS? FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > Also are you doing anything externally to mess with the sql db files? No. > Another possibility is a slow filesystem, we used to have people get this problem on suse linux. CentOS release 5.5 (Final) > You could try running your db dir in a ramdisk. These problems appeared after the upgrade version of FS. Thank you in advance... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/05672bd9/attachment.html From krice at freeswitch.org Sat Jan 7 07:14:23 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 06 Jan 2012 22:14:23 -0600 Subject: [Freeswitch-users] Why database is locked? In-Reply-To: Message-ID: Blow away all the files in freeswitch/db and restart freeswitch Save the voicemail.db file if you don?t wanna lose all the voicemails and give that a try. That will force a reset of the all the databases and should clear any errors... On 1/6/12 10:00 PM, "Valery Kalinin" wrote: >> > Are you running the latest version of FS? > > FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > >> > Also are you doing anything externally to mess with the sql db files? > > No. > >> > Another possibility is a slow filesystem, we used to have people get this > problem on suse linux. > > CentOS release 5.5 (Final) > >> > You could try running your db dir in a ramdisk. > > These problems appeared after the upgrade version of FS. > > > Thank you in advance... > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/dd32977e/attachment.html From vetali100 at gmail.com Sat Jan 7 07:54:00 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Fri, 6 Jan 2012 20:54:00 -0800 Subject: [Freeswitch-users] 407 undefined? sip port 5060 In-Reply-To: <1325907915.75948.YahooMailNeo@web65315.mail.ac2.yahoo.com> References: <1325907915.75948.YahooMailNeo@web65315.mail.ac2.yahoo.com> Message-ID: Ask your provider to send sip packets to 5080, instead of 5060. Every good provider should allow you to select the port YOU want, and not THEY want. If it will not help, then you need to make external profile listen to 5060, instead of 5080. So, you will be sending and receiving sip messages between you and provider, using port 5060 (which is usually used only for your internal profile and your clients). To change this, you need to modify conf/sip_profiles/external.xml replace to "5060" In this case, you will have to reconfigure internal profile to be on a different port, say 5062, and ask your clients to add :5062 to the sip server address when they configure their softphone/sip adapter. Maybe another option would be to consider provider as one of your internal clients, and allow password-less communication from his IP, but this is way too crazy to consider as a good option. Vitalie 2012/1/6 Rodney > I currently have external profile ipkall working on port 5080 but now i am > testing RNK carrier services and they are sending to me on port 5060. I do > not notice anything on the console but they are saying they are sending to > my server and getting this response: > > 2012-01-06 21:52:46.0 000000000 614??????? 6174530953 ( > DID-IN:RNK) (H1 usinteractive-nextdi) 407 undefined > > > > I have the following gateway setup in C:\Program > Files\FreeSWITCH\conf\sip_profiles\external > > > > > > > > > > > > > > > It is exactly the same as my IPkall except for the gateway name and proxy > port. How do I make sure I can accept on this port. Is there somewhere else > I need to be adding this 5060 port information. > > thank you. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/bcb77b2f/attachment-0001.html From msc at freeswitch.org Sat Jan 7 09:11:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Jan 2012 22:11:23 -0800 Subject: [Freeswitch-users] Global Variable Substitution In-Reply-To: References: Message-ID: Gold star for you! Thanks for paying the "voluntary" wiki tax. :P -MC On Thu, Jan 5, 2012 at 5:32 PM, Brian Wiese < brian.wiese.freeswitch at gmail.com> wrote: > Wikified: > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables#Search_and_replace > > :) > > On Thu, Jan 5, 2012 at 5:36 PM, Michael Collins > wrote: > > Nicely done! Be sure to wikify this one. If you have any questions about > > updating the wiki just email me off list. > > > > -MC > > > > > > On Thu, Jan 5, 2012 at 1:19 PM, Brian Wiese > > wrote: > >> > >> Michael: > >> > >> Here's what I came up with, and it works really well! I can embed > >> this in a dialplan or use it at the CLI: > >> > >> lua ~stream:write(tostring(string.gsub("string_to_search", > >> "string_to_find", "replacement_string"))) > >> > >> It's awesome! > >> > >> Thanks for your help! > >> > >> ~Brian > >> > >> On Tue, Jan 3, 2012 at 6:16 PM, Brian Wiese > >> wrote: > >> > Michael: > >> > > >> > I do think that mod_xml_curl is a little more than what I need here. > >> > I'll try a Lua script and see where I get. I'll try to call a Lua > >> > script and return the bridge string back to the dial plan. > >> > > >> > I'll let you know how my scripting fu works... > >> > > >> > ~Brian > >> > > >> > On Tue, Jan 3, 2012 at 6:10 PM, Michael Collins > >> > wrote: > >> >> This kinda sounds like a problem in need of mod_xml_curl. If that > seems > >> >> like > >> >> too much of a hassle then I would fall back to a mod_lua or mod_perl > >> >> script > >> >> to do the regex stuff. How is your scripting fu? > >> >> > >> >> -MC > >> >> > >> >> > >> >> On Tue, Jan 3, 2012 at 3:54 PM, Brian Wiese > >> >> wrote: > >> >>> > >> >>> Michael: > >> >>> > >> >>> I sure can! > >> >>> > >> >>> What I want to do is create template configurations that I can > deploy > >> >>> to multiple servers. Part of the requirement of incoming calls is > to > >> >>> configure which extension(s) ring immediately, 6-second delay, and > >> >>> 12-second delay. I have created groups for these extensions, and by > >> >>> using group_call I can get the full dial string for each group... > >> >>> perfect! Now, I just need a way to delay some of these extensions > by > >> >>> 6 or 12 seconds. I ultimately want to inject leg variables into the > >> >>> dial string for each extension, so when the group_call is expanded > >> >>> each of them expand with the extra leg variable I define. > >> >>> > >> >>> Hope that helps... clear as mud? :) > >> >>> > >> >>> ~Brian > >> >>> > >> >>> On Tue, Jan 3, 2012 at 4:30 PM, Michael Collins > > >> >>> wrote: > >> >>> > Can you expand upon this question a bit? I'm curious if there's a > >> >>> > less > >> >>> > hackish way of doing what you want to do. Under what circumstances > >> >>> > do > >> >>> > you > >> >>> > need to add the leg variables? Also, can you give us the big > >> >>> > picture? > >> >>> > What's > >> >>> > the problem you're solving? > >> >>> > > >> >>> > -MC > >> >>> > > >> >>> > On Mon, Jan 2, 2012 at 8:56 AM, Brian Wiese > >> >>> > wrote: > >> >>> >> > >> >>> >> Hi Everyone. > >> >>> >> > >> >>> >> I thought I read somewhere that this was possible, but I can't > find > >> >>> >> it > >> >>> >> now... > >> >>> >> > >> >>> >> I need a way to find-and-replace within a variable. So, for > >> >>> >> example, > >> >>> >> I want to take variables that have values like this: > >> >>> >> > >> >>> >> 123abc123abc > >> >>> >> abc123abc123abc > >> >>> >> > >> >>> >> ...and do a find/replace of the "abc" with "xyz" so the variables > >> >>> >> would now return: > >> >>> >> > >> >>> >> 123xyz123xyz > >> >>> >> xyz123xyz123xyz > >> >>> >> > >> >>> >> The use case I've run into is that I need to add leg variables to > >> >>> >> group_call. In my case, group_call can return any number of > >> >>> >> members, > >> >>> >> so I figured I would just replace the first "[" with > >> >>> >> "[variable-I-want-to-set=...". > >> >>> >> > >> >>> >> Thanks for the help! > >> >>> >> > >> >>> >> ~Brian > >> >>> >> > >> >>> >> > >> >>> >> > >> >>> >> > _________________________________________________________________________ > >> >>> >> Professional FreeSWITCH Consulting Services: > >> >>> >> consulting at freeswitch.org > >> >>> >> http://www.freeswitchsolutions.com > >> >>> >> > >> >>> >> > >> >>> >> > >> >>> >> > >> >>> >> Official FreeSWITCH Sites > >> >>> >> http://www.freeswitch.org > >> >>> >> http://wiki.freeswitch.org > >> >>> >> http://www.cluecon.com > >> >>> >> > >> >>> >> FreeSWITCH-users mailing list > >> >>> >> FreeSWITCH-users at lists.freeswitch.org > >> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> >> > >> >>> >> > >> >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> >> http://www.freeswitch.org > >> >>> > > >> >>> > > >> >>> > > >> >>> > > >> >>> > > >> >>> > > _________________________________________________________________________ > >> >>> > Professional FreeSWITCH Consulting Services: > >> >>> > consulting at freeswitch.org > >> >>> > http://www.freeswitchsolutions.com > >> >>> > > >> >>> > > >> >>> > > >> >>> > > >> >>> > Official FreeSWITCH Sites > >> >>> > http://www.freeswitch.org > >> >>> > http://wiki.freeswitch.org > >> >>> > http://www.cluecon.com > >> >>> > > >> >>> > FreeSWITCH-users mailing list > >> >>> > FreeSWITCH-users at lists.freeswitch.org > >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > > >> >>> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> > http://www.freeswitch.org > >> >>> > > >> >>> > >> >>> > >> >>> > _________________________________________________________________________ > >> >>> Professional FreeSWITCH Consulting Services: > >> >>> consulting at freeswitch.org > >> >>> http://www.freeswitchsolutions.com > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> Official FreeSWITCH Sites > >> >>> http://www.freeswitch.org > >> >>> http://wiki.freeswitch.org > >> >>> http://www.cluecon.com > >> >>> > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/bc67479a/attachment-0001.html From msc at freeswitch.org Sat Jan 7 09:17:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Jan 2012 22:17:06 -0800 Subject: [Freeswitch-users] in ivr menu In-Reply-To: References: Message-ID: Use an absolute path value for the greeting sounds. When you use a relative path (i.e. you don't start with a forward slash) then it will assume that you want to use a sound file somewhere in the sounds directory structure. Relative: greet-short="ivr/ivr-menu.wav" Absolute: greet-short="/full/path/to/file/ivr-menu.wav" -MC P.S. - this didn't seem to be mentioned on the XML IVR menu page ( http://wiki.freeswitch.org/wiki/IVR_Menu) so if you find this information useful would you mind adding this tidbit there? On Fri, Jan 6, 2012 at 2:28 AM, amit nakum wrote: > Hi, > > I want to play wav files from the different path. As now it is playing the > wav files from: > "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/" > path. > > Please tell me how to play wav files from other path in ivr menu using > greet-short parameter . > > Thanks in advance.... > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/42e14782/attachment.html From msc at freeswitch.org Sat Jan 7 09:25:17 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 6 Jan 2012 22:25:17 -0800 Subject: [Freeswitch-users] Javascript vs Lua In-Reply-To: References: Message-ID: Use Lua. It's easy to learn, it's lightweight, and it scales WAY WAY better than JavaScript/Spidermonkey. In the amount of time it will take you to debug an esoteric js error you can learn Lua. In fact, chapter 7 of the FS book has a nice, gentle intro to Lua. (Full disclosure: I authored chapter 7 of the FreeSWITCH book. I am unabashedly biased in favor of Lua.) -MC Amazon link: http://amzn.to/cNym68 On Fri, Jan 6, 2012 at 2:45 PM, Der Engel wrote: > Hello, > > Just begun to play with FreeSWITCH and was wondering what scripting > language to use to extend its functionality, by reading the wiki I > understand that the recommended one is Lua but I don't know the > language yet, I do know Javascript, so my question is, will FreeSWTCH > support Javascript as a first class citizen in the future (by looking > at the source the spidermonkey lib hasn't been update in years) or > should I just get started in learning Lua for future > support/conformance sake? > > Thanks, > Der > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120106/7ccd0f98/attachment.html From anita.hall at simmortel.com Sat Jan 7 09:32:27 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Sat, 7 Jan 2012 12:02:27 +0530 Subject: [Freeswitch-users] dialer In-Reply-To: References: Message-ID: Check this out http://www.hellohunter.com/dialer_api.php And, if you are more technically inclined, this http://www.sinapticode.com/freeswitch-tips-creating-dialer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/4bb4bf63/attachment.html From anita.hall at simmortel.com Sat Jan 7 11:32:44 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Sat, 7 Jan 2012 14:02:44 +0530 Subject: [Freeswitch-users] Max calls Limit? Message-ID: Is there any config variable somewhere which is limiting the max call limit on FS to 1000? Set-up: Taking FS for max call load on my machine. User freeswitch is calling internal gateway and playing demo-ivr User freeswitch defined in vars.xml Internal Gateway defined in directory/default/freeswitch.xml After 1000 calls, it is giving the same error -ERR DESTINATION_OUT_OF_ORDER Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by the time 1000 calls are being done, but is it not a little strange that every time the resources get exhausted at precisely 1000 calls! regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/74667b0b/attachment.html From rhow at exemail.com.au Sat Jan 7 13:25:29 2012 From: rhow at exemail.com.au (Ryan How) Date: Sat, 07 Jan 2012 18:25:29 +0800 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it Message-ID: <4F081D99.7000406@exemail.com.au> Hi, I've just installed freeswitch on windows using the binary install (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). Whenever a call is made the log fills up with about 42 "Invalid UTF-8 character to ampersand, skip it". I haven't changed any config from the default. Any pointers ? Thanks! Ryan From freeswitch at earthspike.net Sat Jan 7 14:34:17 2012 From: freeswitch at earthspike.net (John) Date: Sat, 07 Jan 2012 11:34:17 +0000 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <4F081D99.7000406@exemail.com.au> References: <4F081D99.7000406@exemail.com.au> Message-ID: <4F082DB9.60509@earthspike.net> On 07/01/12 10:25, Ryan How wrote: > Hi, > > I've just installed freeswitch on windows using the binary install > (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). > Whenever a call is made the log fills up with about 42 "Invalid UTF-8 > character to ampersand, skip it". I haven't changed any config from the > default. Any pointers ? > > Thanks! > > Ryan > Me too. It's 42 lines and always follows a 'say': 2012-01-07 01:53:34.468472 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] I upgraded yesterday from git-9423216 2011-08-15 14-48-05 +0500 to git-1901a41 2012-01-06 11-17-30 -0600 and the only thing I did along the way was enable mod_shout which I reckon is probably irrelevant. John From juanito1982 at gmail.com Sat Jan 7 17:45:54 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 7 Jan 2012 15:45:54 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F0582AA.9060708@gmx.net> References: <4F0582AA.9060708@gmx.net> Message-ID: I have the same problem. An external gateway is returning 486 to FS but FS returns 480 to the caller... I am testing following dialplan: 2012/1/5 Peter P GMX > Hello, > > I have a strange phenomen: > > When a target UA is busy, it returns "486 Busy" to Freeswitch. But > Freeswitch then returns "480 Temporarily Unavailable" to the called party. > Where does this come from and how can I change this behaviour? > > See (anonymized) SIP trace with ngrep: > > UA to Freeswitch: > ======================== > U 2012/01/04 13:59:44.928775 :5060 -> :5080 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP > :5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. > From: "026xxxxxxxx" >;tag=py094Kv7vr03a. > To: > ;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. > Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. > CSeq: 22504928 INVITE. > User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). > Content-Length: 0. > > Freeswitch to Caller: > ======================== > U 2012/01/04 13:59:44.930387 :5060 -> :5060 > SIP/2.0 480 Temporarily Unavailable. > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-4896-2830DFA. > From: > ;user=phone>;tag=13517-HB-08a98588-2622da197. > To: ;user=phone>;tag=XQtc5US24QgDa. > Call-ID: 13517-SG-08a98587-0a352e121 at sip.provider.de. > CSeq: 134781549 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 > 13-52-28 -0600. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: precondition, path, replaces. > Allow-Events: talk, hold, refer. > Content-Length: 0. > P-Asserted-Identity: "069xxxxxxxx" >. > > Best regards > Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/a6bf95ef/attachment-0001.html From paul at cupis.co.uk Sat Jan 7 18:06:10 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 07 Jan 2012 15:06:10 +0000 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> Message-ID: <4F085F62.1000501@cupis.co.uk> On 07/01/12 14:45, Juan Antonio Iba?ez Santorum wrote: > I have the same problem. An external gateway is returning 486 to FS but FS > returns 480 to the caller... I am testing following dialplan: > > > > > > Sounds like a bad interaction with some continue_on_foo configuration. Does it help if you do: after the action-bridge? Regards, From anthony.minessale at gmail.com Sat Jan 7 18:44:06 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Jan 2012 09:44:06 -0600 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <4F081D99.7000406@exemail.com.au> References: <4F081D99.7000406@exemail.com.au> Message-ID: If you check the git log on the conf dir, copy the xml files that were changed recently by moc. There was a patch to enable utf8 support which exposed the invalid files. IIRC it was the de Lang templates. On Jan 7, 2012 4:27 AM, "Ryan How" wrote: > Hi, > > I've just installed freeswitch on windows using the binary install > (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). > Whenever a call is made the log fills up with about 42 "Invalid UTF-8 > character to ampersand, skip it". I haven't changed any config from the > default. Any pointers ? > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/60353cf3/attachment.html From ljjimenez at gmail.com Sat Jan 7 19:24:25 2012 From: ljjimenez at gmail.com (Luis Jimenez) Date: Sat, 7 Jan 2012 12:24:25 -0400 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: References: <4F081D99.7000406@exemail.com.au> Message-ID: Hello, i have the same problem, but i dont understand the solution, please advice me. Thanks in advance Luis Jimenez On Sat, Jan 7, 2012 at 11:44 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If you check the git log on the conf dir, copy the xml files that were > changed recently by moc. There was a patch to enable utf8 support which > exposed the invalid files. IIRC it was the de Lang templates. > On Jan 7, 2012 4:27 AM, "Ryan How" wrote: > >> Hi, >> >> I've just installed freeswitch on windows using the binary install >> (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). >> Whenever a call is made the log fills up with about 42 "Invalid UTF-8 >> character to ampersand, skip it". I haven't changed any config from the >> default. Any pointers ? >> >> Thanks! >> >> Ryan >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/25da1437/attachment.html From juanito1982 at gmail.com Sat Jan 7 19:32:20 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 7 Jan 2012 17:32:20 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F085F62.1000501@cupis.co.uk> References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> Message-ID: No, same result, 480 instead 486 2012/1/7 Paul Cupis > On 07/01/12 14:45, Juan Antonio Iba?ez Santorum wrote: > > I have the same problem. An external gateway is returning 486 to FS but > FS > > returns 480 to the caller... I am testing following dialplan: > > > > > > > > data="sofia/gateway/elastix16/8888"/> > > > > > > Sounds like a bad interaction with some continue_on_foo configuration. > > Does it help if you do: > > > > after the action-bridge? > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/0a51f17f/attachment.html From sherifomran2000 at yahoo.com Sat Jan 7 19:39:55 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 7 Jan 2012 08:39:55 -0800 (PST) Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: Message-ID: <1325954395.40624.YahooMailClassic@web110805.mail.gq1.yahoo.com> I think you have to add en_US to your profile nano /etc/profile export LANGUAGE=en_US export LANG=en_US export LC_ALL=en_US regards, Sherif Omran --- On Sat, 1/7/12, Luis Jimenez wrote: From: Luis Jimenez Subject: Re: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it To: "FreeSWITCH Users Help" Date: Saturday, January 7, 2012, 6:24 PM Hello, i have the same problem, but i dont understand the solution, please advice me. Thanks in advance Luis Jimenez On Sat, Jan 7, 2012 at 11:44 AM, Anthony Minessale wrote: If you check the git log on the conf dir, copy the xml files that were changed recently by moc.? There was a patch to enable utf8 support which exposed the invalid files. IIRC it was the de Lang templates. On Jan 7, 2012 4:27 AM, "Ryan How" wrote: Hi, I've just installed freeswitch on windows using the binary install (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). Whenever a call is made the log fills up with about 42 "Invalid UTF-8 character to ampersand, skip it". I haven't changed any config from the default. Any pointers ? Thanks! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/0c17c4d0/attachment-0001.html From juanito1982 at gmail.com Sat Jan 7 19:45:49 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 7 Jan 2012 17:45:49 +0100 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <4F081D99.7000406@exemail.com.au> References: <4F081D99.7000406@exemail.com.au> Message-ID: Try to update to last git. There are some not supported chars at german language files. Regards 2012/1/7 Ryan How > Hi, > > I've just installed freeswitch on windows using the binary install > (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). > Whenever a call is made the log fills up with about 42 "Invalid UTF-8 > character to ampersand, skip it". I haven't changed any config from the > default. Any pointers ? > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/0beb6682/attachment.html From wstephen80 at gmail.com Sat Jan 7 20:20:10 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Sat, 7 Jan 2012 18:20:10 +0100 Subject: [Freeswitch-users] Out of event dispatch threads! Slowing things down. Message-ID: Hi, my system is showing in the log the following critical errors: 2012-01-07 18:13:34.978566 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:35.981506 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:36.975496 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:37.974385 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:38.977286 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:39.983208 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:40.999186 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:42.003136 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:27.210316 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:28.214168 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:29.211205 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:14:30.208196 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:31.210143 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break The CPU is running at 40%-50% of load and the number of sessions in below the limit, the same for sessions per seconds. There is any parameter that I have to adjust? Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/b5b58899/attachment.html From herman.griffin at gmail.com Sat Jan 7 20:28:30 2012 From: herman.griffin at gmail.com (Herman Griffin) Date: Sat, 7 Jan 2012 09:28:30 -0800 Subject: [Freeswitch-users] Need help using api_hangup_hook and session_in_hangup_hook Message-ID: Hello, I'm using the api_hangup_hook and session_in_hangup variables with a pyrun. How do I access the session object from inside the python module? --------------------------------- Here are some logs : Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] Dialplan: sofia/external/3105795721 at 72.37.252.18 Action set(session_in_hangup_hook=true) Dialplan: sofia/external/3105795721 at 72.37.252.18 Action set(api_hangup_hook=pyrun emergency.hangup) Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] . . 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:281 Call python script 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:284 Finished calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:293 Error calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:164 Python Error by calling script "emergency.hangup": Message: global name 'session' is not defined Exception: None Traceback (most recent call last) File: "/usr/local/freeswitch/scripts/python/emergency/hangup.py", line 34, in runtime ---------------------------------------------------------- Here is the simple script python module emergency.hangup: from freeswitch import * def runtime(arg): consoleLog("info", print(dir(session))) Thanks, Herman From notlikeme75 at yahoo.com Sat Jan 7 20:41:17 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 7 Jan 2012 09:41:17 -0800 (PST) Subject: [Freeswitch-users] 407 undefined? sip port 5060 In-Reply-To: References: Message-ID: <1325958077.65697.YahooMailNeo@web65308.mail.ac2.yahoo.com> Vitalie, I asked my provider to switch up to port 5080 and everything is negotiating perfectly. thanks for your help. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Friday, January 6, 2012 11:54 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 58 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. 407 undefined? sip port 5060 (Rodney) ? 2.? Why database is locked? (Valery Kalinin) ? 3. Re: Why database is locked? (Ken Rice) ? 4. Re: 407 undefined? sip port 5060 (Vitalie Colosov) I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response:? 2012-01-06 21:52:46.0 000000000?? 614??????? ? ???6174530953?(????????? DID-IN:RNK) (H1 usinteractive-nextdi)?? 407?????? undefined? I have the following gateway setup in?C:\Program Files\FreeSWITCH\conf\sip_profiles\external ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. thank you. > Are you running the latest version of FS? FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > Also are you doing anything externally to mess with the sql db files? No. > Another possibility is a slow filesystem, we used to have people get this problem on suse linux. CentOS release 5.5 (Final) > You could try running your db dir in a ramdisk. These problems appeared after the upgrade version of FS. Thank you in advance... Re: [Freeswitch-users] ?Why database is locked? Blow away all the files in freeswitch/db and restart freeswitch Save the voicemail.db file if you don?t wanna lose all the voicemails and give that a try. That will force a reset of the all the databases and should clear any errors... On 1/6/12 10:00 PM, "Valery Kalinin" wrote: > Are you running the latest version of FS? > >FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > >> Also are you doing anything externally to mess with the sql db files? > >No. > >> Another possibility is a slow filesystem, we used to have people get this >problem on suse linux. > >CentOS release 5.5 (Final) > >> You could try running your db dir in a ramdisk. > >These problems appeared after the upgrade version of FS. > > >Thank you in advance... > >>________________________________ >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Ask your provider to send sip packets to 5080, instead of 5060. Every good provider should allow you to select the port YOU want, and not THEY want. If it will not help, then?you need to make?external?profile listen to 5060, instead of 5080. So, you will be sending and receiving sip messages between you and provider, using port 5060 (which is usually used only for your internal profile and your clients). To change this, you need to modify conf/sip_profiles/external.xml replace to "5060" In this case, you will have to reconfigure internal profile to be on a different port, say 5062, and ask your clients to add :5062 to the sip server address when they configure their softphone/sip adapter. Maybe another option would be to consider provider as one of your internal clients, and allow password-less communication from his IP, but this is way too crazy to consider as a good option. Vitalie 2012/1/6 Rodney I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response:? > > >2012-01-06 21:52:46.0 000000000?? 614??????? ? ???6174530953?(????????? DID-IN:RNK) (H1 usinteractive-nextdi)?? 407?????? undefined? > > > > > > > >I have the following gateway setup in?C:\Program Files\FreeSWITCH\conf\sip_profiles\external > > > > > >? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? > > > >It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. > > >thank you. > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/bb7cea62/attachment-0001.html From notlikeme75 at yahoo.com Sat Jan 7 20:48:31 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 7 Jan 2012 09:48:31 -0800 (PST) Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: References: Message-ID: <1325958511.1936.YahooMailNeo@web65303.mail.ac2.yahoo.com> this is doing the same for me but only when they press my option for the Conference Call Count extension. did have any problems with the november msi (english windows version) still hoping it gets fixed. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Saturday, January 7, 2012 12:41 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 62 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: Invalid UTF-8 character to ampersand, skip it ? ? ? (Juan Antonio Iba?ez Santorum) ? 2. Out of event dispatch threads! Slowing things??? down. ? ? ? (Stephen Wilde) ? 3. Need help using api_hangup_hook and??? session_in_hangup_hook ? ? ? (Herman Griffin) ? 4. 407 undefined? sip port 5060 (Rodney) Try to update to last git. There are some not supported chars at german language files. Regards 2012/1/7 Ryan How Hi, > >I've just installed freeswitch on windows using the binary install >(FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). >Whenever a call is made the log fills up with about 42 "Invalid UTF-8 >character to ampersand, skip it". I haven't changed any config from the >default. Any pointers ? > >Thanks! > >Ryan > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Hi, my system is showing in the log the following critical errors: 2012-01-07 18:13:34.978566 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:35.981506 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:36.975496 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:37.974385 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:38.977286 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:39.983208 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:13:40.999186 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:13:42.003136 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:27.210316 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:28.214168 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:29.211205 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-07 18:14:30.208196 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. 2012-01-07 18:14:31.210143 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break The CPU is running at 40%-50% of load and the number of sessions in below the limit, the same for sessions per seconds. There is any parameter that I have to adjust? StephenHello, I'm using the api_hangup_hook and session_in_hangup variables with a pyrun. How do I access the session object from inside the python module? --------------------------------- Here are some logs : Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] Dialplan: sofia/external/3105795721 at 72.37.252.18 Action set(session_in_hangup_hook=true) Dialplan: sofia/external/3105795721 at 72.37.252.18 Action set(api_hangup_hook=pyrun emergency.hangup) Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] . . 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:281 Call python script 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:284 Finished calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:293 Error calling python script 2012-01-07 09:21:57.693488 [ERR] mod_python.c:164 Python Error by calling script "emergency.hangup": Message: global name 'session' is not defined Exception: None Traceback (most recent call last) ??? File: "/usr/local/freeswitch/scripts/python/emergency/hangup.py", line 34, in runtime ---------------------------------------------------------- Here is the simple script python module emergency.hangup: from freeswitch import * def runtime(arg): ? ? consoleLog("info", print(dir(session))) Thanks, Herman Vitalie, I asked my provider to switch up to port 5080 and everything is negotiating perfectly. thanks for your help. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Friday, January 6, 2012 11:54 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 58 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. 407 undefined? sip port 5060 (Rodney) ? 2.? Why database is locked? (Valery Kalinin) ? 3. Re: Why database is locked? (Ken Rice) ? 4. Re: 407 undefined? sip port 5060 (Vitalie Colosov) I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response:? 2012-01-06 21:52:46.0 000000000?? 614??????? ? ???6174530953?(????????? DID-IN:RNK) (H1 usinteractive-nextdi)?? 407?????? undefined? I have the following gateway setup in?C:\Program Files\FreeSWITCH\conf\sip_profiles\external ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. thank you. > Are you running the latest version of FS? FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > Also are you doing anything externally to mess with the sql db files? No. > Another possibility is a slow filesystem, we used to have people get this problem on suse linux. CentOS release 5.5 (Final) > You could try running your db dir in a ramdisk. These problems appeared after the upgrade version of FS. Thank you in advance... Re: [Freeswitch-users] ?Why database is locked? Blow away all the files in freeswitch/db and restart freeswitch Save the voicemail.db file if you don?t wanna lose all the voicemails and give that a try. That will force a reset of the all the databases and should clear any errors... On 1/6/12 10:00 PM, "Valery Kalinin" wrote: > Are you running the latest version of FS? > >FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > >> Also are you doing anything externally to mess with the sql db files? > >No. > >> Another possibility is a slow filesystem, we used to have people get this >problem on suse linux. > >CentOS release 5.5 (Final) > >> You could try running your db dir in a ramdisk. > >These problems appeared after the upgrade version of FS. > > >Thank you in advance... > >>________________________________ >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Ask your provider to send sip packets to 5080, instead of 5060. Every good provider should allow you to select the port YOU want, and not THEY want. If it will not help, then?you need to make?external?profile listen to 5060, instead of 5080. So, you will be sending and receiving sip messages between you and provider, using port 5060 (which is usually used only for your internal profile and your clients). To change this, you need to modify conf/sip_profiles/external.xml replace to "5060" In this case, you will have to reconfigure internal profile to be on a different port, say 5062, and ask your clients to add :5062 to the sip server address when they configure their softphone/sip adapter. Maybe another option would be to consider provider as one of your internal clients, and allow password-less communication from his IP, but this is way too crazy to consider as a good option. Vitalie 2012/1/6 Rodney I currently have external profile ipkall working on port 5080 but now i am testing RNK carrier services and they are sending to me on port 5060. I do not notice anything on the console but they are saying they are sending to my server and getting this response:? > > >2012-01-06 21:52:46.0 000000000?? 614??????? ? ???6174530953?(????????? DID-IN:RNK) (H1 usinteractive-nextdi)?? 407?????? undefined? > > > > > > > >I have the following gateway setup in?C:\Program Files\FreeSWITCH\conf\sip_profiles\external > > > > > >? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? ? >? ? > > > >It is exactly the same as my IPkall except for the gateway name and proxy port. How do I make sure I can accept on this port. Is there somewhere else I need to be adding this 5060 port information. > > >thank you. > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/1b169a57/attachment-0001.html From paul at cupis.co.uk Sat Jan 7 20:54:34 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 07 Jan 2012 17:54:34 +0000 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> Message-ID: <4F0886DA.9040100@cupis.co.uk> On 07/01/12 16:32, Juan Antonio Iba?ez Santorum wrote: > No, same result, 480 instead 486 Can you show us the full log of the call, please? http://pastebin.freeswitch.org/ From juanito1982 at gmail.com Sat Jan 7 21:50:55 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 7 Jan 2012 19:50:55 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F0886DA.9040100@cupis.co.uk> References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> <4F0886DA.9040100@cupis.co.uk> Message-ID: Of course: Caller: 192.168.2.2 FS: 192.168.2.238 Gateway (Asterisk): 192.168.2.239 http://pastebin.freeswitch.org/18099 Regards 2012/1/7 Paul Cupis > On 07/01/12 16:32, Juan Antonio Iba?ez Santorum wrote: > > No, same result, 480 instead 486 > > Can you show us the full log of the call, please? > > http://pastebin.freeswitch.org/ > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/a4a507ca/attachment.html From brad at tech21.com Sat Jan 7 22:15:01 2012 From: brad at tech21.com (Brad Mina) Date: Sat, 7 Jan 2012 11:15:01 -0800 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <4F082DB9.60509@earthspike.net> References: <4F081D99.7000406@exemail.com.au> <4F082DB9.60509@earthspike.net> Message-ID: <66FC4F3B-E0BF-4364-810C-D94BEB3B3126@tech21.com> This is a known problem, the easy fix is deleting the German phrase macros from the lang/de/ directory. As of now there is no permanant fix until Moc or someone finds an easy patch. Sent from my iPhone On Jan 7, 2012, at 3:34 AM, John wrote: > On 07/01/12 10:25, Ryan How wrote: >> Hi, >> >> I've just installed freeswitch on windows using the binary install >> (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). >> Whenever a call is made the log fills up with about 42 "Invalid UTF-8 >> character to ampersand, skip it". I haven't changed any config from >> the >> default. Any pointers ? >> >> Thanks! >> >> Ryan >> > Me too. It's 42 lines and always follows a 'say': > > 2012-01-07 01:53:34.468472 [DEBUG] switch_ivr_play_say.c:67 No > language > specified - Using [en] > > I upgraded yesterday from git-9423216 2011-08-15 14-48-05 +0500 to > git-1901a41 2012-01-06 11-17-30 -0600 and the only thing I did along > the > way was enable mod_shout which I reckon is probably irrelevant. > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From paul at cupis.co.uk Sat Jan 7 22:34:24 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 07 Jan 2012 19:34:24 +0000 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> <4F0886DA.9040100@cupis.co.uk> Message-ID: <4F089E40.1080105@cupis.co.uk> Can you try adding: before your bridge action, please? Regards, From juanito1982 at gmail.com Sat Jan 7 22:45:41 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Sat, 7 Jan 2012 20:45:41 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F089E40.1080105@cupis.co.uk> References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> <4F0886DA.9040100@cupis.co.uk> <4F089E40.1080105@cupis.co.uk> Message-ID: No news. FS still sends 480 instead 486 setting that variable. 2012/1/7 Paul Cupis > Can you try adding: > > > > before your bridge action, please? > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/76e3ac7c/attachment.html From sherifomran2000 at yahoo.com Sat Jan 7 23:31:50 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 7 Jan 2012 12:31:50 -0800 (PST) Subject: [Freeswitch-users] Max calls Limit? In-Reply-To: Message-ID: <1325968310.32667.YahooMailClassic@web110807.mail.gq1.yahoo.com> Yes con/autoload/switch.xml --- On Sat, 1/7/12, Anita Hall wrote: From: Anita Hall Subject: [Freeswitch-users] Max calls Limit? To: "FreeSWITCH Users Help" Date: Saturday, January 7, 2012, 10:32 AM Is there any config variable somewhere which is limiting the max call limit on FS to 1000? Set-up: Taking FS for max call load on my machine. User freeswitch is calling internal gateway and playing demo-ivr User freeswitch defined in vars.xml ? ? ? ? ? ? ? Internal Gateway defined in directory/default/freeswitch.xml ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? After 1000 calls, it is giving the same error -ERR DESTINATION_OUT_OF_ORDER Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by the time 1000 calls are being done, but is it not a little strange that every time the resources get exhausted at precisely 1000 calls! regards, Anita -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/bddb6c3e/attachment-0001.html From krice at freeswitch.org Sun Jan 8 00:18:42 2012 From: krice at freeswitch.org (Ken Rice) Date: Sat, 07 Jan 2012 15:18:42 -0600 Subject: [Freeswitch-users] Max calls Limit? In-Reply-To: <1325968310.32667.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: This is also adjustable from the FS CLI via the fsctl max_sessions ### command... Replace ### with the max number of sessions you want. Note this is MAX Sessions, not max calls... So if you are doing 2 legged calls, the number of calls you can do is effective 1/2 that setting since each call leg is a session. Setting it from the CLI is not kept across restarts of freeswitch for that see switch.xml This setting should be set at a sane level for your particular hardware as yes it does keep freeswitch starving the server of memory and causes FS to start failing calls at that point. The similar setting max_sps is for max sessions per second and this is again to keep you from melting down your hardware. The current setting is sufficient for 99% of installs however if you are doing something out of the norm ie: high volume calling you may need to adjust it up K On 1/7/12 2:31 PM, "Sherif Omran" wrote: > Yes > con/autoload/switch.xml > > > --- On Sat, 1/7/12, Anita Hall wrote: >> >> From: Anita Hall >> Subject: [Freeswitch-users] Max calls Limit? >> To: "FreeSWITCH Users Help" >> Date: Saturday, January 7, 2012, 10:32 AM >> >> Is there any config variable somewhere which is limiting the max call limit >> on FS to 1000? >> >> Set-up: >> Taking FS for max call load on my machine. >> >> User freeswitch is calling internal gateway and playing demo-ivr >> >> User freeswitch defined in vars.xml >> >> >> >> >> >> >> >> >> Internal Gateway defined in directory/default/freeswitch.xml >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> After 1000 calls, it is giving the same error >> -ERR DESTINATION_OUT_OF_ORDER >> >> Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by the >> time 1000 calls are being done, but is it not a little strange that every >> time the resources get exhausted at precisely 1000 calls! >> >> >> regards, >> Anita >> >> >> -----Inline Attachment Follows----- >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/fedf8b70/attachment.html From sherifomran2000 at yahoo.com Sun Jan 8 01:06:34 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 7 Jan 2012 14:06:34 -0800 (PST) Subject: [Freeswitch-users] vBilling Beta Program!! - Review In-Reply-To: Message-ID: <1325973994.20601.YahooMailClassic@web110802.mail.gq1.yahoo.com> Dear Guys, I installed the vbilling sucessfully (no errors) in the freeswitch log. Here are my recommendations: 1- vBilling creates a file called freeswitch.xml by default auto-install script. If you want to use it you have to add the following line in between lines 176 and 177 because it does not load any profiles by default Alternatively: I would recommend to create a file called vBilling.xml with the following contents
?
In freeswitch.conf Add the following line ??? after the line saying ?
In Switch.conf Now it should be correctly installed. If you run ./freeswitch -nonat you should not find any errors. Then run sofia status and you should see your profiles loaded. However, 1) in my case, when i do a call and i check it being registered in the log/cdr-csv/Master.csv? , I still can not find it in the vBilling system? did i miss any configuration? Should I create a user first in vBilling system before testing a call? 2) ZRTP function stops? any clue? regards, Sherif Omran --- On Wed, 1/4/12, nbhatti wrote: From: nbhatti Subject: Re: [Freeswitch-users] vBilling Beta Program!! - Review To: freeswitch-users at lists.freeswitch.org Date: Wednesday, January 4, 2012, 9:14 PM And you would also have to change .htaccess On Wed, Jan 4, 2012 at 4:08 AM, sherif omran [via freeswitch-users] <[hidden email]> wrote: Hi every body, I had a chance 3 days to install vBilling manually, which was not so trivial and would like to share? my experience with you. Installation script is clearly written, any developer with some linux experience can follow it and do the required changes. In fact, I did not use the installation script because I have custom modules enabled. Also, I use the freeswitch xml tree, which is different that what the script do. Additionally, I have BlueBox installed and they should work together. The installation script creates a custom freeswitch.xml file with the required modules. If these freeswitch modules were not previously installed, you have to install them manually. No need to reinstall freeswitch again using (make install) enable them in the freeswitch source and compile using make modulename-install Regarding the installation path: I used Centos 6 Server The software uses the /var/www/html/ folder as the base path. However, since i have bluebox installed, I tried to do the required changes in (freeswitch.xml) and install vBilling in /var/www/html/vBilling. However, after contacting Muhammed, he recommended to enabled mod-rewrite for the apache server. After checking out, I found that it was already enabled for centos using the .htaccess file. I could log to the front page but 404 error pops, if i enter the password leading to another path.? I checked the login php function and corrected the path. At least i could see that after login click, it was trying to call a page from the correct subfolder. However, I would not recommend to install it in a subfolder because it will not work properly. May be this needs some additional investement. When i placed it in the root web server folder, and made the required adjustments it worked fine. During the installation, I had to create a database using the given script, however I changed the username and password in the mysql tag. It returned that i could not login to the frontpage, since the password in encrypted. I had then to install phpmyadmin to revert the changes and change the password from the frontpage. It is now running fine. However, I still did not start playing with its reporting functions or any call log but it looks very promising. If i install some script, I usually check for spying functions such as sending back precious information without a permission to the author. However, the software is really clean. I did not find any spying functions or backholes. I checked the PHP files as well. Finally, i would like to thank Muhammed Bhatti for sharing this nice software with us. kind regards, Sherif Omran +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Sherif Omran Dr. sc. nat. (Title from University of Zurich) Signal and Image Processing, Acoustics, Artificial Intelligence Engineer and Neural Scientist. Design, Modeling and Simulation. Expert in Biomedical devices and Cochlear Implants. Telecommunication Consultant and ERICSSON Certified Engineer. Munich - Germany e-mail: [hidden email] +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ --- On Tue, 1/3/12, Muhammad Naseer Bhatti <[hidden email]> wrote: From: Muhammad Naseer Bhatti <[hidden email]> Subject: Re: [Freeswitch-users] vBilling Beta Program!! To: "FreeSWITCH Users Help" <[hidden email]> Date: Tuesday, January 3, 2012, 8:16 PM Sherif, have you installed manually? If so, you would have to enable mod_rewrite in your apache configuration. On Tue, Jan 3, 2012 at 9:08 PM, Sherif Omran wrote: Can you give more details how it does not work? I have the same situation. I can reach the frontpage and when i give the username i get 404. Is this the case you have? Do you have Centos or Redhat ? regards, Sherif --- On Tue, 1/3/12, Zenny wrote: From: Zenny Subject: Re: [Freeswitch-users] vBilling Beta Program!! To: "FreeSWITCH Users Help" Date: Tuesday, January 3, 2012, 11:18 AM looks promising, but the user login does not work. Best of luck, Mr. Bhatti and Happy New Year 2012, though belated to all freeswitchers! On 1/3/12, nbhatti wrote: > Yes, it will support prepaid calling card and many more features soon. > > On Tue, Jan 3, 2012 at 3:17 AM, dfretes [via freeswitch-users] > wrote: >> It looks great! >> >> Will vBilling support batch user/prepaid calling card creation? >> >> ________________________________ >> If you reply to this email, your message will be added to the discussion >> below: >> http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145235.html >> To unsubscribe from vBilling Beta Program!!, click here. >> NAML > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7145871.html > Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at ... http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at ... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at ... http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at ... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at ... http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at ... http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: [hidden email] http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list [hidden email] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org If you reply to this email, your message will be added to the discussion below: http://freeswitch-users.2379917.n2.nabble.com/vBilling-Beta-Program-tp6760999p7148890.html To unsubscribe from vBilling Beta Program!!, click here. NAML View this message in context: Re: vBilling Beta Program!! - Review Sent from the freeswitch-users mailing list archive at Nabble.com. -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/4ea076ca/attachment-0001.html From kris at kriskinc.com Sun Jan 8 03:24:41 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 7 Jan 2012 19:24:41 -0500 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> <4F0886DA.9040100@cupis.co.uk> Message-ID: Why are you executing hangup ${originate_disposition} after your bridge? You'll see that FreeSWITCH is passing originate_disposition=success, probably because the remote end is returning early media (which FreeSWITCH considers a successful bridge). Do one (or both) of these: 1) Add ignore_early_media=true to your bridge line. 2) Remove the hangup ${originate_disposition} after your bridge and let FreeSWITCH run of things to execute on its own. Depending on your specific goals there are better options but these actions should isolate your problem. 2012/1/7 Juan Antonio Iba?ez Santorum : > Of course: > > Caller: 192.168.2.2 > FS: 192.168.2.238 > Gateway (Asterisk): 192.168.2.239 > > http://pastebin.freeswitch.org/18099 > > Regards > > > 2012/1/7 Paul Cupis >> >> On 07/01/12 16:32, Juan Antonio Iba?ez Santorum wrote: >> > No, same result, 480 instead 486 >> >> Can you show us the full log of the call, please? >> >> http://pastebin.freeswitch.org/ >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From anthony.minessale at gmail.com Sun Jan 8 03:30:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 7 Jan 2012 18:30:07 -0600 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <66FC4F3B-E0BF-4364-810C-D94BEB3B3126@tech21.com> References: <4F081D99.7000406@exemail.com.au> <4F082DB9.60509@earthspike.net> <66FC4F3B-E0BF-4364-810C-D94BEB3B3126@tech21.com> Message-ID: The ones in tree are fixed now but you need to manually install them On Jan 7, 2012 1:18 PM, "Brad Mina" wrote: > This is a known problem, the easy fix is deleting the German phrase > macros from the lang/de/ directory. > > As of now there is no permanant fix until Moc or someone finds an easy > patch. > > Sent from my iPhone > > On Jan 7, 2012, at 3:34 AM, John wrote: > > > On 07/01/12 10:25, Ryan How wrote: > >> Hi, > >> > >> I've just installed freeswitch on windows using the binary install > >> (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). > >> Whenever a call is made the log fills up with about 42 "Invalid UTF-8 > >> character to ampersand, skip it". I haven't changed any config from > >> the > >> default. Any pointers ? > >> > >> Thanks! > >> > >> Ryan > >> > > Me too. It's 42 lines and always follows a 'say': > > > > 2012-01-07 01:53:34.468472 [DEBUG] switch_ivr_play_say.c:67 No > > language > > specified - Using [en] > > > > I upgraded yesterday from git-9423216 2011-08-15 14-48-05 +0500 to > > git-1901a41 2012-01-06 11-17-30 -0600 and the only thing I did along > > the > > way was enable mod_shout which I reckon is probably irrelevant. > > > > John > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/4705e084/attachment.html From freeswitch at earthspike.net Sun Jan 8 04:34:19 2012 From: freeswitch at earthspike.net (John) Date: Sun, 08 Jan 2012 01:34:19 +0000 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: References: <4F081D99.7000406@exemail.com.au> Message-ID: <4F08F29B.6080105@earthspike.net> This is what I did (there are 2 files concerned), and assumes you built from git (ie, used make current): cd /usr/local/src/freeswitch/conf/lang/de/demo (or wherever that is in your source tree) git checkout -- demo.xml cp demo.xml /usr/local/freeswitch/conf/lang/de/demo/ (or wherever your FS is installed) cd /usr/local/src/freeswitch/conf/lang/de/vm git checkout -- tts.xml cp tts.xml /usr/local/freeswitch/conf/lang/de/vm/ If you do not run FS as root, you may need to check and correct the file ownership. Enter fs_cli, hit F6 to reload the XML, then check it now works by doing something that needs phrases (like calling your voicemail). John On 07/01/12 16:24, Luis Jimenez wrote: > Hello, i have the same problem, but i dont understand the solution, > please advice me. > > Thanks in advance > > Luis Jimenez > > > On Sat, Jan 7, 2012 at 11:44 AM, Anthony Minessale > > wrote: > > If you check the git log on the conf dir, copy the xml files that > were changed recently by moc. There was a patch to enable utf8 > support which exposed the invalid files. IIRC it was the de Lang > templates. > > On Jan 7, 2012 4:27 AM, "Ryan How" > wrote: > > Hi, > > I've just installed freeswitch on windows using the binary install > (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 > -0600). > Whenever a call is made the log fills up with about 42 > "Invalid UTF-8 > character to ampersand, skip it". I haven't changed any config > from the > default. Any pointers ? > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/7409ac40/attachment.html From nvitaly at gmail.com Sun Jan 8 04:45:34 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Sat, 7 Jan 2012 20:45:34 -0500 Subject: [Freeswitch-users] IVR (Calling card application) development Message-ID: <8343AF3F-140F-4C0D-8885-A8CE7B1EE71E@gmail.com> Hello, I am choosing between ESL and mod_erlang, I will have around 1k simultaneous calls with 10 min average duration. I will definitely write it in erlang, but question is - how supported is mod_erlang ? Is anyone currently use it in production for heavy load application ? Thank you Vitaly From nbhatti at gmail.com Sun Jan 8 08:52:11 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 8 Jan 2012 08:52:11 +0300 Subject: [Freeswitch-users] vBilling Beta Program!! - Review In-Reply-To: <1325973994.20601.YahooMailClassic@web110802.mail.gq1.yahoo.com> References: <1325973994.20601.YahooMailClassic@web110802.mail.gq1.yahoo.com> Message-ID: Hello Sherif, You don't need to include any other configuration files unless you really know what you are doing. You can create profiles from vBilling interface. Navigate to FreeSWITCH -> New Profile and fill in the form. We are not using mod_cdr_csv. We are storing all the cdr(s) in the database using mod_xml_cdr. If you create your own profiles and gateways, I am afraid you won't be able to use vBilling. For any vBilling issues, please post them to http://forum.vbilling.org/ and someone would help you out. Thanks. On Sun, Jan 8, 2012 at 1:06 AM, Sherif Omran wrote: > Dear Guys, > > I installed the vbilling sucessfully (no errors) in the freeswitch log. > Here are my recommendations: > > 1- vBilling creates a file called freeswitch.xml by default auto-install > script. If you want to use it you have to add the following line in between > lines 176 and 177 because it does not load any profiles by default > > > > Alternatively: I would recommend to create a file called vBilling.xml with > the following contents > > > > > > > > > data="vBilling_xml_curl_binding=configuration|directory"/> > > > > > > > > >
> > > > > > bindings="$${vBilling_xml_curl_binding}"/> > > > >
> >
> > > > data="/usr/local/freeswitch/scripts/vBilling.luac"/> > > > > >
> > > > In freeswitch.conf > > Add the following line > > > > after the line saying > >
> > In Switch.conf > > > > > > Now it should be correctly installed. If you run ./freeswitch -nonat > you should not find any errors. Then run sofia status and you should see > your profiles loaded. > > > However, > > 1) in my case, when i do a call and i check it being registered in the > log/cdr-csv/Master.csv , I still can not find it in the vBilling system? > did i miss any configuration? Should I create a user first in vBilling > system before testing a call? > > 2) ZRTP function stops? > > any clue? > > regards, > Sherif Omran > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/ddb5094b/attachment.html From rhow at exemail.com.au Sun Jan 8 09:23:41 2012 From: rhow at exemail.com.au (Ryan How) Date: Sun, 08 Jan 2012 14:23:41 +0800 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <66FC4F3B-E0BF-4364-810C-D94BEB3B3126@tech21.com> References: <4F081D99.7000406@exemail.com.au> <4F082DB9.60509@earthspike.net> <66FC4F3B-E0BF-4364-810C-D94BEB3B3126@tech21.com> Message-ID: <4F09366D.706@exemail.com.au> Thanks. Did the trick!. I don't know German anyway :) On 8/01/2012 3:15 AM, Brad Mina wrote: > This is a known problem, the easy fix is deleting the German phrase > macros from the lang/de/ directory. > > As of now there is no permanant fix until Moc or someone finds an easy > patch. > > Sent from my iPhone > > On Jan 7, 2012, at 3:34 AM, John wrote: > >> On 07/01/12 10:25, Ryan How wrote: >>> Hi, >>> >>> I've just installed freeswitch on windows using the binary install >>> (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). >>> Whenever a call is made the log fills up with about 42 "Invalid UTF-8 >>> character to ampersand, skip it". I haven't changed any config from >>> the >>> default. Any pointers ? >>> >>> Thanks! >>> >>> Ryan >>> >> Me too. It's 42 lines and always follows a 'say': >> >> 2012-01-07 01:53:34.468472 [DEBUG] switch_ivr_play_say.c:67 No >> language >> specified - Using [en] >> >> I upgraded yesterday from git-9423216 2011-08-15 14-48-05 +0500 to >> git-1901a41 2012-01-06 11-17-30 -0600 and the only thing I did along >> the >> way was enable mod_shout which I reckon is probably irrelevant. >> >> John >> >> _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sdevoy at bizfocused.com Sun Jan 8 04:26:48 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 7 Jan 2012 20:26:48 -0500 Subject: [Freeswitch-users] Answering Service, Call Forwarding, Status Indicator Message-ID: <013001cccda4$97a4eac0$c6eec040$@com> Hi All, I have a customer that has multiple DIDs that deposit incoming callers into a call queue (after a brief message) and then have several extensions process calls from the queue. There are some good examples of how to do that in FS around (thanks to those who posted them). But this customer also needs to forward all those DIDs to an answering service outside work hours. They don't want Time-Of-Day rules because if there are emergency weather conditions, they work overtime and want the phone open. I am setting up Call Forwarding via DialPlan (i.e. *72 to enable and *73 to disable.). I can do that and have the incoming calls forwarded to the answering service number because there are some nice examples of that too (also thanks). I would like to be able to have some visual indication that all calls are forwarded on all of the phones that usually join the call queue . I would love some ideas. All Phones are Cisco SPA504G (4 line each). Here are some ideas I have had, but don't know if they can be done. They are listed in order of my preference. . Have each phone setup with a line button dedicated as a Busy Line Indicator for a dummy extension. Then make that extension "busy" when calls are forwarded. This seems like the best choice, but I don't know how to have a dummy sip device or how to mark it busy or on hook via dial plan. I know I can configure a line button as a BLI for an extension on the 504Gs. This seems ideal, all of the phones would get a red indicator whenever calls are forwarded. . Have the Message Waiting Indicator FLASH when calls are forwarded. Not sure of the feasibility on this one or how actual messages waiting would impact it. . Have the Phones LCD display show the status. Seems easy enough if the user presses a button to an XML service, but to dynamically update the screen from the server when the call forwarding status changes? . I CAN have them dial *nn and have FS "speak" the current status to them. I would prefer something that does not require them to take an action. Clearly a suboptimal solution; I think if they forgot to "unforward" the phones, they are not likely to remember to check the forward status. In the longer term, I hope I think a mini php "Queue Administrator" app for these users would be fun to write. The idea being they login with a browser using their extension and "secret". The web page then shows Call Forwarding Status and number of calls in queue. They can click to join queue processing (or leave it). Or, click to forward all incoming calls (with a passcode). It would of course display caller ID information about queued calls. That could lead to some add on work to allow Customer lookup in a customer DB, or "pop" someone out of the middle of a queue, etc. Thanks for your time and I look forward to your thoughts. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/2e65dd12/attachment-0001.html From dstein at ieee.org Sun Jan 8 04:50:50 2012 From: dstein at ieee.org (David Stein) Date: Sat, 07 Jan 2012 17:50:50 -0800 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines Message-ID: <4F08F67A.8010503@ieee.org> Hello. I need to detect answering machine beeps on 1,000+ channels simultaneously. Unfortunately, this is a real requirement, not a pseudo-requirement that can be engineered away. This is for a mass emergency notification system for government, K-12, and higher education. In the past, I have done this with proprietary hardware (Dialogic and Aculab), which handles this sort of thing without overloading the host CPU, as the hardware's DSPs handle the detections. I want to see if I can do the same thing with FreeSWITCH. Alas, I find the following warning in the wiki page for mod_avmd: AVMD (and VMD) are both very CPU intensive. You need to be aware of this fact when using it. It will drastically reduce your call capacity if you do not manage it correctly. On the other hand it is a very useful tool, and if managed properly will be a great aid for calls needing to do Voice Mail Detection. Eric states, "You can expect about ~50 simultaneous instances on an Intel i7 920 CPU." So, it sounds like VMD and AVMD won't work for what I need to do. I also don't think that the tone_detect application will work, as this requires specific frequencies (as opposed to ranges of frequencies), and answering machines and voicemail come with beeps at all sorts of frequencies. Does anyone know anything, either free or commercial, that I can use with FreeSWITCH to do this many simultaneous detections? I know of commercial software-only platforms (e.g., Aculab's Prosody S) that claim to be able to this, so it seems like it should be possible. Thanks in advance, David Stein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120107/9d276fef/attachment-0001.html From sherifomran2000 at yahoo.com Sun Jan 8 14:57:35 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 8 Jan 2012 03:57:35 -0800 (PST) Subject: [Freeswitch-users] compiling mod_perl fail help required. In-Reply-To: <4F09366D.706@exemail.com.au> Message-ID: <1326023855.5065.YahooMailClassic@web110815.mail.gq1.yahoo.com> Hello I am trying to compile mod-perl but i get the following, i would appreciate a help best regards, Sheirf Omran Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. Compiling freeswitch_perl.cpp... g++ -w -DMULTIPLICITY -DEMBED_PERL -I/usr/local/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o freeswitch_perl.o freeswitch_perl.cpp In file included from freeswitch_perl.cpp:2: freeswitch_perl.h:13:20: error: EXTERN.h: No such file or directory freeswitch_perl.h:14:18: error: perl.h: No such file or directory In file included from freeswitch_perl.cpp:2: freeswitch_perl.h:22: error: ISO C++ forbids declaration of 'PerlInterpreter' with no type freeswitch_perl.h:22: error: expected ';' before '*' token freeswitch_perl.h:23: error: ISO C++ forbids declaration of 'PerlInterpreter' with no type freeswitch_perl.h:23: error: expected ';' before '*' token freeswitch_perl.h:26: error: ISO C++ forbids declaration of 'SV' with no type freeswitch_perl.h:26: error: expected ';' before '*' token freeswitch_perl.h:39: error: 'SV' has not been declared freeswitch_perl.h:49: error: 'PerlInterpreter' has not been declared In file included from freeswitch_perl.cpp:3: mod_perl_extra.h:3: error: variable or field 'mod_perl_conjure_event' declared void mod_perl_extra.h:3: error: 'PerlInterpreter' was not declared in this scope mod_perl_extra.h:3: error: 'my_perl' was not declared in this scope mod_perl_extra.h:3: error: expected primary-expression before '*' token mod_perl_extra.h:3: error: 'event' was not declared in this scope mod_perl_extra.h:3: error: expected primary-expression before 'const' mod_perl_extra.h:4: error: variable or field 'mod_perl_conjure_stream' declared void mod_perl_extra.h:4: error: 'PerlInterpreter' was not declared in this scope mod_perl_extra.h:4: error: 'my_perl' was not declared in this scope mod_perl_extra.h:4: error: expected primary-expression before '*' token mod_perl_extra.h:4: error: 'stream' was not declared in this scope mod_perl_extra.h:4: error: expected primary-expression before 'const' freeswitch_perl.cpp:5: error: 'STRLEN' does not name a type freeswitch_perl.cpp: In constructor 'PERL::Session::Session()': freeswitch_perl.cpp:13: error: 'my_perl' was not declared in this scope freeswitch_perl.cpp: In constructor 'PERL::Session::Session(char*, CoreSession*)': freeswitch_perl.cpp:18: error: 'my_perl' was not declared in this scope freeswitch_perl.cpp: In constructor 'PERL::Session::Session(switch_core_session_t*)': freeswitch_perl.cpp:34: error: 'my_perl' was not declared in this scope freeswitch_perl.cpp: At global scope: freeswitch_perl.cpp:88: error: variable or field 'setPERL' declared void freeswitch_perl.cpp:88: error: 'PerlInterpreter' was not declared in this scope freeswitch_perl.cpp:88: error: 'pi' was not declared in this scope freeswitch_perl.cpp:94: error: variable or field 'setME' declared void freeswitch_perl.cpp:94: error: 'SV' was not declared in this scope freeswitch_perl.cpp:94: error: 'p' was not declared in this scope make[4]: *** [freeswitch_perl.o] Error 1 make[3]: *** [all] Error 1 make[2]: *** [mod_perl-all] Error 1 make[1]: *** [mod_perl] Error 2 make: *** [mod_perl] Error 2 [root at sip freeswitch]#? perl -MExtUtils::Embed -e xsinit Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/dbeb5eb1/attachment.html From notlikeme75 at yahoo.com Sun Jan 8 15:19:46 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 8 Jan 2012 04:19:46 -0800 (PST) Subject: [Freeswitch-users] valet park hanging up on 1 side Message-ID: <1326025186.19988.YahooMailNeo@web65306.mail.ac2.yahoo.com> i am trying to allow a "phone roulette" situation where two callers from the ivr can be randomly be joined together. I need the following options 2 for next caller 0 go back to ivr menu the matchup seems to work but once in a bridge the caller who press 2 moves on but the other caller gets hung up on from the whole system and has to call back. is there a better way to match callers in this scenario without the hangup. here is the extension: ? ? ? ? ? ? ? ? ? ? ? ? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/13d29dca/attachment.html From fieldpeak at gmail.com Sun Jan 8 16:34:39 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Sun, 8 Jan 2012 21:34:39 +0800 Subject: [Freeswitch-users] Dynamic specify the outbound GW within source code Message-ID: Dear friends, i have FS for PSTN outbound call using below dial plan, While, now i need dynamically specify the outbound GW?s IP address according to the return result of the external command before routing in the source code , e.g. if the external command return FS the IP address of OB GW 6.7.8.9, then however, i don't know which function i should call within the source code to realize it, could anybody help to advise, P.S. i know there is existing module ?mod_xml_curl? can realize similar function, however, I could not use it for this case? thanks a lot! Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/5a26f930/attachment.html From avi at avimarcus.net Sun Jan 8 16:39:33 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 8 Jan 2012 15:39:33 +0200 Subject: [Freeswitch-users] Dynamic specify the outbound GW within source code In-Reply-To: References: Message-ID: I'm not quite sure of the use case. Do any of these help? 1) specify a server, not an IP, and then let DNS determine where it goes. 2) use a small lua script to set the channel variable based on whatever you need - an sql query, some logic.. and then use that variable in the bridge string. Those help? If not, please explain more what problem you are trying to solve. -Avi On Sun, Jan 8, 2012 at 3:34 PM, fieldpeak wrote: > Dear friends, > > i have FS for PSTN outbound call using below dial plan, > > > > > > > > While, now i need dynamically specify the outbound GW?s IP address > according to the return result of the external command before routing in > the source code , e.g. if the external command return FS the IP address of > OB GW 6.7.8.9, then > > > > however, i don't know which function i should call within the source code > to realize it, could anybody help to advise, > > P.S. i know there is existing module ?mod_xml_curl? can realize similar > function, however, I could not use it for this case? > > > thanks a lot! > > Regards, > Charles > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/fe0ff6ca/attachment-0001.html From notlikeme75 at yahoo.com Sun Jan 8 22:42:33 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 8 Jan 2012 11:42:33 -0800 (PST) Subject: [Freeswitch-users] error loading module dll open error /sym error Message-ID: <1326051753.45021.YahooMailNeo@web65306.mail.ac2.yahoo.com> I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/21fdd090/attachment.html From jeff at jefflenk.com Sun Jan 8 23:40:06 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 8 Jan 2012 12:40:06 -0800 (PST) Subject: [Freeswitch-users] error loading module dll open error /sym error In-Reply-To: <1326051753.45021.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1326051753.45021.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: <1326055206932-7165750.post@n2.nabble.com> mod_java is not currently supported on windows so I'm not sure how you built that. the spidermonkey error is because thats a sub module of mod_spidermonkey not loadable directly by fs. make sure you are not mixing 32 and 64 version of dlls. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-loading-module-dll-open-error-sym-error-tp7165636p7165750.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Sun Jan 8 23:54:01 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 8 Jan 2012 20:54:01 +0000 Subject: [Freeswitch-users] error loading module dll open error /sym error In-Reply-To: <1326051753.45021.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1326051753.45021.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: <1FFF97C269757C458224B7C895F35F1502C29C@cantor.std.visionutv.se> Have you built the modules yourself? mod_spidermonkey_teletone.dll is only loaded by mod_spidermonkey.dll, nothing you should load youself. mod_java.dll and mod_shout.dll I've never tried myself, I've never even tried to build them. Is this modules you will actually be using? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Rodney [notlikeme75 at yahoo.com] Skickat: den 8 januari 2012 20:42 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] error loading module dll open error /sym error I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks. !DSPAM:4f09f21e32761579542154! From errotan at elder.hu Mon Jan 9 00:22:21 2012 From: errotan at elder.hu (=?ISO-8859-2?Q?Pusk=E1s_Zsolt?=) Date: Sun, 08 Jan 2012 22:22:21 +0100 Subject: [Freeswitch-users] MPL v2 Message-ID: <4F0A090D.3040309@elder.hu> Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 9 01:42:12 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sun, 8 Jan 2012 22:42:12 +0000 (GMT) Subject: [Freeswitch-users] Witholding number on demand Message-ID: <1326062532.76793.YahooMailNeo@web29405.mail.ird.yahoo.com> Hi, I'm a bit of a newbie to Freeswitch at the moment, still trying to wrap my head around its power. My current pet project is fairly simple, trying to mimick the prefix based number withhold features typically offered by carriers (e.g dialling 141 in the UK, or, I believe *67 will withhold your number from the recipient). ?Basically I want to set the RPID/Privacy Flags etc. if a certain prefix is dialled ahead of the main number. My current dialplan is along the following lines : ? ? ? ? ? ? ? ? ? ? more extensions here..... I've found the prefix dialling example here?http://wiki.freeswitch.org/wiki/Prefix_dialing, but I can't see how I can integrate it with a syntax similar to the above where I am already checking for other prefixes?? Thanks in advance Bob From curriegrad2004 at gmail.com Mon Jan 9 03:53:35 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 8 Jan 2012 16:53:35 -0800 Subject: [Freeswitch-users] MPL v2 In-Reply-To: <4F0A090D.3040309@elder.hu> References: <4F0A090D.3040309@elder.hu> Message-ID: No, can't change the license version yet. This matter may require Tony to comment on. On 2012-01-08 1:32 PM, "Pusk?s Zsolt" wrote: > Hi. > > A new version of the MPL license just came out ( > http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better > than the last one? I'm trying to find the differences but this kind of > text is to hard for me :) > > Any plans for changing FreeSWITCH license to the new one ? > > Zsolt > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/b7741171/attachment.html From lists at redbonez.net Mon Jan 9 04:33:01 2012 From: lists at redbonez.net (Adam Ford) Date: Sun, 8 Jan 2012 18:33:01 -0700 Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] Message-ID: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> I can't seem to find any info on the problem I am having by searching the archives, so I apologize if this has been answered in the past (found several about MANDATORY_IE_MISSING but they were different situations). I am trying to setup FreeSWITCH using a FreeTDM + libpri + DAHDI + foneBridge2 stack. Outgoing calls work great, but I am running into the 'MANDATORY_IE_MISSING' problem with incoming calls. I am running the latest git version as of this morning, and completely default configuration with the exception of FreeTDM/DAHDI configuration and a modification of the default inbound_did dialplan to pass my DID 5530 to the default extension 1001. Below is what I get in the log, I highlighted in RED as soon as the call appears to start failing - 2012-01-08 18:04:54.946902 [NOTICE] ftmod_libpri.c:1363 -- Ring on channel 1:1 (from ********** to 5530) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1394 RING event with complete indicator (or overlap receive disabled) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1395 [s1c1][1:1] Changed state from DOWN to RING 2012-01-08 18:04:54.946902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for RING 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [RING] 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from DOWN to RING in 0ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [START] 2012-01-08 18:04:54.946902 [DEBUG] ftdm_io.c:3131 [s1c1][1:1] Enabled software DTMF detector 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:407 Set codec PCMU 20ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1740 Connect inbound channel FreeTDM/1:1/5530 2012-01-08 18:04:54.946902 [NOTICE] switch_channel.c:924 New Channel FreeTDM/1:1/5530 [f11ea7a0-3a5d-11e1-a157-018f03d45a74] 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1846 (FreeTDM/1:1/5530) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:435 (FreeTDM/1:1/5530) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT going to sleep 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_channel.c:1884 (FreeTDM/1:1/5530) Callstate Change DOWN -> RINGING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state RING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Changed state from RING to PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROCEED] 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from RING to PROCEED in 55ms 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->5530 in context public Dialplan: FreeTDM/1:1/5530 parsing [public->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->outside_call] continue=true Dialplan: FreeTDM/1:1/5530 Absolute Condition [outside_call] Dialplan: FreeTDM/1:1/5530 Action set(outside_call=true) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [public->call_debug] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 parsing [public->public_extensions] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [public_extensions] destination_number(5530) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->public_did] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [public_did] destination_number(5530) =~ /^(5530)$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action set(domain_name=xx.xx.xx.xxx) Dialplan: FreeTDM/1:1/5530 Action transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 set(outside_call=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [outside_call]=[true] EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 set(domain_name=xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [domain_name]=[xx.xx.xx.xxx] EXECUTE FreeTDM/1:1/5530 transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr.c:1711 (FreeTDM/1:1/5530) State Change CS_EXECUTE -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [NOTICE] switch_ivr.c:1717 Transfer FreeTDM/1:1/5530 to XML[1001 at default] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state PROCEED 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->1001 in context default Dialplan: FreeTDM/1:1/5530 parsing [default->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->tod_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->holiday_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->group-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->intercept-ext] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->redial] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: FreeTDM/1:1/5530 Absolute Condition [global] Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-2] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-1] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call_return] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->del-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->add-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-simo] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-order] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->extension-intercom] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->Local_Extension] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action export(dialed_extension=1001) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: FreeTDM/1:1/5530 Action set(ringback=${us-ring}) Dialplan: FreeTDM/1:1/5530 Action set(transfer_ringback=local_stream://moh) Dialplan: FreeTDM/1:1/5530 Action set(call_timeout=30) Dialplan: FreeTDM/1:1/5530 Action set(hangup_after_bridge=true) Dialplan: FreeTDM/1:1/5530 Action set(continue_on_fail=true) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: FreeTDM/1:1/5530 Action answer() Dialplan: FreeTDM/1:1/5530 Action sleep(1000) Dialplan: FreeTDM/1:1/5530 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-spymap/**********/f11ea7a0-3a5d-11e1-a157-018f03d45 a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/**********/1001) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/global/f11ea7a0-3a5d-11e1-a157-018f03d45a 74) EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 export(dialed_extension=1001) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1091 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE FreeTDM/1:1/5530 bind_meta_app(1 b s execute_extension::dx XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04 -55.wav) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04 -55.wav EXECUTE FreeTDM/1:1/5530 bind_meta_app(3 b s execute_extension::cf XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE FreeTDM/1:1/5530 set(ringback=%(2000,4000,440,480)) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [ringback]=[%(2000,4000,440,480)] EXECUTE FreeTDM/1:1/5530 set(transfer_ringback=local_stream://moh) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [transfer_ringback]=[local_stream://moh] EXECUTE FreeTDM/1:1/5530 set(call_timeout=30) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [call_timeout]=[30] EXECUTE FreeTDM/1:1/5530 set(hangup_after_bridge=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [hangup_after_bridge]=[true] EXECUTE FreeTDM/1:1/5530 set(continue_on_fail=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [continue_on_fail]=[true] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-call_return/1001/**********) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/1001/f11ea7a0-3a5d-11e1-a157-018f03d4 5a74) EXECUTE FreeTDM/1:1/5530 set(called_party_callgroup=techsupport) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [called_party_callgroup]=[techsupport] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/techsupport/f11ea7a0-3a5d-11e1-a157-0 18f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/global/f11ea7a0-3a5d-11e1-a157-018f03 d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/techsupport/f11ea7a0-3a5d-11e1-a157-018f0 3d45a74) EXECUTE FreeTDM/1:1/5530 bridge(user/1001 at xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [f1292a54-3a5d-11e1-a15e-018f03d45a74] 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:4674 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1884 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change DOWN -> RINGING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA going to sleep 2012-01-08 18:04:55.006902 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [calling][0] 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1668 Got a FACILITY event on span 1:1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1697 FACILITY subcommand 2 is not implemented, ignoring 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1700 FACILITY subcommand 2 handler returned -1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1703 Caught Event on span 1 11 (FACILITY) 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [proceeding][180] 2012-01-08 18:04:55.126907 [NOTICE] sofia.c:5574 Ring-Ready sofia/internal/sip:1001 at xx.xx.xx.xxx:52564! 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Indicating PROGRESS_MEDIA in state PROCEED 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROCEED to PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS] 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROCEED to PROGRESS in 22ms 2012-01-08 18:04:55.146908 [ERR] ftmod_libpri.c:132 XXX Progress message requested but no information is provided 2012-01-08 18:04:55.146908 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROGRESS to PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS_MEDIA] 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS to PROGRESS_MEDIA in 55ms 2012-01-08 18:04:55.206902 [INFO] ftmod_zt.c:656 Setting echo cancel to 64 taps for 1:1 2012-01-08 18:04:55.206902 [WARNING] ftmod_zt.c:661 Echo cancel not available for 1:1 2012-01-08 18:04:55.206902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.206902 [NOTICE] switch_ivr_originate.c:1115 Pre-Answer FreeTDM/1:1/5530! 2012-01-08 18:04:55.206902 [DEBUG] switch_channel.c:2930 (FreeTDM/1:1/5530) Callstate Change RINGING -> EARLY 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2012-01-08 18:04:55.206902 [DEBUG] switch_core_codec.c:116 FreeTDM/1:1/5530 Push codec L16:70 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 1:1 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1078 [s1c1][1:1] Changed state from PROGRESS_MEDIA to TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [TERMINATING] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS_MEDIA to TERMINATING in 0ms 2012-01-08 18:04:55.226909 [DEBUG] ftdm_io.c:5565 [s1c1][1:1] Scheduling safety hangup timer 2012-01-08 18:04:55.226909 [DEBUG] mod_freetdm.c:2416 got clear channel sig [STOP] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2846 (FreeTDM/1:1/5530) Callstate Change EARLY -> HANGUP 2012-01-08 18:04:55.226909 [NOTICE] mod_freetdm.c:2441 Hangup FreeTDM/1:1/5530 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2869 Send signal FreeTDM/1:1/5530 [KILL] 2012-01-08 18:04:55.226909 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_codec.c:141 FreeTDM/1:1/5530 Restore previous codec PCMU:0. 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2846 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change RINGING -> HANGUP 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2869 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [KILL] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [INFO] mod_dptools.c:2900 Originate Failed. Cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:2285 FreeTDM/1:1/5530 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 hanging up, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:530 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP ENTER 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:605 [s1c1][1:1] Changed state from TERMINATING to HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY going to sleep 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from TERMINATING to HANGUP in 15ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:929 [s1c1][1:1] Changed state from HANGUP to HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP_COMPLETE] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP to HANGUP_COMPLETE in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:939 [s1c1][1:1] Changed state from HANGUP_COMPLETE to DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [DOWN] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP_COMPLETE to DOWN in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2930 [s1c1][1:1] DTMF debug is already disabled 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2962 [s1c1][1:1] No need to disable input dump 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2993 [s1c1][1:1] No need to disable output dump 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [RELEASED] 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:6185 Cleared call with id 1 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2735 [s1c1][1:1] channel done 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:704 -- Closed channel 1:1 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:624 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP EXIT 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 FreeTDM/1:1/5530 Standard HANGUP, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (FreeTDM/1:1/5530) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 FreeTDM/1:1/5530 Standard REPORTING, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (FreeTDM/1:1/5530) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 1 (FreeTDM/1:1/5530) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 1 (FreeTDM/1:1/5530) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel FreeTDM/1:1/5530 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (FreeTDM/1:1/5530) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (FreeTDM/1:1/5530) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 FreeTDM/1:1/5530 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY going to sleep Full log (including loading FreeSWITCH) at http://pastebin.com/vQUHu0pR freeswitch/conf/freetdm.conf - [span zt PRI] trunk_type => T1 b-channel=1-23 d-channel=24 freeswitch/conf/autoload_confg/freetdm.conf.xml - /etc/dahdi/system.conf - loadzone = us defaultzone=us dynamic=ethmf,eth0/00:50:c2:65:d7:59/0,24,1 bchan=1-23 dchan=24 I am guessing it is a configuration issue, though this same config is currently working in production with FreeSWITCH 1.0.6 + OpenZAP + Libpri + DAHDI + foneBridge2. Any help is greatly appreciated. -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/7367b2f3/attachment-0001.html From olimonkey at gmail.com Mon Jan 9 04:33:29 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Mon, 9 Jan 2012 09:33:29 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Message-ID: Alright, so I tried to set ignore_early_media=true, but the phone isn't being answered at all anymore. So i went back to the CISCO and turned on debugging. I'm now getting a different sequence of events. I still get: INVITE from FS 100 TRYING from CISCO 183 SESSION PROGRESS from CISCO then, when I pick up the phone I get: 503 SERVICE UNAVAILABLE from CISCO ACK from FS I wonder what's going on now. I'll have to play around with the "supervisiory answer dual-tone". CISCO voice-port config: voice-port 0/3/0 supervisory disconnect dualtone mid-call supervisory answer dualtone output attenuation -3 no comfort-noise cptone AU timeouts call-disconnect 5 timeouts wait-release 5 connection plar opx 1001 impedance complex1 caller-id enable Cheers, Oliver On Sat, Jan 7, 2012 at 9:28 AM, Oliver Schenk wrote: > No, at the time I was no longer using "ignore_early_media= true", > because initially it didn't work. > I was actually thinking of putting this property back in again, thanks > a lot! I'll try it first thing Monday. > > > With regard to EXECUTE log line, either I don't have the right level > of logging turned on or something else is going on, because I've never > seen any such log entries before with my managed application; yet my > IVR app definitely does get executed. I don't think I have the DEBUG > logging output level turned on so that could explain it... > > > Thanks all, > > Oliver > > > > On Fri, Jan 6, 2012 at 11:21 PM, Peter Olsson > wrote: >> Are you still using ignore_early_media=true - this must be set for this to work correctly. >> >> You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. >> >> Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" >> >> Park application is only executed after the call was answered. >> >> /Peter >> >> ________________________________________ >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >> Skickat: den 6 januari 2012 12:04 >> Till: FreeSWITCH Users Help >> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >> >> Because I'm using an FXO card with voice, I added something to my >> CISCO conf. Many others had the same thing: >> >> >> voice-port 0/3/0 >> ? ... >> ? supervisory disconnect dualtone mid-call >> ? supervisory answer dualtone ? ?<---- ADDED THIS ONE >> ? ... >> >> >> >> Once I added this, the FS output now just showed the following while >> the phone was ringing: >> >> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >> sofia/internal/109212xxxx at 192.168.x.x >> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >> Call" <109212xxxx> >> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >> sofia/internal/109212xxxx at 192.168.x.x! >> >> >> Where as previous it would show the above and also show the following: >> >> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >> <0000000000> to "Outbound Call" <109212xxxx> >> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >> <1092122856> >> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >> >> >> >> BUT, the IVR still started playing even before I pick up the phone. >> Hmmmm.....so why is FS still starting the managed application when the >> call has not been answered yet. Are we all sure that the managed >> application should not be executed until the call "has been answered" >> shows up in the log file? >> >> >> Will have to keep testing on monday as I don't have access to the >> CISCO from where i am now. I'll have to see whether the CISCO changes >> had any impact on the times at which the SIP messages are sent back >> and forth. Especially the 200 OK message. >> >> >> Thanks again for help, maybe getting somewhere now...... >> >> Oliver >> >> >> >> >> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >> wrote: >>> If it sends 200 OK right after 183, this IS the problem. >>> >>> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. >>> >>> Seems to me there is something broken in the Cisco. >>> >>> /Peter >>> >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>> Skickat: den 6 januari 2012 06:55 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> >>> I've tried looking at disable-early-media configuration command, but >>> that didn't work and I doubt that has anything to do with the CISCO >>> sending a 200 OK right after a 183 SESSION PROGRESS. >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >>>> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >>>> is usually RINGING (generate ringback locally) while a 183 has media... aka >>>> early media and usually provides ringback inband. >>>> >>>> /b >>>> >>>> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>>> >>>> Shouldn't there be a ?180 RINGING ?somewhere in there? >>>> >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >>>> >>>> I just noticed something else, if I don't pick up the phone at all. >>>> >>>> The IVR just keeps playing until the menu timeout kicks in. >>>> >>>> >>>> So here is a CISCO SIP log: >>>> >>>> http://pastebin.com/Y9sYkuxi >>>> >>>> >>>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>>> >>>> I hope the CISCO log is readable, it's a bit long because I just did >>>> >>>> "debug ccsip all". >>>> >>>> >>>> >>>> >>>> In this test I didn't bother picking up the phone at all, but I can >>>> >>>> see that FS answered anyway and the IVR kept playing until it timed >>>> >>>> out. >>>> >>>> I'm not an expert, but here is what I picked out of it: >>>> >>>> >>>> At 00:08:10 we get a >>>> >>>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>>> >>>> >>>> the further down at the same timestamp we get >>>> >>>> Sent: "SIP/2.0 100 Trying" >>>> >>>> >>>> At 00:08:13 we get a >>>> >>>> Sent: "SIP/2.0 183 Session Progress" >>>> >>>> >>>> At 00:18:13 we get a >>>> >>>> Sent: "SIP/2.0 200 OK" >>>> >>>> >>>> Then at the same timestamp we get: >>>> >>>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>> >>>> >>>> >>>> >>>> Once the IVR times out at 00:09:16 we get >>>> >>>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>> >>>> >>>> And then the reply right after >>>> >>>> Sent: "SIP/2.0 200 OK" >>>> >>>> >>>> >>>> >>>> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>>> >>>> after the "INVITE" is received. >>>> >>>> >>>> >>>> >>>> The part that is beyond my field of expertise so far is WHY? >>>> >>>> >>>> >>>> >>>> Thanks, >>>> >>>> >>>> >>>> Oliver >>>> >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>>> >>>> By the way: >>>> >>>> >>>> I tried {ignore_early_media=true} as well, but as I think we >>>> >>>> determined, my problem is probably with the CISCO telling FS that the >>>> >>>> call has been answered when really it hasn't yet. >>>> >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>>> >>>> Thanks for the help so far. >>>> >>>> >>>> >>>> Here is a pastebin of FreeSWITCH output: >>>> >>>> http://pastebin.com/i6Qgc7ws >>>> >>>> >>>> Notice how the "has been answered" log message comes immediately >>>> >>>> (within a few milliseconds) after the call was originated. I think >>>> >>>> this would suggest that the CISCO is immediately sending a 200 OK, as >>>> >>>> you suggested. I also turned on CISCO debugging, but I'm just trying >>>> >>>> to figure out how to get the information regarding SIP messages back >>>> >>>> to Freeswitch. I'll run the test again and see if I can get some >>>> >>>> useful CISCO debug. >>>> >>>> >>>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>>> >>>> SIP debugging? >>>> >>>> >>>> >>>> Thanks! >>>> >>>> >>>> >>>> >>>> >>>> 2012/1/6 Gustavo M?rsico : >>>> >>>> I think I've a similar problem related to callcenter app. When I made an >>>> originate like this: >>>> >>>> >>>> originate loopback/2500/default/XML &bridge(user/2001) >>>> >>>> >>>> 2500 is an extension that leads to a callcenter application. In this case, >>>> we dial first to the queue and when an agent answered we call to the >>>> customer. As far as I know >>>> >>>> When the A-leg reaches to the queue, without selecting an agent, the call is >>>> automatically sent to the B-leg. As far as I see, there is a pre-answer >>>> method that fs needs to send the media to A-leg. >>>> >>>> In order to try to avoid this, I tried using ignore_early_media=true as part >>>> of the originate in A-leg and/or B-leg, with no luck. >>>> >>>> >>>> originate {ignore_early_media=true}loopback/2500/default/XML >>>> &bridge({ignore_early_media=true}user/2001) >>>> >>>> >>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>>> destination_number(2500) =~ /^(2500)$/ break=on-false >>>> >>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>> >>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>>> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >>>> loopback/2500-b [BREAK] >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>> CHANNEL KILL >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>>> (loopback/2500-b) State ROUTING going to sleep >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>>> (loopback/2500-b) Running State Change CS_EXECUTE >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>>> (loopback/2500-b) State EXECUTE >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>>> CHANNEL EXECUTE >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>>> loopback/2500-b Standard EXECUTE >>>> >>>> EXECUTE loopback/2500-b set(open=true) >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>> [open]=[true] >>>> >>>> EXECUTE loopback/2500-b >>>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>> >>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>> >>>> EXECUTE loopback/2500-b >>>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>> >>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>> >>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>> [ignore_early_media]=[true] >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >>>> callcenter Requires media! pre_answering channel loopback/2500-b >>>> >>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>>> loopback/2500-a! >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >>>> Callstate Change RINGING -> EARLY >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>> loopback/2500-b [BREAK] >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>> CHANNEL KILL >>>> >>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >>>> loopback/2500-b! >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >>>> Callstate Change RINGING -> EARLY >>>> >>>> EXECUTE loopback/2500-b callcenter(click2call) >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >>>> Callstate Change EARLY -> ACTIVE >>>> >>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>>> [loopback/2500-a] has been answered >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>> loopback/2500-b [BREAK] >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>> CHANNEL KILL >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >>>> Resulted in Success: [loopback/2500-a] >>>> >>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >>>> Callstate Change EARLY -> ACTIVE >>>> >>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >>>> Flipping CID from "" <0000000000> to "Outbound Call" >>>> >>>> >>>> >>>> >>>> >>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>> >>>> >>>> Also, maybe I should be doing something like this: >>>> >>>> >>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>> >>>> >>>> instead of: >>>> >>>> >>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>> >>>> >>>> >>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>> >>>> know how really...probably not on the right track there. >>>> >>>> >>>> >>>> >>>> >>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>> >>>> *bump* >>>> >>>> >>>> >>>> So I think maybe the way I'm doing the originate is the problem? In my >>>> >>>> call string I'm creating a connection directly from the CISCO >>>> >>>> (192.168.x.x) to the managed application, which may be why it starts >>>> >>>> playing straight away? >>>> >>>> >>>> Maybe I should be originating a call first and then only once I know >>>> >>>> the other side has picked up will I bridge the call to the IVR managed >>>> >>>> application. >>>> >>>> >>>> Problem is I dunno how to tell whether the other person has picked up >>>> >>>> (or even if the cisco is going to tell me) and I don't know how to do >>>> >>>> things to a call once it has been established. >>>> >>>> >>>> >>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>> >>>> out of it there. >>>> >>>> >>>> >>>> Cheers >>>> >>>> >>>> Oliver >>>> >>>> >>>> >>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>> >>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>> >>>> and connecting through a CISCO 2811. Most things now work quite well, >>>> >>>> but I am having a few issues with the way the system answers calls (or >>>> >>>> doesn't answer calls...). >>>> >>>> >>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>> >>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>> >>>> which is then connected to a POTS phone line. >>>> >>>> >>>> >>>> Take the following scenario: >>>> >>>> >>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>> >>>> to freeswitch and originate a call: >>>> >>>> >>>> string callstring = >>>> >>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>> >>>> '&managed(ivrAppName)'"; >>>> >>>> eslConnection.API("originate", callstring); >>>> >>>> >>>> where 192.168.x.x is the CISCO IP. >>>> >>>> >>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>> >>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>> >>>> number (091234567) to make the call. >>>> >>>> >>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>> >>>> >>>> >>>> >>>> These are my current problems: >>>> >>>> >>>> - IVR starts playing before I even pick up the phone. This means that >>>> >>>> if the system calls a mobile phone and the person doesn't pick up, the >>>> >>>> IVR will start playing and eventually the mobile phone will divert to >>>> >>>> voice mail. Obviously I then get a missed call and an sms saying I >>>> >>>> have a new voice mail, which is annoying. Instead I would like it to >>>> >>>> KNOW that no one has picked up, but I don't know how to do this. >>>> >>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>> >>>> has not yet been answered. For some reason however as soon as the >>>> >>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>> >>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>> >>>> doing originate the wrong way or something ... >>>> >>>> >>>> - The phone only rings for about 10 seconds before hanging up. I've >>>> >>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>> >>>> CISCO "ring number". Nothing works, my phone still only rings for >>>> >>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>> >>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>> >>>> starts playing even if no one answers the phone. >>>> >>>> >>>> >>>> >>>> >>>> >>>> CISCO Config for relevant FXO port: >>>> >>>> >>>> voice service voip >>>> >>>> ?allow-connections h323 to h323 >>>> >>>> ?allow-connections h323 to sip >>>> >>>> ?allow-connections sip to h323 >>>> >>>> ?allow-connections sip to sip >>>> >>>> ?no supplementary-service h450.2 >>>> >>>> ?no supplementary-service h450.3 >>>> >>>> ?supplementary-service h450.12 >>>> >>>> ?no supplementary-service sip moved-temporarily >>>> >>>> ?no supplementary-service sip refer >>>> >>>> ?fax protocol cisco >>>> >>>> ?sip >>>> >>>> ?registrar server expires max 3600 min 3600 >>>> >>>> ?no update-callerid >>>> >>>> ?no call service stop >>>> >>>> >>>> voice-port 0/3/2 >>>> >>>> ?output attenuation -3 >>>> >>>> ?no comfort-noise >>>> >>>> ?cptone AU >>>> >>>> ?impedance complex1 >>>> >>>> ?caller-id enable >>>> >>>> ! >>>> >>>> dial-peer voice 100 pots >>>> >>>> ?preference 1 >>>> >>>> ?destination-pattern 1T >>>> >>>> ?port 0/3/2 >>>> >>>> ! >>>> >>>> >>>> >>>> >>>> Many Thanks, >>>> >>>> >>>> Oliver >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> http://www.cluecon.com >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> >>>> consulting at freeswitch.org >>>> >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> >>>> http://www.freeswitch.org >>>> >>>> http://wiki.freeswitch.org >>>> >>>> http://www.cluecon.com >>>> >>>> >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> -- >>>> Brian West >>>> FreeSWITCH Solutions, LLC >>>> Phone: +1 (918) 420-9266 >>>> Fax: ? +1 (918) 420-9267 >>>> brian at freeswitch.org >>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:4f06d49b32762089563979! >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org From curriegrad2004 at gmail.com Mon Jan 9 04:42:43 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 8 Jan 2012 17:42:43 -0800 Subject: [Freeswitch-users] Witholding number on demand In-Reply-To: <1326062532.76793.YahooMailNeo@web29405.mail.ird.yahoo.com> References: <1326062532.76793.YahooMailNeo@web29405.mail.ird.yahoo.com> Message-ID: Check this config set out: git://github.com/curriegrad2004/freeswitch-sample-configs.git The internal.xml file under the light-pbx/dialplan directory of the git repository demonstrates what you really want to do. On Sun, Jan 8, 2012 at 2:42 PM, Bob Smith wrote: > Hi, > > I'm a bit of a newbie to Freeswitch at the moment, still trying to wrap my head around its power. > > My current pet project is fairly simple, trying to mimick the prefix based number withhold features typically offered by carriers (e.g dialling 141 in the UK, or, I believe *67 will withhold your number from the recipient). ?Basically I want to set the RPID/Privacy Flags etc. if a certain prefix is dialled ahead of the main number. > > My current dialplan is along the following lines : > > > > ? ? ? ? ? > ? ? ? ? ? > > > > more extensions here..... > > > I've found the prefix dialling example here?http://wiki.freeswitch.org/wiki/Prefix_dialing, but I can't see how I can integrate it with a syntax similar to the above where I am already checking for other prefixes? > > Thanks in advance > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From notlikeme75 at yahoo.com Mon Jan 9 05:26:24 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 8 Jan 2012 18:26:24 -0800 (PST) Subject: [Freeswitch-users] error loading module dll open error /sym error In-Reply-To: References: Message-ID: <1326075984.52157.YahooMailNeo@web65306.mail.ac2.yahoo.com> okay good to know about java. and since i cant use java i wont be using spidermonkey teletone, i was trying to use the example answering machine script . i want to have voice bulletin board that would allow callers to record a public message and have other callers be able to listen to the last 30 or so. and the mod voicemail has too many options right now for this. as far as the mod shout goes. yes i want to have a folder of mp3s and maybe even an internet stream or too so this is important to me. when you say not to mix 32bit and 64 bit dlls. do you mean if i installed the 64 bit msi not to build 32bit dlls? if not, is it just better for me to always run the 32 bit even on a 64 bit windows server? rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Sunday, January 8, 2012 8:33 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 69 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. error loading module dll open error /sym error (Rodney) ? 2. Re: error loading module dll open error /sym??? error (Jeff Lenk) ? 3. Re: error loading module dll open error /sym error (Peter Olsson) ? 4. MPL v2 (Pusk?s Zsolt) ? 5. Witholding number on demand (Bob Smith) ? 6. Re: MPL v2 (curriegrad2004) ? 7. FreeTDM [MANDATORY_IE_MISSING] (Adam Ford) I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks.mod_java is not currently supported on windows so I'm not sure how you built that. the spidermonkey error is because thats a sub module of mod_spidermonkey not loadable directly by fs. make sure you are not mixing 32 and 64 version of dlls. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-loading-module-dll-open-error-sym-error-tp7165636p7165750.html Sent from the freeswitch-users mailing list archive at Nabble.com. Have you built the modules yourself? mod_spidermonkey_teletone.dll is only loaded by mod_spidermonkey.dll, nothing you should load youself. mod_java.dll and mod_shout.dll I've never tried myself, I've never even tried to build them. Is this modules you will actually be using? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Rodney [notlikeme75 at yahoo.com] Skickat: den 8 januari 2012 20:42 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] error loading module dll open error /sym error I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks. !DSPAM:4f09f21e32761579542154! Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt Hi, I'm a bit of a newbie to Freeswitch at the moment, still trying to wrap my head around its power. My current pet project is fairly simple, trying to mimick the prefix based number withhold features typically offered by carriers (e.g dialling 141 in the UK, or, I believe *67 will withhold your number from the recipient). ?Basically I want to set the RPID/Privacy Flags etc. if a certain prefix is dialled ahead of the main number. My current dialplan is along the following lines : ? ? ? ? ? ? ? ? ? ? more extensions here..... I've found the prefix dialling example here?http://wiki.freeswitch.org/wiki/Prefix_dialing, but I can't see how I can integrate it with a syntax similar to the above where I am already checking for other prefixes?? Thanks in advance Bob No, can't change the license version yet. This matter may require Tony to comment on. On 2012-01-08 1:32 PM, "Pusk?s Zsolt" wrote: Hi. > >A new version of the MPL license just came out ( >http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better >than the last one? I'm trying to find the differences but this kind of >text is to hard for me :) > >Any plans for changing FreeSWITCH license to the new one ? > >Zsolt > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > I can?t seem to find any info on the problem I am having by searching the archives, so I apologize if this has been answered in the past (found several about MANDATORY_IE_MISSING but they were different situations). ? I am trying to setup FreeSWITCH using a FreeTDM + libpri + DAHDI + foneBridge2 stack.? Outgoing calls work great, but I am running into the ?MANDATORY_IE_MISSING? problem with incoming calls.? I am running the latest git version as of this morning, and completely default configuration with the exception of FreeTDM/DAHDI configuration and a modification of the default inbound_did dialplan to pass my DID 5530 to the default extension 1001. ? Below is what I get in the log, I highlighted in RED as soon as the call appears to start failing - ? 2012-01-08 18:04:54.946902 [NOTICE] ftmod_libpri.c:1363 -- Ring on channel 1:1 (from ********** to 5530) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1394 RING event with complete indicator (or overlap receive disabled) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1395 [s1c1][1:1] Changed state from DOWN to RING 2012-01-08 18:04:54.946902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for RING 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [RING] 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from DOWN to RING in 0ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [START] 2012-01-08 18:04:54.946902 [DEBUG] ftdm_io.c:3131 [s1c1][1:1] Enabled software DTMF detector 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:407 Set codec PCMU 20ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1740 Connect inbound channel FreeTDM/1:1/5530 2012-01-08 18:04:54.946902 [NOTICE] switch_channel.c:924 New Channel FreeTDM/1:1/5530 [f11ea7a0-3a5d-11e1-a157-018f03d45a74] 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1846 (FreeTDM/1:1/5530) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:435 (FreeTDM/1:1/5530) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT going to sleep 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_channel.c:1884 (FreeTDM/1:1/5530) Callstate Change DOWN -> RINGING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state RING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Changed state from RING to PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROCEED] 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from RING to PROCEED in 55ms 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->5530 in context public Dialplan: FreeTDM/1:1/5530 parsing [public->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->outside_call] continue=true Dialplan: FreeTDM/1:1/5530 Absolute Condition [outside_call] Dialplan: FreeTDM/1:1/5530 Action set(outside_call=true) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [public->call_debug] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 parsing [public->public_extensions] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [public_extensions] destination_number(5530) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->public_did] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [public_did] destination_number(5530) =~ /^(5530)$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action set(domain_name=xx.xx.xx.xxx) Dialplan: FreeTDM/1:1/5530 Action transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 set(outside_call=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [outside_call]=[true] EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 set(domain_name=xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [domain_name]=[xx.xx.xx.xxx] EXECUTE FreeTDM/1:1/5530 transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr.c:1711 (FreeTDM/1:1/5530) State Change CS_EXECUTE -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [NOTICE] switch_ivr.c:1717 Transfer FreeTDM/1:1/5530 to XML[1001 at default] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state PROCEED 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->1001 in context default Dialplan: FreeTDM/1:1/5530 parsing [default->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->tod_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->holiday_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->group-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->intercept-ext] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->redial] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: FreeTDM/1:1/5530 Absolute Condition [global] Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-2] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-1] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call_return] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->del-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->add-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-simo] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-order] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->extension-intercom] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->Local_Extension] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action export(dialed_extension=1001) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: FreeTDM/1:1/5530 Action set(ringback=${us-ring}) Dialplan: FreeTDM/1:1/5530 Action set(transfer_ringback=local_stream://moh) Dialplan: FreeTDM/1:1/5530 Action set(call_timeout=30) Dialplan: FreeTDM/1:1/5530 Action set(hangup_after_bridge=true) Dialplan: FreeTDM/1:1/5530 Action set(continue_on_fail=true) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: FreeTDM/1:1/5530 Action answer() Dialplan: FreeTDM/1:1/5530 Action sleep(1000) Dialplan: FreeTDM/1:1/5530 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-spymap/**********/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/**********/1001) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/global/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 export(dialed_extension=1001) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1091 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE FreeTDM/1:1/5530 bind_meta_app(1 b s execute_extension::dx XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04-55.wav) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04-55.wav EXECUTE FreeTDM/1:1/5530 bind_meta_app(3 b s execute_extension::cf XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE FreeTDM/1:1/5530 set(ringback=%(2000,4000,440,480)) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [ringback]=[%(2000,4000,440,480)] EXECUTE FreeTDM/1:1/5530 set(transfer_ringback=local_stream://moh) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [transfer_ringback]=[local_stream://moh] EXECUTE FreeTDM/1:1/5530 set(call_timeout=30) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [call_timeout]=[30] EXECUTE FreeTDM/1:1/5530 set(hangup_after_bridge=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [hangup_after_bridge]=[true] EXECUTE FreeTDM/1:1/5530 set(continue_on_fail=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [continue_on_fail]=[true] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-call_return/1001/**********) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/1001/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 set(called_party_callgroup=techsupport) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [called_party_callgroup]=[techsupport] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/techsupport/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/global/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/techsupport/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 bridge(user/1001 at xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [f1292a54-3a5d-11e1-a15e-018f03d45a74] 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:4674 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1884 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change DOWN -> RINGING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA going to sleep 2012-01-08 18:04:55.006902 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [calling][0] 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1668 Got a FACILITY event on span 1:1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1697 FACILITY subcommand 2 is not implemented, ignoring 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1700 FACILITY subcommand 2 handler returned -1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1703 Caught Event on span 1 11 (FACILITY) 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [proceeding][180] 2012-01-08 18:04:55.126907 [NOTICE] sofia.c:5574 Ring-Ready sofia/internal/sip:1001 at xx.xx.xx.xxx:52564! 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Indicating PROGRESS_MEDIA in state PROCEED 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROCEED to PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS] 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROCEED to PROGRESS in 22ms 2012-01-08 18:04:55.146908 [ERR] ftmod_libpri.c:132 XXX Progress message requested but no information is provided 2012-01-08 18:04:55.146908 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROGRESS to PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS_MEDIA] 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS to PROGRESS_MEDIA in 55ms 2012-01-08 18:04:55.206902 [INFO] ftmod_zt.c:656 Setting echo cancel to 64 taps for 1:1 2012-01-08 18:04:55.206902 [WARNING] ftmod_zt.c:661 Echo cancel not available for 1:1 2012-01-08 18:04:55.206902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.206902 [NOTICE] switch_ivr_originate.c:1115 Pre-Answer FreeTDM/1:1/5530! 2012-01-08 18:04:55.206902 [DEBUG] switch_channel.c:2930 (FreeTDM/1:1/5530) Callstate Change RINGING -> EARLY 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2012-01-08 18:04:55.206902 [DEBUG] switch_core_codec.c:116 FreeTDM/1:1/5530 Push codec L16:70 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 1:1 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1078 [s1c1][1:1] Changed state from PROGRESS_MEDIA to TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [TERMINATING] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS_MEDIA to TERMINATING in 0ms 2012-01-08 18:04:55.226909 [DEBUG] ftdm_io.c:5565 [s1c1][1:1] Scheduling safety hangup timer 2012-01-08 18:04:55.226909 [DEBUG] mod_freetdm.c:2416 got clear channel sig [STOP] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2846 (FreeTDM/1:1/5530) Callstate Change EARLY -> HANGUP 2012-01-08 18:04:55.226909 [NOTICE] mod_freetdm.c:2441 Hangup FreeTDM/1:1/5530 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2869 Send signal FreeTDM/1:1/5530 [KILL] 2012-01-08 18:04:55.226909 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_codec.c:141 FreeTDM/1:1/5530 Restore previous codec PCMU:0. 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2846 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change RINGING -> HANGUP 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2869 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [KILL] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [INFO] mod_dptools.c:2900 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:2285 FreeTDM/1:1/5530 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 hanging up, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:530 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP ENTER 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:605 [s1c1][1:1] Changed state from TERMINATING to HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY going to sleep 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from TERMINATING to HANGUP in 15ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:929 [s1c1][1:1] Changed state from HANGUP to HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP_COMPLETE] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP to HANGUP_COMPLETE in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:939 [s1c1][1:1] Changed state from HANGUP_COMPLETE to DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [DOWN] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP_COMPLETE to DOWN in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2930 [s1c1][1:1] DTMF debug is already disabled 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2962 [s1c1][1:1] No need to disable input dump 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2993 [s1c1][1:1] No need to disable output dump 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [RELEASED] 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:6185 Cleared call with id 1 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2735 [s1c1][1:1] channel done 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:704 -- Closed channel 1:1 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:624 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP EXIT 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 FreeTDM/1:1/5530 Standard HANGUP, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (FreeTDM/1:1/5530) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 FreeTDM/1:1/5530 Standard REPORTING, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (FreeTDM/1:1/5530) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 1 (FreeTDM/1:1/5530) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 1 (FreeTDM/1:1/5530) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel FreeTDM/1:1/5530 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (FreeTDM/1:1/5530) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (FreeTDM/1:1/5530) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 FreeTDM/1:1/5530 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY going to sleep ? Full log (including loading FreeSWITCH) at http://pastebin.com/vQUHu0pR ? freeswitch/conf/freetdm.conf ? [span zt PRI] trunk_type => T1 b-channel=1-23 d-channel=24 ? freeswitch/conf/autoload_confg/freetdm.conf.xml ? ??????? ??????????????? ??????? ??????? ??????????????? ??????????????????????? ??????????????????????? ??????????????????????? ??????????????????????? ??????????????????????? ??????????????? ??????? ? /etc/dahdi/system.conf ? loadzone = us defaultzone=us dynamic=ethmf,eth0/00:50:c2:65:d7:59/0,24,1 bchan=1-23 dchan=24 ? ? I am guessing it is a configuration issue, though this same config is currently working in production with FreeSWITCH 1.0.6 + OpenZAP + Libpri + DAHDI + foneBridge2. ? Any help is greatly appreciated. ? -Adam ? ? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/5bbf0ac2/attachment-0001.html From krice at freeswitch.org Mon Jan 9 06:00:35 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 08 Jan 2012 21:00:35 -0600 Subject: [Freeswitch-users] MPL v2 In-Reply-To: Message-ID: If I had to guess after reading the MPL2.0 tony will most likely not change to that version of the license. They changed it to be a little to close to the GPL for my liking On 1/8/12 6:53 PM, "curriegrad2004" wrote: > No, can't change the license version yet. This matter may require Tony to > comment on. > > On 2012-01-08 1:32 PM, "Pusk?s Zsolt" wrote: >> Hi. >> >> A new version of the MPL license just came out ( >> http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better >> than the last one? I'm trying to find the differences but this kind of >> text is to hard for me :) >> >> Any plans for changing FreeSWITCH license to the new one ? >> >> Zsolt >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120108/8c82348b/attachment.html From herman.griffin at gmail.com Mon Jan 9 06:05:43 2012 From: herman.griffin at gmail.com (Herman Griffin) Date: Sun, 8 Jan 2012 19:05:43 -0800 Subject: [Freeswitch-users] Need help using api_hangup_hook and session_in_hangup_hook In-Reply-To: References: Message-ID: To those that are interested, I figured out a solution to this problem through experimentation. I don't know if this is the correct way nor do I know the drawback when compared to pyrun. However, this does provide a Freeswitch Session object to my Python code. Instead of using: I used: The python module looks like this (emergency.hangup). You need a function named fsapi(session,stream,event,args): def fsapi(session,stream,event,args): pass: Herman Griffin www.hermangriffin.com On Sat, Jan 7, 2012 at 9:28 AM, Herman Griffin wrote: > Hello, > > I'm using the api_hangup_hook and session_in_hangup variables with a > pyrun. How do I access the session object from inside the python > module? > > --------------------------------- > Here are some logs : > > Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] > Dialplan: sofia/external/3105795721 at 72.37.252.18 Action > set(session_in_hangup_hook=true) > Dialplan: sofia/external/3105795721 at 72.37.252.18 Action > set(api_hangup_hook=pyrun emergency.hangup) > Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition [emergency] > . > . > 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:281 Call python script > 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:284 Finished calling > python script > 2012-01-07 09:21:57.693488 [ERR] mod_python.c:293 Error calling python script > 2012-01-07 09:21:57.693488 [ERR] mod_python.c:164 Python Error by > calling script "emergency.hangup": > Message: global name 'session' is not defined > Exception: None > > Traceback (most recent call last) > ? ? ? ?File: "/usr/local/freeswitch/scripts/python/emergency/hangup.py", > line 34, in runtime > > > > ---------------------------------------------------------- > Here is the simple script > > python module emergency.hangup: > > from freeswitch import * > > def runtime(arg): > ? ?consoleLog("info", print(dir(session))) > > > > Thanks, > Herman From olimonkey at gmail.com Mon Jan 9 06:19:38 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Mon, 9 Jan 2012 11:19:38 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Message-ID: I had this in my call string: {sip_cid_type=rpid,ignore_early_media=true} Which caused the 503 Service Unavailble error. The CISCO had some errors just prior to the 503 message: Jan 9 01:30:36: //152/4620425F81D5/SIP/Info/ccsip_indicate_rt_packet_stats: Processing stats for callid=152, proc_id=9 Jan 9 01:30:38: //152/4620425F81D5/SIP/Media/sipSPIUpdateRtpSession: stun is disabled for stream:47110C14 Jan 9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: Requesting stats for callid=152 Jan 9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: Stats request failed for callid=152, dstCallID=153, rc=-7 Jan 9 01:30:38: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT Jan 9 01:30:38: //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: ccsip_spi_get_msg_type returned: 3 for event 7 Jan 9 01:30:38: //152/4620425F81D5/SIP/Info/act_recdinvite_disconnect: Performing disconnect Something to do with "Stats request failed for callid". Which then causes a "SIPSPI_EV_CC_CALL_DISCONNECT" event. So what I did is I removed "sip_cid_type=rpid" from my call string. Now, when I pick up the phone I just get a bit of noise, but nothing else. The CISCO does not send any SIP messages at all. It's as if it doesn't even know I picked up the phone. Does that mean the dualtone answer supervision setting in the CISCO is not working? After I hang up the phone I get: CANCEL from FS (probably due to call timeout) then the CISCO sends OK then 487 Request Cancelled then ACK from FS Output in FS just prior to my phone ringing is: 2012-01-09 11:16:03.637084 [NOTICE] switch_channel.c:816 New Channel sofia/internal/1092122856 at 192.168.255.1 [042f895c-4a34-4295-b250-0291a0f8b91b] 2012-01-09 11:16:07.128527 [INFO] sofia.c:740 sofia/internal/1092122856 at 192.168.255.1 Update Callee ID to "Outbound Call" <1092122856> 2012-01-09 11:16:07.128527 [NOTICE] sofia_glue.c:3793 Pre-Answer sofia/internal/1092122856 at 192.168.255.1! When I pick up the phone there is obviously no further output because the CISCO hasn't detected the pickup. Hmmmm!!! Still doing my head in. Anyone? lol On Mon, Jan 9, 2012 at 9:33 AM, Oliver Schenk wrote: > Alright, so I tried to set ignore_early_media=true, but the phone > isn't being answered at all anymore. So i went back to the CISCO and > turned on debugging. I'm now getting a different sequence of events. > > > I still get: > > INVITE from FS > 100 TRYING from CISCO > 183 SESSION PROGRESS from CISCO > > then, when I pick up the phone I get: > > 503 SERVICE UNAVAILABLE from CISCO > ACK from FS > > > I wonder what's going on now. > > I'll have to play around with the "supervisiory answer dual-tone". > > > > CISCO voice-port config: > > > voice-port 0/3/0 > ?supervisory disconnect dualtone mid-call > ?supervisory answer dualtone > ?output attenuation -3 > ?no comfort-noise > ?cptone AU > ?timeouts call-disconnect 5 > ?timeouts wait-release 5 > ?connection plar opx 1001 > ?impedance complex1 > ?caller-id enable > > > > Cheers, > > Oliver > > > > On Sat, Jan 7, 2012 at 9:28 AM, Oliver Schenk wrote: >> No, at the time I was no longer using "ignore_early_media= true", >> because initially it didn't work. >> I was actually thinking of putting this property back in again, thanks >> a lot! I'll try it first thing Monday. >> >> >> With regard to EXECUTE log line, either I don't have the right level >> of logging turned on or something else is going on, because I've never >> seen any such log entries before with my managed application; yet my >> IVR app definitely does get executed. I don't think I have the DEBUG >> logging output level turned on so that could explain it... >> >> >> Thanks all, >> >> Oliver >> >> >> >> On Fri, Jan 6, 2012 at 11:21 PM, Peter Olsson >> wrote: >>> Are you still using ignore_early_media=true - this must be set for this to work correctly. >>> >>> You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. >>> >>> Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" >>> >>> Park application is only executed after the call was answered. >>> >>> /Peter >>> >>> ________________________________________ >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>> Skickat: den 6 januari 2012 12:04 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>> >>> Because I'm using an FXO card with voice, I added something to my >>> CISCO conf. Many others had the same thing: >>> >>> >>> voice-port 0/3/0 >>> ? ... >>> ? supervisory disconnect dualtone mid-call >>> ? supervisory answer dualtone ? ?<---- ADDED THIS ONE >>> ? ... >>> >>> >>> >>> Once I added this, the FS output now just showed the following while >>> the phone was ringing: >>> >>> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >>> sofia/internal/109212xxxx at 192.168.x.x >>> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >>> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >>> Call" <109212xxxx> >>> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >>> sofia/internal/109212xxxx at 192.168.x.x! >>> >>> >>> Where as previous it would show the above and also show the following: >>> >>> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >>> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >>> <0000000000> to "Outbound Call" <109212xxxx> >>> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >>> <1092122856> >>> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >>> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >>> >>> >>> >>> BUT, the IVR still started playing even before I pick up the phone. >>> Hmmmm.....so why is FS still starting the managed application when the >>> call has not been answered yet. Are we all sure that the managed >>> application should not be executed until the call "has been answered" >>> shows up in the log file? >>> >>> >>> Will have to keep testing on monday as I don't have access to the >>> CISCO from where i am now. I'll have to see whether the CISCO changes >>> had any impact on the times at which the SIP messages are sent back >>> and forth. Especially the 200 OK message. >>> >>> >>> Thanks again for help, maybe getting somewhere now...... >>> >>> Oliver >>> >>> >>> >>> >>> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >>> wrote: >>>> If it sends 200 OK right after 183, this IS the problem. >>>> >>>> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. >>>> >>>> Seems to me there is something broken in the Cisco. >>>> >>>> /Peter >>>> >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>>> Skickat: den 6 januari 2012 06:55 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>>> >>>> I've tried looking at disable-early-media configuration command, but >>>> that didn't work and I doubt that has anything to do with the CISCO >>>> sending a 200 OK right after a 183 SESSION PROGRESS. >>>> >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >>>>> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >>>>> is usually RINGING (generate ringback locally) while a 183 has media... aka >>>>> early media and usually provides ringback inband. >>>>> >>>>> /b >>>>> >>>>> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>>>> >>>>> Shouldn't there be a ?180 RINGING ?somewhere in there? >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >>>>> >>>>> I just noticed something else, if I don't pick up the phone at all. >>>>> >>>>> The IVR just keeps playing until the menu timeout kicks in. >>>>> >>>>> >>>>> So here is a CISCO SIP log: >>>>> >>>>> http://pastebin.com/Y9sYkuxi >>>>> >>>>> >>>>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>>>> >>>>> I hope the CISCO log is readable, it's a bit long because I just did >>>>> >>>>> "debug ccsip all". >>>>> >>>>> >>>>> >>>>> >>>>> In this test I didn't bother picking up the phone at all, but I can >>>>> >>>>> see that FS answered anyway and the IVR kept playing until it timed >>>>> >>>>> out. >>>>> >>>>> I'm not an expert, but here is what I picked out of it: >>>>> >>>>> >>>>> At 00:08:10 we get a >>>>> >>>>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>>>> >>>>> >>>>> the further down at the same timestamp we get >>>>> >>>>> Sent: "SIP/2.0 100 Trying" >>>>> >>>>> >>>>> At 00:08:13 we get a >>>>> >>>>> Sent: "SIP/2.0 183 Session Progress" >>>>> >>>>> >>>>> At 00:18:13 we get a >>>>> >>>>> Sent: "SIP/2.0 200 OK" >>>>> >>>>> >>>>> Then at the same timestamp we get: >>>>> >>>>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>> >>>>> >>>>> >>>>> >>>>> Once the IVR times out at 00:09:16 we get >>>>> >>>>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>> >>>>> >>>>> And then the reply right after >>>>> >>>>> Sent: "SIP/2.0 200 OK" >>>>> >>>>> >>>>> >>>>> >>>>> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>>>> >>>>> after the "INVITE" is received. >>>>> >>>>> >>>>> >>>>> >>>>> The part that is beyond my field of expertise so far is WHY? >>>>> >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> >>>>> >>>>> >>>>> Oliver >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>>>> >>>>> By the way: >>>>> >>>>> >>>>> I tried {ignore_early_media=true} as well, but as I think we >>>>> >>>>> determined, my problem is probably with the CISCO telling FS that the >>>>> >>>>> call has been answered when really it hasn't yet. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>>>> >>>>> Thanks for the help so far. >>>>> >>>>> >>>>> >>>>> Here is a pastebin of FreeSWITCH output: >>>>> >>>>> http://pastebin.com/i6Qgc7ws >>>>> >>>>> >>>>> Notice how the "has been answered" log message comes immediately >>>>> >>>>> (within a few milliseconds) after the call was originated. I think >>>>> >>>>> this would suggest that the CISCO is immediately sending a 200 OK, as >>>>> >>>>> you suggested. I also turned on CISCO debugging, but I'm just trying >>>>> >>>>> to figure out how to get the information regarding SIP messages back >>>>> >>>>> to Freeswitch. I'll run the test again and see if I can get some >>>>> >>>>> useful CISCO debug. >>>>> >>>>> >>>>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>>>> >>>>> SIP debugging? >>>>> >>>>> >>>>> >>>>> Thanks! >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> 2012/1/6 Gustavo M?rsico : >>>>> >>>>> I think I've a similar problem related to callcenter app. When I made an >>>>> originate like this: >>>>> >>>>> >>>>> originate loopback/2500/default/XML &bridge(user/2001) >>>>> >>>>> >>>>> 2500 is an extension that leads to a callcenter application. In this case, >>>>> we dial first to the queue and when an agent answered we call to the >>>>> customer. As far as I know >>>>> >>>>> When the A-leg reaches to the queue, without selecting an agent, the call is >>>>> automatically sent to the B-leg. As far as I see, there is a pre-answer >>>>> method that fs needs to send the media to A-leg. >>>>> >>>>> In order to try to avoid this, I tried using ignore_early_media=true as part >>>>> of the originate in A-leg and/or B-leg, with no luck. >>>>> >>>>> >>>>> originate {ignore_early_media=true}loopback/2500/default/XML >>>>> &bridge({ignore_early_media=true}user/2001) >>>>> >>>>> >>>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>>>> destination_number(2500) =~ /^(2500)$/ break=on-false >>>>> >>>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>>> >>>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>>>> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >>>>> loopback/2500-b [BREAK] >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>> CHANNEL KILL >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>>>> (loopback/2500-b) State ROUTING going to sleep >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>>>> (loopback/2500-b) Running State Change CS_EXECUTE >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>>>> (loopback/2500-b) State EXECUTE >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>>>> CHANNEL EXECUTE >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>>>> loopback/2500-b Standard EXECUTE >>>>> >>>>> EXECUTE loopback/2500-b set(open=true) >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>> [open]=[true] >>>>> >>>>> EXECUTE loopback/2500-b >>>>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>> >>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>>> >>>>> EXECUTE loopback/2500-b >>>>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>> >>>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>>> >>>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>> [ignore_early_media]=[true] >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >>>>> callcenter Requires media! pre_answering channel loopback/2500-b >>>>> >>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>>>> loopback/2500-a! >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >>>>> Callstate Change RINGING -> EARLY >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>> loopback/2500-b [BREAK] >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>> CHANNEL KILL >>>>> >>>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >>>>> loopback/2500-b! >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >>>>> Callstate Change RINGING -> EARLY >>>>> >>>>> EXECUTE loopback/2500-b callcenter(click2call) >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >>>>> Callstate Change EARLY -> ACTIVE >>>>> >>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>>>> [loopback/2500-a] has been answered >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>> loopback/2500-b [BREAK] >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>> CHANNEL KILL >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >>>>> Resulted in Success: [loopback/2500-a] >>>>> >>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >>>>> Callstate Change EARLY -> ACTIVE >>>>> >>>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >>>>> Flipping CID from "" <0000000000> to "Outbound Call" >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>>> >>>>> >>>>> Also, maybe I should be doing something like this: >>>>> >>>>> >>>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>>> >>>>> >>>>> instead of: >>>>> >>>>> >>>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>>> >>>>> >>>>> >>>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>>> >>>>> know how really...probably not on the right track there. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>> >>>>> *bump* >>>>> >>>>> >>>>> >>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>> >>>>> call string I'm creating a connection directly from the CISCO >>>>> >>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>> >>>>> playing straight away? >>>>> >>>>> >>>>> Maybe I should be originating a call first and then only once I know >>>>> >>>>> the other side has picked up will I bridge the call to the IVR managed >>>>> >>>>> application. >>>>> >>>>> >>>>> Problem is I dunno how to tell whether the other person has picked up >>>>> >>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>> >>>>> things to a call once it has been established. >>>>> >>>>> >>>>> >>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>> >>>>> out of it there. >>>>> >>>>> >>>>> >>>>> Cheers >>>>> >>>>> >>>>> Oliver >>>>> >>>>> >>>>> >>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>> >>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>> >>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>> >>>>> but I am having a few issues with the way the system answers calls (or >>>>> >>>>> doesn't answer calls...). >>>>> >>>>> >>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>> >>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>> >>>>> which is then connected to a POTS phone line. >>>>> >>>>> >>>>> >>>>> Take the following scenario: >>>>> >>>>> >>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>> >>>>> to freeswitch and originate a call: >>>>> >>>>> >>>>> string callstring = >>>>> >>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>> >>>>> '&managed(ivrAppName)'"; >>>>> >>>>> eslConnection.API("originate", callstring); >>>>> >>>>> >>>>> where 192.168.x.x is the CISCO IP. >>>>> >>>>> >>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>> >>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>> >>>>> number (091234567) to make the call. >>>>> >>>>> >>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>> >>>>> >>>>> >>>>> >>>>> These are my current problems: >>>>> >>>>> >>>>> - IVR starts playing before I even pick up the phone. This means that >>>>> >>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>> >>>>> IVR will start playing and eventually the mobile phone will divert to >>>>> >>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>> >>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>> >>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>> >>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>> >>>>> has not yet been answered. For some reason however as soon as the >>>>> >>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>> >>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>> >>>>> doing originate the wrong way or something ... >>>>> >>>>> >>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>> >>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>> >>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>> >>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>> >>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>> >>>>> starts playing even if no one answers the phone. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> CISCO Config for relevant FXO port: >>>>> >>>>> >>>>> voice service voip >>>>> >>>>> ?allow-connections h323 to h323 >>>>> >>>>> ?allow-connections h323 to sip >>>>> >>>>> ?allow-connections sip to h323 >>>>> >>>>> ?allow-connections sip to sip >>>>> >>>>> ?no supplementary-service h450.2 >>>>> >>>>> ?no supplementary-service h450.3 >>>>> >>>>> ?supplementary-service h450.12 >>>>> >>>>> ?no supplementary-service sip moved-temporarily >>>>> >>>>> ?no supplementary-service sip refer >>>>> >>>>> ?fax protocol cisco >>>>> >>>>> ?sip >>>>> >>>>> ?registrar server expires max 3600 min 3600 >>>>> >>>>> ?no update-callerid >>>>> >>>>> ?no call service stop >>>>> >>>>> >>>>> voice-port 0/3/2 >>>>> >>>>> ?output attenuation -3 >>>>> >>>>> ?no comfort-noise >>>>> >>>>> ?cptone AU >>>>> >>>>> ?impedance complex1 >>>>> >>>>> ?caller-id enable >>>>> >>>>> ! >>>>> >>>>> dial-peer voice 100 pots >>>>> >>>>> ?preference 1 >>>>> >>>>> ?destination-pattern 1T >>>>> >>>>> ?port 0/3/2 >>>>> >>>>> ! >>>>> >>>>> >>>>> >>>>> >>>>> Many Thanks, >>>>> >>>>> >>>>> Oliver >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> http://wiki.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> >>>>> consulting at freeswitch.org >>>>> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> http://wiki.freeswitch.org >>>>> >>>>> http://www.cluecon.com >>>>> >>>>> >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> -- >>>>> Brian West >>>>> FreeSWITCH Solutions, LLC >>>>> Phone: +1 (918) 420-9266 >>>>> Fax: ? +1 (918) 420-9267 >>>>> brian at freeswitch.org >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:4f06d49b32762089563979! >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org From curriegrad2004 at gmail.com Mon Jan 9 09:15:36 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 8 Jan 2012 22:15:36 -0800 Subject: [Freeswitch-users] MPL v2 In-Reply-To: References: Message-ID: Unfortunately in the MPL 1.1, the license allows them to use a later version of the license if they choose. However with the current situation of the files, it's at MPL 1.1. And I hope they leave it at MPL 1.1. However for the contributions, I beg to differ... On Sun, Jan 8, 2012 at 7:00 PM, Ken Rice wrote: > If I had to guess after reading the MPL2.0 tony will most likely not change > to that version of the license. > > They changed it to be a little to close to the GPL for my liking > > > > On 1/8/12 6:53 PM, "curriegrad2004" wrote: > > No, can't change the license version yet. This matter may require Tony to > comment on. > > On 2012-01-08 1:32 PM, "Pusk?s Zsolt" wrote: > > Hi. > > A new version of the MPL license just came out ( > http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better > than the last one? I'm trying to find the differences but this kind of > text is to hard for me :) > > Any plans for changing FreeSWITCH license to the new one ? > > Zsolt > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From olimonkey at gmail.com Mon Jan 9 09:25:03 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Mon, 9 Jan 2012 14:25:03 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Message-ID: Ok, this is starting to become 100% CISCO issue so I might have to move this topic to a CISCO forum instead. I played around with a few CISCO parameters. I added this to the voice port: voice-port 0/3/2 supervisory disconnect dualtone mid-call supervisory answer dualtone no battery-reversal Funny thing happens now. When I answer the call I have to make a noise into the phone before the CISCO actually picks up! I just had to shake my head there...and start bashing my head against the keyboard. Wouldn't that be funny... Instruction Manual: "if you hear nothing, please snap your fingers into the phone receiver and you'll hear the IVR start playing." So anyway, I tried to set: supervisory answer dualtone to supervisory answer dualtone sensitivity high That didn't make any difference. So next I added: no comfort-noise That didn't make any difference. Then I tried: battery-reversal answer As soon as I do that the CISCO once again causes FS to start playing the IVR before I even pick up the phone. So back to square one. Starting to run out of options here... On Mon, Jan 9, 2012 at 11:19 AM, Oliver Schenk wrote: > I had this in my call string: > > {sip_cid_type=rpid,ignore_early_media=true} > > > Which caused the 503 Service Unavailble error. The CISCO had some > errors just prior to the 503 message: > > Jan ?9 01:30:36: > //152/4620425F81D5/SIP/Info/ccsip_indicate_rt_packet_stats: Processing > stats for callid=152, proc_id=9 > Jan ?9 01:30:38: //152/4620425F81D5/SIP/Media/sipSPIUpdateRtpSession: > stun is disabled for stream:47110C14 > Jan ?9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: > Requesting stats for callid=152 > Jan ?9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: > Stats request failed for callid=152, dstCallID=153, rc=-7 > Jan ?9 01:30:38: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued > event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT > Jan ?9 01:30:38: > //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: > ccsip_spi_get_msg_type returned: 3 for event 7 > Jan ?9 01:30:38: > //152/4620425F81D5/SIP/Info/act_recdinvite_disconnect: Performing > disconnect > > > > Something to do with "Stats request failed for callid". Which then > causes a "SIPSPI_EV_CC_CALL_DISCONNECT" event. > > > > So what I did is I removed "sip_cid_type=rpid" from my call string. > > > Now, when I pick up the phone I just get a bit of noise, but nothing > else. The CISCO does not send any SIP messages at all. It's as if it > doesn't even know I picked up the phone. Does that mean the dualtone > answer supervision setting in the CISCO is not working? > > > > After I hang up the phone I get: > > > CANCEL from FS ?(probably due to call timeout) > then the CISCO sends OK > then 487 Request Cancelled > then ACK from FS > > > Output in FS just prior to my phone ringing is: > > 2012-01-09 11:16:03.637084 [NOTICE] switch_channel.c:816 New Channel > sofia/internal/1092122856 at 192.168.255.1 > [042f895c-4a34-4295-b250-0291a0f8b91b] > 2012-01-09 11:16:07.128527 [INFO] sofia.c:740 > sofia/internal/1092122856 at 192.168.255.1 Update Callee ID to "Outbound > Call" <1092122856> > 2012-01-09 11:16:07.128527 [NOTICE] sofia_glue.c:3793 Pre-Answer > sofia/internal/1092122856 at 192.168.255.1! > > When I pick up the phone there is obviously no further output because > the CISCO hasn't detected the pickup. > > > Hmmmm!!! Still doing my head in. > > > Anyone? lol > > > > > On Mon, Jan 9, 2012 at 9:33 AM, Oliver Schenk wrote: >> Alright, so I tried to set ignore_early_media=true, but the phone >> isn't being answered at all anymore. So i went back to the CISCO and >> turned on debugging. I'm now getting a different sequence of events. >> >> >> I still get: >> >> INVITE from FS >> 100 TRYING from CISCO >> 183 SESSION PROGRESS from CISCO >> >> then, when I pick up the phone I get: >> >> 503 SERVICE UNAVAILABLE from CISCO >> ACK from FS >> >> >> I wonder what's going on now. >> >> I'll have to play around with the "supervisiory answer dual-tone". >> >> >> >> CISCO voice-port config: >> >> >> voice-port 0/3/0 >> ?supervisory disconnect dualtone mid-call >> ?supervisory answer dualtone >> ?output attenuation -3 >> ?no comfort-noise >> ?cptone AU >> ?timeouts call-disconnect 5 >> ?timeouts wait-release 5 >> ?connection plar opx 1001 >> ?impedance complex1 >> ?caller-id enable >> >> >> >> Cheers, >> >> Oliver >> >> >> >> On Sat, Jan 7, 2012 at 9:28 AM, Oliver Schenk wrote: >>> No, at the time I was no longer using "ignore_early_media= true", >>> because initially it didn't work. >>> I was actually thinking of putting this property back in again, thanks >>> a lot! I'll try it first thing Monday. >>> >>> >>> With regard to EXECUTE log line, either I don't have the right level >>> of logging turned on or something else is going on, because I've never >>> seen any such log entries before with my managed application; yet my >>> IVR app definitely does get executed. I don't think I have the DEBUG >>> logging output level turned on so that could explain it... >>> >>> >>> Thanks all, >>> >>> Oliver >>> >>> >>> >>> On Fri, Jan 6, 2012 at 11:21 PM, Peter Olsson >>> wrote: >>>> Are you still using ignore_early_media=true - this must be set for this to work correctly. >>>> >>>> You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. >>>> >>>> Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" >>>> >>>> Park application is only executed after the call was answered. >>>> >>>> /Peter >>>> >>>> ________________________________________ >>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>>> Skickat: den 6 januari 2012 12:04 >>>> Till: FreeSWITCH Users Help >>>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>>> >>>> Because I'm using an FXO card with voice, I added something to my >>>> CISCO conf. Many others had the same thing: >>>> >>>> >>>> voice-port 0/3/0 >>>> ? ... >>>> ? supervisory disconnect dualtone mid-call >>>> ? supervisory answer dualtone ? ?<---- ADDED THIS ONE >>>> ? ... >>>> >>>> >>>> >>>> Once I added this, the FS output now just showed the following while >>>> the phone was ringing: >>>> >>>> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >>>> sofia/internal/109212xxxx at 192.168.x.x >>>> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >>>> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >>>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >>>> Call" <109212xxxx> >>>> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >>>> sofia/internal/109212xxxx at 192.168.x.x! >>>> >>>> >>>> Where as previous it would show the above and also show the following: >>>> >>>> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >>>> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >>>> <0000000000> to "Outbound Call" <109212xxxx> >>>> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >>>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >>>> <1092122856> >>>> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >>>> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >>>> >>>> >>>> >>>> BUT, the IVR still started playing even before I pick up the phone. >>>> Hmmmm.....so why is FS still starting the managed application when the >>>> call has not been answered yet. Are we all sure that the managed >>>> application should not be executed until the call "has been answered" >>>> shows up in the log file? >>>> >>>> >>>> Will have to keep testing on monday as I don't have access to the >>>> CISCO from where i am now. I'll have to see whether the CISCO changes >>>> had any impact on the times at which the SIP messages are sent back >>>> and forth. Especially the 200 OK message. >>>> >>>> >>>> Thanks again for help, maybe getting somewhere now...... >>>> >>>> Oliver >>>> >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >>>> wrote: >>>>> If it sends 200 OK right after 183, this IS the problem. >>>>> >>>>> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. >>>>> >>>>> Seems to me there is something broken in the Cisco. >>>>> >>>>> /Peter >>>>> >>>>> ________________________________________ >>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>>>> Skickat: den 6 januari 2012 06:55 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>>>> >>>>> I've tried looking at disable-early-media configuration command, but >>>>> that didn't work and I doubt that has anything to do with the CISCO >>>>> sending a 200 OK right after a 183 SESSION PROGRESS. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >>>>>> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >>>>>> is usually RINGING (generate ringback locally) while a 183 has media... aka >>>>>> early media and usually provides ringback inband. >>>>>> >>>>>> /b >>>>>> >>>>>> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>>>>> >>>>>> Shouldn't there be a ?180 RINGING ?somewhere in there? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >>>>>> >>>>>> I just noticed something else, if I don't pick up the phone at all. >>>>>> >>>>>> The IVR just keeps playing until the menu timeout kicks in. >>>>>> >>>>>> >>>>>> So here is a CISCO SIP log: >>>>>> >>>>>> http://pastebin.com/Y9sYkuxi >>>>>> >>>>>> >>>>>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>>>>> >>>>>> I hope the CISCO log is readable, it's a bit long because I just did >>>>>> >>>>>> "debug ccsip all". >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> In this test I didn't bother picking up the phone at all, but I can >>>>>> >>>>>> see that FS answered anyway and the IVR kept playing until it timed >>>>>> >>>>>> out. >>>>>> >>>>>> I'm not an expert, but here is what I picked out of it: >>>>>> >>>>>> >>>>>> At 00:08:10 we get a >>>>>> >>>>>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>>>>> >>>>>> >>>>>> the further down at the same timestamp we get >>>>>> >>>>>> Sent: "SIP/2.0 100 Trying" >>>>>> >>>>>> >>>>>> At 00:08:13 we get a >>>>>> >>>>>> Sent: "SIP/2.0 183 Session Progress" >>>>>> >>>>>> >>>>>> At 00:18:13 we get a >>>>>> >>>>>> Sent: "SIP/2.0 200 OK" >>>>>> >>>>>> >>>>>> Then at the same timestamp we get: >>>>>> >>>>>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Once the IVR times out at 00:09:16 we get >>>>>> >>>>>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>>> >>>>>> >>>>>> And then the reply right after >>>>>> >>>>>> Sent: "SIP/2.0 200 OK" >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>>>>> >>>>>> after the "INVITE" is received. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> The part that is beyond my field of expertise so far is WHY? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Thanks, >>>>>> >>>>>> >>>>>> >>>>>> Oliver >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>>>>> >>>>>> By the way: >>>>>> >>>>>> >>>>>> I tried {ignore_early_media=true} as well, but as I think we >>>>>> >>>>>> determined, my problem is probably with the CISCO telling FS that the >>>>>> >>>>>> call has been answered when really it hasn't yet. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>>>>> >>>>>> Thanks for the help so far. >>>>>> >>>>>> >>>>>> >>>>>> Here is a pastebin of FreeSWITCH output: >>>>>> >>>>>> http://pastebin.com/i6Qgc7ws >>>>>> >>>>>> >>>>>> Notice how the "has been answered" log message comes immediately >>>>>> >>>>>> (within a few milliseconds) after the call was originated. I think >>>>>> >>>>>> this would suggest that the CISCO is immediately sending a 200 OK, as >>>>>> >>>>>> you suggested. I also turned on CISCO debugging, but I'm just trying >>>>>> >>>>>> to figure out how to get the information regarding SIP messages back >>>>>> >>>>>> to Freeswitch. I'll run the test again and see if I can get some >>>>>> >>>>>> useful CISCO debug. >>>>>> >>>>>> >>>>>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>>>>> >>>>>> SIP debugging? >>>>>> >>>>>> >>>>>> >>>>>> Thanks! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> 2012/1/6 Gustavo M?rsico : >>>>>> >>>>>> I think I've a similar problem related to callcenter app. When I made an >>>>>> originate like this: >>>>>> >>>>>> >>>>>> originate loopback/2500/default/XML &bridge(user/2001) >>>>>> >>>>>> >>>>>> 2500 is an extension that leads to a callcenter application. In this case, >>>>>> we dial first to the queue and when an agent answered we call to the >>>>>> customer. As far as I know >>>>>> >>>>>> When the A-leg reaches to the queue, without selecting an agent, the call is >>>>>> automatically sent to the B-leg. As far as I see, there is a pre-answer >>>>>> method that fs needs to send the media to A-leg. >>>>>> >>>>>> In order to try to avoid this, I tried using ignore_early_media=true as part >>>>>> of the originate in A-leg and/or B-leg, with no luck. >>>>>> >>>>>> >>>>>> originate {ignore_early_media=true}loopback/2500/default/XML >>>>>> &bridge({ignore_early_media=true}user/2001) >>>>>> >>>>>> >>>>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>>>>> destination_number(2500) =~ /^(2500)$/ break=on-false >>>>>> >>>>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>>>> >>>>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>>>>> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >>>>>> loopback/2500-b [BREAK] >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>> CHANNEL KILL >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>>>>> (loopback/2500-b) State ROUTING going to sleep >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>>>>> (loopback/2500-b) Running State Change CS_EXECUTE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>>>>> (loopback/2500-b) State EXECUTE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>>>>> CHANNEL EXECUTE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>>>>> loopback/2500-b Standard EXECUTE >>>>>> >>>>>> EXECUTE loopback/2500-b set(open=true) >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>> [open]=[true] >>>>>> >>>>>> EXECUTE loopback/2500-b >>>>>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>>> >>>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>>>> >>>>>> EXECUTE loopback/2500-b >>>>>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>>> >>>>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>>>> >>>>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>> [ignore_early_media]=[true] >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >>>>>> callcenter Requires media! pre_answering channel loopback/2500-b >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>>>>> loopback/2500-a! >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >>>>>> Callstate Change RINGING -> EARLY >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>>> loopback/2500-b [BREAK] >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>> CHANNEL KILL >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >>>>>> loopback/2500-b! >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >>>>>> Callstate Change RINGING -> EARLY >>>>>> >>>>>> EXECUTE loopback/2500-b callcenter(click2call) >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >>>>>> Callstate Change EARLY -> ACTIVE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>>>>> [loopback/2500-a] has been answered >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>>> loopback/2500-b [BREAK] >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>> CHANNEL KILL >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >>>>>> Resulted in Success: [loopback/2500-a] >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >>>>>> Callstate Change EARLY -> ACTIVE >>>>>> >>>>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >>>>>> Flipping CID from "" <0000000000> to "Outbound Call" >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>>>> >>>>>> >>>>>> Also, maybe I should be doing something like this: >>>>>> >>>>>> >>>>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>>>> >>>>>> >>>>>> instead of: >>>>>> >>>>>> >>>>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>>>> >>>>>> >>>>>> >>>>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>>>> >>>>>> know how really...probably not on the right track there. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>>> >>>>>> *bump* >>>>>> >>>>>> >>>>>> >>>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>>> >>>>>> call string I'm creating a connection directly from the CISCO >>>>>> >>>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>>> >>>>>> playing straight away? >>>>>> >>>>>> >>>>>> Maybe I should be originating a call first and then only once I know >>>>>> >>>>>> the other side has picked up will I bridge the call to the IVR managed >>>>>> >>>>>> application. >>>>>> >>>>>> >>>>>> Problem is I dunno how to tell whether the other person has picked up >>>>>> >>>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>>> >>>>>> things to a call once it has been established. >>>>>> >>>>>> >>>>>> >>>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>>> >>>>>> out of it there. >>>>>> >>>>>> >>>>>> >>>>>> Cheers >>>>>> >>>>>> >>>>>> Oliver >>>>>> >>>>>> >>>>>> >>>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>>> >>>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>>> >>>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>>> >>>>>> but I am having a few issues with the way the system answers calls (or >>>>>> >>>>>> doesn't answer calls...). >>>>>> >>>>>> >>>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>>> >>>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>>> >>>>>> which is then connected to a POTS phone line. >>>>>> >>>>>> >>>>>> >>>>>> Take the following scenario: >>>>>> >>>>>> >>>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>>> >>>>>> to freeswitch and originate a call: >>>>>> >>>>>> >>>>>> string callstring = >>>>>> >>>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>>> >>>>>> '&managed(ivrAppName)'"; >>>>>> >>>>>> eslConnection.API("originate", callstring); >>>>>> >>>>>> >>>>>> where 192.168.x.x is the CISCO IP. >>>>>> >>>>>> >>>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>>> >>>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>>> >>>>>> number (091234567) to make the call. >>>>>> >>>>>> >>>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> These are my current problems: >>>>>> >>>>>> >>>>>> - IVR starts playing before I even pick up the phone. This means that >>>>>> >>>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>>> >>>>>> IVR will start playing and eventually the mobile phone will divert to >>>>>> >>>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>>> >>>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>>> >>>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>>> >>>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>>> >>>>>> has not yet been answered. For some reason however as soon as the >>>>>> >>>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>>> >>>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>>> >>>>>> doing originate the wrong way or something ... >>>>>> >>>>>> >>>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>>> >>>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>>> >>>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>>> >>>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>>> >>>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>>> >>>>>> starts playing even if no one answers the phone. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> CISCO Config for relevant FXO port: >>>>>> >>>>>> >>>>>> voice service voip >>>>>> >>>>>> ?allow-connections h323 to h323 >>>>>> >>>>>> ?allow-connections h323 to sip >>>>>> >>>>>> ?allow-connections sip to h323 >>>>>> >>>>>> ?allow-connections sip to sip >>>>>> >>>>>> ?no supplementary-service h450.2 >>>>>> >>>>>> ?no supplementary-service h450.3 >>>>>> >>>>>> ?supplementary-service h450.12 >>>>>> >>>>>> ?no supplementary-service sip moved-temporarily >>>>>> >>>>>> ?no supplementary-service sip refer >>>>>> >>>>>> ?fax protocol cisco >>>>>> >>>>>> ?sip >>>>>> >>>>>> ?registrar server expires max 3600 min 3600 >>>>>> >>>>>> ?no update-callerid >>>>>> >>>>>> ?no call service stop >>>>>> >>>>>> >>>>>> voice-port 0/3/2 >>>>>> >>>>>> ?output attenuation -3 >>>>>> >>>>>> ?no comfort-noise >>>>>> >>>>>> ?cptone AU >>>>>> >>>>>> ?impedance complex1 >>>>>> >>>>>> ?caller-id enable >>>>>> >>>>>> ! >>>>>> >>>>>> dial-peer voice 100 pots >>>>>> >>>>>> ?preference 1 >>>>>> >>>>>> ?destination-pattern 1T >>>>>> >>>>>> ?port 0/3/2 >>>>>> >>>>>> ! >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Many Thanks, >>>>>> >>>>>> >>>>>> Oliver >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> >>>>>> consulting at freeswitch.org >>>>>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> http://wiki.freeswitch.org >>>>>> >>>>>> http://www.cluecon.com >>>>>> >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> -- >>>>>> Brian West >>>>>> FreeSWITCH Solutions, LLC >>>>>> Phone: +1 (918) 420-9266 >>>>>> Fax: ? +1 (918) 420-9267 >>>>>> brian at freeswitch.org >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> !DSPAM:4f06d49b32762089563979! >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org From fieldpeak at gmail.com Mon Jan 9 09:50:34 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Mon, 9 Jan 2012 14:50:34 +0800 Subject: [Freeswitch-users] Dynamic specify the outbound GW within source code In-Reply-To: References: Message-ID: Hi Avi, Thanks so much for your kindly reply. Actually, now i'm using mod_nibble for billing, i write a function "check_billing_before_routing" in nibble_state_handler, in this func("check_billing_before_routing"), it will call an external command, this command will query the backend database if the caller has enough money to contiue the call, the mod_nibblebill will contiue the call or hangup the call according to the result of the external command. i have realize all above, it works well. switch_state_handler_table_t nibble_state_handler = { /* on_init */ NULL, /* on_routing */ check_billing_before_routing, /* Need to add a check here for anything in their account before routing */ /* on_execute */ sched_billing, /* Turn on heartbeat for this session and do an initial account check */ /* on_hangup */ process_hangup, /* On hangup - most important place to go bill */ /* on_exch_media */ NULL, /* on_soft_exec */ NULL, /* on_consume_med */ NULL, /* on_hibernate */ NULL, /* on_reset */ NULL, /* on_park */ NULL, /* on_reporting */ NULL, /* on_destroy */ NULL }; For PSTN call, i use dial plan below, "1.2.3.4" is the PSTN-GW Now, as we add one more PSTN-GW for outbound call, and the FS have to route call to the specific GW accoring to result of the external command (the external command will return the IP address of GW as well), i can think out the FS own function like "switch_channel_set_variable(channel, "caller_id_number")" can configure the value of variable, however, what variable should i use for this case, could you please advise, thank you very much! Regards, Charels 2012/1/8 Avi Marcus > I'm not quite sure of the use case. Do any of these help? > 1) specify a server, not an IP, and then let DNS determine where it goes. > 2) use a small lua script to set the channel variable based on whatever > you need - an sql query, some logic.. and then use that variable in the > bridge string. > > Those help? If not, please explain more what problem you are trying to > solve. > > -Avi > > > On Sun, Jan 8, 2012 at 3:34 PM, fieldpeak wrote: > >> Dear friends, >> >> i have FS for PSTN outbound call using below dial plan, >> >> >> >> >> >> >> >> While, now i need dynamically specify the outbound GW?s IP address >> according to the return result of the external command before routing in >> the source code , e.g. if the external command return FS the IP address of >> OB GW 6.7.8.9, then >> >> >> >> however, i don't know which function i should call within the source code >> to realize it, could anybody help to advise, >> >> P.S. i know there is existing module ?mod_xml_curl? can realize similar >> function, however, I could not use it for this case? >> >> >> thanks a lot! >> >> Regards, >> Charles >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/53e1176d/attachment.html From peter.olsson at visionutveckling.se Mon Jan 9 09:58:52 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 9 Jan 2012 06:58:52 +0000 Subject: [Freeswitch-users] error loading module dll open error /sym error In-Reply-To: <1326075984.52157.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: , <1326075984.52157.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: <12ED64F5-4310-485B-B443-227B11E932F8@visionutveckling.se> mod_java is for Java. mod_spidermonkey is for Javascript, so you can still use Javascript. mod_spidermonkey_teletone will be loaded by mod_spidermonkey, so it should work. About mixing 32 and 64 bit, you can't load a 64 bit dll from 32 application and vice versa. As long as you stick to one platform it will work. It's always better to use 64 bit if possible. /Peter ----- Reply message ----- Fr?n: "Rodney" Datum: m?n, jan 9, 2012 03:48 Rubrik: [Freeswitch-users] error loading module dll open error /sym error Till: "freeswitch-users at lists.freeswitch.org" okay good to know about java. and since i cant use java i wont be using spidermonkey teletone, i was trying to use the example answering machine script . i want to have voice bulletin board that would allow callers to record a public message and have other callers be able to listen to the last 30 or so. and the mod voicemail has too many options right now for this. as far as the mod shout goes. yes i want to have a folder of mp3s and maybe even an internet stream or too so this is important to me. when you say not to mix 32bit and 64 bit dlls. do you mean if i installed the 64 bit msi not to build 32bit dlls? if not, is it just better for me to always run the 32 bit even on a 64 bit windows server? rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Sunday, January 8, 2012 8:33 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 69 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. error loading module dll open error /sym error (Rodney) 2. Re: error loading module dll open error /sym error (Jeff Lenk) 3. Re: error loading module dll open error /sym error (Peter Olsson) 4. MPL v2 (Pusk?s Zsolt) 5. Witholding number on demand (Bob Smith) 6. Re: MPL v2 (curriegrad2004) 7. FreeTDM [MANDATORY_IE_MISSING] (Adam Ford) I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks. mod_java is not currently supported on windows so I'm not sure how you built that. the spidermonkey error is because thats a sub module of mod_spidermonkey not loadable directly by fs. make sure you are not mixing 32 and 64 version of dlls. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-loading-module-dll-open-error-sym-error-tp7165636p7165750.html Sent from the freeswitch-users mailing list archive at Nabble.com. Have you built the modules yourself? mod_spidermonkey_teletone.dll is only loaded by mod_spidermonkey.dll, nothing you should load youself. mod_java.dll and mod_shout.dll I've never tried myself, I've never even tried to build them. Is this modules you will actually be using? /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Rodney [notlikeme75 at yahoo.com] Skickat: den 8 januari 2012 20:42 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] error loading module dll open error /sym error I am using Freeswitch Version 1.0.head (git-a2ea9e5 2012-01-05 16-12-35 -0800) on windows 64 when I try to load most modules not pre loaded I get errors. even though the dlls are in the mod folder. here are some sample errors [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_java.dll dll open error 1261 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_shout.dll dll open error 1931 [CRIT] switch_loadable_module.c:1281 Error Loading module C:\Program Files\Freeswitch\mod\mod_spidermonkey_teletone.dll dll sym error 1271 since this is happening on most modules is there a windows permission problem or should i be looking in another place? thanks. Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt Hi, I'm a bit of a newbie to Freeswitch at the moment, still trying to wrap my head around its power. My current pet project is fairly simple, trying to mimick the prefix based number withhold features typically offered by carriers (e.g dialling 141 in the UK, or, I believe *67 will withhold your number from the recipient). Basically I want to set the RPID/Privacy Flags etc. if a certain prefix is dialled ahead of the main number. My current dialplan is along the following lines : more extensions here..... I've found the prefix dialling example here http://wiki.freeswitch.org/wiki/Prefix_dialing, but I can't see how I can integrate it with a syntax similar to the above where I am already checking for other prefixes? Thanks in advance Bob No, can't change the license version yet. This matter may require Tony to comment on. On 2012-01-08 1:32 PM, "Pusk?s Zsolt" > wrote: Hi. A new version of the MPL license just came out ( http://www.opensource.org/licenses/MPL-2.0 ). Is the new version better than the last one? I'm trying to find the differences but this kind of text is to hard for me :) Any plans for changing FreeSWITCH license to the new one ? Zsolt _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org I can?t seem to find any info on the problem I am having by searching the archives, so I apologize if this has been answered in the past (found several about MANDATORY_IE_MISSING but they were different situations). I am trying to setup FreeSWITCH using a FreeTDM + libpri + DAHDI + foneBridge2 stack. Outgoing calls work great, but I am running into the ?MANDATORY_IE_MISSING? problem with incoming calls. I am running the latest git version as of this morning, and completely default configuration with the exception of FreeTDM/DAHDI configuration and a modification of the default inbound_did dialplan to pass my DID 5530 to the default extension 1001. Below is what I get in the log, I highlighted in RED as soon as the call appears to start failing - 2012-01-08 18:04:54.946902 [NOTICE] ftmod_libpri.c:1363 -- Ring on channel 1:1 (from ********** to 5530) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1394 RING event with complete indicator (or overlap receive disabled) 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:1395 [s1c1][1:1] Changed state from DOWN to RING 2012-01-08 18:04:54.946902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for RING 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [RING] 2012-01-08 18:04:54.946902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from DOWN to RING in 0ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [START] 2012-01-08 18:04:54.946902 [DEBUG] ftdm_io.c:3131 [s1c1][1:1] Enabled software DTMF detector 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:407 Set codec PCMU 20ms 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1740 Connect inbound channel FreeTDM/1:1/5530 2012-01-08 18:04:54.946902 [NOTICE] switch_channel.c:924 New Channel FreeTDM/1:1/5530 [f11ea7a0-3a5d-11e1-a157-018f03d45a74] 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:1846 (FreeTDM/1:1/5530) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_INIT 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:435 (FreeTDM/1:1/5530) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:401 (FreeTDM/1:1/5530) State INIT going to sleep 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:54.946902 [DEBUG] switch_channel.c:1884 (FreeTDM/1:1/5530) Callstate Change DOWN -> RINGING 2012-01-08 18:04:54.946902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state RING 2012-01-08 18:04:54.946902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Changed state from RING to PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROCEED 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROCEED] 2012-01-08 18:04:55.006902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from RING to PROCEED in 55ms 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->5530 in context public Dialplan: FreeTDM/1:1/5530 parsing [public->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->outside_call] continue=true Dialplan: FreeTDM/1:1/5530 Absolute Condition [outside_call] Dialplan: FreeTDM/1:1/5530 Action set(outside_call=true) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [public->call_debug] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 parsing [public->public_extensions] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [public_extensions] destination_number(5530) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [public->public_did] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [public_did] destination_number(5530) =~ /^(5530)$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action set(domain_name=xx.xx.xx.xxx) Dialplan: FreeTDM/1:1/5530 Action transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 set(outside_call=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [outside_call]=[true] EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 set(domain_name=xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [domain_name]=[xx.xx.xx.xxx] EXECUTE FreeTDM/1:1/5530 transfer(1001 XML default) 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr.c:1711 (FreeTDM/1:1/5530) State Change CS_EXECUTE -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [NOTICE] switch_ivr.c:1717 Transfer FreeTDM/1:1/5530 to XML[1001 at default] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:458 FreeTDM/1:1/5530 CHANNEL ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:461 [s1c1][1:1] Indicating PROCEED in state PROCEED 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:104 FreeTDM/1:1/5530 Standard ROUTING 2012-01-08 18:04:55.006902 [INFO] mod_dialplan_xml.c:481 Processing ********** <**********>->1001 in context default Dialplan: FreeTDM/1:1/5530 parsing [default->unloop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->tod_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [tod_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->holiday_example] continue=true Dialplan: FreeTDM/1:1/5530 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->group-intercept] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->intercept-ext] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->redial] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->global] continue=true Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: FreeTDM/1:1/5530 Absolute Condition [global] Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-2] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->snom-demo-1] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->eavesdrop] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call_return] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->del-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->add-group] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-simo] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->call-group-order] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->extension-intercom] continue=false Dialplan: FreeTDM/1:1/5530 Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 parsing [default->Local_Extension] continue=false Dialplan: FreeTDM/1:1/5530 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action export(dialed_extension=1001) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: FreeTDM/1:1/5530 Action set(ringback=${us-ring}) Dialplan: FreeTDM/1:1/5530 Action set(transfer_ringback=local_stream://moh) Dialplan: FreeTDM/1:1/5530 Action set(call_timeout=30) Dialplan: FreeTDM/1:1/5530 Action set(hangup_after_bridge=true) Dialplan: FreeTDM/1:1/5530 Action set(continue_on_fail=true) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: FreeTDM/1:1/5530 Action answer() Dialplan: FreeTDM/1:1/5530 Action sleep(1000) Dialplan: FreeTDM/1:1/5530 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-spymap/**********/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/**********/1001) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/global/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Sun, 08 Jan 2012 18:04:55 -0700) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Sun, 08 Jan 2012 18:04:55 -0700] EXECUTE FreeTDM/1:1/5530 export(dialed_extension=1001) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1091 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE FreeTDM/1:1/5530 bind_meta_app(1 b s execute_extension::dx XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04-55.wav) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/**********.2012-01-08-18-04-55.wav EXECUTE FreeTDM/1:1/5530 bind_meta_app(3 b s execute_extension::cf XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-01-08 18:04:55.006902 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE FreeTDM/1:1/5530 set(ringback=%(2000,4000,440,480)) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [ringback]=[%(2000,4000,440,480)] EXECUTE FreeTDM/1:1/5530 set(transfer_ringback=local_stream://moh) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [transfer_ringback]=[local_stream://moh] EXECUTE FreeTDM/1:1/5530 set(call_timeout=30) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [call_timeout]=[30] EXECUTE FreeTDM/1:1/5530 set(hangup_after_bridge=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [hangup_after_bridge]=[true] EXECUTE FreeTDM/1:1/5530 set(continue_on_fail=true) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [continue_on_fail]=[true] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-call_return/1001/**********) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/1001/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 set(called_party_callgroup=techsupport) 2012-01-08 18:04:55.006902 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [called_party_callgroup]=[techsupport] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/techsupport/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/global/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/techsupport/f11ea7a0-3a5d-11e1-a157-018f03d45a74) EXECUTE FreeTDM/1:1/5530 bridge(user/1001 at xx.xx.xx.xxx) 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at xx.xx.xx.xxx] 2012-01-08 18:04:55.006902 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [f1292a54-3a5d-11e1-a15e-018f03d45a74] 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:4674 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_NEW -> CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_INIT 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA INIT 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_INIT -> CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State INIT going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_channel.c:1884 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change DOWN -> RINGING 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING 2012-01-08 18:04:55.006902 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA ROUTING 2012-01-08 18:04:55.006902 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State ROUTING going to sleep 2012-01-08 18:04:55.006902 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA 2012-01-08 18:04:55.006902 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State CONSUME_MEDIA going to sleep 2012-01-08 18:04:55.006902 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [calling][0] 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1668 Got a FACILITY event on span 1:1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1697 FACILITY subcommand 2 is not implemented, ignoring 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1700 FACILITY subcommand 2 handler returned -1 2012-01-08 18:04:55.046916 [DEBUG] ftmod_libpri.c:1703 Caught Event on span 1 11 (FACILITY) 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.126907 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 entering state [proceeding][180] 2012-01-08 18:04:55.126907 [NOTICE] sofia.c:5574 Ring-Ready sofia/internal/sip:1001 at xx.xx.xx.xxx:52564! 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Indicating PROGRESS_MEDIA in state PROCEED 2012-01-08 18:04:55.126907 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROCEED to PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS] 2012-01-08 18:04:55.146908 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROCEED to PROGRESS in 22ms 2012-01-08 18:04:55.146908 [ERR] ftmod_libpri.c:132 XXX Progress message requested but no information is provided 2012-01-08 18:04:55.146908 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROGRESS to PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS_MEDIA 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS_MEDIA] 2012-01-08 18:04:55.206902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS to PROGRESS_MEDIA in 55ms 2012-01-08 18:04:55.206902 [INFO] ftmod_zt.c:656 Setting echo cancel to 64 taps for 1:1 2012-01-08 18:04:55.206902 [WARNING] ftmod_zt.c:661 Echo cancel not available for 1:1 2012-01-08 18:04:55.206902 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.206902 [NOTICE] switch_ivr_originate.c:1115 Pre-Answer FreeTDM/1:1/5530! 2012-01-08 18:04:55.206902 [DEBUG] switch_channel.c:2930 (FreeTDM/1:1/5530) Callstate Change RINGING -> EARLY 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2012-01-08 18:04:55.206902 [DEBUG] switch_core_codec.c:116 FreeTDM/1:1/5530 Push codec L16:70 2012-01-08 18:04:55.206902 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 1:1 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:1078 [s1c1][1:1] Changed state from PROGRESS_MEDIA to TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for TERMINATING 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [TERMINATING] 2012-01-08 18:04:55.226909 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS_MEDIA to TERMINATING in 0ms 2012-01-08 18:04:55.226909 [DEBUG] ftdm_io.c:5565 [s1c1][1:1] Scheduling safety hangup timer 2012-01-08 18:04:55.226909 [DEBUG] mod_freetdm.c:2416 got clear channel sig [STOP] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2846 (FreeTDM/1:1/5530) Callstate Change EARLY -> HANGUP 2012-01-08 18:04:55.226909 [NOTICE] mod_freetdm.c:2441 Hangup FreeTDM/1:1/5530 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2012-01-08 18:04:55.226909 [DEBUG] switch_channel.c:2869 Send signal FreeTDM/1:1/5530 [KILL] 2012-01-08 18:04:55.226909 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_codec.c:141 FreeTDM/1:1/5530 Restore previous codec PCMU:0. 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2846 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change RINGING -> HANGUP 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_channel.c:2869 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [KILL] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2012-01-08 18:04:55.246902 [INFO] mod_dptools.c:2900 Originate Failed. Cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:2285 FreeTDM/1:1/5530 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_HANGUP 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 hanging up, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:530 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP ENTER 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:605 [s1c1][1:1] Changed state from TERMINATING to HANGUP 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 SOFIA DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at xx.xx.xx.xxx:52564 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:52564) State DESTROY going to sleep 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from TERMINATING to HANGUP in 15ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:929 [s1c1][1:1] Changed state from HANGUP to HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP_COMPLETE 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP_COMPLETE] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP to HANGUP_COMPLETE in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:939 [s1c1][1:1] Changed state from HANGUP_COMPLETE to DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for DOWN 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [DOWN] 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP_COMPLETE to DOWN in 0ms 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2930 [s1c1][1:1] DTMF debug is already disabled 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2962 [s1c1][1:1] No need to disable input dump 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2993 [s1c1][1:1] No need to disable output dump 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:2416 got clear channel sig [RELEASED] 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:6185 Cleared call with id 1 2012-01-08 18:04:55.246902 [DEBUG] ftdm_io.c:2735 [s1c1][1:1] channel done 2012-01-08 18:04:55.246902 [DEBUG] ftmod_libpri.c:704 -- Closed channel 1:1 2012-01-08 18:04:55.246902 [DEBUG] mod_freetdm.c:624 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP EXIT 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:47 FreeTDM/1:1/5530 Standard HANGUP, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:393 (FreeTDM/1:1/5530) State Change CS_HANGUP -> CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:79 FreeTDM/1:1/5530 Standard REPORTING, cause: MANDATORY_IE_MISSING 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING going to sleep 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:387 (FreeTDM/1:1/5530) State Change CS_REPORTING -> CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_session.c:1380 Session 1 (FreeTDM/1:1/5530) Locked, Waiting on external entities 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1398 Session 1 (FreeTDM/1:1/5530) Ended 2012-01-08 18:04:55.246902 [NOTICE] switch_core_session.c:1400 Close Channel FreeTDM/1:1/5530 [CS_DESTROY] 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:491 (FreeTDM/1:1/5530) Callstate Change HANGUP -> DOWN 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:494 (FreeTDM/1:1/5530) Running State Change CS_DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:86 FreeTDM/1:1/5530 Standard DESTROY 2012-01-08 18:04:55.246902 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY going to sleep Full log (including loading FreeSWITCH) at http://pastebin.com/vQUHu0pR freeswitch/conf/freetdm.conf ? [span zt PRI] trunk_type => T1 b-channel=1-23 d-channel=24 freeswitch/conf/autoload_confg/freetdm.conf.xml ? /etc/dahdi/system.conf ? loadzone = us defaultzone=us dynamic=ethmf,eth0/00:50:c2:65:d7:59/0,24,1 bchan=1-23 dchan=24 I am guessing it is a configuration issue, though this same config is currently working in production with FreeSWITCH 1.0.6 + OpenZAP + Libpri + DAHDI + foneBridge2. Any help is greatly appreciated. -Adam _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f0a532832769871211918! From amit.nakum2009 at gmail.com Mon Jan 9 10:06:33 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Mon, 9 Jan 2012 12:36:33 +0530 Subject: [Freeswitch-users] ivr no digit press Message-ID: Dear All, How to travel next ivr menu if max-failuers=3 over,and also suggest how i can play some file while no digit is press or enter in ivrs. Thanks in advance. amit -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/724294f2/attachment.html From miha at softnet.si Mon Jan 9 10:20:38 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 09 Jan 2012 08:20:38 +0100 Subject: [Freeswitch-users] Password in dialplan In-Reply-To: <4F06B394.6000808@softnet.si> References: <4F054EFA.2000004@softnet.si> <4F06B394.6000808@softnet.si> Message-ID: <4F0A9546.2030407@softnet.si> Hi, can anyone can help me how can I get same password from user/dir also for radius authorization? Thanks! regards, Miha On 1/6/2012 9:40 AM, Miha Zoubek wrote: > Hi Michael, > > this I have in my dialplan: > data="USERNAME=${dialed_extension}"/> > > > > > in user/dir: > > user id="013108500"> > > > > When user make call we authenticate it with freeradius. I would like > that, the same password for user login is also for radius > authentication, but I do not know how to link this with dial plan > (that the dial plan will parse password from user dir for auth_function. > > Thanks! > > Regards, > Miha > > On 1/5/2012 4:49 PM, Michael Collins wrote: >> How is this user being authenticated? >> -MC >> >> On Wed, Jan 4, 2012 at 11:19 PM, Miha Zoubek > > wrote: >> >> Hi, >> >> in dial plan I have this line (for radius): >> >> > data="PASSWD=$${default_password}"/> >> >> Deafult password is set to 1234. As I do not wont default >> password for radius, I would just like to have here password >> which is set for user in directory. >> >> >> >> How to set in dialplan that the password will be taken from >> user/dir? >> >> > data="PASSWD=$${password}"/> Like this? >> >> Thanks! >> >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/c0002e35/attachment.html From tech at worldtecgroup.cz Mon Jan 9 05:45:06 2012 From: tech at worldtecgroup.cz (WTG Technical) Date: Mon, 9 Jan 2012 03:45:06 +0100 Subject: [Freeswitch-users] IVR (Calling card application) development In-Reply-To: <8343AF3F-140F-4C0D-8885-A8CE7B1EE71E@gmail.com> References: <8343AF3F-140F-4C0D-8885-A8CE7B1EE71E@gmail.com> Message-ID: <007201ccce78$b8a39f30$29eadd90$@cz> Hi Vitaly, Why not just use embedded lua? It is fast and good for ivr scripting. FYI http://wiki.freeswitch.org/wiki/Examples_db_agent_login_lua ----- S pozdravem WTG Technical Department http://www.wtg.cz -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitaly Nikolaev Sent: Sunday, January 08, 2012 2:46 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] IVR (Calling card application) development Hello, I am choosing between ESL and mod_erlang, I will have around 1k simultaneous calls with 10 min average duration. I will definitely write it in erlang, but question is - how supported is mod_erlang ? Is anyone currently use it in production for heavy load application ? Thank you Vitaly _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 9 11:26:36 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 9 Jan 2012 08:26:36 +0000 (GMT) Subject: [Freeswitch-users] Witholding number on demand Message-ID: <1326097596.53653.YahooMailNeo@web29406.mail.ird.yahoo.com> Thanks very much for this, that's great ! Bob >Check this config set out:? >git://github.com/curriegrad2004/freeswitch-sample-configs.git? >The internal.xml file under the light-pbx/dialplan directory of the >git repository demonstrates what you really want to do. From miha at softnet.si Mon Jan 9 12:41:16 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 09 Jan 2012 10:41:16 +0100 Subject: [Freeswitch-users] Mod_rad_auth Message-ID: <4F0AB63C.4040802@softnet.si> Hi, just curious. Is mod_rad_auth automatic reject user if the password for radius is wrong or must I set condition is dialplan? I am asking this because I noticed that radius auth is working but not rejecting users if the password is wrong. Is this problem with my setting or is this ok and I must reject it with dialplan? Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/469a9d7c/attachment.html From anita.hall at simmortel.com Mon Jan 9 12:55:23 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 9 Jan 2012 15:25:23 +0530 Subject: [Freeswitch-users] Max calls Limit? In-Reply-To: References: <1325968310.32667.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: Thanks guys, I was able to do more than 1200 calls on a lowly AMD machine, after which the 2 GB RAM went out and 1 GB swap got used. Is there a way to automatically get a sense of the call "quality"? Something like, find if UDP packets are getting dropped :) regards, Anita On Sun, Jan 8, 2012 at 2:48 AM, Ken Rice wrote: > This is also adjustable from the FS CLI via the fsctl max_sessions ### > command... Replace ### with the max number of sessions you want. > > Note this is MAX Sessions, not max calls... So if you are doing 2 legged > calls, the number of calls you can do is effective 1/2 that setting since > each call leg is a session. > > Setting it from the CLI is not kept across restarts of freeswitch for that > see switch.xml > > This setting should be set at a sane level for your particular hardware as > yes it does keep freeswitch starving the server of memory and causes FS to > start failing calls at that point. > > The similar setting max_sps is for max sessions per second and this is > again to keep you from melting down your hardware. The current setting is > sufficient for 99% of installs however if you are doing something out of > the norm ie: high volume calling you may need to adjust it up > > K > > > > On 1/7/12 2:31 PM, "Sherif Omran" wrote: > > Yes > con/autoload/switch.xml > > > --- On *Sat, 1/7/12, Anita Hall * wrote: > > > From: Anita Hall > Subject: [Freeswitch-users] Max calls Limit? > To: "FreeSWITCH Users Help" > Date: Saturday, January 7, 2012, 10:32 AM > > Is there any config variable somewhere which is limiting the max call > limit on FS to 1000? > > Set-up: > Taking FS for max call load on my machine. > > User freeswitch is calling internal gateway and playing demo-ivr > > User freeswitch defined in vars.xml > > > > data="default_provider_from_domain=10.60.20.145"/> > > > > > Internal Gateway defined in directory/default/freeswitch.xml > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > After 1000 calls, it is giving the same error > -ERR DESTINATION_OUT_OF_ORDER > > Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by > the time 1000 calls are being done, but is it not a little strange that > every time the resources get exhausted at precisely 1000 calls! > > > regards, > Anita > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org < > /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/bfa65666/attachment-0001.html From anita.hall at simmortel.com Mon Jan 9 13:06:06 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 9 Jan 2012 15:36:06 +0530 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: <4F08F67A.8010503@ieee.org> References: <4F08F67A.8010503@ieee.org> Message-ID: Hi David Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? If not, you can distribute the load between 2 or more machines. Have you tried this? (from Wiki) For best results in getting accurate detection as well a saving system resources (CPU & memory), you should do the following: - Wait until AFTER the call has been answered before starting AVMD. In many cases you may find it best to wait a few seconds after the call is answered before starting AVMD. - Starting AVMD before the call is answered will often result in false beeps and wastes CPU resources. - Once you have received the AVMD event, you should explicitly stop AVMD, even though it will not return another event. regards, Anita On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: > Hello. > > I need to detect answering machine beeps on 1,000+ channels > simultaneously. Unfortunately, this is a real requirement, not a > pseudo-requirement that can be engineered away. This is for a mass > emergency notification system for government, K-12, and higher education. > > In the past, I have done this with proprietary hardware (Dialogic and > Aculab), which handles this sort of thing without overloading the host CPU, > as the hardware's DSPs handle the detections. I want to see if I can do > the same thing with FreeSWITCH. > > Alas, I find the following warning in the wiki page for mod_avmd: > > AVMD (and VMD) are both very CPU intensive. You need to be aware of this > fact when using it. It will drastically reduce your call capacity if you do > not manage it correctly. On the other hand it is a very useful tool, and if > managed properly will be a great aid for calls needing to do Voice Mail > Detection. > > Eric states, "You can expect about ~50 simultaneous instances on an > Intel i7 920 CPU." > > So, it sounds like VMD and AVMD won't work for what I need to do. I also > don't think that the tone_detect application will work, as this requires > specific frequencies (as opposed to ranges of frequencies), and answering > machines and voicemail come with beeps at all sorts of frequencies. > > Does anyone know anything, either free or commercial, that I can use with > FreeSWITCH to do this many simultaneous detections? I know of commercial > software-only platforms (e.g., Aculab's Prosody S) that claim to be able > to this, so it seems like it should be possible. > > Thanks in advance, > David Stein > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/45cce706/attachment.html From avi at avimarcus.net Mon Jan 9 14:49:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 9 Jan 2012 13:49:31 +0200 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: <4F08F67A.8010503@ieee.org> Message-ID: Yes, even with no other way of making it work more efficiently... 1 machine can handle 50 calls that have actually picked up, before sending it to the recording. So your actual window is maximum 30? seconds per calls that picks up that mod_avmd needs to be running. So then that's 50 per 30 second windows so 100 calls per minute. That's 1,000 actual connected calls within 10 minutes. Is that within your threshold? For using ONE machine? (What happens if a person picks up? On silence it just starts the recording..? Or do you just have a looping first 30 seconds so it's ok if they miss the start?) -Avi On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall wrote: > Hi David > > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? If > not, you can distribute the load between 2 or more machines. > > Have you tried this? (from Wiki) > > For best results in getting accurate detection as well a saving system > resources (CPU & memory), you should do the following: > > Wait until AFTER the call has been answered before starting AVMD. In many > cases you may find it best to wait a few seconds after the call is answered > before starting AVMD. > > Starting AVMD before the call is answered will often result in false beeps > and wastes CPU resources. > > Once you have received the AVMD event, you should explicitly stop AVMD, even > though it will not return another event. > > > > regards, > Anita > > > > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: >> >> Hello. >> >> I need to detect answering machine beeps on 1,000+ channels >> simultaneously.? Unfortunately, this is a real requirement, not a >> pseudo-requirement that can be engineered away.? This is for a mass >> emergency notification system for government, K-12, and higher education.. >> >> In the past, I have done this with proprietary hardware (Dialogic and >> Aculab), which handles this sort of thing without overloading the host CPU, >> as the hardware's DSPs handle the detections.? I want to see if I can do the >> same thing with FreeSWITCH. >> >> Alas, I find the following warning in the wiki page for mod_avmd: >> >> AVMD (and VMD) are both very CPU intensive. You need to be aware of this >> fact when using it. It will drastically reduce your call capacity if you do >> not manage it correctly. On the other hand it is a very useful tool, and if >> managed properly will be a great aid for calls needing to do Voice Mail >> Detection. >> >> ??? Eric states, "You can expect about ~50 simultaneous instances on an >> Intel i7 920 CPU." >> >> So, it sounds like VMD and AVMD won't work for what I need to do.? I also >> don't think that the tone_detect application will work, as this requires >> specific frequencies (as opposed to ranges of frequencies), and answering >> machines and voicemail come with beeps at all sorts of frequencies. >> >> Does anyone know anything, either free or commercial, that I can use with >> FreeSWITCH to do this many simultaneous detections?? I know of commercial >> software-only platforms (e.g., Aculab's? Prosody S) that claim to be able to >> this, so it seems like it should be possible. >> >> Thanks in advance, >> David Stein >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From arnuld at phonologies.com Mon Jan 9 16:29:26 2012 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Mon, 9 Jan 2012 18:59:26 +0530 Subject: [Freeswitch-users] Connecting to FreeSWITCH from a C Program Message-ID: <05caf55cdc7b529576cc517e8b56161b.squirrel@webmail1.web.com> I want to connect to FreeSWITCH from a C Program and make some calls and know when a particular call is over. An example would be connectin usingh telnet and then we can make calls using "api originate" and recieve events like CHANNEL_DESTROY to know that a call is over. Want to do something similar from my C program -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From nvitaly at gmail.com Mon Jan 9 17:00:48 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Mon, 9 Jan 2012 09:00:48 -0500 Subject: [Freeswitch-users] IVR (Calling card application) development In-Reply-To: <007201ccce78$b8a39f30$29eadd90$@cz> References: <8343AF3F-140F-4C0D-8885-A8CE7B1EE71E@gmail.com> <007201ccce78$b8a39f30$29eadd90$@cz> Message-ID: Hello, > db_conn = assert(env:connect("dbname=mydb user=freeswitch password='t0psecret'")) If it 1k calls I will have 1k connections to databases, and 1k lua scripts in memory. I know there are ways around but separation calling card logic from freeswitch will give more freedom. Thank you PS: Il craft small mod_erlang based test IVR and try it with call simulator. On Sun, Jan 8, 2012 at 9:45 PM, WTG Technical wrote: > Hi Vitaly, > > Why not just use embedded lua? It is fast and good for ivr scripting. > > FYI http://wiki.freeswitch.org/wiki/Examples_db_agent_login_lua > > > ----- > S pozdravem > WTG Technical Department > http://www.wtg.cz > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitaly > Nikolaev > Sent: Sunday, January 08, 2012 2:46 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] IVR (Calling card application) development > > Hello, > > I am choosing between ESL and mod_erlang, I will have around 1k > simultaneous > calls with 10 min average duration. > > I will definitely write it in erlang, but question is - how supported is > mod_erlang ? Is anyone currently use it in production for heavy load > application ? > > Thank you > Vitaly > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -- Vitaly Nikolaev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/c6ff8b32/attachment.html From odermann at googlemail.com Mon Jan 9 17:33:04 2012 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Jan 2012 15:33:04 +0100 Subject: [Freeswitch-users] User Configuration: Question about F_REC and misc. Message-ID: hi, we would like to allow SIP-phones to connect to our fs directly. therefor we would like to be able to set some options for each user/phone. the most important setting is regarding the "record key" (F_REC), which one can find on some/most/all SIP-phones: 1.) is it possible to DISABLE the key for individual users? if we press the F_REC key on our snom, fs starts to record. we can find options to disable some keys, but we did not find an option to disable the F_REC key. this option is very important, because we can not allow everybody to record calls. 2.) addionally we would like to be able to play an audio-file, when the record key is pressed (to let the other side know, that the call is beeing recorded). misc. questions about user configuration options: can we set the DTMF-type (like info and/or rfc2833) or the allowed codec (like G722,PCMA at 20i) to override the global settings? thanks a lot for your help! kind regards dennis From odermann at googlemail.com Mon Jan 9 17:40:27 2012 From: odermann at googlemail.com (Dennis) Date: Mon, 9 Jan 2012 15:40:27 +0100 Subject: [Freeswitch-users] voicemail_inject (Mod voicemail) problems Message-ID: hi, we have problems getting "voicemail_inject" to work. whatever we do, we alway get the same error message (fs newest version). for example we do: voicemail_inject 1000 at 192.168.2.2 /usr/local/freeswitch/recordings/1000_2012-01-07-21-47-44.wav 1001 test we will always get: Error: [group=[@domain]|domain=|[@]] [] [] what are we doing wrong? thanks a lot for your help! kind regards dennis From krice at freeswitch.org Mon Jan 9 17:52:43 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 09 Jan 2012 08:52:43 -0600 Subject: [Freeswitch-users] Connecting to FreeSWITCH from a C Program In-Reply-To: <05caf55cdc7b529576cc517e8b56161b.squirrel@webmail1.web.com> Message-ID: The ESL Library... See fs_cli and ESL .... On 1/9/12 7:29 AM, "Arnuld Uttre (Phonologies)" wrote: > I want to connect to FreeSWITCH from a C Program and make some calls and > know when a particular call is over. > > An example would be connectin usingh telnet and then we can make calls > using "api originate" and recieve events like CHANNEL_DESTROY to know > that a call is over. Want to do something similar from my C program > > > From curriegrad2004 at gmail.com Mon Jan 9 18:22:30 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 9 Jan 2012 07:22:30 -0800 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: <4F08F67A.8010503@ieee.org> Message-ID: Since an answering machine creates a tone, have you looked into mod_spandsp's tone_detect feature? I think it does what you want it to do. On 2012-01-09 3:50 AM, "Avi Marcus" wrote: > Yes, even with no other way of making it work more efficiently... 1 > machine can handle 50 calls that have actually picked up, before > sending it to the recording. > So your actual window is maximum 30? seconds per calls that picks up > that mod_avmd needs to be running. So then that's 50 per 30 second > windows so 100 calls per minute. That's 1,000 actual connected calls > within 10 minutes. Is that within your threshold? For using ONE > machine? > (What happens if a person picks up? On silence it just starts the > recording..? Or do you just have a looping first 30 seconds so it's ok > if they miss the start?) > > -Avi > > > On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall > wrote: > > Hi David > > > > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? > If > > not, you can distribute the load between 2 or more machines. > > > > Have you tried this? (from Wiki) > > > > For best results in getting accurate detection as well a saving system > > resources (CPU & memory), you should do the following: > > > > Wait until AFTER the call has been answered before starting AVMD. In many > > cases you may find it best to wait a few seconds after the call is > answered > > before starting AVMD. > > > > Starting AVMD before the call is answered will often result in false > beeps > > and wastes CPU resources. > > > > Once you have received the AVMD event, you should explicitly stop AVMD, > even > > though it will not return another event. > > > > > > > > regards, > > Anita > > > > > > > > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: > >> > >> Hello. > >> > >> I need to detect answering machine beeps on 1,000+ channels > >> simultaneously. Unfortunately, this is a real requirement, not a > >> pseudo-requirement that can be engineered away. This is for a mass > >> emergency notification system for government, K-12, and higher > education.. > >> > >> In the past, I have done this with proprietary hardware (Dialogic and > >> Aculab), which handles this sort of thing without overloading the host > CPU, > >> as the hardware's DSPs handle the detections. I want to see if I can > do the > >> same thing with FreeSWITCH. > >> > >> Alas, I find the following warning in the wiki page for mod_avmd: > >> > >> AVMD (and VMD) are both very CPU intensive. You need to be aware of this > >> fact when using it. It will drastically reduce your call capacity if > you do > >> not manage it correctly. On the other hand it is a very useful tool, > and if > >> managed properly will be a great aid for calls needing to do Voice Mail > >> Detection. > >> > >> Eric states, "You can expect about ~50 simultaneous instances on an > >> Intel i7 920 CPU." > >> > >> So, it sounds like VMD and AVMD won't work for what I need to do. I > also > >> don't think that the tone_detect application will work, as this requires > >> specific frequencies (as opposed to ranges of frequencies), and > answering > >> machines and voicemail come with beeps at all sorts of frequencies. > >> > >> Does anyone know anything, either free or commercial, that I can use > with > >> FreeSWITCH to do this many simultaneous detections? I know of > commercial > >> software-only platforms (e.g., Aculab's Prosody S) that claim to be > able to > >> this, so it seems like it should be possible. > >> > >> Thanks in advance, > >> David Stein > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/1dbaf303/attachment.html From krice at freeswitch.org Mon Jan 9 18:26:23 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 09 Jan 2012 09:26:23 -0600 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: Message-ID: Detecting the tone is exactly what mod_vmd and mod_avmd do now... Not exactly the most efficient way to detect this sort of thing, but it does work K On 1/9/12 9:22 AM, "curriegrad2004" wrote: > Since an answering machine creates a tone, have you looked into mod_spandsp's > tone_detect feature? I think it does what you want it to do. > > On 2012-01-09 3:50 AM, "Avi Marcus" wrote: >> Yes, even with no other way of making it work more efficiently... 1 >> machine can handle 50 calls that have actually picked up, before >> sending it to the recording. >> So your actual window is maximum 30? seconds per calls that picks up >> that mod_avmd needs to be running. So then that's 50 per 30 second >> windows so 100 calls per minute. That's 1,000 actual connected calls >> within 10 minutes. Is that within your threshold? For using ONE >> machine? >> (What happens if a person picks up? On silence it just starts the >> recording..? Or do you just have a looping first 30 seconds so it's ok >> if they miss the start?) >> >> -Avi >> >> >> On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall wrote: >>> > Hi David >>> > >>> > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? >>> If >>> > not, you can distribute the load between 2 or more machines. >>> > >>> > Have you tried this? (from Wiki) >>> > >>> > For best results in getting accurate detection as well a saving system >>> > resources (CPU & memory), you should do the following: >>> > >>> > Wait until AFTER the call has been answered before starting AVMD. In many >>> > cases you may find it best to wait a few seconds after the call is >>> answered >>> > before starting AVMD. >>> > >>> > Starting AVMD before the call is answered will often result in false beeps >>> > and wastes CPU resources. >>> > >>> > Once you have received the AVMD event, you should explicitly stop AVMD, >>> even >>> > though it will not return another event. >>> > >>> > >>> > >>> > regards, >>> > Anita >>> > >>> > >>> > >>> > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: >>>> >> >>>> >> Hello. >>>> >> >>>> >> I need to detect answering machine beeps on 1,000+ channels >>>> >> simultaneously.? Unfortunately, this is a real requirement, not a >>>> >> pseudo-requirement that can be engineered away.? This is for a mass >>>> >> emergency notification system for government, K-12, and higher >>>> education.. >>>> >> >>>> >> In the past, I have done this with proprietary hardware (Dialogic and >>>> >> Aculab), which handles this sort of thing without overloading the host >>>> CPU, >>>> >> as the hardware's DSPs handle the detections.? I want to see if I can do >>>> the >>>> >> same thing with FreeSWITCH. >>>> >> >>>> >> Alas, I find the following warning in the wiki page for mod_avmd: >>>> >> >>>> >> AVMD (and VMD) are both very CPU intensive. You need to be aware of this >>>> >> fact when using it. It will drastically reduce your call capacity if you do >>>> >> not manage it correctly. On the other hand it is a very useful tool, and if >>>> >> managed properly will be a great aid for calls needing to do Voice Mail >>>> >> Detection. >>>> >> >>>> >> ??? Eric states, "You can expect about ~50 simultaneous instances on an >>>> >> Intel i7 920 CPU." >>>> >> >>>> >> So, it sounds like VMD and AVMD won't work for what I need to do.? I >>>> also >>>> >> don't think that the tone_detect application will work, as this requires >>>> >> specific frequencies (as opposed to ranges of frequencies), and >>>> answering >>>> >> machines and voicemail come with beeps at all sorts of frequencies. >>>> >> >>>> >> Does anyone know anything, either free or commercial, that I can use >>>> with >>>> >> FreeSWITCH to do this many simultaneous detections?? I know of >>>> commercial >>>> >> software-only platforms (e.g., Aculab's? Prosody S) that claim to be >>>> able to >>>> >> this, so it seems like it should be possible. >>>> >> >>>> >> Thanks in advance, >>>> >> David Stein >>>> >> >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://wiki.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/1af0c873/attachment-0001.html From roland at haenel.me Mon Jan 9 18:51:01 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Mon, 9 Jan 2012 16:51:01 +0100 Subject: [Freeswitch-users] "expire" for outgoing registrations not handled correctly? Message-ID: Hi, It seems that I'm running into a timing issue with outgoing (gateway) registrations, which results in some milliseconds in that I can't send calls over that gateway. I have configured a gateway like this: ^M ^M If I look at the "200 OK" messages for the SIP REGISTER packet, the server sends me an "Expires: 30", so FreeSwitch should re-REGISTER every 30 seconds. However, that's what happening: 2012-01-09 16:37:00.804865 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:37:31.804880 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:38:03.804894 [NOTICE] sofia_reg.c:407 Registering duro 2012-01-09 16:38:35.824877 [NOTICE] sofia_reg.c:407 Registering duro *2012-01-09 16:39:06.825090 [NOTICE] sofia_reg.c:407 Registering duro* 2012-01-09 16:39:39.824881 [NOTICE] sofia_reg.c:407 Registering duro You can see that it's *approximately* 30 seconds betwee registrations, but in fact every time it's just a little more than 30 secs. This alone makes me feel a little bit strange, since my registration might have been expired during this time window (and consequently, maybe I'm loosing incoming calls because of that). Even for outgoing calls, this seems to be an issue, let's take a look at the 16:39:06.825090 re-REGISTER. We can see that at 16:39:05 (no micro-seconds here) the gateway becomes "UNREGED": Content-Length: 542 Content-Type: text/event-plain Event-Subclass: sofia::gateway_state Event-Name: CUSTOM Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 FreeSWITCH-Hostname: lap597 FreeSWITCH-Switchname: lap597 FreeSWITCH-IPv4: 192.168.232.164 FreeSWITCH-IPv6: ::1 *Event-Date-Local: 2012-01-09 16:39:05* Event-Date-GMT: Mon, 09 Jan 2012 15:39:05 GMT Event-Date-Timestamp: 1326123545824882 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event Event-Calling-Line-Number: 151 Gateway: duro State: UNREGED Ping-Status: UP But the new REGISTER is done a full second later (this is not just a display issue, I saw those messages scrolling by and there was just a second pause between them): Event-Subclass: sofia::gateway_state Event-Name: CUSTOM Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 FreeSWITCH-Hostname: lap597 FreeSWITCH-Switchname: lap597 FreeSWITCH-IPv4: 192.168.232.164 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-01-09 16:39:06 *Event-Date-GMT: Mon, 09 Jan 2012 15:39:06 GMT* Event-Date-Timestamp: 1326123546825090 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event Event-Calling-Line-Number: 151 Gateway: duro State: TRYING Ping-Status: DOWN Then, the gateway gets re-registered in a couple of milliseconds and everything is fine again. *It seems that this behaviour effectively creates a timespan of 1-2 seconds, where the gateway is not available for outgoing (and maybe also incoming) calls.* Does anyone of you know this issue? I wonder because this looks just to easy to be wrong in this fashion? Greetings, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/06b4c720/attachment.html From stkn at freeswitch.org Mon Jan 9 18:52:39 2012 From: stkn at freeswitch.org (Stefan Knoblich) Date: Mon, 09 Jan 2012 16:52:39 +0100 Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] In-Reply-To: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> References: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> Message-ID: <4F0B0D47.3020107@freeswitch.org> On 09.01.2012 02:33, Adam Ford wrote: > Below is what I get in the log, I highlighted in RED as soon as the call > appears to start failing - Please enable libpri debug logging (ftdm libpri debug SPAN q931_all) and capture the log output again. -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From gabler at abx.de Mon Jan 9 11:30:49 2012 From: gabler at abx.de (Andreas Gabler) Date: Mon, 9 Jan 2012 08:30:49 +0000 (UTC) Subject: [Freeswitch-users] Overlap dialing problem References: Message-ID: Hello ?tefan, did you found a solution for this problem? Kind regards, Andreas Gabler From msc at freeswitch.org Mon Jan 9 19:02:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 08:02:52 -0800 Subject: [Freeswitch-users] Bridge answered In-Reply-To: References: Message-ID: During the call or when the call is done? During the call I believe you can just check the endpoint_disposition var. After the call you can use an api_hangup_hook and have access to the entire suite of channel variables. -MC 2012/1/6 Juan Antonio Iba?ez Santorum > Hello, > > Is there any way to get if an originate-bridge has been answered and > duration doing the originate-bridge from an already answered channel? Could > be considered hangup cause NORMAL_CLEARING=ANSWER? > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/d420a0da/attachment.html From msc at freeswitch.org Mon Jan 9 19:04:54 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 08:04:54 -0800 Subject: [Freeswitch-users] Invalid UTF-8 character to ampersand, skip it In-Reply-To: <1325958511.1936.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1325958511.1936.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: It's been fixed. The encoding in the German lang XML files was not UTF-8. You can just delete conf/lang/de/* and be done with it. -MC On Sat, Jan 7, 2012 at 9:48 AM, Rodney wrote: > this is doing the same for me but only when they press my option for the > Conference Call Count extension. did have any problems with the november > msi (english windows version) still hoping it gets fixed. > > ------------------------------ > *From:* "freeswitch-users-request at lists.freeswitch.org" < > freeswitch-users-request at lists.freeswitch.org> > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Saturday, January 7, 2012 12:41 PM > *Subject:* FreeSWITCH-users Digest, Vol 67, Issue 62 > > ----- Forwarded Message ----- > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Invalid UTF-8 character to ampersand, skip it > (Juan Antonio Iba?ez Santorum) > 2. Out of event dispatch threads! Slowing things down. > (Stephen Wilde) > 3. Need help using api_hangup_hook and session_in_hangup_hook > (Herman Griffin) > 4. 407 undefined? sip port 5060 (Rodney) > Try to update to last git. There are some not supported chars at german > language files. > > Regards > > 2012/1/7 Ryan How > > Hi, > > I've just installed freeswitch on windows using the binary install > (FreeSWITCH Version 1.0.head (git-6ab3f56 2011-12-28 16-52-30 -0600). > Whenever a call is made the log fills up with about 42 "Invalid UTF-8 > character to ampersand, skip it". I haven't changed any config from the > default. Any pointers ? > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > Hi, > my system is showing in the log the following critical errors: > > 2012-01-07 18:13:34.978566 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:13:35.981506 [CRIT] switch_event.c:339 Event system > overloading. Taking a 10 second break > 2012-01-07 18:13:36.975496 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:13:37.974385 [CRIT] switch_event.c:339 Event system > overloading. Taking a 10 second break > 2012-01-07 18:13:38.977286 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:13:39.983208 [CRIT] switch_event.c:339 Event system > overloading. Taking a 10 second break > 2012-01-07 18:13:40.999186 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:13:42.003136 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:14:27.210316 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:14:28.214168 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:14:29.211205 [CRIT] switch_event.c:339 Event system > overloading. Taking a 10 second break > 2012-01-07 18:14:30.208196 [CRIT] switch_event.c:360 Out of event > dispatch threads! Slowing things down. > 2012-01-07 18:14:31.210143 [CRIT] switch_event.c:339 Event system > overloading. Taking a 10 second break > > The CPU is running at 40%-50% of load and the number of sessions in below > the limit, the same for sessions per seconds. > > There is any parameter that I have to adjust? > > Stephen > Hello, > > I'm using the api_hangup_hook and session_in_hangup variables with a > pyrun. How do I access the session object from inside the python > module? > > --------------------------------- > Here are some logs : > > Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition > [emergency] > Dialplan: sofia/external/3105795721 at 72.37.252.18 Action > set(session_in_hangup_hook=true) > Dialplan: sofia/external/3105795721 at 72.37.252.18 Action > set(api_hangup_hook=pyrun emergency.hangup) > Dialplan: sofia/external/3105795721 at 72.37.252.18 Absolute Condition > [emergency] > . > . > 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:281 Call python script > 2012-01-07 09:21:57.693488 [DEBUG] mod_python.c:284 Finished calling > python script > 2012-01-07 09:21:57.693488 [ERR] mod_python.c:293 Error calling python > script > 2012-01-07 09:21:57.693488 [ERR] mod_python.c:164 Python Error by > calling script "emergency.hangup": > Message: global name 'session' is not defined > Exception: None > > Traceback (most recent call last) > File: "/usr/local/freeswitch/scripts/python/emergency/hangup.py", > line 34, in runtime > > > > ---------------------------------------------------------- > Here is the simple script > > python module emergency.hangup: > > from freeswitch import * > > def runtime(arg): > consoleLog("info", print(dir(session))) > > > > Thanks, > Herman > > > Vitalie, > > I asked my provider to switch up to port 5080 and everything is > negotiating perfectly. thanks for your help. > > > > ------------------------------ > *From:* "freeswitch-users-request at lists.freeswitch.org" < > freeswitch-users-request at lists.freeswitch.org> > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Friday, January 6, 2012 11:54 PM > *Subject:* FreeSWITCH-users Digest, Vol 67, Issue 58 > > ----- Forwarded Message ----- > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. 407 undefined? sip port 5060 (Rodney) > 2. Why database is locked? (Valery Kalinin) > 3. Re: Why database is locked? (Ken Rice) > 4. Re: 407 undefined? sip port 5060 (Vitalie Colosov) > I currently have external profile ipkall working on port 5080 but now i am > testing RNK carrier services and they are sending to me on port 5060. I do > not notice anything on the console but they are saying they are sending to > my server and getting this response: > > 2012-01-06 21:52:46.0 000000000 614??????? 6174530953 ( > DID-IN:RNK) (H1 usinteractive-nextdi) 407 undefined > > > > I have the following gateway setup in C:\Program > Files\FreeSWITCH\conf\sip_profiles\external > > > > > > > > > > > > > > > It is exactly the same as my IPkall except for the gateway name and proxy > port. How do I make sure I can accept on this port. Is there somewhere else > I need to be adding this 5060 port information. > > thank you. > > > > Are you running the latest version of FS? > > FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > > > Also are you doing anything externally to mess with the sql db files? > > No. > > > Another possibility is a slow filesystem, we used to have people get this > problem on suse linux. > > CentOS release 5.5 (Final) > > > You could try running your db dir in a ramdisk. > > These problems appeared after the upgrade version of FS. > > > Thank you in advance... > Blow away all the files in freeswitch/db and restart freeswitch > > Save the voicemail.db file if you don?t wanna lose all the voicemails and > give that a try. > > That will force a reset of the all the databases and should clear any > errors... > > > On 1/6/12 10:00 PM, "Valery Kalinin" wrote: > > > Are you running the latest version of FS? > > FreeSWITCH Version 1.0.head (git-4cd616c 2011-12-15 23-36-20 -0500) > > > Also are you doing anything externally to mess with the sql db files? > > No. > > > Another possibility is a slow filesystem, we used to have people get this > problem on suse linux. > > CentOS release 5.5 (Final) > > > You could try running your db dir in a ramdisk. > > These problems appeared after the upgrade version of FS. > > > Thank you in advance... > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > Ask your provider to send sip packets to 5080, instead of 5060. > Every good provider should allow you to select the port YOU want, and not > THEY want. > > If it will not help, then you need to make external profile listen to > 5060, instead of 5080. > So, you will be sending and receiving sip messages between you and > provider, using port 5060 (which is usually used only for your internal > profile and your clients). > > To change this, you need to modify conf/sip_profiles/external.xml > replace to "5060" > > In this case, you will have to reconfigure internal profile to be on a > different port, say 5062, and ask your clients to add :5062 to the sip > server address when they configure their softphone/sip adapter. > > Maybe another option would be to consider provider as one of your internal > clients, and allow password-less communication from his IP, but this is way > too crazy to consider as a good option. > > Vitalie > > > > > > 2012/1/6 Rodney > > I currently have external profile ipkall working on port 5080 but now i am > testing RNK carrier services and they are sending to me on port 5060. I do > not notice anything on the console but they are saying they are sending to > my server and getting this response: > > 2012-01-06 21:52:46.0 000000000 614??????? 6174530953 ( > DID-IN:RNK) (H1 usinteractive-nextdi) 407 undefined > > > > I have the following gateway setup in C:\Program > Files\FreeSWITCH\conf\sip_profiles\external > > > > > > > > > > > > > > > It is exactly the same as my IPkall except for the gateway name and proxy > port. How do I make sure I can accept on this port. Is there somewhere else > I need to be adding this 5060 port information. > > thank you. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/fafa5db3/attachment-0001.html From msc at freeswitch.org Mon Jan 9 19:21:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 08:21:22 -0800 Subject: [Freeswitch-users] IVR (Calling card application) development In-Reply-To: References: <8343AF3F-140F-4C0D-8885-A8CE7B1EE71E@gmail.com> <007201ccce78$b8a39f30$29eadd90$@cz> Message-ID: I'm assuming you're talking about mod_erlang_event. That was developed by user Andrew Thompson (IRC: Vagabond) and is used extensively by the guys at 2600hz. I'd say it's as well supported as any community-contributed FS module. -MC On Mon, Jan 9, 2012 at 6:00 AM, Vitaly Nikolaev wrote: > Hello, > > > db_conn = assert(env:connect("dbname=mydb user=freeswitch > password='t0psecret'")) > > If it 1k calls I will have 1k connections to databases, and 1k lua scripts > in memory. I know there are ways around but > separation calling card logic from freeswitch will give more freedom. > > > Thank you > > PS: Il craft small mod_erlang based test IVR and try it with call > simulator. > > > > > > On Sun, Jan 8, 2012 at 9:45 PM, WTG Technical wrote: > >> Hi Vitaly, >> >> Why not just use embedded lua? It is fast and good for ivr scripting. >> >> FYI http://wiki.freeswitch.org/wiki/Examples_db_agent_login_lua >> >> >> ----- >> S pozdravem >> WTG Technical Department >> http://www.wtg.cz >> >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Vitaly >> Nikolaev >> Sent: Sunday, January 08, 2012 2:46 AM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] IVR (Calling card application) development >> >> Hello, >> >> I am choosing between ESL and mod_erlang, I will have around 1k >> simultaneous >> calls with 10 min average duration. >> >> I will definitely write it in erlang, but question is - how supported is >> mod_erlang ? Is anyone currently use it in production for heavy load >> application ? >> >> Thank you >> Vitaly >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > -- > Vitaly Nikolaev > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/ba2b3027/attachment.html From msc at freeswitch.org Mon Jan 9 19:48:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 08:48:45 -0800 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: Message-ID: FWIW, When experimenting with outbound voice broadcasting and trying to detect SIT tones, I stumbled across a factoid that might help, depending on your locale. In North America, whenever I hit an answering machine or a cell phone voicemail system the beep almost always was detected by my SIT tone detects. It was a total accident that I even found it. IIRC, the 1776.7 Hz tone was the one that almost always detected an answering machine/voicemail beep. No guarantees that it will work, but it's probably worth investigating. If it handles 90% or more of your targets then it's at least as good as any other type of detection you'll find. -MC On Mon, Jan 9, 2012 at 7:26 AM, Ken Rice wrote: > Detecting the tone is exactly what mod_vmd and mod_avmd do now... Not > exactly the most efficient way to detect this sort of thing, but it does > work > > K > > > > On 1/9/12 9:22 AM, "curriegrad2004" wrote: > > Since an answering machine creates a tone, have you looked into > mod_spandsp's tone_detect feature? I think it does what you want it to do. > > On 2012-01-09 3:50 AM, "Avi Marcus" wrote: > > Yes, even with no other way of making it work more efficiently... 1 > machine can handle 50 calls that have actually picked up, before > sending it to the recording. > So your actual window is maximum 30? seconds per calls that picks up > that mod_avmd needs to be running. So then that's 50 per 30 second > windows so 100 calls per minute. That's 1,000 actual connected calls > within 10 minutes. Is that within your threshold? For using ONE > machine? > (What happens if a person picks up? On silence it just starts the > recording..? Or do you just have a looping first 30 seconds so it's ok > if they miss the start?) > > -Avi > > > On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall > wrote: > > Hi David > > > > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? > If > > not, you can distribute the load between 2 or more machines. > > > > Have you tried this? (from Wiki) > > > > For best results in getting accurate detection as well a saving system > > resources (CPU & memory), you should do the following: > > > > Wait until AFTER the call has been answered before starting AVMD. In many > > cases you may find it best to wait a few seconds after the call is > answered > > before starting AVMD. > > > > Starting AVMD before the call is answered will often result in false > beeps > > and wastes CPU resources. > > > > Once you have received the AVMD event, you should explicitly stop AVMD, > even > > though it will not return another event. > > > > > > > > regards, > > Anita > > > > > > > > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: > >> > >> Hello. > >> > >> I need to detect answering machine beeps on 1,000+ channels > >> simultaneously. Unfortunately, this is a real requirement, not a > >> pseudo-requirement that can be engineered away. This is for a mass > >> emergency notification system for government, K-12, and higher > education.. > >> > >> In the past, I have done this with proprietary hardware (Dialogic and > >> Aculab), which handles this sort of thing without overloading the host > CPU, > >> as the hardware's DSPs handle the detections. I want to see if I can > do the > >> same thing with FreeSWITCH. > >> > >> Alas, I find the following warning in the wiki page for mod_avmd: > >> > >> AVMD (and VMD) are both very CPU intensive. You need to be aware of this > >> fact when using it. It will drastically reduce your call capacity if > you do > >> not manage it correctly. On the other hand it is a very useful tool, > and if > >> managed properly will be a great aid for calls needing to do Voice Mail > >> Detection. > >> > >> Eric states, "You can expect about ~50 simultaneous instances on an > >> Intel i7 920 CPU." > >> > >> So, it sounds like VMD and AVMD won't work for what I need to do. I > also > >> don't think that the tone_detect application will work, as this requires > >> specific frequencies (as opposed to ranges of frequencies), and > answering > >> machines and voicemail come with beeps at all sorts of frequencies. > >> > >> Does anyone know anything, either free or commercial, that I can use > with > >> FreeSWITCH to do this many simultaneous detections? I know of > commercial > >> software-only platforms (e.g., Aculab's Prosody S) that claim to be > able to > >> this, so it seems like it should be possible. > >> > >> Thanks in advance, > >> David Stein > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/3413ba2e/attachment-0001.html From anthony.minessale at gmail.com Mon Jan 9 19:58:25 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Jan 2012 10:58:25 -0600 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: Message-ID: Nicer box with all the money saved from free software? =p On Mon, Jan 9, 2012 at 10:48 AM, Michael Collins wrote: > FWIW, > > When experimenting with outbound voice broadcasting and trying to detect > SIT tones, I stumbled across a factoid that might help, depending on your > locale. In North America, whenever I hit an answering machine or a cell > phone voicemail system the beep almost always was detected by my SIT tone > detects. It was a total accident that I even found it. IIRC, the 1776.7 Hz > tone was the one that almost always detected an answering machine/voicemail > beep. > > No guarantees that it will work, but it's probably worth investigating. If > it handles 90% or more of your targets then it's at least as good as any > other type of detection you'll find. > > -MC > > > On Mon, Jan 9, 2012 at 7:26 AM, Ken Rice wrote: > >> Detecting the tone is exactly what mod_vmd and mod_avmd do now... Not >> exactly the most efficient way to detect this sort of thing, but it does >> work >> >> K >> >> >> >> On 1/9/12 9:22 AM, "curriegrad2004" wrote: >> >> Since an answering machine creates a tone, have you looked into >> mod_spandsp's tone_detect feature? I think it does what you want it to do. >> >> On 2012-01-09 3:50 AM, "Avi Marcus" wrote: >> >> Yes, even with no other way of making it work more efficiently... 1 >> machine can handle 50 calls that have actually picked up, before >> sending it to the recording. >> So your actual window is maximum 30? seconds per calls that picks up >> that mod_avmd needs to be running. So then that's 50 per 30 second >> windows so 100 calls per minute. That's 1,000 actual connected calls >> within 10 minutes. Is that within your threshold? For using ONE >> machine? >> (What happens if a person picks up? On silence it just starts the >> recording..? Or do you just have a looping first 30 seconds so it's ok >> if they miss the start?) >> >> -Avi >> >> >> On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall >> wrote: >> > Hi David >> > >> > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls >> ? If >> > not, you can distribute the load between 2 or more machines. >> > >> > Have you tried this? (from Wiki) >> > >> > For best results in getting accurate detection as well a saving system >> > resources (CPU & memory), you should do the following: >> > >> > Wait until AFTER the call has been answered before starting AVMD. In >> many >> > cases you may find it best to wait a few seconds after the call is >> answered >> > before starting AVMD. >> > >> > Starting AVMD before the call is answered will often result in false >> beeps >> > and wastes CPU resources. >> > >> > Once you have received the AVMD event, you should explicitly stop AVMD, >> even >> > though it will not return another event. >> > >> > >> > >> > regards, >> > Anita >> > >> > >> > >> > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: >> >> >> >> Hello. >> >> >> >> I need to detect answering machine beeps on 1,000+ channels >> >> simultaneously. Unfortunately, this is a real requirement, not a >> >> pseudo-requirement that can be engineered away. This is for a mass >> >> emergency notification system for government, K-12, and higher >> education.. >> >> >> >> In the past, I have done this with proprietary hardware (Dialogic and >> >> Aculab), which handles this sort of thing without overloading the host >> CPU, >> >> as the hardware's DSPs handle the detections. I want to see if I can >> do the >> >> same thing with FreeSWITCH. >> >> >> >> Alas, I find the following warning in the wiki page for mod_avmd: >> >> >> >> AVMD (and VMD) are both very CPU intensive. You need to be aware of >> this >> >> fact when using it. It will drastically reduce your call capacity if >> you do >> >> not manage it correctly. On the other hand it is a very useful tool, >> and if >> >> managed properly will be a great aid for calls needing to do Voice Mail >> >> Detection. >> >> >> >> Eric states, "You can expect about ~50 simultaneous instances on an >> >> Intel i7 920 CPU." >> >> >> >> So, it sounds like VMD and AVMD won't work for what I need to do. I >> also >> >> don't think that the tone_detect application will work, as this >> requires >> >> specific frequencies (as opposed to ranges of frequencies), and >> answering >> >> machines and voicemail come with beeps at all sorts of frequencies. >> >> >> >> Does anyone know anything, either free or commercial, that I can use >> with >> >> FreeSWITCH to do this many simultaneous detections? I know of >> commercial >> >> software-only platforms (e.g., Aculab's Prosody S) that claim to be >> able to >> >> this, so it seems like it should be possible. >> >> >> >> Thanks in advance, >> >> David Stein >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/32768c11/attachment.html From avi at avimarcus.net Mon Jan 9 20:07:09 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 9 Jan 2012 19:07:09 +0200 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: Message-ID: That wikified somewhere? And how do you handle someone picking up? wait for silence? -Avi On Mon, Jan 9, 2012 at 6:48 PM, Michael Collins wrote: > FWIW, > > When experimenting with outbound voice broadcasting and trying to detect SIT > tones, I stumbled across a factoid that might help, depending on your > locale. In North America, whenever I hit an answering machine or a cell > phone voicemail system the beep almost always was detected by my SIT tone > detects. It was a total accident that I even found it. IIRC, the 1776.7 Hz > tone was the one that almost always detected an answering machine/voicemail > beep. > > No guarantees that it will work, but it's probably worth investigating. If > it handles 90% or more of your targets then it's at least as good as any > other type of detection you'll find. > > -MC > > > On Mon, Jan 9, 2012 at 7:26 AM, Ken Rice wrote: >> >> Detecting the tone is exactly what mod_vmd and mod_avmd do now... Not >> exactly the most efficient way to detect this sort of thing, but it does >> work >> >> K >> >> >> >> On 1/9/12 9:22 AM, "curriegrad2004" wrote: >> >> Since an answering machine creates a tone, have you looked into >> mod_spandsp's tone_detect feature? I think it does what you want it to do. >> >> On 2012-01-09 3:50 AM, "Avi Marcus" wrote: >> >> Yes, even with no other way of making it work more efficiently... 1 >> machine can handle 50 calls that have actually picked up, before >> sending it to the recording. >> So your actual window is maximum 30? seconds per calls that picks up >> that mod_avmd needs to be running. So then that's 50 per 30 second >> windows so 100 calls per minute. That's 1,000 actual connected calls >> within 10 minutes. Is that within your threshold? For using ONE >> machine? >> (What happens if a person picks up? On silence it just starts the >> recording..? Or do you just have a looping first 30 seconds so it's ok >> if they miss the start?) >> >> -Avi >> >> >> On Mon, Jan 9, 2012 at 12:06 PM, Anita Hall >> wrote: >> > Hi David >> > >> > Question: Do you want to use only 1 machine for doing AMD on 1k+ calls ? >> > If >> > not, you can distribute the load between 2 or more machines. >> > >> > Have you tried this? (from Wiki) >> > >> > For best results in getting accurate detection as well a saving system >> > resources (CPU & memory), you should do the following: >> > >> > Wait until AFTER the call has been answered before starting AVMD. In >> > many >> > cases you may find it best to wait a few seconds after the call is >> > answered >> > before starting AVMD. >> > >> > Starting AVMD before the call is answered will often result in false >> > beeps >> > and wastes CPU resources. >> > >> > Once you have received the AVMD event, you should explicitly stop AVMD, >> > even >> > though it will not return another event. >> > >> > >> > >> > regards, >> > Anita >> > >> > >> > >> > On Sun, Jan 8, 2012 at 7:20 AM, David Stein wrote: >> >> >> >> Hello. >> >> >> >> I need to detect answering machine beeps on 1,000+ channels >> >> simultaneously.? Unfortunately, this is a real requirement, not a >> >> pseudo-requirement that can be engineered away.? This is for a mass >> >> emergency notification system for government, K-12, and higher >> >> education.. >> >> >> >> In the past, I have done this with proprietary hardware (Dialogic and >> >> Aculab), which handles this sort of thing without overloading the host >> >> CPU, >> >> as the hardware's DSPs handle the detections.? I want to see if I can >> >> do the >> >> same thing with FreeSWITCH. >> >> >> >> Alas, I find the following warning in the wiki page for mod_avmd: >> >> >> >> AVMD (and VMD) are both very CPU intensive. You need to be aware of >> >> this >> >> fact when using it. It will drastically reduce your call capacity if >> >> you do >> >> not manage it correctly. On the other hand it is a very useful tool, >> >> and if >> >> managed properly will be a great aid for calls needing to do Voice Mail >> >> Detection. >> >> >> >> ??? Eric states, "You can expect about ~50 simultaneous instances on an >> >> Intel i7 920 CPU." >> >> >> >> So, it sounds like VMD and AVMD won't work for what I need to do.? I >> >> also >> >> don't think that the tone_detect application will work, as this >> >> requires >> >> specific frequencies (as opposed to ranges of frequencies), and >> >> answering >> >> machines and voicemail come with beeps at all sorts of frequencies. >> >> >> >> Does anyone know anything, either free or commercial, that I can use >> >> with >> >> FreeSWITCH to do this many simultaneous detections?? I know of >> >> commercial >> >> software-only platforms (e.g., Aculab's? Prosody S) that claim to be >> >> able to >> >> this, so it seems like it should be possible. >> >> >> >> Thanks in advance, >> >> David Stein >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From vetali100 at gmail.com Mon Jan 9 20:29:53 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Mon, 9 Jan 2012 09:29:53 -0800 Subject: [Freeswitch-users] Max calls Limit? In-Reply-To: References: <1325968310.32667.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: Very good tool, which gives you lost packets, jitter and MOS for each call leg: http://www.voipmonitor.org/ 2012/1/9 Anita Hall > Thanks guys, I was able to do more than 1200 calls on a lowly AMD machine, > after which the 2 GB RAM went out and 1 GB swap got used. > > Is there a way to automatically get a sense of the call "quality"? > Something like, find if UDP packets are getting dropped :) > > regards, > Anita > > > > > On Sun, Jan 8, 2012 at 2:48 AM, Ken Rice wrote: > >> This is also adjustable from the FS CLI via the fsctl max_sessions ### >> command... Replace ### with the max number of sessions you want. >> >> Note this is MAX Sessions, not max calls... So if you are doing 2 legged >> calls, the number of calls you can do is effective 1/2 that setting since >> each call leg is a session. >> >> Setting it from the CLI is not kept across restarts of freeswitch for >> that see switch.xml >> >> This setting should be set at a sane level for your particular hardware >> as yes it does keep freeswitch starving the server of memory and causes FS >> to start failing calls at that point. >> >> The similar setting max_sps is for max sessions per second and this is >> again to keep you from melting down your hardware. The current setting is >> sufficient for 99% of installs however if you are doing something out of >> the norm ie: high volume calling you may need to adjust it up >> >> K >> >> >> >> On 1/7/12 2:31 PM, "Sherif Omran" wrote: >> >> Yes >> con/autoload/switch.xml >> >> >> --- On *Sat, 1/7/12, Anita Hall * wrote: >> >> >> From: Anita Hall >> Subject: [Freeswitch-users] Max calls Limit? >> To: "FreeSWITCH Users Help" >> Date: Saturday, January 7, 2012, 10:32 AM >> >> Is there any config variable somewhere which is limiting the max call >> limit on FS to 1000? >> >> Set-up: >> Taking FS for max call load on my machine. >> >> User freeswitch is calling internal gateway and playing demo-ivr >> >> User freeswitch defined in vars.xml >> >> >> >> > data="default_provider_from_domain=10.60.20.145"/> >> >> >> >> >> Internal Gateway defined in directory/default/freeswitch.xml >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> After 1000 calls, it is giving the same error >> -ERR DESTINATION_OUT_OF_ORDER >> >> Yes, my CPU (quad-core AMD) and RAM (2GB only) are under heavy stress by >> the time 1000 calls are being done, but is it not a little strange that >> every time the resources get exhausted at precisely 1000 calls! >> >> >> regards, >> Anita >> >> >> -----Inline Attachment Follows----- >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org < >> /mc/compose?to=FreeSWITCH-users at lists.freeswitch.org> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/0466665d/attachment.html From anthony.minessale at gmail.com Mon Jan 9 20:42:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Jan 2012 11:42:22 -0600 Subject: [Freeswitch-users] voicemail_inject (Mod voicemail) problems In-Reply-To: References: Message-ID: is 192.168.2.2 a valid domain in your user directory? On Mon, Jan 9, 2012 at 8:40 AM, Dennis wrote: > hi, > > we have problems getting "voicemail_inject" to work. whatever we do, > we alway get the same error message (fs newest version). > > for example we do: voicemail_inject 1000 at 192.168.2.2 > /usr/local/freeswitch/recordings/1000_2012-01-07-21-47-44.wav 1001 > test > > we will always get: Error: > [group=[@domain]|domain=|[@]] > [] [] > > what are we doing wrong? > > > thanks a lot for your help! > > > kind regards > dennis > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/dd391244/attachment.html From msc at freeswitch.org Mon Jan 9 20:57:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 09:57:12 -0800 Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines In-Reply-To: References: Message-ID: On Mon, Jan 9, 2012 at 9:07 AM, Avi Marcus wrote: > That wikified somewhere? > And how do you handle someone picking up? wait for silence? > > No, just random knowledge flying around between my ears. Also, I was making assumptions - if the SIT tone was detected in early media then I assumed it was a traditional SIT; if the SIT tone was detected after the b leg answered then I assumed it was some sort of answering machine or voicemail. It was never meant to be scientific - I was just experimenting with voice broadcasting in the very early days of FreeSWITCH and OpenZAP. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/61c8b10a/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 9 22:38:34 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 9 Jan 2012 19:38:34 +0000 (GMT) Subject: [Freeswitch-users] NAT Confusion Message-ID: <1326137914.97220.YahooMailNeo@web29404.mail.ird.yahoo.com> Hi, Newbie here (again !) ....? Hope someone here can help. I'm having a few NAT related issues and the sparse, sometimes out-of-date wikis on the freeswitch site are not helping.... ;-( My layout is as follows ... Freewitch Server ? ? ? .....**INTERNET**........[FIREWALL].....[BRIA 3 SOFTPHONE] (Public IP, No NAT) ?.......................................NAT..............Internal IP Firewall traversal method in Bria is set to "none" as I've no ICE/STUN/TURN server. Freeswitch is started with the -nonat -nonatmap options because it is on a public IP with no NAT. (1) Outbound Calls I can dial out. I can hear the remote end. The remote end cannot hear me. (2) Inbound Calls These do not work at all. Console output (excluding debug level messages, and with IP and Phone Numbers changed to protect the innocent) 2012-01-09 19:22:14.076501 [NOTICE] switch_channel.c:924 New Channel sofia/internal/02079460949 at 10.0.0.134 [3c4ec25c-3af7-11e1-96ae-155d65675ea5] 2012-01-09 19:22:14.076501 [INFO] mod_dialplan_xml.c:481 Processing 02079460949 <02079460949>->01174960949 in context public 2012-01-09 19:22:14.096468 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/02079460949 at 10.0.0.134 to XML[1000 at default] 2012-01-09 19:22:14.096468 [INFO] mod_dialplan_xml.c:481 Processing 02079460949 <02079460949>->1000 in context default 2012-01-09 19:22:14.096468 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1000 at 10.14.2.2:48101 [3c4f8cb4-3af7-11e1-96b5-155d65675ea5] 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-09 19:22:14.156503 [NOTICE] sofia.c:6246 Hangup sofia/internal/sip:1000 at 10.14.2.2:48101 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.156503 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.156503 [INFO] mod_dptools.c:2900 Originate Failed. ?Cause: SERVICE_NOT_IMPLEMENTED 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1398 Session 27 (sofia/internal/sip:1000 at 10.14.2.2:48101) Ended 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1000 at 10.14.2.2:48101 [CS_DESTROY] 2012-01-09 19:22:14.156503 [NOTICE] mod_dptools.c:3019 Hangup sofia/internal/02079460949 at 10.0.0.134 [CS_EXECUTE] [SERVICE_NOT_IMPLEMENTED] 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1398 Session 26 (sofia/internal/02079460949 at 10.0.0.134) Ended 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/02079460949 at 10.0.0.134 [CS_DESTROY] The above would suggest I need to boot Freeswitch with NAT turned on ? ?But I'm confused as to why ? ?Because Freeswitch has no need to discover its public IP address. Thanks Bob From msc at freeswitch.org Mon Jan 9 22:48:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jan 2012 11:48:35 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call This Week: Sangoma Message-ID: 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 placeon the wiki. Talk to you all on Wednesday! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/2422e3e0/attachment.html From ccesario at tecnomega.com.br Tue Jan 10 01:16:52 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Mon, 9 Jan 2012 20:16:52 -0200 Subject: [Freeswitch-users] Xfer Call Back Message-ID: <4F0B6754.1090409@tecnomega.com.br> Hello, In Att Xfer exists any method to call return to original call if called dont attend the call ? Eg: A transfer to B and hold phone, if B dont attend call the call back to caller . Grets Carlos From anthony.minessale at gmail.com Tue Jan 10 03:26:13 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 9 Jan 2012 18:26:13 -0600 Subject: [Freeswitch-users] "expire" for outgoing registrations not handled correctly? In-Reply-To: References: Message-ID: It might be because we only check for expiration every 30 seconds. update to latest GIT and I added a profile param: registration-thread-frequency The default is 30 so you can try setting it to 10 or 5 or something. 2012/1/9 Roland H?nel > Hi, > > It seems that I'm running into a timing issue with outgoing (gateway) > registrations, which results in some milliseconds in that I can't send > calls over that gateway. > > I have configured a gateway like this: > > > > > > > > ^M > > > > ^M > > If I look at the "200 OK" messages for the SIP REGISTER packet, the server > sends me an "Expires: 30", so FreeSwitch should re-REGISTER every 30 > seconds. > > However, that's what happening: > > 2012-01-09 16:37:00.804865 [NOTICE] sofia_reg.c:407 Registering duro > 2012-01-09 16:37:31.804880 [NOTICE] sofia_reg.c:407 Registering duro > 2012-01-09 16:38:03.804894 [NOTICE] sofia_reg.c:407 Registering duro > 2012-01-09 16:38:35.824877 [NOTICE] sofia_reg.c:407 Registering duro > *2012-01-09 16:39:06.825090 [NOTICE] sofia_reg.c:407 Registering duro* > 2012-01-09 16:39:39.824881 [NOTICE] sofia_reg.c:407 Registering duro > > > You can see that it's *approximately* 30 seconds betwee registrations, > but in fact every time it's just a little more than 30 secs. This alone > makes me feel a little bit strange, since my registration might have been > expired during this time window (and consequently, maybe I'm loosing > incoming calls because of that). > > Even for outgoing calls, this seems to be an issue, let's take a look at > the 16:39:06.825090 re-REGISTER. > > We can see that at 16:39:05 (no micro-seconds here) the gateway becomes > "UNREGED": > > Content-Length: 542 > Content-Type: text/event-plain > > Event-Subclass: sofia::gateway_state > Event-Name: CUSTOM > Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 > FreeSWITCH-Hostname: lap597 > FreeSWITCH-Switchname: lap597 > FreeSWITCH-IPv4: 192.168.232.164 > FreeSWITCH-IPv6: ::1 > *Event-Date-Local: 2012-01-09 16:39:05* > Event-Date-GMT: Mon, 09 Jan 2012 15:39:05 GMT > Event-Date-Timestamp: 1326123545824882 > Event-Calling-File: sofia_reg.c > Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event > Event-Calling-Line-Number: 151 > Gateway: duro > State: UNREGED > Ping-Status: UP > > > But the new REGISTER is done a full second later (this is not just a > display issue, I saw those messages scrolling by and there was just a > second pause between them): > > Event-Subclass: sofia::gateway_state > Event-Name: CUSTOM > Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 > FreeSWITCH-Hostname: lap597 > FreeSWITCH-Switchname: lap597 > FreeSWITCH-IPv4: 192.168.232.164 > FreeSWITCH-IPv6: ::1 > Event-Date-Local: 2012-01-09 16:39:06 > *Event-Date-GMT: Mon, 09 Jan 2012 15:39:06 GMT* > Event-Date-Timestamp: 1326123546825090 > Event-Calling-File: sofia_reg.c > Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event > Event-Calling-Line-Number: 151 > Gateway: duro > State: TRYING > Ping-Status: DOWN > > > Then, the gateway gets re-registered in a couple of milliseconds and > everything is fine again. > > *It seems that this behaviour effectively creates a timespan of 1-2 > seconds, where the gateway is not available for outgoing (and maybe also > incoming) calls.* > > Does anyone of you know this issue? I wonder because this looks just to > easy to be wrong in this fashion? > > Greetings, > Roland > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/50bee15f/attachment-0001.html From sunwood360 at gmail.com Tue Jan 10 04:27:55 2012 From: sunwood360 at gmail.com (envelopes envelopes) Date: Mon, 9 Jan 2012 17:27:55 -0800 Subject: [Freeswitch-users] NAT Confusion In-Reply-To: <1326137914.97220.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1326137914.97220.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: Why not pick up a public stun server and config it in your softphone, most likely it will address your issue. On Jan 9, 2012 11:40 AM, "Bob Smith" wrote: > Hi, > > Newbie here (again !) .... > > Hope someone here can help. > > I'm having a few NAT related issues and the sparse, sometimes out-of-date > wikis on the freeswitch site are not helping.... ;-( > > My layout is as follows ... > > > Freewitch Server .....**INTERNET**........[FIREWALL].....[BRIA 3 > SOFTPHONE] > (Public IP, No NAT) > .......................................NAT..............Internal IP > > Firewall traversal method in Bria is set to "none" as I've no > ICE/STUN/TURN server. > > Freeswitch is started with the -nonat -nonatmap options because it is on a > public IP with no NAT. > > (1) Outbound Calls > > I can dial out. > I can hear the remote end. > The remote end cannot hear me. > > (2) Inbound Calls > > These do not work at all. > > Console output (excluding debug level messages, and with IP and Phone > Numbers changed to protect the innocent) > > 2012-01-09 19:22:14.076501 [NOTICE] switch_channel.c:924 New Channel > sofia/internal/02079460949 at 10.0.0.134[3c4ec25c-3af7-11e1-96ae-155d65675ea5] > 2012-01-09 19:22:14.076501 [INFO] mod_dialplan_xml.c:481 Processing > 02079460949 <02079460949>->01174960949 in context public > 2012-01-09 19:22:14.096468 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/02079460949 at 10.0.0.134 to XML[1000 at default] > 2012-01-09 19:22:14.096468 [INFO] mod_dialplan_xml.c:481 Processing > 02079460949 <02079460949>->1000 in context default > 2012-01-09 19:22:14.096468 [NOTICE] switch_channel.c:924 New Channel > sofia/internal/sip:1000 at 10.14.2.2:48101[3c4f8cb4-3af7-11e1-96b5-155d65675ea5] > 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping > disabled > 2012-01-09 19:22:14.096468 [INFO] switch_nat.c:590 NAT port mapping > disabled > 2012-01-09 19:22:14.156503 [NOTICE] sofia.c:6246 Hangup sofia/internal/ > sip:1000 at 10.14.2.2:48101 [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED] > 2012-01-09 19:22:14.156503 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [user] cause: [SERVICE_NOT_IMPLEMENTED] > 2012-01-09 19:22:14.156503 [INFO] mod_dptools.c:2900 Originate Failed. > Cause: SERVICE_NOT_IMPLEMENTED > 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1398 Session 27 > (sofia/internal/sip:1000 at 10.14.2.2:48101) Ended > 2012-01-09 19:22:14.156503 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/sip:1000 at 10.14.2.2:48101 [CS_DESTROY] > 2012-01-09 19:22:14.156503 [NOTICE] mod_dptools.c:3019 Hangup > sofia/internal/02079460949 at 10.0.0.134 [CS_EXECUTE] > [SERVICE_NOT_IMPLEMENTED] > 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1398 Session 26 > (sofia/internal/02079460949 at 10.0.0.134) Ended > 2012-01-09 19:22:14.176780 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/02079460949 at 10.0.0.134 [CS_DESTROY] > > > The above would suggest I need to boot Freeswitch with NAT turned on ? > But I'm confused as to why ? Because Freeswitch has no need to discover > its public IP address. > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120109/b2a930e4/attachment.html From rhow at exemail.com.au Tue Jan 10 04:34:41 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 09:34:41 +0800 Subject: [Freeswitch-users] Any got a windows java esl.jar and lib built ? Message-ID: <4F0B95B1.90208@exemail.com.au> Hi, Has anyone got a windows esl.jar and corrsponding lib built?. I'm using the 2011 12 28 build of the windows binary. The instructions for building the ESL from source on the wiki are all for linux systems. There is also the pure java esl_client.jar, but it hasn't had updates since aug 2010 and the documentation looks pretty sparse!. Or doesn't it really matter, is it pretty stable and just like a drop in for the core version? I've generally learnt from spending hours and hours and hours messing with things, unless it looks supported or active it is going to take a lot of time to get it to work, and even then it probably won't work properly... Thanks! Ryan From bwibowo at gmail.com Tue Jan 10 06:08:02 2012 From: bwibowo at gmail.com (bwibowo at gmail.com) Date: Tue, 10 Jan 2012 03:08:02 +0000 Subject: [Freeswitch-users] (no subject) Message-ID: <1093484023-1326164883-cardhu_decombobulator_blackberry.rim.net-1568862817-@b27.c2.bise3.blackberry> From bwibowo at gmail.com Tue Jan 10 06:09:50 2012 From: bwibowo at gmail.com (bwibowo at gmail.com) Date: Tue, 10 Jan 2012 03:09:50 +0000 Subject: [Freeswitch-users] Ota in sip Message-ID: <1284558287-1326164994-cardhu_decombobulator_blackberry.rim.net-392685695-@b27.c2.bise3.blackberry> Dear all Is there any implementation of OTA in sip. Purpose is to upload new parameter when user connected to sip server. Thx Budi From olimonkey at gmail.com Tue Jan 10 06:31:01 2012 From: olimonkey at gmail.com (Oliver Schenk) Date: Tue, 10 Jan 2012 11:31:01 +0800 Subject: [Freeswitch-users] CISCO 2811 Freeswitch IVR In-Reply-To: References: <103FB5D0-E69B-49EA-AF35-704CAC2D768A@freeswitch.org> <1FFF97C269757C458224B7C895F35F1502BDFB@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F1502BEBC@cantor.std.visionutv.se> Message-ID: https://supportforums.cisco.com/message/3529491 On Mon, Jan 9, 2012 at 2:25 PM, Oliver Schenk wrote: > Ok, this is starting to become 100% CISCO issue so I might have to > move this topic to a CISCO forum instead. I played around with a few > CISCO parameters. > > > I added this to the voice port: > > voice-port 0/3/2 > ?supervisory disconnect dualtone mid-call > ?supervisory answer dualtone > ?no battery-reversal > > > > Funny thing happens now. When I answer the call I have to make a noise > into the phone before the CISCO actually picks up! I just had to shake > my head there...and start bashing my head against the keyboard. > Wouldn't that be funny... Instruction Manual: "if you hear nothing, > please snap your fingers into the phone receiver and you'll hear the > IVR start playing." > > > So anyway, I tried to set: > > ? supervisory answer dualtone > > to > > ? supervisory answer dualtone sensitivity high > > > That didn't make any difference. > > > So next I added: > > ?no comfort-noise > > > That didn't make any difference. > > > Then I tried: > > ? battery-reversal answer > > > As soon as I do that the CISCO once again causes FS to start playing > the IVR before I even pick up the phone. So back to square one. > > > Starting to run out of options here... > > > > > > > > > > > > On Mon, Jan 9, 2012 at 11:19 AM, Oliver Schenk wrote: >> I had this in my call string: >> >> {sip_cid_type=rpid,ignore_early_media=true} >> >> >> Which caused the 503 Service Unavailble error. The CISCO had some >> errors just prior to the 503 message: >> >> Jan ?9 01:30:36: >> //152/4620425F81D5/SIP/Info/ccsip_indicate_rt_packet_stats: Processing >> stats for callid=152, proc_id=9 >> Jan ?9 01:30:38: //152/4620425F81D5/SIP/Media/sipSPIUpdateRtpSession: >> stun is disabled for stream:47110C14 >> Jan ?9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: >> Requesting stats for callid=152 >> Jan ?9 01:30:38: //152/4620425F81D5/SIP/Info/ccsip_call_statistics: >> Stats request failed for callid=152, dstCallID=153, rc=-7 >> Jan ?9 01:30:38: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued >> event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT >> Jan ?9 01:30:38: >> //-1/xxxxxxxxxxxx/SIP/Info/ccsip_process_sipspi_queue_event: >> ccsip_spi_get_msg_type returned: 3 for event 7 >> Jan ?9 01:30:38: >> //152/4620425F81D5/SIP/Info/act_recdinvite_disconnect: Performing >> disconnect >> >> >> >> Something to do with "Stats request failed for callid". Which then >> causes a "SIPSPI_EV_CC_CALL_DISCONNECT" event. >> >> >> >> So what I did is I removed "sip_cid_type=rpid" from my call string. >> >> >> Now, when I pick up the phone I just get a bit of noise, but nothing >> else. The CISCO does not send any SIP messages at all. It's as if it >> doesn't even know I picked up the phone. Does that mean the dualtone >> answer supervision setting in the CISCO is not working? >> >> >> >> After I hang up the phone I get: >> >> >> CANCEL from FS ?(probably due to call timeout) >> then the CISCO sends OK >> then 487 Request Cancelled >> then ACK from FS >> >> >> Output in FS just prior to my phone ringing is: >> >> 2012-01-09 11:16:03.637084 [NOTICE] switch_channel.c:816 New Channel >> sofia/internal/1092122856 at 192.168.255.1 >> [042f895c-4a34-4295-b250-0291a0f8b91b] >> 2012-01-09 11:16:07.128527 [INFO] sofia.c:740 >> sofia/internal/1092122856 at 192.168.255.1 Update Callee ID to "Outbound >> Call" <1092122856> >> 2012-01-09 11:16:07.128527 [NOTICE] sofia_glue.c:3793 Pre-Answer >> sofia/internal/1092122856 at 192.168.255.1! >> >> When I pick up the phone there is obviously no further output because >> the CISCO hasn't detected the pickup. >> >> >> Hmmmm!!! Still doing my head in. >> >> >> Anyone? lol >> >> >> >> >> On Mon, Jan 9, 2012 at 9:33 AM, Oliver Schenk wrote: >>> Alright, so I tried to set ignore_early_media=true, but the phone >>> isn't being answered at all anymore. So i went back to the CISCO and >>> turned on debugging. I'm now getting a different sequence of events. >>> >>> >>> I still get: >>> >>> INVITE from FS >>> 100 TRYING from CISCO >>> 183 SESSION PROGRESS from CISCO >>> >>> then, when I pick up the phone I get: >>> >>> 503 SERVICE UNAVAILABLE from CISCO >>> ACK from FS >>> >>> >>> I wonder what's going on now. >>> >>> I'll have to play around with the "supervisiory answer dual-tone". >>> >>> >>> >>> CISCO voice-port config: >>> >>> >>> voice-port 0/3/0 >>> ?supervisory disconnect dualtone mid-call >>> ?supervisory answer dualtone >>> ?output attenuation -3 >>> ?no comfort-noise >>> ?cptone AU >>> ?timeouts call-disconnect 5 >>> ?timeouts wait-release 5 >>> ?connection plar opx 1001 >>> ?impedance complex1 >>> ?caller-id enable >>> >>> >>> >>> Cheers, >>> >>> Oliver >>> >>> >>> >>> On Sat, Jan 7, 2012 at 9:28 AM, Oliver Schenk wrote: >>>> No, at the time I was no longer using "ignore_early_media= true", >>>> because initially it didn't work. >>>> I was actually thinking of putting this property back in again, thanks >>>> a lot! I'll try it first thing Monday. >>>> >>>> >>>> With regard to EXECUTE log line, either I don't have the right level >>>> of logging turned on or something else is going on, because I've never >>>> seen any such log entries before with my managed application; yet my >>>> IVR app definitely does get executed. I don't think I have the DEBUG >>>> logging output level turned on so that could explain it... >>>> >>>> >>>> Thanks all, >>>> >>>> Oliver >>>> >>>> >>>> >>>> On Fri, Jan 6, 2012 at 11:21 PM, Peter Olsson >>>> wrote: >>>>> Are you still using ignore_early_media=true - this must be set for this to work correctly. >>>>> >>>>> You will see a EXECUTE log line when FS executes the application, with ignore_early_media enabled it shouldn't execute until the call has been answered. I just tried it myself, and it works as expected. >>>>> >>>>> Example "originate {ignore_early_media=true}sofia/internal/number at host &park()" >>>>> >>>>> Park application is only executed after the call was answered. >>>>> >>>>> /Peter >>>>> >>>>> ________________________________________ >>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>>>> Skickat: den 6 januari 2012 12:04 >>>>> Till: FreeSWITCH Users Help >>>>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>>>> >>>>> Because I'm using an FXO card with voice, I added something to my >>>>> CISCO conf. Many others had the same thing: >>>>> >>>>> >>>>> voice-port 0/3/0 >>>>> ? ... >>>>> ? supervisory disconnect dualtone mid-call >>>>> ? supervisory answer dualtone ? ?<---- ADDED THIS ONE >>>>> ? ... >>>>> >>>>> >>>>> >>>>> Once I added this, the FS output now just showed the following while >>>>> the phone was ringing: >>>>> >>>>> 2012-01-05 16:19:31.644440 [NOTICE] switch_channel.c:816 New Channel >>>>> sofia/internal/109212xxxx at 192.168.x.x >>>>> [69e3f13d-1e2a-409e-97a4-b5526ea6e4ec] >>>>> 2012-01-05 16:19:35.124882 [INFO] sofia.c:740 >>>>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "Outbound >>>>> Call" <109212xxxx> >>>>> 2012-01-05 16:19:35.126883 [NOTICE] sofia_glue.c:3793 Pre-Answer >>>>> sofia/internal/109212xxxx at 192.168.x.x! >>>>> >>>>> >>>>> Where as previous it would show the above and also show the following: >>>>> >>>>> 2012-01-05 16:19:35.127883 [INFO] switch_channel.c:2456 >>>>> sofia/internal/109212xxxx at 192.168.x.x Flipping CID from "" >>>>> <0000000000> to "Outbound Call" <109212xxxx> >>>>> 2012-01-05 16:19:35.137384 [INFO] sofia.c:740 >>>>> sofia/internal/109212xxxx at 192.168.x.x Update Callee ID to "109212xxxx" >>>>> <1092122856> >>>>> 2012-01-05 16:19:35.138384 [NOTICE] sofia.c:5296 Channel >>>>> [sofia/internal/109212xxxx at 192.168.x.x] has been answered >>>>> >>>>> >>>>> >>>>> BUT, the IVR still started playing even before I pick up the phone. >>>>> Hmmmm.....so why is FS still starting the managed application when the >>>>> call has not been answered yet. Are we all sure that the managed >>>>> application should not be executed until the call "has been answered" >>>>> shows up in the log file? >>>>> >>>>> >>>>> Will have to keep testing on monday as I don't have access to the >>>>> CISCO from where i am now. I'll have to see whether the CISCO changes >>>>> had any impact on the times at which the SIP messages are sent back >>>>> and forth. Especially the 200 OK message. >>>>> >>>>> >>>>> Thanks again for help, maybe getting somewhere now...... >>>>> >>>>> Oliver >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jan 6, 2012 at 4:20 PM, Peter Olsson >>>>> wrote: >>>>>> If it sends 200 OK right after 183, this IS the problem. >>>>>> >>>>>> 200 OK means that the call was answered, it should not be sent until the call was actually picked up in the remote end. When 200 OK arrives to FS it will execute your app, and you will start playing the files. >>>>>> >>>>>> Seems to me there is something broken in the Cisco. >>>>>> >>>>>> /Peter >>>>>> >>>>>> ________________________________________ >>>>>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Oliver Schenk [olimonkey at gmail.com] >>>>>> Skickat: den 6 januari 2012 06:55 >>>>>> Till: FreeSWITCH Users Help >>>>>> ?mne: Re: [Freeswitch-users] CISCO 2811 Freeswitch IVR >>>>>> >>>>>> I've tried looking at disable-early-media configuration command, but >>>>>> that didn't work and I doubt that has anything to do with the CISCO >>>>>> sending a 200 OK right after a 183 SESSION PROGRESS. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jan 6, 2012 at 9:20 AM, Brian West wrote: >>>>>>> Thats what the 183 is.. 180 vs 183 are kinda sketchy in some devices.. 180 >>>>>>> is usually RINGING (generate ringback locally) while a 183 has media... aka >>>>>>> early media and usually provides ringback inband. >>>>>>> >>>>>>> /b >>>>>>> >>>>>>> On Jan 5, 2012, at 7:13 PM, Oliver Schenk wrote: >>>>>>> >>>>>>> Shouldn't there be a ?180 RINGING ?somewhere in there? >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 6, 2012 at 8:25 AM, Oliver Schenk wrote: >>>>>>> >>>>>>> I just noticed something else, if I don't pick up the phone at all. >>>>>>> >>>>>>> The IVR just keeps playing until the menu timeout kicks in. >>>>>>> >>>>>>> >>>>>>> So here is a CISCO SIP log: >>>>>>> >>>>>>> http://pastebin.com/Y9sYkuxi >>>>>>> >>>>>>> >>>>>>> The FS server is 192.168.x.50 and the CISCO is 192.168.x.1. >>>>>>> >>>>>>> I hope the CISCO log is readable, it's a bit long because I just did >>>>>>> >>>>>>> "debug ccsip all". >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> In this test I didn't bother picking up the phone at all, but I can >>>>>>> >>>>>>> see that FS answered anyway and the IVR kept playing until it timed >>>>>>> >>>>>>> out. >>>>>>> >>>>>>> I'm not an expert, but here is what I picked out of it: >>>>>>> >>>>>>> >>>>>>> At 00:08:10 we get a >>>>>>> >>>>>>> Received: "INVITE sip:109212xxxx at 192.168.x.1 SIP/2.0" >>>>>>> >>>>>>> >>>>>>> the further down at the same timestamp we get >>>>>>> >>>>>>> Sent: "SIP/2.0 100 Trying" >>>>>>> >>>>>>> >>>>>>> At 00:08:13 we get a >>>>>>> >>>>>>> Sent: "SIP/2.0 183 Session Progress" >>>>>>> >>>>>>> >>>>>>> At 00:18:13 we get a >>>>>>> >>>>>>> Sent: "SIP/2.0 200 OK" >>>>>>> >>>>>>> >>>>>>> Then at the same timestamp we get: >>>>>>> >>>>>>> Received: "ACK sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Once the IVR times out at 00:09:16 we get >>>>>>> >>>>>>> Received: "BYE sip:109212xxxx at 192.168.x.1:5060 SIP/2.0" >>>>>>> >>>>>>> >>>>>>> And then the reply right after >>>>>>> >>>>>>> Sent: "SIP/2.0 200 OK" >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> So I think you were right, the CISCO is sending back an "OK" 3 seconds >>>>>>> >>>>>>> after the "INVITE" is received. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> The part that is beyond my field of expertise so far is WHY? >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Thanks, >>>>>>> >>>>>>> >>>>>>> >>>>>>> Oliver >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 6, 2012 at 8:04 AM, Oliver Schenk wrote: >>>>>>> >>>>>>> By the way: >>>>>>> >>>>>>> >>>>>>> I tried {ignore_early_media=true} as well, but as I think we >>>>>>> >>>>>>> determined, my problem is probably with the CISCO telling FS that the >>>>>>> >>>>>>> call has been answered when really it hasn't yet. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Fri, Jan 6, 2012 at 8:01 AM, Oliver Schenk wrote: >>>>>>> >>>>>>> Thanks for the help so far. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Here is a pastebin of FreeSWITCH output: >>>>>>> >>>>>>> http://pastebin.com/i6Qgc7ws >>>>>>> >>>>>>> >>>>>>> Notice how the "has been answered" log message comes immediately >>>>>>> >>>>>>> (within a few milliseconds) after the call was originated. I think >>>>>>> >>>>>>> this would suggest that the CISCO is immediately sending a 200 OK, as >>>>>>> >>>>>>> you suggested. I also turned on CISCO debugging, but I'm just trying >>>>>>> >>>>>>> to figure out how to get the information regarding SIP messages back >>>>>>> >>>>>>> to Freeswitch. I'll run the test again and see if I can get some >>>>>>> >>>>>>> useful CISCO debug. >>>>>>> >>>>>>> >>>>>>> Which "debug ccsip" commands are relevant to what I want for the CISCO >>>>>>> >>>>>>> SIP debugging? >>>>>>> >>>>>>> >>>>>>> >>>>>>> Thanks! >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> 2012/1/6 Gustavo M?rsico : >>>>>>> >>>>>>> I think I've a similar problem related to callcenter app. When I made an >>>>>>> originate like this: >>>>>>> >>>>>>> >>>>>>> originate loopback/2500/default/XML &bridge(user/2001) >>>>>>> >>>>>>> >>>>>>> 2500 is an extension that leads to a callcenter application. In this case, >>>>>>> we dial first to the queue and when an agent answered we call to the >>>>>>> customer. As far as I know >>>>>>> >>>>>>> When the A-leg reaches to the queue, without selecting an agent, the call is >>>>>>> automatically sent to the B-leg. As far as I see, there is a pre-answer >>>>>>> method that fs needs to send the media to A-leg. >>>>>>> >>>>>>> In order to try to avoid this, I tried using ignore_early_media=true as part >>>>>>> of the originate in A-leg and/or B-leg, with no luck. >>>>>>> >>>>>>> >>>>>>> originate {ignore_early_media=true}loopback/2500/default/XML >>>>>>> &bridge({ignore_early_media=true}user/2001) >>>>>>> >>>>>>> >>>>>>> Dialplan: loopback/2500-b Regex (PASS) [CallCenter_Click2Call] >>>>>>> destination_number(2500) =~ /^(2500)$/ break=on-false >>>>>>> >>>>>>> Dialplan: loopback/2500-b Action set(ignore_early_media=true) >>>>>>> >>>>>>> Dialplan: loopback/2500-b Action callcenter(click2call) >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:154 >>>>>>> (loopback/2500-b) State Change CS_ROUTING -> CS_EXECUTE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:1180 Send signal >>>>>>> loopback/2500-b [BREAK] >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>>> CHANNEL KILL >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:410 >>>>>>> (loopback/2500-b) State ROUTING going to sleep >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:362 >>>>>>> (loopback/2500-b) Running State Change CS_EXECUTE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:417 >>>>>>> (loopback/2500-b) State EXECUTE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:375 loopback/2500-b >>>>>>> CHANNEL EXECUTE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_state_machine.c:192 >>>>>>> loopback/2500-b Standard EXECUTE >>>>>>> >>>>>>> EXECUTE loopback/2500-b set(open=true) >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>>> [open]=[true] >>>>>>> >>>>>>> EXECUTE loopback/2500-b >>>>>>> hash(insert/10.8.0.70-spymap/0000000000/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>>>> >>>>>>> EXECUTE loopback/2500-b hash(insert/10.8.0.70-last_dial/0000000000/2500) >>>>>>> >>>>>>> EXECUTE loopback/2500-b >>>>>>> hash(insert/10.8.0.70-last_dial/global/fef7d864-b3c7-407c-a6e1-94386642bfbb) >>>>>>> >>>>>>> EXECUTE loopback/2500-b set(RFC2822_DATE=Thu, 05 Jan 2012 13:36:08 -0300) >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>>> [RFC2822_DATE]=[Thu, 05 Jan 2012 13:36:08 -0300] >>>>>>> >>>>>>> EXECUTE loopback/2500-b set(ignore_early_media=true) >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_dptools.c:1286 loopback/2500-b SET >>>>>>> [ignore_early_media]=[true] >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:2133 Application >>>>>>> callcenter Requires media! pre_answering channel loopback/2500-b >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:760 Pre-Answer >>>>>>> loopback/2500-a! >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-a) >>>>>>> Callstate Change RINGING -> EARLY >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>>>> loopback/2500-b [BREAK] >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>>> CHANNEL KILL >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [NOTICE] switch_core_session.c:2135 Pre-Answer >>>>>>> loopback/2500-b! >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:2930 (loopback/2500-b) >>>>>>> Callstate Change RINGING -> EARLY >>>>>>> >>>>>>> EXECUTE loopback/2500-b callcenter(click2call) >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-a) >>>>>>> Callstate Change EARLY -> ACTIVE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [NOTICE] mod_loopback.c:755 Channel >>>>>>> [loopback/2500-a] has been answered >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_core_session.c:729 Send signal >>>>>>> loopback/2500-b [BREAK] >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] mod_loopback.c:475 loopback/2500-b >>>>>>> CHANNEL KILL >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_ivr_originate.c:3266 Originate >>>>>>> Resulted in Success: [loopback/2500-a] >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [DEBUG] switch_channel.c:3188 (loopback/2500-b) >>>>>>> Callstate Change EARLY -> ACTIVE >>>>>>> >>>>>>> 2012-01-05 13:36:08.541517 [INFO] switch_channel.c:2708 loopback/2500-a >>>>>>> Flipping CID from "" <0000000000> to "Outbound Call" >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Jan 5, 2012, at 4:17 AM, Oliver Schenk wrote: >>>>>>> >>>>>>> >>>>>>> Also, maybe I should be doing something like this: >>>>>>> >>>>>>> >>>>>>> sofia/gateway/mygatewayname/1091234567 '&managed(ivrAppName)' >>>>>>> >>>>>>> >>>>>>> instead of: >>>>>>> >>>>>>> >>>>>>> sofia/internal/1091234567 at 192.168.x.x '&managed(ivrAppName)' >>>>>>> >>>>>>> >>>>>>> >>>>>>> but, I don't really have the CISCO configured as a gateway, nor do I >>>>>>> >>>>>>> know how really...probably not on the right track there. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, Jan 5, 2012 at 3:06 PM, Oliver Schenk wrote: >>>>>>> >>>>>>> *bump* >>>>>>> >>>>>>> >>>>>>> >>>>>>> So I think maybe the way I'm doing the originate is the problem? In my >>>>>>> >>>>>>> call string I'm creating a connection directly from the CISCO >>>>>>> >>>>>>> (192.168.x.x) to the managed application, which may be why it starts >>>>>>> >>>>>>> playing straight away? >>>>>>> >>>>>>> >>>>>>> Maybe I should be originating a call first and then only once I know >>>>>>> >>>>>>> the other side has picked up will I bridge the call to the IVR managed >>>>>>> >>>>>>> application. >>>>>>> >>>>>>> >>>>>>> Problem is I dunno how to tell whether the other person has picked up >>>>>>> >>>>>>> (or even if the cisco is going to tell me) and I don't know how to do >>>>>>> >>>>>>> things to a call once it has been established. >>>>>>> >>>>>>> >>>>>>> >>>>>>> I'm currently reading the Dialplan wiki page, hoping to get something >>>>>>> >>>>>>> out of it there. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Cheers >>>>>>> >>>>>>> >>>>>>> Oliver >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Tue, Jan 3, 2012 at 11:46 AM, Oliver Schenk wrote: >>>>>>> >>>>>>> I've been battling while creating an IVR using FreeSWITCH mod_managed >>>>>>> >>>>>>> and connecting through a CISCO 2811. Most things now work quite well, >>>>>>> >>>>>>> but I am having a few issues with the way the system answers calls (or >>>>>>> >>>>>>> doesn't answer calls...). >>>>>>> >>>>>>> >>>>>>> I have FreeSWITCH running as a windows service on Windows Server 2008, >>>>>>> >>>>>>> which is connected via LAN to a CISCO 2811 with a 4 port FXO card, >>>>>>> >>>>>>> which is then connected to a POTS phone line. >>>>>>> >>>>>>> >>>>>>> >>>>>>> Take the following scenario: >>>>>>> >>>>>>> >>>>>>> 1. Managed .NET application creates a call string and uses ESL to talk >>>>>>> >>>>>>> to freeswitch and originate a call: >>>>>>> >>>>>>> >>>>>>> string callstring = >>>>>>> >>>>>>> "{bridge_answer_timeout=20,ignore_early_media=true,call_timeout=20}sofia/internal/1091234567 at 192.168.x.x >>>>>>> >>>>>>> '&managed(ivrAppName)'"; >>>>>>> >>>>>>> eslConnection.API("originate", callstring); >>>>>>> >>>>>>> >>>>>>> where 192.168.x.x is the CISCO IP. >>>>>>> >>>>>>> >>>>>>> 2. The CISCO sees that the phone number (1091234567) starts with a 1 >>>>>>> >>>>>>> so it uses FXO port 1 and strips the 1 and uses the remaining phone >>>>>>> >>>>>>> number (091234567) to make the call. >>>>>>> >>>>>>> >>>>>>> 3. My phone rings, I pick up and I can hear my IVR playing. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> These are my current problems: >>>>>>> >>>>>>> >>>>>>> - IVR starts playing before I even pick up the phone. This means that >>>>>>> >>>>>>> if the system calls a mobile phone and the person doesn't pick up, the >>>>>>> >>>>>>> IVR will start playing and eventually the mobile phone will divert to >>>>>>> >>>>>>> voice mail. Obviously I then get a missed call and an sms saying I >>>>>>> >>>>>>> have a new voice mail, which is annoying. Instead I would like it to >>>>>>> >>>>>>> KNOW that no one has picked up, but I don't know how to do this. >>>>>>> >>>>>>> Somehow the CISCO needs to be able to tell FreeSWITCH that the call >>>>>>> >>>>>>> has not yet been answered. For some reason however as soon as the >>>>>>> >>>>>>> CISCO starts calling FreeSWITCH thinks the call is already connected. >>>>>>> >>>>>>> It doesn't know that the CISCO is actually still ringing. Maybe I'm >>>>>>> >>>>>>> doing originate the wrong way or something ... >>>>>>> >>>>>>> >>>>>>> - The phone only rings for about 10 seconds before hanging up. I've >>>>>>> >>>>>>> tried "call_timeout", "bridge_answer_timeout". I've also tried setting >>>>>>> >>>>>>> CISCO "ring number". Nothing works, my phone still only rings for >>>>>>> >>>>>>> about 10 seconds. I don't know if this is a FreeSWITCH issue or a >>>>>>> >>>>>>> CISCO issue. I'm leaning towards CISCO, because FreeSWITCH IVR just >>>>>>> >>>>>>> starts playing even if no one answers the phone. >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> CISCO Config for relevant FXO port: >>>>>>> >>>>>>> >>>>>>> voice service voip >>>>>>> >>>>>>> ?allow-connections h323 to h323 >>>>>>> >>>>>>> ?allow-connections h323 to sip >>>>>>> >>>>>>> ?allow-connections sip to h323 >>>>>>> >>>>>>> ?allow-connections sip to sip >>>>>>> >>>>>>> ?no supplementary-service h450.2 >>>>>>> >>>>>>> ?no supplementary-service h450.3 >>>>>>> >>>>>>> ?supplementary-service h450.12 >>>>>>> >>>>>>> ?no supplementary-service sip moved-temporarily >>>>>>> >>>>>>> ?no supplementary-service sip refer >>>>>>> >>>>>>> ?fax protocol cisco >>>>>>> >>>>>>> ?sip >>>>>>> >>>>>>> ?registrar server expires max 3600 min 3600 >>>>>>> >>>>>>> ?no update-callerid >>>>>>> >>>>>>> ?no call service stop >>>>>>> >>>>>>> >>>>>>> voice-port 0/3/2 >>>>>>> >>>>>>> ?output attenuation -3 >>>>>>> >>>>>>> ?no comfort-noise >>>>>>> >>>>>>> ?cptone AU >>>>>>> >>>>>>> ?impedance complex1 >>>>>>> >>>>>>> ?caller-id enable >>>>>>> >>>>>>> ! >>>>>>> >>>>>>> dial-peer voice 100 pots >>>>>>> >>>>>>> ?preference 1 >>>>>>> >>>>>>> ?destination-pattern 1T >>>>>>> >>>>>>> ?port 0/3/2 >>>>>>> >>>>>>> ! >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Many Thanks, >>>>>>> >>>>>>> >>>>>>> Oliver >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> >>>>>>> consulting at freeswitch.org >>>>>>> >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> http://wiki.freeswitch.org >>>>>>> >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> >>>>>>> consulting at freeswitch.org >>>>>>> >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> http://wiki.freeswitch.org >>>>>>> >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Brian West >>>>>>> FreeSWITCH Solutions, LLC >>>>>>> Phone: +1 (918) 420-9266 >>>>>>> Fax: ? +1 (918) 420-9267 >>>>>>> brian at freeswitch.org >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>>> >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at freeswitch.org >>>>>>> http://www.freeswitchsolutions.com >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> Official FreeSWITCH Sites >>>>>>> http://www.freeswitch.org >>>>>>> http://wiki.freeswitch.org >>>>>>> http://www.cluecon.com >>>>>>> >>>>>>> FreeSWITCH-users mailing list >>>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> !DSPAM:4f06d49b32762089563979! >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org From rhow at exemail.com.au Tue Jan 10 07:02:22 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 12:02:22 +0800 Subject: [Freeswitch-users] Calling multiple phones at once Message-ID: <4F0BB84E.8080504@exemail.com.au> Hi, How do I set it up in freeswitch to call mulitple phones at once (internal and via a gateway) that works with call_timeout. I can't seem to find the appropriate wiki page. Call groups kinda work, but then the phones need to register. My setup with bridge the timeout doesn't work After 30 seconds all the phones start ringing again. And when the external number hangs up it immediately starts ringing again. With the call groups it seems to work correctly. Also, using the user/1000 at domain notation, multiple registrations doesn't work, only a single device rings. Thanks! From rhow at exemail.com.au Tue Jan 10 07:54:17 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 12:54:17 +0800 Subject: [Freeswitch-users] Calling multiple phones at once In-Reply-To: <4F0BB84E.8080504@exemail.com.au> References: <4F0BB84E.8080504@exemail.com.au> Message-ID: <4F0BC479.105@exemail.com.au> Ok, to further this, Using call groups From the cli doing the following: group insert:02 at 192.168.2.7:[sip_cid_type=none]sofia/gateway/sip.pennytel.com/0447123456 group insert:02 at 192.168.2.7:[sip_cid_type=none]sofia/internal/1000 at 192.168.2.7 The external number doesn't get called (and it goes to voicemail!, It looks like it just transfers to extension 1000) However if I use group insert:02 at 192.168.2.7:[sip_cid_type=none]sofia/gateway/sip.pennytel.com/0447123456 group insert:02 at 192.168.2.7:[sip_cid_type=none]user/1000 at 192.168.2.7 The external number gets dialed, BUT only a single registered phone on 1000 gets dialed, not all of them. So, I'm still stuck as how to do this. The only option I can see is to ditch the multiple registrations and use the group like group insert:02 at 192.168.2.7:[sip_cid_type=none]sofia/gateway/sip.pennytel.com/0447123456 group insert:02 at 192.168.2.7:[sip_cid_type=none]user/1001 at 192.168.2.7 group insert:02 at 192.168.2.7:[sip_cid_type=none]user/1002 at 192.168.2.7 group insert:02 at 192.168.2.7:[sip_cid_type=none]user/1003 at 192.168.2.7 But the multiple registrations is a nice feature... I would like to use it if I can so I can just keep adding phones on that extension and it doesn't matter where I am. On 10/01/2012 12:02 PM, Ryan How wrote: > Hi, > > How do I set it up in freeswitch to call mulitple phones at once > (internal and via a gateway) that works with call_timeout. > > I can't seem to find the appropriate wiki page. > > Call groups kinda work, but then the phones need to register. > > My setup with bridge the timeout doesn't work > > > > > data="user/1000@$${domain},sofia/gateway/sip.pennytel.com/0447123456"/> > > > > > After 30 seconds all the phones start ringing again. And when the > external number hangs up it immediately starts ringing again. > > With the call groups it seems to work correctly. > > Also, using the user/1000 at domain notation, multiple registrations > doesn't work, only a single device rings. > > Thanks! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From lakindia89 at gmail.com Tue Jan 10 08:00:47 2012 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Tue, 10 Jan 2012 10:30:47 +0530 Subject: [Freeswitch-users] Overlap dialing problem In-Reply-To: References: Message-ID: Long time ago, I've tried overlap dial with FreeSWITCH. I didn't had any issues with the min_digits. The only problem that I had with overlap was "DialTone", which is not generated to the phone, when it is off hook. 2011/7/13 ?tefan ?udai > Hi, > > does FreeSwitch support for overlap dialing or did somebody try to use it? > > I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured > my freetdm.conf.xml > with this two lines: > > > > > But when I'm dialing number (which has 14 digits) I hear disconnect tone > after dialing first 5 digits. I think that reason is because FreeTDM > module is waiting number with minimum digits equals to "0" (see my > min_digits). > I have also try set min_digits to 20 and then dial same number with 14 > digits but result was same. So, I looked to log file and found that > FreeTDM module got in thist case STATUS CONFIRM message but he was still > waiting for next digits and his time expired. > > Can you help me please? > > Thanks a lot. > Stefan > > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/bdd58268/attachment.html From manjiri05_deshpande at yahoo.co.in Tue Jan 10 08:41:10 2012 From: manjiri05_deshpande at yahoo.co.in (Manjiri Deshpande) Date: Tue, 10 Jan 2012 11:11:10 +0530 (IST) Subject: [Freeswitch-users] ignoring 180 and 183 w/o sdp Message-ID: <1326174070.8668.YahooMailNeo@web95910.mail.in.yahoo.com> Hi, ? To avoid unnecessary ring back,I want to avoid 180 and 183 w/o SDP from a particular m/c?address.\ I tried using ignore_early_media and sip_ignore_183nosdp.But still FS is not ignoring 180 and forwarding 180 to inbound leg. ? What happens is sip_ignore_183nosdp is ignoring 183,but sending 180 instead? of this. Please help me to find out any solution for this. ? Manjiri -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/9f2f9ea7/attachment.html From avi at avimarcus.net Tue Jan 10 09:19:36 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 10 Jan 2012 08:19:36 +0200 Subject: [Freeswitch-users] Ota in sip In-Reply-To: <1284558287-1326164994-cardhu_decombobulator_blackberry.rim.net-392685695-@b27.c2.bise3.blackberry> References: <1284558287-1326164994-cardhu_decombobulator_blackberry.rim.net-392685695-@b27.c2.bise3.blackberry> Message-ID: Are you referring to provisioning, the ability for the client phones to get new configurations from the server? Most hardphones support such an option, and Cudatel, fusionpbx and I think blue.box / 2600hz comes with a large list of phones they support, but it's not part of sip and it's certainly not standardized.. -Avi On Tue, Jan 10, 2012 at 5:09 AM, wrote: > Dear all > Is there any implementation of OTA in sip. > Purpose is to upload new parameter when user connected to sip server. > > Thx > > Budi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rhow at exemail.com.au Tue Jan 10 10:07:58 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 15:07:58 +0800 Subject: [Freeswitch-users] Voicemail skip greeting Message-ID: <4F0BE3CE.1020903@exemail.com.au> Hi, I have a voicemail set up. I want it to not do the "Record your message at the tone, to end recording....". I just want it to go "My message... BEEEEEP". I've put skip_instructions=true in the dial plan, but it seems to have no effect. Anything I am doing wrong? Thanks! Ryan From rhow at exemail.com.au Tue Jan 10 10:22:44 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 15:22:44 +0800 Subject: [Freeswitch-users] Notify of a missed call (no voicemail left) Message-ID: <4F0BE744.4040207@exemail.com.au> Hi, Is there any way to send an email notification of a missed call, no voicemail left?. Or even if a voicemail is left I guess, that isn't a biggy... just so if someone rings and hangs up, or rings and voicemail answers but no message. I'm thinking something that goes in here like: But that wouldn't capture when they hang up before voicemail answering would it ? I guess I could do something like application="socket" and then write an event socket program that sends an email. If there isn't already something out there that does it. Otherwise I'll leave this for another day coz it is a bit beyond my current magic ability, I need to level up :P Thanks again! Ryan From avi at avimarcus.net Tue Jan 10 10:34:35 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 10 Jan 2012 09:34:35 +0200 Subject: [Freeswitch-users] Notify of a missed call (no voicemail left) In-Reply-To: <4F0BE744.4040207@exemail.com.au> References: <4F0BE744.4040207@exemail.com.au> Message-ID: As of Jan 4th, mod_voicemail now sets the channel variable "voicemail_message_len" with the message length of the voicemail that was left. So now all you need is a post-call processor. E.g. you can have your xml_cdr script check - if the call had normal_clearing with last_app of bridge, then someone picked up. (I think... I didn't get around to making this yet.) If it was last_app of voicemail, and voicemail_message_len is set, then an email already went out via the voicemail application. Otherwise it's a missed call and run you script. Or I suppose you could do this as a lua script in the hangup hook with session_in_hangup hook or something. If someone can wikify dialplan code and a lua script we can get it working... I suppose the email would go to the same place as the voicemail goes to. It would be based on a separate user variable about "mail on missed call" though.. -Avi Marcus On Tue, Jan 10, 2012 at 9:22 AM, Ryan How wrote: > Hi, > > Is there any way to send an email notification of a missed call, no > voicemail left?. Or even if a voicemail is left I guess, that isn't a > biggy... just so if someone rings and hangs up, or rings and voicemail > answers but no message. > > I'm thinking something that goes in here like: application="email" data="something here"/> > > data="user/${dialed_extension}@${domain_name}"/> > > > > > > > But that wouldn't capture when they hang up before voicemail answering > would it ? > > I guess I could do something like application="socket" and then write an > event socket program that sends an email. If there isn't already > something out there that does it. Otherwise I'll leave this for another > day coz it is a bit beyond my current magic ability, I need to level up :P > > Thanks again! > > Ryan > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tha_tux at hotmail.com Tue Jan 10 11:30:41 2012 From: tha_tux at hotmail.com (Tux Tux) Date: Tue, 10 Jan 2012 09:30:41 +0100 Subject: [Freeswitch-users] European CNAM database Message-ID: Hi, Can someone tell me a good European CNAM database with a API to combine with mod_cidlookup. At http://www.voip-info.org/wiki/view/CNAM reffered from http://wiki.freeswitch.org/wiki/Mod_cidlookup I can't find one. Googeling "European CNAM Database" isn't finding any good to. Both commercial and free databases are okay. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/f849e6cd/attachment.html From davidwaf at gmail.com Tue Jan 10 11:45:44 2012 From: davidwaf at gmail.com (David Wafula) Date: Tue, 10 Jan 2012 10:45:44 +0200 Subject: [Freeswitch-users] Any got a windows java esl.jar and lib built ? In-Reply-To: <4F0B95B1.90208@exemail.com.au> References: <4F0B95B1.90208@exemail.com.au> Message-ID: I use the client version instead (http://wiki.freeswitch.org/wiki/Java_ESL_Client), but probably that is not what you want for your needs. But works for me nicely. Admittedly am on Linux, but its Java, so target platform is not an issue. On Tue, Jan 10, 2012 at 3:34 AM, Ryan How wrote: > Hi, > > Has anyone got a windows esl.jar and corrsponding lib built?. I'm using > the 2011 12 28 build of the windows binary. The instructions for > building the ESL from source on the wiki are all for linux systems. > > There is also the pure java esl_client.jar, but it hasn't had updates > since aug 2010 and the documentation looks pretty sparse!. Or doesn't it > really matter, is it pretty stable and just like a drop in for the core > version? I've generally learnt from spending hours and hours and hours > messing with things, unless it looks supported or active it is going to > take a lot of time to get it to work, and even then it probably won't > work properly... > > > Thanks! > > Ryan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Wafula From ccesario at tecnomega.com.br Tue Jan 10 13:12:58 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Tue, 10 Jan 2012 08:12:58 -0200 Subject: [Freeswitch-users] Xfer Call Back In-Reply-To: <4F0B6754.1090409@tecnomega.com.br> References: <4F0B6754.1090409@tecnomega.com.br> Message-ID: <4F0C0F2A.9050304@tecnomega.com.br> As this .... http://pastebin.com/yHiAi62M Any idea?! Its is possible?! Greats Carlos Em 09-01-2012 20:16, Carlos Cesario escreveu: > Hello, > > In Att Xfer exists any method to call return to original call if called > dont attend the call ? > > Eg: A transfer to B and hold phone, if B dont attend call the call back > to caller . > > Grets > > Carlos > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From roland at haenel.me Tue Jan 10 13:22:13 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Tue, 10 Jan 2012 11:22:13 +0100 Subject: [Freeswitch-users] "expire" for outgoing registrations not handled correctly? In-Reply-To: References: Message-ID: Hello Anthony, Thanks. I have tested the registration-thread-frequency, but it seems this does not address the issue I'm having here. But the good news is: I just found the bug with the help of Alexandr Dubovikov... In sofia_reg.c, sofia_reg_check_gateway(...) it reads: if (gateway_ptr->freq >= 60) { gateway_ptr->expires = now + (gateway_ptr->freq - 15); } else { if (gateway_ptr->freq < 30 && gateway_ptr->freq >= 5) { gateway_ptr->expires = now + (gateway_ptr->freq - 5); } else { gateway_ptr->expires = now + (gateway_ptr->freq); } } this results in a re-REGISTER 15 seconds ahead of expiration if the register timer is more than 60 seconds. We are a 5 seconds ahead of expiration if the register timer is between 5 and 29 seconds. And the winner is... if the register timer is between 30 and 59 seconds, we have no "ahead". Obviously a bug, since the code is difficult to read with all those else conditions. I created a patch, verified that is works and attached it to this mail. Let me know if I need to do more (open an issue etc.) to make sure this patch gets applied... Thanks again, Roland 2012/1/10 Anthony Minessale > It might be because we only check for expiration every 30 seconds. > > update to latest GIT and I added a profile > param: registration-thread-frequency > > The default is 30 so you can try setting it to 10 or 5 or something. > > > > > > 2012/1/9 Roland H?nel > >> Hi, >> >> It seems that I'm running into a timing issue with outgoing (gateway) >> registrations, which results in some milliseconds in that I can't send >> calls over that gateway. >> >> I have configured a gateway like this: >> >> >> >> >> >> >> >> ^M >> >> >> >> ^M >> >> If I look at the "200 OK" messages for the SIP REGISTER packet, the >> server sends me an "Expires: 30", so FreeSwitch should re-REGISTER every 30 >> seconds. >> >> However, that's what happening: >> >> 2012-01-09 16:37:00.804865 [NOTICE] sofia_reg.c:407 Registering duro >> 2012-01-09 16:37:31.804880 [NOTICE] sofia_reg.c:407 Registering duro >> 2012-01-09 16:38:03.804894 [NOTICE] sofia_reg.c:407 Registering duro >> 2012-01-09 16:38:35.824877 [NOTICE] sofia_reg.c:407 Registering duro >> *2012-01-09 16:39:06.825090 [NOTICE] sofia_reg.c:407 Registering duro* >> 2012-01-09 16:39:39.824881 [NOTICE] sofia_reg.c:407 Registering duro >> >> >> You can see that it's *approximately* 30 seconds betwee registrations, >> but in fact every time it's just a little more than 30 secs. This alone >> makes me feel a little bit strange, since my registration might have been >> expired during this time window (and consequently, maybe I'm loosing >> incoming calls because of that). >> >> Even for outgoing calls, this seems to be an issue, let's take a look at >> the 16:39:06.825090 re-REGISTER. >> >> We can see that at 16:39:05 (no micro-seconds here) the gateway becomes >> "UNREGED": >> >> Content-Length: 542 >> Content-Type: text/event-plain >> >> Event-Subclass: sofia::gateway_state >> Event-Name: CUSTOM >> Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 >> FreeSWITCH-Hostname: lap597 >> FreeSWITCH-Switchname: lap597 >> FreeSWITCH-IPv4: 192.168.232.164 >> FreeSWITCH-IPv6: ::1 >> *Event-Date-Local: 2012-01-09 16:39:05* >> Event-Date-GMT: Mon, 09 Jan 2012 15:39:05 GMT >> Event-Date-Timestamp: 1326123545824882 >> Event-Calling-File: sofia_reg.c >> Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event >> Event-Calling-Line-Number: 151 >> Gateway: duro >> State: UNREGED >> Ping-Status: UP >> >> >> But the new REGISTER is done a full second later (this is not just a >> display issue, I saw those messages scrolling by and there was just a >> second pause between them): >> >> Event-Subclass: sofia::gateway_state >> Event-Name: CUSTOM >> Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 >> FreeSWITCH-Hostname: lap597 >> FreeSWITCH-Switchname: lap597 >> FreeSWITCH-IPv4: 192.168.232.164 >> FreeSWITCH-IPv6: ::1 >> Event-Date-Local: 2012-01-09 16:39:06 >> *Event-Date-GMT: Mon, 09 Jan 2012 15:39:06 GMT* >> Event-Date-Timestamp: 1326123546825090 >> Event-Calling-File: sofia_reg.c >> Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event >> Event-Calling-Line-Number: 151 >> Gateway: duro >> State: TRYING >> Ping-Status: DOWN >> >> >> Then, the gateway gets re-registered in a couple of milliseconds and >> everything is fine again. >> >> *It seems that this behaviour effectively creates a timespan of 1-2 >> seconds, where the gateway is not available for outgoing (and maybe also >> incoming) calls.* >> >> Does anyone of you know this issue? I wonder because this looks just to >> easy to be wrong in this fashion? >> >> Greetings, >> Roland >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/ed47843f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: next-expire-mod.patch Type: application/octet-stream Size: 988 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/ed47843f/attachment-0001.obj From ayobami at programmer.net Tue Jan 10 14:36:58 2012 From: ayobami at programmer.net (ayobami) Date: Tue, 10 Jan 2012 03:36:58 -0800 (PST) Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call Message-ID: <1326195418755-7171721.post@n2.nabble.com> I need to get the physical IP addresses of two parties involved in a call, I have tried to monitor events being generated during the call life cycle, from both parties involved in the call, its only returning the physical IP address of the Freeswitch box, is their any settings that I need to set so that appropriate IP addresses will be returned, thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addresses-of-The-Parties-Involved-in-a-call-tp7171721p7171721.html Sent from the freeswitch-users mailing list archive at Nabble.com. From engineerzuhairraza at gmail.com Tue Jan 10 14:47:50 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 10 Jan 2012 15:47:50 +0400 Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call In-Reply-To: <1326195418755-7171721.post@n2.nabble.com> References: <1326195418755-7171721.post@n2.nabble.com> Message-ID: See http://wiki.freeswitch.org/wiki/Channel_Variables#SIP_related_variables Regards, Zohair Raza On Tue, Jan 10, 2012 at 3:36 PM, ayobami wrote: > I need to get the physical IP addresses of two parties involved in a call, > I > have tried to monitor events being > generated during the call life cycle, from both parties involved in the > call, its only returning the physical IP address of the Freeswitch box, is > their any settings that I need to set so that appropriate IP addresses will > be returned, thanks > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addresses-of-The-Parties-Involved-in-a-call-tp7171721p7171721.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/91c110c8/attachment.html From arnuld at phonologies.com Tue Jan 10 15:00:34 2012 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Tue, 10 Jan 2012 17:30:34 +0530 Subject: [Freeswitch-users] running freeswitch binary on more than one computer Message-ID: I have several computers and all of them have exactly same OS, CentOS 5.5 . I built freeswitch on one computer with some --prefix=/home/arnuld/foo . After "make install" was done, I copied the installed directory to other machine. When I try to run it says "cwitch_xml.c couldn't find /home/arnuld/foo" Is it not feasbile to move the binary around ? Why freeswitch hardcodes the --prefix into binary ? -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From rhow at exemail.com.au Tue Jan 10 15:34:37 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 10 Jan 2012 20:34:37 +0800 Subject: [Freeswitch-users] Any got a windows java esl.jar and lib built ? In-Reply-To: References: <4F0B95B1.90208@exemail.com.au> Message-ID: <4F0C305D.2050900@exemail.com.au> Hi, Do you find it to work reliably?. I'll give it a go. It just looked like it hadn't been updated in quite a while, and the roadmap has a lot of todos still on it... Thanks, Ryan On 10/01/2012 4:45 PM, David Wafula wrote: > I use the client version instead > (http://wiki.freeswitch.org/wiki/Java_ESL_Client), but probably that > is not what you want for your needs. But works for me nicely. > Admittedly am on Linux, but its Java, so target platform is not an > issue. > > > On Tue, Jan 10, 2012 at 3:34 AM, Ryan How wrote: >> Hi, >> >> Has anyone got a windows esl.jar and corrsponding lib built?. I'm using >> the 2011 12 28 build of the windows binary. The instructions for >> building the ESL from source on the wiki are all for linux systems. >> >> There is also the pure java esl_client.jar, but it hasn't had updates >> since aug 2010 and the documentation looks pretty sparse!. Or doesn't it >> really matter, is it pretty stable and just like a drop in for the core >> version? I've generally learnt from spending hours and hours and hours >> messing with things, unless it looks supported or active it is going to >> take a lot of time to get it to work, and even then it probably won't >> work properly... >> >> >> Thanks! >> >> Ryan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From miha at softnet.si Tue Jan 10 15:37:29 2012 From: miha at softnet.si (Miha Zoubek) Date: Tue, 10 Jan 2012 13:37:29 +0100 Subject: [Freeswitch-users] PBX Message-ID: <4F0C3109.3090201@softnet.si> Hi, is it possible to have on freeswitch more than one PBX? If it is, where can I find this threat on wiki. Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From gmaruzz at gmail.com Tue Jan 10 15:37:39 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 10 Jan 2012 13:37:39 +0100 Subject: [Freeswitch-users] running freeswitch binary on more than one computer In-Reply-To: References: Message-ID: because it needs to finds its directories, conf, etc etc use a prefix that you use in all machines -giovanni On Tue, Jan 10, 2012 at 1:00 PM, Arnuld Uttre (Phonologies) wrote: > I have several computers and all of them have exactly same OS, CentOS 5.5 > . I built freeswitch on one computer with some --prefix=/home/arnuld/foo > . > > After "make install" was done, I copied the installed directory to other > machine. When I try to run it says "cwitch_xml.c couldn't find > /home/arnuld/foo" > > Is it not feasbile to move the binary around ? ?Why freeswitch hardcodes > the --prefix into binary ? > > > > -- > Arnuld Uttre > Systems Software Engineer > > arnuld at Phonologies.COM > http://www.phonologies.com > > Phonologies (India) Private Limited > West Wing, Marri Deep, M. C. H. No. 12-5-4, > Lallaguda, Secunderabad 500017, INDIA. > Ph:+91-40-2701 8993 / 36 > Fax:+91-40-2701 8992 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From juanito1982 at gmail.com Tue Jan 10 15:43:16 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 10 Jan 2012 13:43:16 +0100 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> <4F085F62.1000501@cupis.co.uk> <4F0886DA.9040100@cupis.co.uk> Message-ID: What about if not possible to ignore early media? 2012/1/8 Kristian Kielhofner > Why are you executing hangup ${originate_disposition} after your bridge? > > You'll see that FreeSWITCH is passing originate_disposition=success, > probably because the remote end is returning early media (which > FreeSWITCH considers a successful bridge). > > Do one (or both) of these: > 1) Add ignore_early_media=true to your bridge line. > > 2) Remove the hangup ${originate_disposition} after your bridge and > let FreeSWITCH run of things to execute on its own. > > Depending on your specific goals there are better options but these > actions should isolate your problem. > > 2012/1/7 Juan Antonio Iba?ez Santorum : > > Of course: > > > > Caller: 192.168.2.2 > > FS: 192.168.2.238 > > Gateway (Asterisk): 192.168.2.239 > > > > http://pastebin.freeswitch.org/18099 > > > > Regards > > > > > > 2012/1/7 Paul Cupis > >> > >> On 07/01/12 16:32, Juan Antonio Iba?ez Santorum wrote: > >> > No, same result, 480 instead 486 > >> > >> Can you show us the full log of the call, please? > >> > >> http://pastebin.freeswitch.org/ > >> > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/0e91833f/attachment-0001.html From rajkumar.kmry at gmail.com Tue Jan 10 08:01:23 2012 From: rajkumar.kmry at gmail.com (rajkumar k) Date: Mon, 9 Jan 2012 21:01:23 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH hangs at start up Message-ID: <33108439.post@talk.nabble.com> I installed freeswitch and tried to start, but it hangs while running check_ip task. Does anyone know about this? # freeswitch 2012-01-10 10:26:31.984000 [INFO] switch_event.c:631 Activate Eventing Engine. 2012-01-10 10:26:32.060000 [DEBUG] switch_event.c:610 Create event dispatch thread 0 2012-01-10 10:26:47.768000 [INFO] switch_nat.c:419 Scanning for NAT 2012-01-10 10:26:47.772000 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2012-01-10 10:26:47.780000 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2012-01-10 10:26:47.780000 [DEBUG] switch_nat.c:424 Checking for UPnP 2012-01-10 10:26:59.796000 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2012-01-10 10:26:59.912000 [INFO] switch_core_sqldb.c:1898 Opening DB 2012-01-10 10:27:01.848000 [NOTICE] switch_scheduler.c:166 Starting task thread 2012-01-10 10:27:01.884000 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1326171421 2012-01-10 10:27:01.884000 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1326171421 -- View this message in context: http://old.nabble.com/FreeSWITCH-hangs-at-start-up-tp33108439p33108439.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jehudi.maes at gmail.com Tue Jan 10 11:53:01 2012 From: jehudi.maes at gmail.com (Jehudi Maes) Date: Tue, 10 Jan 2012 09:53:01 +0100 Subject: [Freeswitch-users] Transfer out of lua script on fax detection - novice user Message-ID: I am having trouble with fax detection while executing a lua script: I am setting the fax detection in the dialplan: and continue by executing a script: when a fax is detected (this works fine) during the execution of the script, the script exits and the call is hangup. How can I avoid the hangup and continue with the fax receiving ? Kind regards, Jehudi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/ec2c7afb/attachment.html From anita.hall at simmortel.com Tue Jan 10 15:54:24 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 10 Jan 2012 18:24:24 +0530 Subject: [Freeswitch-users] PBX In-Reply-To: <4F0C3109.3090201@softnet.si> References: <4F0C3109.3090201@softnet.si> Message-ID: Miha, perhaps you are looking for these http://wiki.freeswitch.org/wiki/Multi-tenant http://wiki.freeswitch.org/wiki/Multiple_Companies Basically, you have to start by making different domain-a.conf.xml and domain-b.conf.xml in directory/ regards, Anita On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek wrote: > Hi, > > is it possible to have on freeswitch more than one PBX? If it is, where > can I find this threat on wiki. > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/f8c774bf/attachment.html From davidwaf at gmail.com Tue Jan 10 15:54:48 2012 From: davidwaf at gmail.com (David Wafula) Date: Tue, 10 Jan 2012 14:54:48 +0200 Subject: [Freeswitch-users] Any got a windows java esl.jar and lib built ? In-Reply-To: <4F0C305D.2050900@exemail.com.au> References: <4F0B95B1.90208@exemail.com.au> <4F0C305D.2050900@exemail.com.au> Message-ID: I only use it for listening/controlling conference events, and i cant complain. On git that is about month old. On Tue, Jan 10, 2012 at 2:34 PM, Ryan How wrote: > Hi, > > Do you find it to work reliably?. I'll give it a go. It just looked like > it hadn't been updated in quite a while, and the roadmap has a lot of > todos still on it... > > Thanks, > > Ryan > > > On 10/01/2012 4:45 PM, David Wafula wrote: >> I ?use the client version instead >> (http://wiki.freeswitch.org/wiki/Java_ESL_Client), but probably that >> is not what you want for your needs. But works for me nicely. >> Admittedly am on Linux, but its Java, so target platform is not an >> issue. >> >> >> On Tue, Jan 10, 2012 at 3:34 AM, Ryan How ?wrote: >>> Hi, >>> >>> Has anyone got a windows esl.jar and corrsponding lib built?. I'm using >>> the 2011 12 28 build of the windows binary. The instructions for >>> building the ESL from source on the wiki are all for linux systems. >>> >>> There is also the pure java esl_client.jar, but it hasn't had updates >>> since aug 2010 and the documentation looks pretty sparse!. Or doesn't it >>> really matter, is it pretty stable and just like a drop in for the core >>> version? I've generally learnt from spending hours and hours and hours >>> messing with things, unless it looks supported or active it is going to >>> take a lot of time to get it to work, and even then it probably won't >>> work properly... >>> >>> >>> Thanks! >>> >>> Ryan >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Wafula From anita.hall at simmortel.com Tue Jan 10 16:01:10 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 10 Jan 2012 18:31:10 +0530 Subject: [Freeswitch-users] in ivr menu In-Reply-To: References: Message-ID: Added to Wiki. regards, Anita On Sat, Jan 7, 2012 at 11:47 AM, Michael Collins wrote: > Use an absolute path value for the greeting sounds. When you use a > relative path (i.e. you don't start with a forward slash) then it will > assume that you want to use a sound file somewhere in the sounds directory > structure. > > Relative: > greet-short="ivr/ivr-menu.wav" > > Absolute: > greet-short="/full/path/to/file/ivr-menu.wav" > > -MC > > P.S. - this didn't seem to be mentioned on the XML IVR menu page ( > http://wiki.freeswitch.org/wiki/IVR_Menu) so if you find this information > useful would you mind adding this tidbit there? > > > On Fri, Jan 6, 2012 at 2:28 AM, amit nakum wrote: > >> Hi, >> >> I want to play wav files from the different path. As now it is playing the >> wav files from: >> "/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/" >> path. >> >> Please tell me how to play wav files from other path in ivr menu using >> greet-short parameter . >> >> Thanks in advance.... >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/1c27343f/attachment.html From miha at softnet.si Tue Jan 10 16:15:20 2012 From: miha at softnet.si (Miha Zoubek) Date: Tue, 10 Jan 2012 14:15:20 +0100 Subject: [Freeswitch-users] PBX In-Reply-To: References: <4F0C3109.3090201@softnet.si> Message-ID: <4F0C39E8.1090604@softnet.si> On 1/10/2012 1:54 PM, Anita Hall wrote: > Miha, perhaps you are looking for these > > http://wiki.freeswitch.org/wiki/Multi-tenant > http://wiki.freeswitch.org/wiki/Multiple_Companies > > Basically, you have to start by making different domain-a.conf.xml and > domain-b.conf.xml in directory/ > > regards, > Anita > > > > On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek > wrote: > > Hi, > > is it possible to have on freeswitch more than one PBX? If it is, > where > can I find this threat on wiki. > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Thanks Anita! BR, miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/26f2e4c9/attachment-0001.html From roland at haenel.me Tue Jan 10 16:32:56 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Tue, 10 Jan 2012 14:32:56 +0100 Subject: [Freeswitch-users] "expire" for outgoing registrations not handled correctly? In-Reply-To: References: Message-ID: Sorry, it's me again :-| There is an additional issue to be fixed. Now that FreeSwitch re-REGISTERs soon enough, I'm still facing the problem that when the gateway state drops to REG_STATE_UNREGED, the status of the gateway is set to SOFIA_GATEWAY_DOWN. This is incorrect because it creates a short downtime for the gateway even if the re-REGISTER succeeds. The patch attached now corrects the status, that is, when the gateway re-REGISTERED, the status is not changed. Finally the registration will succeed (then, the status is forced UP) or fail (then, the status is forced DOWN). A gateway with continuous re-registrations now keeps the SOFIA_GATEWAY_UP status all the time, until a REGISTER fails. Greetings, Roland 2012/1/10 Roland H?nel > Hello Anthony, > > Thanks. I have tested the registration-thread-frequency, but it seems this > does not address the issue I'm having here. > > But the good news is: I just found the bug with the help of Alexandr > Dubovikov... > > In sofia_reg.c, sofia_reg_check_gateway(...) it reads: > > if (gateway_ptr->freq >= 60) { > gateway_ptr->expires = now + (gateway_ptr->freq - > 15); > } else { > if (gateway_ptr->freq < 30 && gateway_ptr->freq >= 5) > { > gateway_ptr->expires = now + > (gateway_ptr->freq - 5); > } else { > gateway_ptr->expires = now + > (gateway_ptr->freq); > } > } > > this results in a re-REGISTER 15 seconds ahead of expiration if the > register timer is more than 60 seconds. We are a 5 seconds ahead of > expiration if the register timer is between 5 and 29 seconds. And the > winner is... if the register timer is between 30 and 59 seconds, we have no > "ahead". Obviously a bug, since the code is difficult to read with all > those else conditions. > > I created a patch, verified that is works and attached it to this mail. > Let me know if I need to do more (open an issue etc.) to make sure this > patch gets applied... > > Thanks again, > Roland > > > 2012/1/10 Anthony Minessale > >> It might be because we only check for expiration every 30 seconds. >> >> update to latest GIT and I added a profile >> param: registration-thread-frequency >> >> The default is 30 so you can try setting it to 10 or 5 or something. >> >> >> >> >> >> 2012/1/9 Roland H?nel >> >>> Hi, >>> >>> It seems that I'm running into a timing issue with outgoing (gateway) >>> registrations, which results in some milliseconds in that I can't send >>> calls over that gateway. >>> >>> I have configured a gateway like this: >>> >>> >>> >>> >>> >>> >>> >>> ^M >>> >>> >>> >>> ^M >>> >>> If I look at the "200 OK" messages for the SIP REGISTER packet, the >>> server sends me an "Expires: 30", so FreeSwitch should re-REGISTER every 30 >>> seconds. >>> >>> However, that's what happening: >>> >>> 2012-01-09 16:37:00.804865 [NOTICE] sofia_reg.c:407 Registering duro >>> 2012-01-09 16:37:31.804880 [NOTICE] sofia_reg.c:407 Registering duro >>> 2012-01-09 16:38:03.804894 [NOTICE] sofia_reg.c:407 Registering duro >>> 2012-01-09 16:38:35.824877 [NOTICE] sofia_reg.c:407 Registering duro >>> *2012-01-09 16:39:06.825090 [NOTICE] sofia_reg.c:407 Registering duro* >>> 2012-01-09 16:39:39.824881 [NOTICE] sofia_reg.c:407 Registering duro >>> >>> >>> You can see that it's *approximately* 30 seconds betwee registrations, >>> but in fact every time it's just a little more than 30 secs. This alone >>> makes me feel a little bit strange, since my registration might have been >>> expired during this time window (and consequently, maybe I'm loosing >>> incoming calls because of that). >>> >>> Even for outgoing calls, this seems to be an issue, let's take a look at >>> the 16:39:06.825090 re-REGISTER. >>> >>> We can see that at 16:39:05 (no micro-seconds here) the gateway becomes >>> "UNREGED": >>> >>> Content-Length: 542 >>> Content-Type: text/event-plain >>> >>> Event-Subclass: sofia::gateway_state >>> Event-Name: CUSTOM >>> Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 >>> FreeSWITCH-Hostname: lap597 >>> FreeSWITCH-Switchname: lap597 >>> FreeSWITCH-IPv4: 192.168.232.164 >>> FreeSWITCH-IPv6: ::1 >>> *Event-Date-Local: 2012-01-09 16:39:05* >>> Event-Date-GMT: Mon, 09 Jan 2012 15:39:05 GMT >>> Event-Date-Timestamp: 1326123545824882 >>> Event-Calling-File: sofia_reg.c >>> Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event >>> Event-Calling-Line-Number: 151 >>> Gateway: duro >>> State: UNREGED >>> Ping-Status: UP >>> >>> >>> But the new REGISTER is done a full second later (this is not just a >>> display issue, I saw those messages scrolling by and there was just a >>> second pause between them): >>> >>> Event-Subclass: sofia::gateway_state >>> Event-Name: CUSTOM >>> Core-UUID: 243269fe-d280-4f83-aca1-abb88f2ad258 >>> FreeSWITCH-Hostname: lap597 >>> FreeSWITCH-Switchname: lap597 >>> FreeSWITCH-IPv4: 192.168.232.164 >>> FreeSWITCH-IPv6: ::1 >>> Event-Date-Local: 2012-01-09 16:39:06 >>> *Event-Date-GMT: Mon, 09 Jan 2012 15:39:06 GMT* >>> Event-Date-Timestamp: 1326123546825090 >>> Event-Calling-File: sofia_reg.c >>> Event-Calling-Function: sofia_reg_fire_custom_gateway_state_event >>> Event-Calling-Line-Number: 151 >>> Gateway: duro >>> State: TRYING >>> Ping-Status: DOWN >>> >>> >>> Then, the gateway gets re-registered in a couple of milliseconds and >>> everything is fine again. >>> >>> *It seems that this behaviour effectively creates a timespan of 1-2 >>> seconds, where the gateway is not available for outgoing (and maybe also >>> incoming) calls.* >>> >>> Does anyone of you know this issue? I wonder because this looks just to >>> easy to be wrong in this fashion? >>> >>> Greetings, >>> Roland >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Gru?, > Roland > > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/1762718d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: next-expire-mod2.patch Type: application/octet-stream Size: 2244 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/1762718d/attachment-0001.obj From curriegrad2004 at gmail.com Tue Jan 10 16:46:20 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 10 Jan 2012 05:46:20 -0800 Subject: [Freeswitch-users] PBX In-Reply-To: References: <4F0C3109.3090201@softnet.si> Message-ID: A quick and dirty way of doing it is to create a seperate context. On 2012-01-10 4:55 AM, "Anita Hall" wrote: > Miha, perhaps you are looking for these > > http://wiki.freeswitch.org/wiki/Multi-tenant > http://wiki.freeswitch.org/wiki/Multiple_Companies > > Basically, you have to start by making different domain-a.conf.xml and > domain-b.conf.xml in directory/ > > regards, > Anita > > > > On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek wrote: > >> Hi, >> >> is it possible to have on freeswitch more than one PBX? If it is, where >> can I find this threat on wiki. >> >> Regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/dfe910fc/attachment.html From acosgrov at gmail.com Tue Jan 10 16:50:01 2012 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Tue, 10 Jan 2012 08:50:01 -0500 Subject: [Freeswitch-users] Transfer out of lua script on fax detection - novice user In-Reply-To: References: Message-ID: Jehudi, I just went through this exercise. You will need to invoke set_zombie_exec from either the xml or while in your Lua script. Take a look at the wiki page - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_zombie_exec Anthony C. On Jan 10, 2012, at 3:53 AM, Jehudi Maes wrote: > I am having trouble with fax detection while executing a lua script: I am setting the fax detection in the dialplan: > > > > and continue by executing a script: > > > > when a fax is detected (this works fine) during the execution of the script, the script exits and the call is hangup. How can I avoid the hangup and continue with the fax receiving ? > > Kind regards, > > Jehudi > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/96be955a/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Jan 10 16:51:24 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 10 Jan 2012 13:51:24 +0000 (GMT) Subject: [Freeswitch-users] NAT Confusion Message-ID: <1326203484.94913.YahooMailNeo@web29406.mail.ird.yahoo.com> Because by all accounts, judging by the Wiki Entry, usign a public server is not really a wise thing to do in production? .... the " http://dumbme.mbit.com.au/trixbox/stun_servers.htm" link on teh Wiki is broken, and the Wiki also says "Note: stun.fwd.org is gone; also stun.freeswitch.org is never guaranteed to be up and running so use it in production at your own risk."....? so I can only assume that most public servers are similarly provided on a "best endeavours" basis and prone to disappear at any time. Further, I now have an IPsec VPN between me and the firewall, hence eliminating NAT concerns, and I am still seeing the same issues as originally described with things partially working (even with Any/Any rules applied to firewalls). >Why not pick up a public stun server and config it in your softphone, most >likely it will address your issue. From acosgrov at gmail.com Tue Jan 10 16:54:28 2012 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Tue, 10 Jan 2012 08:54:28 -0500 Subject: [Freeswitch-users] Transfer out of lua script on fax detection - novice user In-Reply-To: References: Message-ID: <4D49164B-8C3B-4023-83DE-FCC129C81B3A@gmail.com> Sorry I took a second look at your description and I answered a little prematurely... sounds like your script is exiting too early. Can you pastebin the script? http://pastebin.freeswitch.org Anthony C On Jan 10, 2012, at 8:50 AM, Anthony Cosgrove wrote: > Jehudi, > > I just went through this exercise. You will need to invoke set_zombie_exec from either the xml or while in your Lua script. Take a look at the wiki page - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_zombie_exec > > > Anthony C. > > On Jan 10, 2012, at 3:53 AM, Jehudi Maes wrote: > >> I am having trouble with fax detection while executing a lua script: I am setting the fax detection in the dialplan: >> >> >> >> and continue by executing a script: >> >> >> >> when a fax is detected (this works fine) during the execution of the script, the script exits and the call is hangup. How can I avoid the hangup and continue with the fax receiving ? >> >> Kind regards, >> >> Jehudi >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/422c17a3/attachment.html From nvitaly at gmail.com Tue Jan 10 17:13:59 2012 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Tue, 10 Jan 2012 09:13:59 -0500 Subject: [Freeswitch-users] NAT Confusion In-Reply-To: <1326203484.94913.YahooMailNeo@web29406.mail.ird.yahoo.com> References: <1326203484.94913.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: Agree on STUN, most NAT problems can be resolved without it, freeswitch can handle phones behind NAT nicely. Make traces (tcpdump) at server and phone network, check all RTP streams find where it get blocked (misdirected). most likely it is network related problem. On Tue, Jan 10, 2012 at 8:51 AM, Bob Smith wrote: > Because by all accounts, judging by the Wiki Entry, usign a public server > is not really a wise thing to do in production .... the " > http://dumbme.mbit.com.au/trixbox/stun_servers.htm" link on teh Wiki is > broken, and the Wiki also says "Note: stun.fwd.org is gone; also > stun.freeswitch.org is never guaranteed to be up and running so use it in > production at your own risk.".... so I can only assume that most public > servers are similarly provided on a "best endeavours" basis and prone to > disappear at any time. > > Further, I now have an IPsec VPN between me and the firewall, hence > eliminating NAT concerns, and I am still seeing the same issues as > originally described with things partially working (even with Any/Any rules > applied to firewalls). > > > >Why not pick up a public stun server and config it in your softphone, most > >likely it will address your issue. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -- Vitaly Nikolaev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/4ea9118a/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Tue Jan 10 17:26:49 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 10 Jan 2012 15:26:49 +0100 Subject: [Freeswitch-users] European CNAM database In-Reply-To: References: Message-ID: <4F0C4AA9.80001@puzzled.xs4all.nl> On 10-01-12 09:30, Tux Tux wrote: > Hi, > > Can someone tell me a good European CNAM database with a API to combine > with mod_cidlookup. > At http://www.voip-info.org/wiki/view/CNAM reffered from > http://wiki.freeswitch.org/wiki/Mod_cidlookup I can't find one. > > Googeling "European CNAM Database" isn't finding any good to. > > Both commercial and free databases are okay. Afaik CNAM does not exist in Europe. The second result of the Google search you did also mentions that. Regards, Patrick From freeswitch at earthspike.net Tue Jan 10 17:46:07 2012 From: freeswitch at earthspike.net (John) Date: Tue, 10 Jan 2012 14:46:07 +0000 Subject: [Freeswitch-users] audo sync issues with record_session to mp3 In-Reply-To: <6AEC73649FA6431CA9EF84A54F378594@ws4> References: <6AEC73649FA6431CA9EF84A54F378594@ws4> Message-ID: <4F0C4F2F.3050901@earthspike.net> I've just enabled session recording using MP3 encoding and have the same symptoms. In my case it is between a BRI ISDN connection and a SIP phone, both running with G.711. I haven't tried recording between extensions which would be a purer test, nor have I attempted using WAV. Are there any suggestions for how to fix this? Should I register a bug on Jira? John On 02/01/12 15:01, Frank @ Impact wrote: > We have the same problem. We are running git from 12/30/11. our aleg is a > sip channel coming to FS and the bleg is a sip channels leaving FS. > > I noticed this problem really when we started using mp3 instead of > wav. With wav, it really was not noticeable for us in a 10-15minute call. > But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is so > far out of sync, it sounds like 2 different calls. > > The relevant dialplan is > > > >
> > > > > > > > > data="/mnt/rd/recordfile.mp3"/> > data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> > > > > >
> > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gabriel > Gunderson > Sent: Thursday, October 20, 2011 3:07 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] audo sync issues with record_session to mp3 > > On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott wrote: >> Longer calls, after about 10 minutes start to introduce sync issues >> between the A-leg and the B-leg. >> >> We are running record_session on the A-leg, and it seems to get ahead of >> the B-leg. >> >> For example the caller on the A-leg will be heard to answer a question >> whilst the person on the B-leg is asking it. > What's on the other end of each leg? That might help us figure this out. > > Gabe > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Hector.Geraldino at ip-soft.net Tue Jan 10 18:55:46 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Tue, 10 Jan 2012 10:55:46 -0500 Subject: [Freeswitch-users] Any got a windows java esl.jar and lib built ? In-Reply-To: <4F0C305D.2050900@exemail.com.au> References: <4F0B95B1.90208@exemail.com.au> <4F0C305D.2050900@exemail.com.au> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507859@NY1-EXMB-01.ip-soft.net> I do recommend you to use it; no dependencies on system libs, based on the jboss/netty implementation of the new java-nio API, and so far I haven't had any issues with it (even in environments with heavy load) -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan How Sent: Tuesday, January 10, 2012 7:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Any got a windows java esl.jar and lib built ? Hi, Do you find it to work reliably?. I'll give it a go. It just looked like it hadn't been updated in quite a while, and the roadmap has a lot of todos still on it... Thanks, Ryan On 10/01/2012 4:45 PM, David Wafula wrote: > I use the client version instead > (http://wiki.freeswitch.org/wiki/Java_ESL_Client), but probably that > is not what you want for your needs. But works for me nicely. > Admittedly am on Linux, but its Java, so target platform is not an > issue. > > > On Tue, Jan 10, 2012 at 3:34 AM, Ryan How wrote: >> Hi, >> >> Has anyone got a windows esl.jar and corrsponding lib built?. I'm using >> the 2011 12 28 build of the windows binary. The instructions for >> building the ESL from source on the wiki are all for linux systems. >> >> There is also the pure java esl_client.jar, but it hasn't had updates >> since aug 2010 and the documentation looks pretty sparse!. Or doesn't it >> really matter, is it pretty stable and just like a drop in for the core >> version? I've generally learnt from spending hours and hours and hours >> messing with things, unless it looks supported or active it is going to >> take a lot of time to get it to work, and even then it probably won't >> work properly... >> >> >> Thanks! >> >> Ryan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From notlikeme75 at yahoo.com Tue Jan 10 19:01:06 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 10 Jan 2012 08:01:06 -0800 (PST) Subject: [Freeswitch-users] xml cdr stopped working Message-ID: <1326211266.10849.YahooMailNeo@web65307.mail.ac2.yahoo.com> my?C:\Program Files\FreeSWITCH\log\cdr-csv is updated with master.csv with the latest calls from today ?but the?C:\Program Files\FreeSWITCH\log\xml_cdr\archive\2012\Jan stop at jan 8th.? consequently preventing me from viewing them in the fusionpbx php. did i?accidentally?turn something off ? anyone ever have this problem. please help to get them back. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/c273e17f/attachment.html From bdfoster at endigotech.com Tue Jan 10 19:59:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 11:59:52 -0500 Subject: [Freeswitch-users] Voicemail skip greeting In-Reply-To: <4F0BE3CE.1020903@exemail.com.au> References: <4F0BE3CE.1020903@exemail.com.au> Message-ID: Try using skip greeting and skip instructions. On Jan 10, 2012 2:09 AM, "Ryan How" wrote: > Hi, > > I have a voicemail set up. I want it to not do the "Record your message > at the tone, to end recording....". I just want it to go "My message... > BEEEEEP". > > I've put skip_instructions=true in the dial plan, but it seems to have > no effect. Anything I am doing wrong? > > > > data="user/${dialed_extension}@${domain_name}"/> > > > > > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/fb4d7649/attachment.html From juanito1982 at gmail.com Tue Jan 10 20:04:41 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 10 Jan 2012 18:04:41 +0100 Subject: [Freeswitch-users] my_thread_global_end() X threads didn't exit Message-ID: Hello! I have some access to MySQL via ODBC in a LUA script. All works OK except I can see messages as 'threads didn't exit() X threads didn't exit' in the log. I tried to update unixODCB and libmyodbc to latest versions but those messages remain. Anyone got to solve this problem? I am working on Debian 6. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/d4c914ee/attachment.html From acosgrov at gmail.com Tue Jan 10 20:25:43 2012 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Tue, 10 Jan 2012 12:25:43 -0500 Subject: [Freeswitch-users] my_thread_global_end() X threads didn't exit In-Reply-To: References: Message-ID: Juan, I was running into the same issue on another distro. You'll need to make sure you are using the thread safe version of libmysqlclient (it will be suffixed by _t). I landed up changing to postgresql because of this. Anthony C. On Jan 10, 2012, at 12:04 PM, Juan Antonio Iba?ez Santorum wrote: > Hello! > > I have some access to MySQL via ODBC in a LUA script. All works OK except I can see messages as 'threads didn't exit() X threads didn't exit' in the log. I tried to update unixODCB and libmyodbc to latest versions but those messages remain. Anyone got to solve this problem? > > I am working on Debian 6. > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From curriegrad2004 at gmail.com Tue Jan 10 20:51:07 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 10 Jan 2012 09:51:07 -0800 Subject: [Freeswitch-users] xml cdr stopped working In-Reply-To: <1326211266.10849.YahooMailNeo@web65307.mail.ac2.yahoo.com> References: <1326211266.10849.YahooMailNeo@web65307.mail.ac2.yahoo.com> Message-ID: You may want to talk with the FusionPBX community about. On 2012-01-10 8:02 AM, "Rodney" wrote: > my C:\Program Files\FreeSWITCH\log\cdr-csv is updated with master.csv with > the latest calls from today but the C:\Program > Files\FreeSWITCH\log\xml_cdr\archive\2012\Jan > > stop at jan 8th. > > consequently preventing me from viewing them in the fusionpbx php. did i > accidentally turn something off ? anyone ever have this problem. please > help to get them back. thanks. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/5369c74f/attachment-0001.html From gkuri at ieee.org Tue Jan 10 21:48:58 2012 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 10 Jan 2012 10:48:58 -0800 Subject: [Freeswitch-users] FreeSWITCH in a University Environment Message-ID: Does anyone know of any Universities running FreeSWITCH, or other open source VoIP for that matter, for local call routing between handsets? We're looking at replacing our old Avaya system or upgrade it, and the forklift upgrade from Avaya is ridiculously expensive (no surprise). We'd like to replace our Avaya system with a combination of OpenSIPS and FreeSWITCH and some Cisco routers for external PSTN access, but it's going to be a tough sell to our CIO, unless we can show someone else has done it already. Any pointers to other Universities would be great. Cheers, Gabe From ifoundthetao at gmail.com Tue Jan 10 21:59:21 2012 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Tue, 10 Jan 2012 12:59:21 -0600 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: <4F0C8A89.8000607@gmail.com> Someone once told me, "Unless it saves me money, makes me money, or makes me look good--I don't want to hear about it." The solution you're proposing does two of the three! I found an article on Sam Houston State University using F/OSS VoIP. So that might be a lead. 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 1/10/2012 12:48 PM, Gabriel Kuri wrote: > Does anyone know of any Universities running FreeSWITCH, or other open > source VoIP for that matter, for local call routing between handsets? > > We're looking at replacing our old Avaya system or upgrade it, and the > forklift upgrade from Avaya is ridiculously expensive (no surprise). > > We'd like to replace our Avaya system with a combination of OpenSIPS > and FreeSWITCH and some Cisco routers for external PSTN access, but > it's going to be a tough sell to our CIO, unless we can show someone > else has done it already. > > Any pointers to other Universities would be great. > > Cheers, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanito1982 at gmail.com Tue Jan 10 22:00:31 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 10 Jan 2012 20:00:31 +0100 Subject: [Freeswitch-users] my_thread_global_end() X threads didn't exit In-Reply-To: References: Message-ID: Which is the library causing the problem? Libmysqlclient or libmyodbc? Were you been able to solve the problem? El 10/01/2012 18:26, "Anthony Cosgrove" escribi?: > Juan, > > I was running into the same issue on another distro. You'll need to make > sure you are using the thread safe version of libmysqlclient (it will be > suffixed by _t). > > I landed up changing to postgresql because of this. > > > Anthony C. > > On Jan 10, 2012, at 12:04 PM, Juan Antonio Iba?ez Santorum wrote: > > > Hello! > > > > I have some access to MySQL via ODBC in a LUA script. All works OK > except I can see messages as 'threads didn't exit() X threads didn't exit' > in the log. I tried to update unixODCB and libmyodbc to latest versions but > those messages remain. Anyone got to solve this problem? > > > > I am working on Debian 6. > > > > Regards > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/016b74fc/attachment.html From avi at avimarcus.net Tue Jan 10 22:23:45 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 10 Jan 2012 21:23:45 +0200 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: <4F0C8A89.8000607@gmail.com> References: <4F0C8A89.8000607@gmail.com> Message-ID: Would *actual businesses* deciding to go with FS not be good enough to show it's worth? -Avi On Tue, Jan 10, 2012 at 8:59 PM, Timothy Bolton wrote: > Someone once told me, "Unless it saves me money, makes me money, or > makes me look good--I don't want to hear about it." > > The solution you're proposing does two of the three! > > I found an article on Sam Houston State University using F/OSS VoIP. So > that might be a lead. > > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > > On 1/10/2012 12:48 PM, Gabriel Kuri wrote: > > Does anyone know of any Universities running FreeSWITCH, or other open > > source VoIP for that matter, for local call routing between handsets? > > > > We're looking at replacing our old Avaya system or upgrade it, and the > > forklift upgrade from Avaya is ridiculously expensive (no surprise). > > > > We'd like to replace our Avaya system with a combination of OpenSIPS > > and FreeSWITCH and some Cisco routers for external PSTN access, but > > it's going to be a tough sell to our CIO, unless we can show someone > > else has done it already. > > > > Any pointers to other Universities would be great. > > > > Cheers, > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/2880a538/attachment.html From acosgrov at gmail.com Tue Jan 10 22:56:05 2012 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Tue, 10 Jan 2012 14:56:05 -0500 Subject: [Freeswitch-users] my_thread_global_end() X threads didn't exit In-Reply-To: References: Message-ID: Libmyodbc needs to link against libmysqlclient_t (run ld on libmyodbc to see dependencies). No never got it solved. My install (Gentoo) reported it was linked against the thread safe version but I still got the error so that's when I moved to pgsql. Anthony C. On Jan 10, 2012, at 2:00 PM, Juan Antonio Iba?ez Santorum wrote: > Which is the library causing the problem? Libmysqlclient or libmyodbc? Were you been able to solve the problem? > > El 10/01/2012 18:26, "Anthony Cosgrove" escribi?: > Juan, > > I was running into the same issue on another distro. You'll need to make sure you are using the thread safe version of libmysqlclient (it will be suffixed by _t). > > I landed up changing to postgresql because of this. > > > Anthony C. > > On Jan 10, 2012, at 12:04 PM, Juan Antonio Iba?ez Santorum wrote: > > > Hello! > > > > I have some access to MySQL via ODBC in a LUA script. All works OK except I can see messages as 'threads didn't exit() X threads didn't exit' in the log. I tried to update unixODCB and libmyodbc to latest versions but those messages remain. Anyone got to solve this problem? > > > > I am working on Debian 6. > > > > Regards > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/37545214/attachment-0001.html From bdfoster at endigotech.com Tue Jan 10 23:00:56 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 15:00:56 -0500 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: <4F0C8A89.8000607@gmail.com> Message-ID: I agree with Avi, actual businesses are using FS with great results. If you aren't too sure about commercial support or not having the personell to maintain FS, might I suggest CudaTel? -bdfoster On Tue, Jan 10, 2012 at 2:23 PM, Avi Marcus wrote: > Would *actual businesses* deciding to go with FS not be good enough to > show it's worth? > -Avi > > > On Tue, Jan 10, 2012 at 8:59 PM, Timothy Bolton wrote: > >> Someone once told me, "Unless it saves me money, makes me money, or >> makes me look good--I don't want to hear about it." >> >> The solution you're proposing does two of the three! >> >> I found an article on Sam Houston State University using F/OSS VoIP. So >> that might be a lead. >> >> 'We who cut mere stones must always be envisioning cathedrals.' >> Quarry Worker's Creed >> >> >> On 1/10/2012 12:48 PM, Gabriel Kuri wrote: >> > Does anyone know of any Universities running FreeSWITCH, or other open >> > source VoIP for that matter, for local call routing between handsets? >> > >> > We're looking at replacing our old Avaya system or upgrade it, and the >> > forklift upgrade from Avaya is ridiculously expensive (no surprise). >> > >> > We'd like to replace our Avaya system with a combination of OpenSIPS >> > and FreeSWITCH and some Cisco routers for external PSTN access, but >> > it's going to be a tough sell to our CIO, unless we can show someone >> > else has done it already. >> > >> > Any pointers to other Universities would be great. >> > >> > Cheers, >> > Gabe >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/8bec7b32/attachment.html From jehudi.maes at gmail.com Tue Jan 10 18:31:09 2012 From: jehudi.maes at gmail.com (Jehudi Maes) Date: Tue, 10 Jan 2012 16:31:09 +0100 Subject: [Freeswitch-users] Transfer out of lua script on fax detection - novice user In-Reply-To: <4D49164B-8C3B-4023-83DE-FCC129C81B3A@gmail.com> References: <4D49164B-8C3B-4023-83DE-FCC129C81B3A@gmail.com> Message-ID: Antony, Thanks for your reply. The script is some IVR script (quite standard). As you say correctly my script is indeed exiting too early, this is also what I would like it to do. When a fax is detected it should the lua script should be ended and the call needs to transfer to the fax extension. It seems that the script exiting is OK but also generates a hangup. Kind regards, Jehudi On Tue, Jan 10, 2012 at 2:54 PM, Anthony Cosgrove wrote: > Sorry I took a second look at your description and I answered a little > prematurely... sounds like your script is exiting too early. Can you > pastebin the script? > > http://pastebin.freeswitch.org > > Anthony C > > On Jan 10, 2012, at 8:50 AM, Anthony Cosgrove wrote: > > Jehudi, > > I just went through this exercise. You will need to invoke set_zombie_exec > from either the xml or while in your Lua script. Take a look at the wiki > page - > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_zombie_exec > > > Anthony C. > > On Jan 10, 2012, at 3:53 AM, Jehudi Maes wrote: > > I am having trouble with fax detection while executing a lua script: I am > setting the fax detection in the dialplan: > > > > and continue by executing a script: > > > > when a fax is detected (this works fine) during the execution of the > script, the script exits and the call is hangup. How can I avoid the hangup > and continue with the fax receiving ? > > Kind regards, > > Jehudi > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/7fcad2ba/attachment.html From cmrienzo at gmail.com Tue Jan 10 23:28:43 2012 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Tue, 10 Jan 2012 15:28:43 -0500 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: <4F0C8A89.8000607@gmail.com> References: <4F0C8A89.8000607@gmail.com> Message-ID: <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> That university has switched back to Cisco. On Jan 10, 2012, at 13:59, Timothy Bolton wrote: > Someone once told me, "Unless it saves me money, makes me money, or > makes me look good--I don't want to hear about it." > > The solution you're proposing does two of the three! > > I found an article on Sam Houston State University using F/OSS VoIP. So > that might be a lead. > > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > > On 1/10/2012 12:48 PM, Gabriel Kuri wrote: >> Does anyone know of any Universities running FreeSWITCH, or other open >> source VoIP for that matter, for local call routing between handsets? >> >> We're looking at replacing our old Avaya system or upgrade it, and the >> forklift upgrade from Avaya is ridiculously expensive (no surprise). >> >> We'd like to replace our Avaya system with a combination of OpenSIPS >> and FreeSWITCH and some Cisco routers for external PSTN access, but >> it's going to be a tough sell to our CIO, unless we can show someone >> else has done it already. >> >> Any pointers to other Universities would be great. >> >> Cheers, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanito1982 at gmail.com Tue Jan 10 23:37:38 2012 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Tue, 10 Jan 2012 21:37:38 +0100 Subject: [Freeswitch-users] my_thread_global_end() X threads didn't exit In-Reply-To: References: Message-ID: Mine is using thread safe version. Noone had that problem before? 2012/1/10 Anthony Cosgrove > Libmyodbc needs to link against libmysqlclient_t (run ld on libmyodbc to > see dependencies). > > No never got it solved. My install (Gentoo) reported it was linked against > the thread safe version but I still got the error so that's when I moved to > pgsql. > > > > Anthony C. > > > On Jan 10, 2012, at 2:00 PM, Juan Antonio Iba?ez Santorum wrote: > > Which is the library causing the problem? Libmysqlclient or libmyodbc? > Were you been able to solve the problem? > El 10/01/2012 18:26, "Anthony Cosgrove" escribi?: > >> Juan, >> >> I was running into the same issue on another distro. You'll need to make >> sure you are using the thread safe version of libmysqlclient (it will be >> suffixed by _t). >> >> I landed up changing to postgresql because of this. >> >> >> Anthony C. >> >> On Jan 10, 2012, at 12:04 PM, Juan Antonio Iba?ez Santorum wrote: >> >> > Hello! >> > >> > I have some access to MySQL via ODBC in a LUA script. All works OK >> except I can see messages as 'threads didn't exit() X threads didn't exit' >> in the log. I tried to update unixODCB and libmyodbc to latest versions but >> those messages remain. Anyone got to solve this problem? >> > >> > I am working on Debian 6. >> > >> > Regards >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/0ef49ae2/attachment-0001.html From msc at freeswitch.org Wed Jan 11 00:22:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 13:22:52 -0800 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> Message-ID: On Tue, Jan 10, 2012 at 12:28 PM, wrote: > That university has switched back to Cisco. > > Yeah, they used Asterisk before FreeSWITCH was available and mature. The problem they ran into with Asterisk at SHSU was they needed people full time to baby sit their * servers and make ridiculous patches all day. They ended up spending a lot of money on people. With Cisco, they get to spend all that money on equipment, licensing, etc. I don't know of any large campuses using FreeSWITCH. Depending on the needs of the campus the CudaTel might work. Bare FreeSWITCH servers w/ OpenSIPS would also be an interesting solution but you'll still need to have someone be a "vendor" for support, etc. FSS will do support contracts on the FreeSWITCH stuff but you still need someone for the OpenSIPS side. I know Bogdan (from OpenSIPS) does that, as does Flavio Goncalves (OpenSIPS trainer, author). It sounds to me like you could save a bundle with OSS but it will take a coordinated effort and you must definitely get experts to spec out the install before hand. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/37c3e91d/attachment.html From msc at freeswitch.org Wed Jan 11 00:28:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 13:28:18 -0800 Subject: [Freeswitch-users] in ivr menu In-Reply-To: References: Message-ID: On Tue, Jan 10, 2012 at 5:01 AM, Anita Hall wrote: > Added to Wiki. > > Thanks Anita! Much appreciated. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/10981500/attachment.html From rsaavedra at ecogizmos.com Wed Jan 11 00:28:28 2012 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Tue, 10 Jan 2012 16:28:28 -0500 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> Message-ID: We have a Trixbox (Asterisk) server in the "Universidad Autonoma del Caribe" in Barranquilla, Colombia with extensions in: Miami, Usa; Cartagena; Santa Marta; Bogot?. We have external PSTN access (2 E1) in all the locations with 200 extensions and follow me to the cellular (one E1). The server have 3 years on production. Best Regards, Ricardo Saavedra > That university has switched back to Cisco. > > On Jan 10, 2012, at 13:59, Timothy Bolton wrote: > >> Someone once told me, "Unless it saves me money, makes me money, or >> makes me look good--I don't want to hear about it." >> >> The solution you're proposing does two of the three! >> >> I found an article on Sam Houston State University using F/OSS VoIP. So >> that might be a lead. >> >> 'We who cut mere stones must always be envisioning cathedrals.' >> Quarry Worker's Creed >> >> >> On 1/10/2012 12:48 PM, Gabriel Kuri wrote: >>> Does anyone know of any Universities running FreeSWITCH, or other open >>> source VoIP for that matter, for local call routing between handsets? >>> >>> We're looking at replacing our old Avaya system or upgrade it, and the >>> forklift upgrade from Avaya is ridiculously expensive (no surprise). >>> >>> We'd like to replace our Avaya system with a combination of OpenSIPS >>> and FreeSWITCH and some Cisco routers for external PSTN access, but >>> it's going to be a tough sell to our CIO, unless we can show someone >>> else has done it already. >>> >>> Any pointers to other Universities would be great. >>> >>> Cheers, >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rhow at exemail.com.au Wed Jan 11 03:44:40 2012 From: rhow at exemail.com.au (Ryan How) Date: Wed, 11 Jan 2012 08:44:40 +0800 Subject: [Freeswitch-users] Voicemail skip greeting In-Reply-To: References: <4F0BE3CE.1020903@exemail.com.au> Message-ID: <4F0CDB78.3010407@exemail.com.au> Hi, I have tried both skip_greeting and skip_instructions and both seem to have no effect. (Or they are doing something different to what I think they should be doing) Perhaps I am not using it right? On 11/01/2012 12:59 AM, Brian Foster wrote: > > Try using skip greeting and skip instructions. > > On Jan 10, 2012 2:09 AM, "Ryan How" > wrote: > > Hi, > > I have a voicemail set up. I want it to not do the "Record your > message > at the tone, to end recording....". I just want it to go "My > message... > BEEEEEP". > > I've put skip_instructions=true in the dial plan, but it seems to have > no effect. Anything I am doing wrong? > > > > data="user/${dialed_extension}@${domain_name}"/> > > > > > > Thanks! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/ebacfb3d/attachment.html From andrew.radke at yuruga.com.au Wed Jan 11 02:19:22 2012 From: andrew.radke at yuruga.com.au (Andrew Radke) Date: Wed, 11 Jan 2012 09:19:22 +1000 Subject: [Freeswitch-users] simple B2BUA install on OpenWRT router Message-ID: Hi all, I would like to create a Freeswitch install on our OpenWRT based router to work as a B2BUA SBC for our sipX PBX. Very simple requirements: authenticate outgoing calls for our ITSP preferably enforce negotiation of G729 due to limited bandwidth no need to relay RTP, just let it get passed through from the phones to the ITSP directly possibly register with our ITSP for inbound calls which would be sent to the PBX (unlikely to be used, but nice as an option) An advantage would also be that it can be started and stopped, etc in line with the state of the connection rather than waiting for a timeout if the Internet is down. I've looked at the Freeswitch documentation but it is heavily aimed at people looking to do more complex setups and you need to learn a lot about it. I'm hoping that someone here could provide me with the basics to get something like this going and then I will document it for both the sipx and freeswitch communities. The existing documents use fairly complex setups and Kamailio and I'd prefer to keep it as simple as possible. Also, it's possible that Freeswitch isn't the right option for this with something like opensips being a better way to achieve what I need. Please tell me if you think this is the case too. Regards, Andrew Radke Yuruga Nursery Pty Ltd Clonal Solutions Australia Pty Ltd PO Box 220 Walkamin Qld 4872 Phone: (07) 4093 3826 Fax: (07) 4093 3869 Email: andrew.radke at yuruga.com.au Web: www.yuruga.com.au -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/172a91b9/attachment-0001.html From msc at freeswitch.org Wed Jan 11 04:16:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 17:16:12 -0800 Subject: [Freeswitch-users] Voicemail skip greeting In-Reply-To: <4F0CDB78.3010407@exemail.com.au> References: <4F0BE3CE.1020903@exemail.com.au> <4F0CDB78.3010407@exemail.com.au> Message-ID: On Tue, Jan 10, 2012 at 4:44 PM, Ryan How wrote: > Hi, > > I have tried both skip_greeting and skip_instructions and both seem to > have no effect. (Or they are doing something different to what I think they > should be doing) > > Perhaps I am not using it right? > It looks like you're using loopback to call voicemail which means you probably need to set skip_xx=true in the dialstring to make sure the right channel picks it up. Try this and let us know: -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/90881be3/attachment.html From john at whitesmiths.com Wed Jan 11 04:52:43 2012 From: john at whitesmiths.com (John O'Brien) Date: Wed, 11 Jan 2012 12:52:43 +1100 Subject: [Freeswitch-users] Freeswitch Process Dies Message-ID: Hi, We have a reasonably recent version of FreeSWITCH running. FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) We are running on CentOS 6 We start FreeSWITCH from a shell script. ulimit -s 240 freeswitch -ncwait -u usrXXX -g grpXXX The issue is that things seem to go along nicely for a day to two. Then without any apparent reason the freeswitch process dies. Usually there is nothing happening when this occurs, system is idle. There appear to be no process remnants we can use for investigation. We would appreciate some suggestions on how we might track the issue down. Is there some simple logging we can turn on etc? Regards, John O'Brien john at whitesmiths.com From darcy at primrose.ws Wed Jan 11 05:04:15 2012 From: darcy at primrose.ws (Darcy) Date: Tue, 10 Jan 2012 21:04:15 -0500 Subject: [Freeswitch-users] using ip and port from contact Message-ID: Hi all. I have a freeswitch sitting on a public IP that has a multitude of devices, mostly spa2102s, spread across the country, all behind routers. Our 8xx voip supplier (iristel) sends us traffic on port 5060 but their switch is behind a firewall. As a result I receive traffic from them on any port the firewall chooses but they want the response to be to port 5060, which is of course in the contact message. This is trunk group is authorized by IP address. Is there a setting I can do that will allow me to return the traffic to them on the ip and port in the contact but not louse up the normal nat traffic from the atas. Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/4ecc6524/attachment.html From bdfoster at endigotech.com Wed Jan 11 05:10:48 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 21:10:48 -0500 Subject: [Freeswitch-users] Dial by IP address Message-ID: I have some phones on my network that are capable of dialing a SIP URI. It seems by default FreeSWITCH doesn't allow this, as it just picks up everything before the @ and puts it as the destination number. Is there a way to allow this? What would be needed in the dialplan? On Jan 10, 2012 9:05 PM, "John O'Brien" wrote: > Hi, > > We have a reasonably recent version of FreeSWITCH running. > FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > We are running on CentOS 6 > > We start FreeSWITCH from a shell script. > > ulimit -s 240 > freeswitch -ncwait -u usrXXX -g grpXXX > > The issue is that things seem to go along nicely for a day to two. > Then without any apparent reason the freeswitch process dies. > Usually there is nothing happening when this occurs, system is idle. > > There appear to be no process remnants we can use for investigation. > > We would appreciate some suggestions on how we might track the issue down. > Is there some simple logging we can turn on etc? > > Regards, > > John O'Brien > john at whitesmiths.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/ec320a8e/attachment.html From krice at freeswitch.org Wed Jan 11 05:18:52 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 10 Jan 2012 20:18:52 -0600 Subject: [Freeswitch-users] Freeswitch Process Dies In-Reply-To: Message-ID: Set ulimit -c unlimited and see if you are dropping a core Also another thing you can do is use the freeswitch.redhat script in the source tree this is init.d compatible it has all the good ulimits set in it K On 1/10/12 7:52 PM, "John O'Brien" wrote: > Hi, > > We have a reasonably recent version of FreeSWITCH running. > FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > We are running on CentOS 6 > > We start FreeSWITCH from a shell script. > > ulimit -s 240 > freeswitch -ncwait -u usrXXX -g grpXXX > > The issue is that things seem to go along nicely for a day to two. > Then without any apparent reason the freeswitch process dies. > Usually there is nothing happening when this occurs, system is idle. > > There appear to be no process remnants we can use for investigation. > > We would appreciate some suggestions on how we might track the issue down. > Is there some simple logging we can turn on etc? > > Regards, > > John O'Brien > john at whitesmiths.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at freeswitch.org Wed Jan 11 05:20:06 2012 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Tue, 10 Jan 2012 23:20:06 -0300 Subject: [Freeswitch-users] Freeswitch Process Dies In-Reply-To: References: Message-ID: There should be a core dump file somewhere on your system. Check on the /usr/loca/freeswitch/bin folder or if you ran FS manually (no init script), check the folder where you started it from. With that core.PID, I can teach you how to extract debug information that is useful for the developers. Another hint is ALWAYS upgrade to the latest git before even sending out email. Chances are it was a temporary problem that was already solved. Jo?o Mesquita On Tue, Jan 10, 2012 at 10:52 PM, John O'Brien wrote: > Hi, > > We have a reasonably recent version of FreeSWITCH running. > FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > We are running on CentOS 6 > > We start FreeSWITCH from a shell script. > > ulimit -s 240 > freeswitch -ncwait -u usrXXX -g grpXXX > > The issue is that things seem to go along nicely for a day to two. > Then without any apparent reason the freeswitch process dies. > Usually there is nothing happening when this occurs, system is idle. > > There appear to be no process remnants we can use for investigation. > > We would appreciate some suggestions on how we might track the issue down. > Is there some simple logging we can turn on etc? > > Regards, > > John O'Brien > john at whitesmiths.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/e308eb07/attachment.html From krice at freeswitch.org Wed Jan 11 05:38:11 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 10 Jan 2012 20:38:11 -0600 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: Message-ID: Hey db don?t hijack threads like that... Start a new thread... This really screws with the thread tracking on all the web archival stuff K On 1/10/12 8:10 PM, "Brian Foster" wrote: > I have some phones on my network that are capable of dialing a SIP URI. It > seems by default FreeSWITCH doesn't allow this, as it just picks up everything > before the @ and puts it as the destination number. Is there a way to allow > this? What would be needed in the dialplan? > > On Jan 10, 2012 9:05 PM, "John O'Brien" wrote: >> Hi, >> >> We have a reasonably recent version of FreeSWITCH running. >> FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) >> We are running on CentOS 6 >> >> We start FreeSWITCH from a shell script. >> >> ulimit -s 240 >> freeswitch -ncwait -u usrXXX -g grpXXX >> >> The issue is that things seem to go along nicely for a day to two. >> Then without any apparent reason the freeswitch process dies. >> Usually there is nothing happening when this occurs, system is idle. >> >> There appear to be no process remnants we can use for investigation. >> >> We would appreciate some suggestions on how we might track the issue down. >> Is there some simple logging we can turn on etc? >> >> Regards, >> >> John O'Brien >> john at whitesmiths.com >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/84d2452a/attachment-0001.html From darcy at primrose.ws Wed Jan 11 05:42:29 2012 From: darcy at primrose.ws (Darcy) Date: Tue, 10 Jan 2012 21:42:29 -0500 Subject: [Freeswitch-users] using ip and port from contact In-Reply-To: References: Message-ID: <5B1D76C36F4B4FF5B9583F50D929D859@DWP> Please disregard this, it ended up being caused by an earlier change to the sip profile ?? to correct another problem, hard to please everyone! From: Darcy Sent: Tuesday, January 10, 2012 9:04 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] using ip and port from contact Hi all. I have a freeswitch sitting on a public IP that has a multitude of devices, mostly spa2102s, spread across the country, all behind routers. Our 8xx voip supplier (iristel) sends us traffic on port 5060 but their switch is behind a firewall. As a result I receive traffic from them on any port the firewall chooses but they want the response to be to port 5060, which is of course in the contact message. This is trunk group is authorized by IP address. Is there a setting I can do that will allow me to return the traffic to them on the ip and port in the contact but not louse up the normal nat traffic from the atas. Darcy -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/28f9908c/attachment.html From bdfoster at endigotech.com Wed Jan 11 05:53:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 21:53:58 -0500 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: References: Message-ID: Huh? I could have sworn I started another thread? On Jan 10, 2012 9:52 PM, "Ken Rice" wrote: > Hey db don?t hijack threads like that... Start a new thread... This > really screws with the thread tracking on all the web archival stuff > > K > > > On 1/10/12 8:10 PM, "Brian Foster" wrote: > > I have some phones on my network that are capable of dialing a SIP URI. It > seems by default FreeSWITCH doesn't allow this, as it just picks up > everything before the @ and puts it as the destination number. Is there a > way to allow this? What would be needed in the dialplan? > > On Jan 10, 2012 9:05 PM, "John O'Brien" wrote: > > Hi, > > We have a reasonably recent version of FreeSWITCH running. > FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > We are running on CentOS 6 > > We start FreeSWITCH from a shell script. > > ulimit -s 240 > freeswitch -ncwait -u usrXXX -g grpXXX > > The issue is that things seem to go along nicely for a day to two. > Then without any apparent reason the freeswitch process dies. > Usually there is nothing happening when this occurs, system is idle. > > There appear to be no process remnants we can use for investigation. > > We would appreciate some suggestions on how we might track the issue down. > Is there some simple logging we can turn on etc? > > Regards, > > John O'Brien > john at whitesmiths.com > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/e46529fc/attachment.html From msc at freeswitch.org Wed Jan 11 05:58:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 18:58:16 -0800 Subject: [Freeswitch-users] compiling mod_perl fail help required. In-Reply-To: <1326023855.5065.YahooMailClassic@web110815.mail.gq1.yahoo.com> References: <4F09366D.706@exemail.com.au> <1326023855.5065.YahooMailClassic@web110815.mail.gq1.yahoo.com> Message-ID: Looks like you don't have the Perl dev libraries installed. -MC On Sun, Jan 8, 2012 at 3:57 AM, Sherif Omran wrote: > Hello > > I am trying to compile mod-perl but i get the following, i would > appreciate a help > > best regards, > Sheirf Omran > > > Can't locate ExtUtils/Embed.pm in @INC (@INC contains: > /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl > /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). > BEGIN failed--compilation aborted. > Can't locate ExtUtils/Embed.pm in @INC (@INC contains: > /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl > /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). > BEGIN failed--compilation aborted. > Compiling freeswitch_perl.cpp... > g++ -w -DMULTIPLICITY -DEMBED_PERL -I/usr/local/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE > -DHAVE_CONFIG_H -c -o freeswitch_perl.o freeswitch_perl.cpp > In file included from freeswitch_perl.cpp:2: > freeswitch_perl.h:13:20: error: EXTERN.h: No such file or directory > freeswitch_perl.h:14:18: error: perl.h: No such file or directory > In file included from freeswitch_perl.cpp:2: > freeswitch_perl.h:22: error: ISO C++ forbids declaration of > 'PerlInterpreter' with no type > freeswitch_perl.h:22: error: expected ';' before '*' token > freeswitch_perl.h:23: error: ISO C++ forbids declaration of > 'PerlInterpreter' with no type > freeswitch_perl.h:23: error: expected ';' before '*' token > freeswitch_perl.h:26: error: ISO C++ forbids declaration of 'SV' with no > type > freeswitch_perl.h:26: error: expected ';' before '*' token > freeswitch_perl.h:39: error: 'SV' has not been declared > freeswitch_perl.h:49: error: 'PerlInterpreter' has not been declared > In file included from freeswitch_perl.cpp:3: > mod_perl_extra.h:3: error: variable or field 'mod_perl_conjure_event' > declared void > mod_perl_extra.h:3: error: 'PerlInterpreter' was not declared in this scope > mod_perl_extra.h:3: error: 'my_perl' was not declared in this scope > mod_perl_extra.h:3: error: expected primary-expression before '*' token > mod_perl_extra.h:3: error: 'event' was not declared in this scope > mod_perl_extra.h:3: error: expected primary-expression before 'const' > mod_perl_extra.h:4: error: variable or field 'mod_perl_conjure_stream' > declared void > mod_perl_extra.h:4: error: 'PerlInterpreter' was not declared in this scope > mod_perl_extra.h:4: error: 'my_perl' was not declared in this scope > mod_perl_extra.h:4: error: expected primary-expression before '*' token > mod_perl_extra.h:4: error: 'stream' was not declared in this scope > mod_perl_extra.h:4: error: expected primary-expression before 'const' > freeswitch_perl.cpp:5: error: 'STRLEN' does not name a type > freeswitch_perl.cpp: In constructor 'PERL::Session::Session()': > freeswitch_perl.cpp:13: error: 'my_perl' was not declared in this scope > freeswitch_perl.cpp: In constructor 'PERL::Session::Session(char*, > CoreSession*)': > freeswitch_perl.cpp:18: error: 'my_perl' was not declared in this scope > freeswitch_perl.cpp: In constructor > 'PERL::Session::Session(switch_core_session_t*)': > freeswitch_perl.cpp:34: error: 'my_perl' was not declared in this scope > freeswitch_perl.cpp: At global scope: > freeswitch_perl.cpp:88: error: variable or field 'setPERL' declared void > freeswitch_perl.cpp:88: error: 'PerlInterpreter' was not declared in this > scope > freeswitch_perl.cpp:88: error: 'pi' was not declared in this scope > freeswitch_perl.cpp:94: error: variable or field 'setME' declared void > freeswitch_perl.cpp:94: error: 'SV' was not declared in this scope > freeswitch_perl.cpp:94: error: 'p' was not declared in this scope > make[4]: *** [freeswitch_perl.o] Error 1 > make[3]: *** [all] Error 1 > make[2]: *** [mod_perl-all] Error 1 > make[1]: *** [mod_perl] Error 2 > make: *** [mod_perl] Error 2 > [root at sip freeswitch]# perl -MExtUtils::Embed -e xsinit > Can't locate ExtUtils/Embed.pm in @INC (@INC contains: > /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl > /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). > BEGIN failed--compilation aborted. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/0a302cde/attachment-0001.html From sunwood360 at gmail.com Wed Jan 11 06:16:05 2012 From: sunwood360 at gmail.com (envelopes envelopes) Date: Tue, 10 Jan 2012 19:16:05 -0800 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> Message-ID: On Jan 10, 2012 1:32 PM, "Michael Collins" wrote: > > > > On Tue, Jan 10, 2012 at 12:28 PM, wrote: >> >> That university has switched back to Cisco. >> > Yeah, they used Asterisk before FreeSWITCH was available and mature. The problem they ran into with Asterisk at SHSU was they needed people full time to baby sit their * servers and make ridiculous patches all day. ^^^^ that is exactly the issue here either : when somebody asks a question on the board, the answer is always: have you upgraded to header version? Or do not use it. Enterprise can't afford patch& upgrade daily. You guys need to take a strategy to release and support. For instance, two release: stable release will has a life span 3 years and be supported. Test release with latest features. They ended up spending a lot of money on people. With Cisco, they get to spend all that money on equipment, licensing, etc. > > I don't know of any large campuses using FreeSWITCH. Depending on the needs of the campus the CudaTel might work. Bare FreeSWITCH servers w/ OpenSIPS would also be an interesting solution but you'll still need to have someone be a "vendor" for support, etc. FSS will do support contracts on the FreeSWITCH stuff but you still need someone for the OpenSIPS side. I know Bogdan (from OpenSIPS) does that, as does Flavio Goncalves (OpenSIPS trainer, author). > > It sounds to me like you could save a bundle with OSS but it will take a coordinated effort and you must definitely get experts to spec out the install before hand. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/ffef6bd6/attachment.html From rhow at exemail.com.au Wed Jan 11 07:01:56 2012 From: rhow at exemail.com.au (Ryan How) Date: Wed, 11 Jan 2012 12:01:56 +0800 Subject: [Freeswitch-users] Voicemail skip greeting In-Reply-To: References: <4F0BE3CE.1020903@exemail.com.au> <4F0CDB78.3010407@exemail.com.au> Message-ID: <4F0D09B4.7060603@exemail.com.au> Thanks! That worked!. I am just using the default config which I am slowly modifying. I don't understand the difference between using bridge loopback or application voicemail. Does loopback drop all the channel variables? Thanks, Ryan On 11/01/2012 9:16 AM, Michael Collins wrote: > > > On Tue, Jan 10, 2012 at 4:44 PM, Ryan How > wrote: > > Hi, > > I have tried both skip_greeting and skip_instructions and both > seem to have no effect. (Or they are doing something different to > what I think they should be doing) > > Perhaps I am not using it right? > > It looks like you're using loopback to call voicemail which means you > probably need to set skip_xx=true in the dialstring to make sure the > right channel picks it up. Try this and let us know: > > data="{skip_instructions=true}loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> > > -MC > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/f2ed15af/attachment.html From bdfoster at endigotech.com Wed Jan 11 07:14:21 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 23:14:21 -0500 Subject: [Freeswitch-users] Dial by IP In-Reply-To: References: Message-ID: I have some phones on my network that are capable of dialing a SIP URI. It seems by default FreeSWITCH doesn't allow this, as it just picks up everything before the @ and puts it as the destination number. Is there a way to allow this? What would be needed in the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/08b5cf0d/attachment.html From msc at freeswitch.org Wed Jan 11 07:16:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 20:16:14 -0800 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> Message-ID: > Enterprise can't afford patch& upgrade daily. You guys need to take a > strategy to release and support. For instance, two release: stable release > will has a life span 3 years and be supported. Test release with latest > features. > We've got some things in the works. Stay tuned! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/04e25b22/attachment.html From bwibowo at gmail.com Wed Jan 11 07:21:26 2012 From: bwibowo at gmail.com (budi wibowo) Date: Wed, 11 Jan 2012 11:21:26 +0700 Subject: [Freeswitch-users] Ota in sip In-Reply-To: References: <1284558287-1326164994-cardhu_decombobulator_blackberry.rim.net-392685695-@b27.c2.bise3.blackberry> Message-ID: any softphone support this function? On Tue, Jan 10, 2012 at 1:19 PM, Avi Marcus wrote: > Are you referring to provisioning, the ability for the client phones > to get new configurations from the server? > Most hardphones support such an option, and Cudatel, fusionpbx and I > think blue.box / 2600hz comes with a large list of phones they > support, but it's not part of sip and it's certainly not > standardized.. > > -Avi > > > On Tue, Jan 10, 2012 at 5:09 AM, wrote: > > Dear all > > Is there any implementation of OTA in sip. > > Purpose is to upload new parameter when user connected to sip server. > > > > Thx > > > > Budi > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/f6b52675/attachment.html From curriegrad2004 at gmail.com Wed Jan 11 07:30:58 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 10 Jan 2012 20:30:58 -0800 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: References: Message-ID: Ken, you're getting old :P No, bd did start a new thread :P On Tue, Jan 10, 2012 at 6:53 PM, Brian Foster wrote: > Huh? I could have sworn I started another thread? > > On Jan 10, 2012 9:52 PM, "Ken Rice" wrote: >> >> Hey db don?t hijack threads like that... Start a new thread... This really >> screws with the thread tracking on all the web archival stuff >> >> K >> >> >> On 1/10/12 8:10 PM, "Brian Foster" wrote: >> >> I have some phones on my network that are capable of dialing a SIP URI. It >> seems by default FreeSWITCH doesn't allow this, as it just picks up >> everything before the @ and puts it as the destination number. Is there a >> way to allow this? What would be needed in the dialplan? >> >> On Jan 10, 2012 9:05 PM, "John O'Brien" wrote: >> >> Hi, >> >> We have a reasonably recent version of FreeSWITCH running. >> FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) >> We are running on CentOS 6 >> >> We start FreeSWITCH from a shell script. >> >> ulimit -s 240 >> freeswitch -ncwait -u usrXXX -g grpXXX >> >> The issue is that things seem to go along nicely for a day to two. >> Then without any apparent reason the freeswitch process dies. >> Usually there is nothing happening when this occurs, system is idle. >> >> There appear to be no process remnants we can use for investigation. >> >> We would appreciate some suggestions on how we might track the issue down. >> Is there some simple logging we can turn on etc? >> >> Regards, >> >> John O'Brien >> john at whitesmiths.com >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed Jan 11 07:45:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jan 2012 20:45:02 -0800 Subject: [Freeswitch-users] Voicemail skip greeting In-Reply-To: <4F0D09B4.7060603@exemail.com.au> References: <4F0BE3CE.1020903@exemail.com.au> <4F0CDB78.3010407@exemail.com.au> <4F0D09B4.7060603@exemail.com.au> Message-ID: On Tue, Jan 10, 2012 at 8:01 PM, Ryan How wrote: > Thanks! > > That worked!. I am just using the default config which I am slowly > modifying. I don't understand the difference between using bridge loopback > or application voicemail. Does loopback drop all the channel variables? > yes, because the bridge app creates an entirely new channel, whereas the voicemail app uses the current channel. When you create a new channel you need to export any desired channel variables over to it. Hope this helps. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/9d9029a7/attachment.html From bdfoster at endigotech.com Wed Jan 11 07:45:06 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jan 2012 23:45:06 -0500 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: References: Message-ID: Well I eventually did but I actually reply, changed the subject, and didn't actually delete the reply portion of the email. I'll do better and not be so lazy next time. ::adds mailing list to address book:: On Jan 10, 2012 11:39 PM, "curriegrad2004" wrote: > Ken, you're getting old :P > > No, bd did start a new thread :P > > On Tue, Jan 10, 2012 at 6:53 PM, Brian Foster > wrote: > > Huh? I could have sworn I started another thread? > > > > On Jan 10, 2012 9:52 PM, "Ken Rice" wrote: > >> > >> Hey db don?t hijack threads like that... Start a new thread... This > really > >> screws with the thread tracking on all the web archival stuff > >> > >> K > >> > >> > >> On 1/10/12 8:10 PM, "Brian Foster" wrote: > >> > >> I have some phones on my network that are capable of dialing a SIP URI. > It > >> seems by default FreeSWITCH doesn't allow this, as it just picks up > >> everything before the @ and puts it as the destination number. Is there > a > >> way to allow this? What would be needed in the dialplan? > >> > >> On Jan 10, 2012 9:05 PM, "John O'Brien" > wrote: > >> > >> Hi, > >> > >> We have a reasonably recent version of FreeSWITCH running. > >> FreeSWITCH version: 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) > >> We are running on CentOS 6 > >> > >> We start FreeSWITCH from a shell script. > >> > >> ulimit -s 240 > >> freeswitch -ncwait -u usrXXX -g grpXXX > >> > >> The issue is that things seem to go along nicely for a day to two. > >> Then without any apparent reason the freeswitch process dies. > >> Usually there is nothing happening when this occurs, system is idle. > >> > >> There appear to be no process remnants we can use for investigation. > >> > >> We would appreciate some suggestions on how we might track the issue > down. > >> Is there some simple logging we can turn on etc? > >> > >> Regards, > >> > >> John O'Brien > >> john at whitesmiths.com > >> > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> ________________________________ > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120110/88ba4275/attachment.html From amit.nakum2009 at gmail.com Wed Jan 11 08:26:53 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Wed, 11 Jan 2012 10:56:53 +0530 Subject: [Freeswitch-users] PBX In-Reply-To: <4F0C3109.3090201@softnet.si> References: <4F0C3109.3090201@softnet.si> Message-ID: dear, it is possible by using concept of Multiple domain.following link will be helpful. 1)http://wiki.freeswitch.org/wiki/Multi-tenant 2)http://wiki.freeswitch.org/wiki/Multiple_Companies Regards Amit On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek wrote: > Hi, > > is it possible to have on freeswitch more than one PBX? If it is, where > can I find this threat on wiki. > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/4905dbe3/attachment.html From krice at freeswitch.org Wed Jan 11 09:16:18 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Jan 2012 00:16:18 -0600 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: Message-ID: Really now? Then how did he quote John OBrien's message in his original post? Oh yeah... That's kinda obvious there now isnt it.... lol On 1/10/12 10:30 PM, "curriegrad2004" wrote: > Ken, you're getting old :P No, bd did start a new thread :P On Tue, Jan 10, > 2012 at 6:53 PM, Brian Foster wrote: > Huh? I could > have sworn I started another thread? > > On Jan 10, 2012 9:52 PM, "Ken Rice" > wrote: >> >> Hey db don?t hijack threads like that... > Start a new thread... This really >> screws with the thread tracking on all > the web archival stuff >> >> K >> >> >> On 1/10/12 8:10 PM, "Brian Foster" > wrote: >> >> I have some phones on my network that > are capable of dialing a SIP URI. It >> seems by default FreeSWITCH doesn't > allow this, as it just picks up >> everything before the @ and puts it as the > destination number. Is there a >> way to allow this? What would be needed in > the dialplan? >> >> On Jan 10, 2012 9:05 PM, "John O'Brien" > wrote: >> >> Hi, >> >> We have a reasonably recent > version of FreeSWITCH running. >> FreeSWITCH version: 1.0.head (git-10df279 > 2011-10-15 07-59-23 -0500) >> We are running on CentOS 6 >> >> We start > FreeSWITCH from a shell script. >> >> ulimit -s 240 >> freeswitch -ncwait -u > usrXXX -g grpXXX >> >> The issue is that things seem to go along nicely for a > day to two. >> Then without any apparent reason the freeswitch process > dies. >> Usually there is nothing happening when this occurs, system is > idle. >> >> There appear to be no process remnants we can use for > investigation. >> >> We would appreciate some suggestions on how we might > track the issue down. >> Is there some simple logging we can turn on > etc? >> >> Regards, >> >> John O'Brien >> > john at whitesmiths.com >> >> >> >> >> >> > _________________________________________________________________________>> > Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The > CudaTel Communication Server >> >> >> Official > FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> > http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> ________________________________ >> > _________________________________________________________________________>> > Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The > CudaTel Communication Server >> >> >> Official > FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> > http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> > _________________________________________________________________________>> > Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The > CudaTel Communication Server >> >> >> Official > FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> > http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > > > _________________________________________________________________________> > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel > Communication Server > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > http://wiki.freeswitch.org > > http://www.cluecon.com > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > _________________________________________________ > ________________________ Professional FreeSWITCH Consulting > Services: consulting at freeswitch.org http://www.freeswitchsolutions.com FreeSW > ITCH-powered IP PBX: The CudaTel Communication > Server Official FreeSWITCH > Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon. > com FreeSWITCH-users mailing > list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman > /listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > ions/freeswitch-users http://www.freeswitch.org From bdfoster at endigotech.com Wed Jan 11 09:28:20 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 11 Jan 2012 01:28:20 -0500 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: References: Message-ID: Ok...ok... I did my Sacrament of Penance, said my Hail Mary's... should be good to go now. Started new thread. On Wed, Jan 11, 2012 at 1:16 AM, Ken Rice wrote: > Really now? Then how did he quote John OBrien's message in his original > post? > > Oh yeah... That's kinda obvious there now isnt it.... lol > > > On 1/10/12 10:30 PM, "curriegrad2004" wrote: > > > Ken, you're getting old :P > > No, bd did start a new thread :P > > On Tue, Jan 10, > > 2012 at 6:53 PM, Brian Foster wrote: > > Huh? I could > > have sworn I started another thread? > > > > On Jan 10, 2012 9:52 PM, "Ken Rice" > > wrote: > >> > >> Hey db don?t hijack threads like that... > > Start a new thread... This really > >> screws with the thread tracking on all > > the web archival stuff > >> > >> K > >> > >> > >> On 1/10/12 8:10 PM, "Brian Foster" > > wrote: > >> > >> I have some phones on my network that > > are capable of dialing a SIP URI. It > >> seems by default FreeSWITCH doesn't > > allow this, as it just picks up > >> everything before the @ and puts it as the > > destination number. Is there a > >> way to allow this? What would be needed in > > the dialplan? > >> > >> On Jan 10, 2012 9:05 PM, "John O'Brien" > > wrote: > >> > >> Hi, > >> > >> We have a reasonably recent > > version of FreeSWITCH running. > >> FreeSWITCH version: 1.0.head (git-10df279 > > 2011-10-15 07-59-23 -0500) > >> We are running on CentOS 6 > >> > >> We start > > FreeSWITCH from a shell script. > >> > >> ulimit -s 240 > >> freeswitch -ncwait -u > > usrXXX -g grpXXX > >> > >> The issue is that things seem to go along nicely for a > > day to two. > >> Then without any apparent reason the freeswitch process > > dies. > >> Usually there is nothing happening when this occurs, system is > > idle. > >> > >> There appear to be no process remnants we can use for > > investigation. > >> > >> We would appreciate some suggestions on how we might > > track the issue down. > >> Is there some simple logging we can turn on > > etc? > >> > >> Regards, > >> > >> John O'Brien > >> > > john at whitesmiths.com > >> > >> > >> > >> > >> > >> > > > _________________________________________________________________________>> > > Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> > > http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The > > CudaTel Communication Server > >> > >> > >> Official > > FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> > > http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> > > FreeSWITCH-users at lists.freeswitch.org > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > > http://www.freeswitch.org > >> > >> > >> ________________________________ > >> > > > _________________________________________________________________________>> > > Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> > > http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The > > CudaTel Communication Server > >> > >> > >> Official > > FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> > > http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> > > FreeSWITCH-users at lists.freeswitch.org > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > > http://www.freeswitch.org > >> > >> > >> > > > _________________________________________________________________________>> > > Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> > > http://www.freeswitchsolutions.com > >> > >> FreeSWITCH-powered IP PBX: The > > CudaTel Communication Server > >> > >> > >> Official > > FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> > > http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> > > FreeSWITCH-users at lists.freeswitch.org > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > > http://www.freeswitch.org > >> > > > > > > > _________________________________________________________________________> > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel > > Communication Server > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > _________________________________________________ > > ________________________ > Professional FreeSWITCH Consulting > > Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > FreeSW > > ITCH-powered IP PBX: The CudaTel Communication > > Server > > > Official FreeSWITCH > > Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon. > > com > > FreeSWITCH-users mailing > > list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman > > /listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opt > > ions/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/3385bf23/attachment.html From gabe at gundy.org Wed Jan 11 11:06:42 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 11 Jan 2012 01:06:42 -0700 Subject: [Freeswitch-users] Dial by IP address In-Reply-To: References: Message-ID: On Tue, Jan 10, 2012 at 7:38 PM, Ken Rice wrote: > Hey db don?t hijack threads like that... Start a new thread... This really > screws with the thread tracking on all the web archival stuff Thanks Ken. I love to see someone cracking the list etiquette whip :) We could also use someone to police emails for lack of trimming. Sometimes emails can grow to be 5 or 6 times the screen height for a single reply. It takes too long to follow conversations and makes it so that people are less likely to help out or respond to your emails. Anyway, carry on people. Gabe P.S. We we'll leave top posting alone; I can see it's too late for that fight :) From miha at softnet.si Wed Jan 11 11:22:52 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 11 Jan 2012 09:22:52 +0100 Subject: [Freeswitch-users] PBX In-Reply-To: References: <4F0C3109.3090201@softnet.si> Message-ID: <4F0D46DC.9000606@softnet.si> Dear, just curious. Softswitch that we are having now due to performance we must have more than one PBX for users (to much load). So for every 1000 users we create new pbx. Is freeswitch also having this problems? Is it better that we create more PBX-s? Thanks! Regards, Miha On 1/11/2012 6:26 AM, amit nakum wrote: > dear, > > it is possible by using concept of Multiple domain.following link will > be helpful. > > 1)http://wiki.freeswitch.org/wiki/Multi-tenant > 2)http://wiki.freeswitch.org/wiki/Multiple_Companies > > Regards > Amit > > On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek > wrote: > > Hi, > > is it possible to have on freeswitch more than one PBX? If it is, > where > can I find this threat on wiki. > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/f86ca777/attachment-0001.html From odermann at googlemail.com Wed Jan 11 11:45:36 2012 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Jan 2012 09:45:36 +0100 Subject: [Freeswitch-users] voicemail_inject (Mod voicemail) problems In-Reply-To: References: Message-ID: hi anthony, we fixed the problem and yes, the mentioned ip is a valid domain. the error was because of a problem with the LUA User Directory (http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh#Lua_Script_2). with the given example the voicebox did not work properly (beside voicemail_inject everything else worked): XML_STRING = [[
]] we made some small changes to the script and it worked: XML_STRING = [[
]] we do not know, if there is a problem with the LUA example or if we do not understand how to use it. kind regards dennis 2012/1/9 Anthony Minessale : > is?192.168.2.2 a valid domain in your user directory? From odermann at googlemail.com Wed Jan 11 11:49:23 2012 From: odermann at googlemail.com (Dennis) Date: Wed, 11 Jan 2012 09:49:23 +0100 Subject: [Freeswitch-users] User Configuration: Question about F_REC and misc. In-Reply-To: References: Message-ID: is there nobody who could tell us, how to diable the "record_key"? at least in germany it is forbidden to record phone calls, without the other side accepting/knowing this. kind regards dennis From miha at softnet.si Wed Jan 11 13:43:55 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 11 Jan 2012 11:43:55 +0100 Subject: [Freeswitch-users] Multi-Tenant Message-ID: <4F0D67EB.2080505@softnet.si> Hi, I have done as is written on wiki, but my phones do not register. I guess that I have some problems with domains as I get this : 2012-01-11 11:31:15.430343 [WARNING] sofia_reg.c:2410 Can't find user [018108500 at xxx.xxx.xxx.xxx] You must define a domain called 'xx.xxx.xxx.xxx' in your directory and add a user with the id="018108500" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2012-01-11 11:31:15.430343 [WARNING] sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile 'internal' for [018108500 at xx.xxx.xxx.xxxi] from ip xx.xxx.xxx.xxx xx.xxx.xxx.xxx is an Ip of my server. In my directory I have set domain kabelnet1. I have created kabelnet1.xml and dir kabelvoip1 in conf/directory. I have also commend in sip_profiles/internal.xml what is written on wiki. My question. If my domain for FS server in freeswitch.test.org and new created directory kabelnet1. Should the phone be registering and domain kabelnet1.freeswitch.test.org? Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/ecae6224/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Wed Jan 11 14:51:17 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Wed, 11 Jan 2012 11:51:17 +0000 (GMT) Subject: [Freeswitch-users] NAT Confusion Message-ID: <1326282677.19722.YahooMailNeo@web29405.mail.ird.yahoo.com> Thanks Vitaly. Reckon I've got it sorted now.... >Make traces (tcpdump) at server and phone network, check all RTP streams >find where it get blocked (misdirected). most likely it is network related >problem. From gb10hkzo-freeswitch at yahoo.co.uk Wed Jan 11 14:57:59 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Wed, 11 Jan 2012 11:57:59 +0000 (GMT) Subject: [Freeswitch-users] Help needing translating Asterisk syntax to Freeswitch Message-ID: <1326283079.37785.YahooMailNeo@web29406.mail.ird.yahoo.com> Hello all, I'm having a little difficulty translating the following Asterisk sip.conf syntax to Freeswitch, hopefully someone can help. An alternative way of asking this question (which might be easier for some to answer) is .... has anyone successfully connected Freeswitch to the PCH INOC-DBA service ? register => 0000*1:password:user at sip-host/0000*1 This is my current attempt at it, but I'm getting registration failed messages on the freeswitch console : ? ? ? ? ? ? ? ? Thanks Bob From gb10hkzo-freeswitch at yahoo.co.uk Wed Jan 11 15:36:17 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Wed, 11 Jan 2012 12:36:17 +0000 (GMT) Subject: [Freeswitch-users] switch_core_sqldb.c:840 SQL ERR Message-ID: <1326285377.82678.YahooMailNeo@web29403.mail.ird.yahoo.com> Hi, I think it may be related to me updating to the latest git point (cfa926d7 ....add sub_host to the sip_registrations table to keep track of the original host). Haven't seen these messages before today, and don't really want to break anything more than it might already be, so could someone point me in the right direction as to what needs to be done ? 2012-01-11 12:30:59.466486 [ERR] switch_core_sqldb.c:840 SQL ERR: [select sip_registrations.sip_user, sip_registrations.sub_host, sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, sip_dialogs.sip_to_host, sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, sip_presence.open_closed,'','' from sip_registrations left join sip_dialogs on sip_dialogs.presence_id = sip_registrations.sip_user || '@' || sip_registrations.sub_host or (sip_dialogs.sip_from_user = sip_registrations.sip_user and sip_dialogs.sip_from_host = sip_registrations.sub_host) left join sip_presence on (sip_registrations.sip_user=sip_presence.sip_user and sip_registrations.orig_server_host=sip_presence.sip_server and sip_registrations.profile_name=sip_presence.profile_name) where sip_dialogs.presence_id='1000 at an.example.com' or (sip_registrations.sip_user='1000' and (sip_registrations.orig_server_host='an.example.com' or sip_registrations.sub_host='an.example.com' or sip_registrations.presence_hosts like '%an.example.com%'))] no such column: sip_presence.sip_server Thanks Bob From beppe.grillo at gmail.com Wed Jan 11 15:43:58 2012 From: beppe.grillo at gmail.com (Giuseppe Grillo) Date: Wed, 11 Jan 2012 13:43:58 +0100 Subject: [Freeswitch-users] Accept-contact in INVITE (RFC-3841) Message-ID: Hi, What should I configure to include the Accept-contact parameter? in INVITE message ? (ref. RFC-3841). Thanks, Giuseppe From Claudio.Cavalera at italtel.it Wed Jan 11 16:39:20 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 11 Jan 2012 14:39:20 +0100 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented Message-ID: Hello, I'm dealing with a segfault on a Centos 6.0 x64. I've tried to start freeswitch with -c and -core flags to get the output after the segfault (althouth I think that these flags do not change the default behaviour). Core files are generated and in /var/log/messages I see things like: ...kernel: freeswitch[22563]: segfault at 7f677c3ed935 ip 00007f677c3ed935 sp 00007f677c144b10 error 14 in libgnutls.so.26.14.12[7f677c21d000+1ff000] I think the problem it's related to mod_dingaling and libgnutls. However in attempt to see the output on the console where freeswitch got started I've tried to follow the steps documented here on the wiki: http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch but it seems crash-protection has been removed from configuration files and the source code (at least on my git of two months ago. Any hints? Kind Regards, Claudio 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. ----------------------------------------------------------------------------------------------------- From gcd at i.ph Wed Jan 11 16:43:48 2012 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 11 Jan 2012 21:43:48 +0800 Subject: [Freeswitch-users] gsmopen - samsung gt-c3303k Message-ID: hi to gsmopen developers, fyi, i get this log message repeatedly with this mobile phone Samsung Champ GT-C3303K: AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is not Motorola, please contact the gsmopen developers. Else, if your phone IS a Motorola, probably a long msg was incoming and ther first part was read and then deleted. The second part is now orphan. If you got this warning repeatedly, and you cannot correctly receive SMSs from this interface, please manually clean all messages (and the residual parts of them) from the cellphone/SIM. Continuing. tks, nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/9bdd69bb/attachment-0001.html From gmaruzz at gmail.com Wed Jan 11 16:59:30 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 11 Jan 2012 14:59:30 +0100 Subject: [Freeswitch-users] gsmopen - samsung gt-c3303k In-Reply-To: References: Message-ID: Hi Nandy, I'm the main developer of mod_gsmopen. Your phone model is not supported by mod_gsmopen at the moment. Mod_gsmopen development will continue starting end of February, focusing on Huawei dongle support (I'm now leaving for a long vacation). If you want your phone model to be supported in the future, in the mean time please fill a Jira issue with all the necessary information as stated in the wiki page for mod_gsmopen, and at end of February I'll look into it. -giovanni On Wed, Jan 11, 2012 at 2:43 PM, Nandy Dagondon wrote: > hi to gsmopen developers, > > fyi, i get this log message repeatedly with this mobile phone Samsung Champ > GT-C3303K: > > AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is not > Motorola, please contact the gsmopen developers. Else, if your phone IS a > Motorola, probably a long msg was incoming and ther first part was read and > then deleted. The second part is now orphan. If you got this warning > repeatedly, and you cannot correctly receive SMSs from this interface, > please manually clean all messages (and the residual parts of them) from the > cellphone/SIM. Continuing. > > tks, > nandy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gcd at i.ph Wed Jan 11 17:06:20 2012 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 11 Jan 2012 22:06:20 +0800 Subject: [Freeswitch-users] gsmopen - samsung gt-c3303k In-Reply-To: References: Message-ID: hi giovanni, ok. tks for the update. hv a nice vacation! -nandy On Wed, Jan 11, 2012 at 9:59 PM, Giovanni Maruzzelli wrote: > Hi Nandy, > > I'm the main developer of mod_gsmopen. > > Your phone model is not supported by mod_gsmopen at the moment. > > Mod_gsmopen development will continue starting end of February, > focusing on Huawei dongle support (I'm now leaving for a long > vacation). > > If you want your phone model to be supported in the future, in the > mean time please fill a Jira issue with all the necessary information > as stated in the wiki page for mod_gsmopen, and at end of February > I'll look into it. > > -giovanni > > On Wed, Jan 11, 2012 at 2:43 PM, Nandy Dagondon wrote: > > hi to gsmopen developers, > > > > fyi, i get this log message repeatedly with this mobile phone Samsung > Champ > > GT-C3303K: > > > > AT+MMGL="HEADER ONLY" does not get OK from the phone. If your phone is > not > > Motorola, please contact the gsmopen developers. Else, if your phone IS a > > Motorola, probably a long msg was incoming and ther first part was read > and > > then deleted. The second part is now orphan. If you got this warning > > repeatedly, and you cannot correctly receive SMSs from this interface, > > please manually clean all messages (and the residual parts of them) from > the > > cellphone/SIM. Continuing. > > > > tks, > > nandy > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/59263d18/attachment.html From dgarcia at anew.com.ve Wed Jan 11 17:12:29 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Wed, 11 Jan 2012 09:42:29 -0430 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> Message-ID: <4F0D98CD.2040204@anew.com.ve> Hi all, About a strategy to release and support. I work for a integrator. We have in our portafolio two big brand: Alcatel and Genesys. They give new minor releases and hotfixes almost every week in their products. Mayor releases are available in one year, sometimes more. About minor releases and hot-fixes, it depend on our local scenario. Not all our customer has issues than involves/require apply hot-fixes each week. We use this approach: 1. If there is an issue, we look in there are an aplicable hotfix. If not, we try to find a workaround or get a estimated date for a hotfix. 2. Each year we offer to our customer access to last binaries to upgrade their platform to the latest release Also, you have to "design" an integral political procedure for upgrades, hot-fixes and roll-back. Because, FS depend on OS, and other components like mediagateway, routers, etc. I have seen in our customer, when they apply a patch in the OS or their pbx, could affect other systems. On 1/10/2012 11:46 PM, Michael Collins wrote: > > Enterprise can't afford patch& upgrade daily. You guys need to > take a strategy to release and support. For instance, two release: > stable release will has a life span 3 years and be supported. Test > release with latest features. > > We've got some things in the works. Stay tuned! > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4734 - Release Date: 01/10/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/1310659e/attachment.html From anita.hall at simmortel.com Wed Jan 11 17:23:34 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Wed, 11 Jan 2012 19:53:34 +0530 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: <4F0D98CD.2040204@anew.com.ve> References: <4F0C8A89.8000607@gmail.com> <4E12C8B6-2CD2-4B9A-AB52-CE757046D001@gmail.com> <4F0D98CD.2040204@anew.com.ve> Message-ID: Some release policy along the lines of Ubuntu LTS will be great! Our client has been running FS for more than 2 years now and they do not have a "stay with the latest git" policy. That would be a pain anyway. They are currently doing around 80 E1 lines at 10 different locations. @MC, eagerly awaited :) regards, Anita On Wed, Jan 11, 2012 at 7:42 PM, Saugort Dario Garcia Tovar < dgarcia at anew.com.ve> wrote: > Hi all, > > About a strategy to release and support. > > I work for a integrator. We have in our portafolio two big brand: Alcatel > and Genesys. They give new minor releases and hotfixes almost every week in > their products. Mayor releases are available in one year, sometimes more. > > About minor releases and hot-fixes, it depend on our local scenario. Not > all our customer has issues than involves/require apply hot-fixes each > week. We use this approach: > 1. If there is an issue, we look in there are an aplicable hotfix. If not, > we try to find a workaround or get a estimated date for a hotfix. > 2. Each year we offer to our customer access to last binaries to upgrade > their platform to the latest release > > Also, you have to "design" an integral political procedure for upgrades, > hot-fixes and roll-back. Because, FS depend on OS, and other components > like mediagateway, routers, etc. I have seen in our customer, when they > apply a patch in the OS or their pbx, could affect other systems. > > > On 1/10/2012 11:46 PM, Michael Collins wrote: > > > Enterprise can't afford patch& upgrade daily. You guys need to take a >> strategy to release and support. For instance, two release: stable release >> will has a life span 3 years and be supported. Test release with latest >> features. >> > We've got some things in the works. Stay tuned! > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4734 - Release Date: 01/10/12 > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/560eed6a/attachment-0001.html From krice at freeswitch.org Wed Jan 11 17:44:47 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Jan 2012 08:44:47 -0600 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: Message-ID: I started to reply to this last night... There is not ?stay with the latest git policy? ... It has long been stated you need to test for your actual deployment make sure it works for you then deploy. On the ?are you running the latest git?? thing. The FreeSWITCH Dev Team is pretty small and pretty much everyone on the dev team has another ?day job?. so them asking can you duplicate the issue on the latest git is a way to reduce the amount of bug finding by asking the reporter to help the dev team. Even better is if you can help identify a previous working version in git and the commit at which it stopped working that?s even better (see git bisect for helping out on this) L On 1/11/12 8:23 AM, "Anita Hall" wrote: > Some release policy along the lines of Ubuntu LTS will be great! > > Our client has been running FS for more than 2 years now and they do not have > a "stay with the latest git" policy. That would be a pain anyway. They are > currently doing around 80 E1 lines at 10 different locations. > > @MC, eagerly awaited :) > > regards, > Anita > > > > On Wed, Jan 11, 2012 at 7:42 PM, Saugort Dario Garcia Tovar > wrote: >> >> Hi all, >> >> About a strategy to release and support. >> >> I work for a integrator. We have in our portafolio two big brand: Alcatel >> and Genesys. They give new minor releases and hotfixes almost every week in >> their products. Mayor releases are available in one year, sometimes more. >> >> About minor releases and hot-fixes, it depend on our local scenario. Not all >> our customer has issues than involves/require apply hot-fixes each week. We >> use this approach: >> 1. If there is an issue, we look in there are an aplicable hotfix. If not, >> we try to find a workaround or get a estimated date for a hotfix. >> 2. Each year we offer to our customer access to last binaries to upgrade >> their platform to the latest release >> >> Also, you have to "design" an integral political procedure for upgrades, >> hot-fixes and roll-back. Because, FS depend on OS, and other components like >> mediagateway, routers, etc. I have seen in our customer, when they apply a >> patch in the OS or their pbx, could affect other systems. >> >> >> On 1/10/2012 11:46 PM, Michael Collins wrote: >>> >>> >>> >>>> >>>> >>>> Enterprise can't afford patch& upgrade daily. You guys need to take a >>>> strategy to release and support. For instance, two release: stable release >>>> will has a life span 3 years and be supported. Test release with latest >>>> features. >>>> >>> We've got some things in the works. Stay tuned! >>> -MC >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> No virus found in this message. >>> Checked by AVG - www.avg.com >>> Version: 2012.0.1901 / Virus Database: 2109/4734 - Release Date: 01/10/12 >>> >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/8767e777/attachment.html From daggelinckxmichel at gmail.com Wed Jan 11 17:45:43 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Wed, 11 Jan 2012 15:45:43 +0100 Subject: [Freeswitch-users] PBX In-Reply-To: <4F0D46DC.9000606@softnet.si> References: <4F0C3109.3090201@softnet.si> <4F0D46DC.9000606@softnet.si> Message-ID: a possible sollution for your load problem might be to use the freeswitch based whistle platform offered by 2600hz.org this allows you to transparantly add capacity to your setup and load balance your multi-tenant pbx setup. Michel On Wed, Jan 11, 2012 at 9:22 AM, Miha Zoubek wrote: > Dear, > > just curious. Softswitch that we are having now due to performance we must > have more than one PBX for users (to much load). > So for every 1000 users we create new pbx. > > Is freeswitch also having this problems? Is it better that we create more > PBX-s? > > > Thanks! > > Regards, > Miha > > > On 1/11/2012 6:26 AM, amit nakum wrote: > > dear, > > it is possible by using concept of Multiple domain.following link will be > helpful. > > 1)http://wiki.freeswitch.org/wiki/Multi-tenant > 2)http://wiki.freeswitch.org/wiki/Multiple_Companies > > Regards > Amit > > On Tue, Jan 10, 2012 at 6:07 PM, Miha Zoubek wrote: > >> Hi, >> >> is it possible to have on freeswitch more than one PBX? If it is, where >> can I find this threat on wiki. >> >> Regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/aa56ec5a/attachment.html From miha at softnet.si Wed Jan 11 17:52:34 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 11 Jan 2012 15:52:34 +0100 Subject: [Freeswitch-users] Multi domains issue Message-ID: <4F0DA232.2080703@softnet.si> Hi, plase help me out why I am getting: 2012-01-11 15:42:56.746075 [WARNING] sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile 'internal' for [0123456 at fs1.test.si@fs1.xxxx.si] from ip xxx.xxx.xxx.xxx I do not know how to fix this or what I am doing wrong. $${domian} I chaged in fs1.test.si http://pastebin.freeswitch.org/18117 In user auth for phone I put 0123456 at fs1.test.si, Thanks! Br, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/f00dbb50/attachment-0001.html From daggelinckxmichel at gmail.com Wed Jan 11 18:00:11 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Wed, 11 Jan 2012 16:00:11 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: <4F0DA232.2080703@softnet.si> References: <4F0DA232.2080703@softnet.si> Message-ID: 0123456 at fs1.test.si@fs1.xxxx.si <--- looks weird to me On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek wrote: > Hi, > > plase help me out why I am getting: > > 2012-01-11 15:42:56.746075 [WARNING] sofia_reg.c:1359 SIP auth failure > (REGISTER) on sofia profile 'internal' for [ > 0123456 at fs1.test.si@fs1.xxxx.si] from ip xxx.xxx.xxx.xxx > > I do not know how to fix this or what I am doing wrong. > $${domian} I chaged in fs1.test.si > > http://pastebin.freeswitch.org/18117 > > In user auth for phone I put 0123456 at fs1.test.si, > > > Thanks! > > Br, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/37c64d4a/attachment.html From avi at avimarcus.net Wed Jan 11 18:05:15 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 11 Jan 2012 17:05:15 +0200 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: libgnutls is a dependency for an older version of mod_dingaling... the newer version uses openssh afaik. It's probably a good idea to recompile from latest. -Avi On Wed, Jan 11, 2012 at 3:39 PM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > Hello, > I'm dealing with a segfault on a Centos 6.0 x64. > I've tried to start freeswitch with -c and -core flags to get the output > after the segfault (althouth I think that these flags do not change the > default behaviour). > > Core files are generated and in /var/log/messages I see things like: > ...kernel: freeswitch[22563]: segfault at 7f677c3ed935 ip > 00007f677c3ed935 sp 00007f677c144b10 error 14 in > libgnutls.so.26.14.12[7f677c21d000+1ff000] > I think the problem it's related to mod_dingaling and libgnutls. > > However in attempt to see the output on the console where freeswitch got > started I've tried to follow the steps documented here on the wiki: > http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch > but it seems crash-protection has been removed from configuration files > and the source code (at least on my git of two months ago. > Any hints? > > Kind Regards, > Claudio > > > 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. > > ----------------------------------------------------------------------------------------------------- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/0673ca95/attachment.html From curriegrad2004 at gmail.com Wed Jan 11 18:26:03 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 11 Jan 2012 07:26:03 -0800 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Unfortunately libgnutls ranges from being relatively stable to just unpredictable. As Avi did mention Tony made a few changes back in mid November and switched dingaling to use OpenSSL instead of gnutls. If > > libgnutls is a dependency for an older version of mod_dingaling... the > newer version uses openssh afaik. It's probably a good idea to recompile > from latest. > > -Avi > > > On Wed, Jan 11, 2012 at 3:39 PM, Cavalera Claudio Luigi < > Claudio.Cavalera at italtel.it> wrote: > >> Hello, >> I'm dealing with a segfault on a Centos 6.0 x64. >> I've tried to start freeswitch with -c and -core flags to get the output >> after the segfault (althouth I think that these flags do not change the >> default behaviour). >> >> Core files are generated and in /var/log/messages I see things like: >> ...kernel: freeswitch[22563]: segfault at 7f677c3ed935 ip >> 00007f677c3ed935 sp 00007f677c144b10 error 14 in >> libgnutls.so.26.14.12[7f677c21d000+1ff000] >> I think the problem it's related to mod_dingaling and libgnutls. >> >> However in attempt to see the output on the console where freeswitch got >> started I've tried to follow the steps documented here on the wiki: >> http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch >> but it seems crash-protection has been removed from configuration files >> and the source code (at least on my git of two months ago. >> Any hints? >> >> Kind Regards, >> Claudio >> >> >> 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. >> >> ----------------------------------------------------------------------------------------------------- >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/bf608302/attachment-0001.html From Claudio.Cavalera at italtel.it Wed Jan 11 20:24:47 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 11 Jan 2012 18:24:47 +0100 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Okay, I'm glad to know this has been fixed and I will update to latest git as soon as possible. However if someone confirm that crash-protection does not exist anymore i will remove that part on the wiki; otherwise please point me on the new feature to get the output on console after a generic segfault. Thanks, Claudio From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: Wednesday, January 11, 2012 4:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] crash-protection documented on the wiki but no more implemented Unfortunately libgnutls ranges from being relatively stable to just unpredictable. As Avi did mention Tony made a few changes back in mid November and switched dingaling to use OpenSSL instead of gnutls. If libgnutls is a dependency for an older version of mod_dingaling... the newer version uses openssh afaik. It's probably a good idea to recompile from latest. -Avi 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. ----------------------------------------------------------------------------------------------------- From msc at freeswitch.org Wed Jan 11 20:38:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jan 2012 09:38:26 -0800 Subject: [Freeswitch-users] valet park hanging up on 1 side In-Reply-To: <1326025186.19988.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1326025186.19988.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: You can set the action to be taken on both legs. Note the syntax for bda: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action Something like this: Tinker with that syntax and see if you can make the other leg also get x-fered instead of disconnected. -MC On Sun, Jan 8, 2012 at 4:19 AM, Rodney wrote: > i am trying to allow a "phone roulette" situation where two callers from > the ivr can be randomly be joined together. I need the following options > > 2 for next caller > 0 go back to ivr menu > > the matchup seems to work but once in a bridge the caller who press 2 > moves on but the other caller gets hung up on from the whole system and has > to call back. is there a better way to match callers in this scenario > without the hangup. here is the extension: > > > > > > > > > data="moderator,0,exec:transfer,399 XML default"/> > data="moderator,2,exec:transfer,533 XML default"/> > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/989fd291/attachment.html From anthony.minessale at gmail.com Wed Jan 11 20:38:53 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Jan 2012 11:38:53 -0600 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: it indeed does not exist anymore. I removed it years ago. On Wed, Jan 11, 2012 at 11:24 AM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > Okay, > I'm glad to know this has been fixed and I will update to latest git as > soon as possible. > However if someone confirm that crash-protection does not exist anymore i > will remove that part on the wiki; otherwise please point me on the new > feature to get the output on console after a generic segfault. > Thanks, > Claudio > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 > Sent: Wednesday, January 11, 2012 4:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] crash-protection documented on the wiki > but no more implemented > > Unfortunately libgnutls ranges from being relatively stable to just > unpredictable. As Avi did mention Tony made a few changes back in mid > November and switched dingaling to use OpenSSL instead of gnutls. > If > libgnutls is a dependency for an older version of mod_dingaling... the > newer version uses openssh afaik. It's probably a good idea to recompile > from latest. > > > -Avi > > > 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. > > ----------------------------------------------------------------------------------------------------- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/df95d84b/attachment.html From leonardo.bidinoto at voicetechnology.com.br Wed Jan 11 21:05:37 2012 From: leonardo.bidinoto at voicetechnology.com.br (Leonardo P. Bidinoto) Date: Wed, 11 Jan 2012 16:05:37 -0200 Subject: [Freeswitch-users] Provider Configuration: Vono Message-ID: Hi All. I was having a problem configuring a vono provider into my FreeSWITCH. Now its working. If someone had the same problem as me, here it's configuration. In external.xml file, insert the below line into settings: Somehow, Vono Provider wont accept the default string that FreeSWITCH send. ex: FreeSWITCH-mod_sofia/1.0.head-git-db5f504 2011-07-17 17-00-38 -0400. and then, the vono.xml file: Hope it helps someone. -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/190a2027/attachment-0001.html From krice at freeswitch.org Wed Jan 11 21:09:56 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Jan 2012 12:09:56 -0600 Subject: [Freeswitch-users] Provider Configuration: Vono In-Reply-To: Message-ID: I?ve seen this repeatedly look at your sofia profile and shorten the UserAgent string... Why these carriers refuse to accept a perfectly valid (or valid and random) useragent string I will never understand K On 1/11/12 12:05 PM, "Leonardo P. Bidinoto" wrote: > Hi All. > > I was having a problem configuring a vono provider into my FreeSWITCH. > Now its working. > If someone had the same problem as me, here it's configuration. > > In external.xml file, insert the below line into settings: > > > Somehow, Vono Provider wont accept the default string that FreeSWITCH send. > ex: FreeSWITCH-mod_sofia/1.0.head-git-db5f504 2011-07-17 17-00-38 -0400. > > and then, the vono.xml file: > > > ?? > ???? > ???? > ???? > ???? > ???? > ???? > ???? > ???? > ???? > ???? > ?? > > > Hope it helps someone. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/8fba5ab3/attachment.html From anthony.minessale at gmail.com Wed Jan 11 21:44:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Jan 2012 12:44:55 -0600 Subject: [Freeswitch-users] switch_core_sqldb.c:840 SQL ERR In-Reply-To: <1326285377.82678.YahooMailNeo@web29403.mail.ird.yahoo.com> References: <1326285377.82678.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: fixed in tree On Wed, Jan 11, 2012 at 6:36 AM, Bob Smith wrote: > Hi, > > I think it may be related to me updating to the latest git point (cfa926d7 > ....add sub_host to the sip_registrations table to keep track of the > original host). > > Haven't seen these messages before today, and don't really want to break > anything more than it might already be, so could someone point me in the > right direction as to what needs to be done ? > > 2012-01-11 12:30:59.466486 [ERR] switch_core_sqldb.c:840 SQL ERR: [select > sip_registrations.sip_user, sip_registrations.sub_host, > sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, > sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, > sip_dialogs.sip_to_host, > sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, > sip_presence.open_closed,'','' from sip_registrations left join sip_dialogs > on sip_dialogs.presence_id = sip_registrations.sip_user || '@' || > sip_registrations.sub_host or (sip_dialogs.sip_from_user = > sip_registrations.sip_user and sip_dialogs.sip_from_host = > sip_registrations.sub_host) left join sip_presence on > (sip_registrations.sip_user=sip_presence.sip_user and > sip_registrations.orig_server_host=sip_presence.sip_server and > sip_registrations.profile_name=sip_presence.profile_name) where > sip_dialogs.presence_id='1000 at an.example.com' or > (sip_registrations.sip_user='1000' and > (sip_registrations.orig_server_host='an.example.com' or > sip_registrations.sub_host='an.example.com' or > sip_registrations.presence_hosts like '%an.example.com%'))] no such > column: sip_presence.sip_server > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/3c7f0c75/attachment.html From justlikeef at gmail.com Wed Jan 11 22:11:23 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Wed, 11 Jan 2012 14:11:23 -0500 Subject: [Freeswitch-users] switch_core_sqldb.c:840 SQL ERR In-Reply-To: References: <1326285377.82678.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: <201201111411.24135.justlikeef@gmail.com> Anthony - Does this mean that the presence issue is likely resolved? Thanks, Rob On Wednesday 11 January 2012 13:44:55 Anthony Minessale wrote: > fixed in tree > > On Wed, Jan 11, 2012 at 6:36 AM, Bob Smith > wrote: > > > Hi, > > > > I think it may be related to me updating to the latest git point (cfa926d7 > > ....add sub_host to the sip_registrations table to keep track of the > > original host). > > > > Haven't seen these messages before today, and don't really want to break > > anything more than it might already be, so could someone point me in the > > right direction as to what needs to be done ? > > > > 2012-01-11 12:30:59.466486 [ERR] switch_core_sqldb.c:840 SQL ERR: [select > > sip_registrations.sip_user, sip_registrations.sub_host, > > sip_registrations.status, sip_registrations.rpid, '', sip_dialogs.uuid, > > sip_dialogs.state, sip_dialogs.direction, sip_dialogs.sip_to_user, > > sip_dialogs.sip_to_host, > > sip_presence.status,sip_presence.rpid,sip_dialogs.presence_id, > > sip_presence.open_closed,'','' from sip_registrations left join sip_dialogs > > on sip_dialogs.presence_id = sip_registrations.sip_user || '@' || > > sip_registrations.sub_host or (sip_dialogs.sip_from_user = > > sip_registrations.sip_user and sip_dialogs.sip_from_host = > > sip_registrations.sub_host) left join sip_presence on > > (sip_registrations.sip_user=sip_presence.sip_user and > > sip_registrations.orig_server_host=sip_presence.sip_server and > > sip_registrations.profile_name=sip_presence.profile_name) where > > sip_dialogs.presence_id='1000 at an.example.com' or > > (sip_registrations.sip_user='1000' and > > (sip_registrations.orig_server_host='an.example.com' or > > sip_registrations.sub_host='an.example.com' or > > sip_registrations.presence_hosts like '%an.example.com%'))] no such > > column: sip_presence.sip_server > > > > Thanks > > > > Bob > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/ffd29965/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 01:58:33 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Wed, 11 Jan 2012 22:58:33 +0000 (GMT) Subject: [Freeswitch-users] switch_core_sqldb.c:840 SQL ERR Message-ID: <1326322713.90557.YahooMailNeo@web29406.mail.ird.yahoo.com> Awsome. Thanks ! >? fixed in tree From msc at freeswitch.org Thu Jan 12 02:42:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jan 2012 15:42:51 -0800 Subject: [Freeswitch-users] Help needing translating Asterisk syntax to Freeswitch In-Reply-To: <1326283079.37785.YahooMailNeo@web29406.mail.ird.yahoo.com> References: <1326283079.37785.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: I'm not familiar with this service, however what I would do is get a sip trace from it working on your Asterisk box and compare that with the sip trace from the freeswitch box. I'll bet that will help you narrow it down. -MC On Wed, Jan 11, 2012 at 3:57 AM, Bob Smith wrote: > Hello all, > > I'm having a little difficulty translating the following Asterisk sip.conf > syntax to Freeswitch, hopefully someone can help. > > An alternative way of asking this question (which might be easier for some > to answer) is .... has anyone successfully connected Freeswitch to the PCH > INOC-DBA service ? > > register => 0000*1:password:user at sip-host/0000*1 > > This is my current attempt at it, but I'm getting registration failed > messages on the freeswitch console : > > > > > > > > > > > > > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/e6e38601/attachment.html From ayobami at programmer.net Thu Jan 12 02:45:28 2012 From: ayobami at programmer.net (ayobami) Date: Wed, 11 Jan 2012 15:45:28 -0800 (PST) Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call In-Reply-To: References: <1326195418755-7171721.post@n2.nabble.com> Message-ID: <1326325528461-7178132.post@n2.nabble.com> thanks for your time, but I search the wiki page, I couldnt get the exact code or setting or where I will tell FS to include the parties making calls IP addresses in the Events being sent -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addresses-of-The-Parties-Involved-in-a-call-tp7171721p7178132.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ayobami at programmer.net Thu Jan 12 02:53:37 2012 From: ayobami at programmer.net (ayobami) Date: Wed, 11 Jan 2012 15:53:37 -0800 (PST) Subject: [Freeswitch-users] How do I make a video call from Fscli Message-ID: <1326326017312-7178147.post@n2.nabble.com> How do I initiate a video call from command line in FS? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-do-I-make-a-video-call-from-Fscli-tp7178147p7178147.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu Jan 12 03:02:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jan 2012 16:02:52 -0800 Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call In-Reply-To: <1326325528461-7178132.post@n2.nabble.com> References: <1326195418755-7171721.post@n2.nabble.com> <1326325528461-7178132.post@n2.nabble.com> Message-ID: Best thing to do when you are trying to learn about channel variables is to go to the fs_cli and then do "uuid_dump " and see all the chan vars that show up. Pick out the ones with the information you need and go from there. -MC On Wed, Jan 11, 2012 at 3:45 PM, ayobami wrote: > thanks for your time, but I search the wiki page, I couldnt get the exact > code or setting or where I will tell FS to include the parties making calls > IP addresses in the Events being sent > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addresses-of-The-Parties-Involved-in-a-call-tp7171721p7178132.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/87ac18b9/attachment.html From anthony.minessale at gmail.com Thu Jan 12 04:00:19 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Jan 2012 19:00:19 -0600 Subject: [Freeswitch-users] audo sync issues with record_session to mp3 In-Reply-To: <4F0C4F2F.3050901@earthspike.net> References: <6AEC73649FA6431CA9EF84A54F378594@ws4> <4F0C4F2F.3050901@earthspike.net> Message-ID: I really wish things like this would go to jira it's very hard to track things from the mailing list...... commit a365fb636ad9e2f4bb5dee43eddc305560699114 Author: Anthony Minessale Date: Wed Jan 11 17:49:35 2012 -0600 mailing list 36bc584d980ce80fe6a6f6e7d7383db9.squirrel at my.tomp.co.uk[Freeswitch-users] audo sync issues with record_session to mp3 I redid the stream recording with timestamps and headers to try to keep it more synced On Tue, Jan 10, 2012 at 8:46 AM, John wrote: > I've just enabled session recording using MP3 encoding and have the same > symptoms. In my case it is between a BRI ISDN connection and a SIP > phone, both running with G.711. I haven't tried recording between > extensions which would be a purer test, nor have I attempted using WAV. > Are there any suggestions for how to fix this? Should I register a bug > on Jira? > > John > > > On 02/01/12 15:01, Frank @ Impact wrote: > > We have the same problem. We are running git from 12/30/11. our aleg > is a > > sip channel coming to FS and the bleg is a sip channels leaving FS. > > > > I noticed this problem really when we started using mp3 instead of > > wav. With wav, it really was not noticeable for us in a 10-15minute > call. > > But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is > so > > far out of sync, it sounds like 2 different calls. > > > > The relevant dialplan is > > > > > > > >
> > > > > > > > > > > > > > > > > > > data="/mnt/rd/recordfile.mp3"/> > > > data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> > > > > > > > > > >
> > > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Gabriel > > Gunderson > > Sent: Thursday, October 20, 2011 3:07 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] audo sync issues with record_session to > mp3 > > > > On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott wrote: > >> Longer calls, after about 10 minutes start to introduce sync issues > >> between the A-leg and the B-leg. > >> > >> We are running record_session on the A-leg, and it seems to get ahead of > >> the B-leg. > >> > >> For example the caller on the A-leg will be heard to answer a question > >> whilst the person on the B-leg is asking it. > > What's on the other end of each leg? That might help us figure this out. > > > > Gabe > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/51c3bc5b/attachment-0001.html From Duane.Gilbert at patlive.com Thu Jan 12 04:24:06 2012 From: Duane.Gilbert at patlive.com (Duane Gilbert) Date: Wed, 11 Jan 2012 20:24:06 -0500 Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call In-Reply-To: References: <1326195418755-7171721.post@n2.nabble.com><1326325528461-7178132.post@n2.nabble.com> Message-ID: This is how I did the talquin stuff...it is bad ass! - Duane From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, January 11, 2012 7:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call Best thing to do when you are trying to learn about channel variables is to go to the fs_cli and then do "uuid_dump " and see all the chan vars that show up. Pick out the ones with the information you need and go from there. -MC On Wed, Jan 11, 2012 at 3:45 PM, ayobami wrote: thanks for your time, but I search the wiki page, I couldnt get the exact code or setting or where I will tell FS to include the parties making calls IP addresses in the Events being sent -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addres ses-of-The-Parties-Involved-in-a-call-tp7171721p7178132.html Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/6068a84d/attachment.html From gcd at i.ph Thu Jan 12 05:25:56 2012 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 12 Jan 2012 10:25:56 +0800 Subject: [Freeswitch-users] audo sync issues with record_session to mp3 In-Reply-To: References: <6AEC73649FA6431CA9EF84A54F378594@ws4> <4F0C4F2F.3050901@earthspike.net> Message-ID: i did this before by recording in WAV then converted them to OGG - all WAV files of the day - after midnight where the usage is low. On Thu, Jan 12, 2012 at 9:00 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I really wish things like this would go to jira it's very hard to track > things from the mailing list...... > > > commit a365fb636ad9e2f4bb5dee43eddc305560699114 > Author: Anthony Minessale > Date: Wed Jan 11 17:49:35 2012 -0600 > > mailing list 36bc584d980ce80fe6a6f6e7d7383db9.squirrel at my.tomp.co.uk[Freeswitch-users] audo sync issues with record_session to mp3 I redid the > stream recording with timestamps and headers to try to keep it more synced > > > > On Tue, Jan 10, 2012 at 8:46 AM, John wrote: > >> I've just enabled session recording using MP3 encoding and have the same >> symptoms. In my case it is between a BRI ISDN connection and a SIP >> phone, both running with G.711. I haven't tried recording between >> extensions which would be a purer test, nor have I attempted using WAV. >> Are there any suggestions for how to fix this? Should I register a bug >> on Jira? >> >> John >> >> >> On 02/01/12 15:01, Frank @ Impact wrote: >> > We have the same problem. We are running git from 12/30/11. our aleg >> is a >> > sip channel coming to FS and the bleg is a sip channels leaving FS. >> > >> > I noticed this problem really when we started using mp3 instead of >> > wav. With wav, it really was not noticeable for us in a 10-15minute >> call. >> > But with mp3, we notice it after just 2-3 minutes. By 10 minutes, it is >> so >> > far out of sync, it sounds like 2 different calls. >> > >> > The relevant dialplan is >> > >> > >> > >> >
>> > >> > >> > >> > >> > >> > >> > >> > >> > > > data="/mnt/rd/recordfile.mp3"/> >> > > > data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> >> > >> > >> > >> > >> >
>> > >> > >> > >> > >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Gabriel >> > Gunderson >> > Sent: Thursday, October 20, 2011 3:07 AM >> > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] audo sync issues with record_session to >> mp3 >> > >> > On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott wrote: >> >> Longer calls, after about 10 minutes start to introduce sync issues >> >> between the A-leg and the B-leg. >> >> >> >> We are running record_session on the A-leg, and it seems to get ahead >> of >> >> the B-leg. >> >> >> >> For example the caller on the A-leg will be heard to answer a question >> >> whilst the person on the B-leg is asking it. >> > What's on the other end of each leg? That might help us figure this out. >> > >> > Gabe >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/f89b3b8b/attachment-0001.html From vetali100 at gmail.com Thu Jan 12 06:33:57 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 11 Jan 2012 19:33:57 -0800 Subject: [Freeswitch-users] How do I get the IP Addresses of The Parties Involved in a call In-Reply-To: <1326195418755-7171721.post@n2.nabble.com> References: <1326195418755-7171721.post@n2.nabble.com> Message-ID: Try this variable: ${network_addr} 2012/1/10 ayobami > I need to get the physical IP addresses of two parties involved in a call, > I > have tried to monitor events being > generated during the call life cycle, from both parties involved in the > call, its only returning the physical IP address of the Freeswitch box, is > their any settings that I need to set so that appropriate IP addresses will > be returned, thanks > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/How-do-I-get-the-IP-Addresses-of-The-Parties-Involved-in-a-call-tp7171721p7171721.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/cb382e36/attachment.html From yehavi.bourvine at gmail.com Thu Jan 12 08:28:04 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Jan 2012 07:28:04 +0200 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: Hello Gabriel, The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX connected aside it with about 200 extensions (we do not put more extensions due to potential problems it might create on the Nortel). We started with Asterisk and quite quickly understood that it lacks some functionality that an organization like us needs, thus we moved to Freeswitch, and we are happy with this decision. We also have AudioCodes ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. We have to do some move soon as our Nortels are almost 20 years old. We are now in the process of estimating how much an open source solution would cost us (needs more man power which we have to estimate the amount of) so we can compare it to a commecrical solution offerings. If you need more information don't hesitate to ask. Regards, __Yehavi: 2012/1/10 Gabriel Kuri > Does anyone know of any Universities running FreeSWITCH, or other open > source VoIP for that matter, for local call routing between handsets? > > We're looking at replacing our old Avaya system or upgrade it, and the > forklift upgrade from Avaya is ridiculously expensive (no surprise). > > We'd like to replace our Avaya system with a combination of OpenSIPS > and FreeSWITCH and some Cisco routers for external PSTN access, but > it's going to be a tough sell to our CIO, unless we can show someone > else has done it already. > > Any pointers to other Universities would be great. > > Cheers, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/2d30d463/attachment.html From krice at freeswitch.org Thu Jan 12 08:39:17 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 11 Jan 2012 23:39:17 -0600 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: Message-ID: Yahavi, This is great to hear about. Have you buy any chance written a white paper on your installation? K On 1/11/12 11:28 PM, "Yehavi Bourvine" wrote: > Hello Gabriel, > ? > ? The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving around > 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX connected > aside it with about 200 extensions (we do not put more extensions due to > potential problems it might create on the Nortel). > ? > ? We started with Asterisk and quite quickly understood that it lacks some > functionality that an organization like us needs, thus we moved to Freeswitch, > and we are happy with this decision. We also have AudioCodes ATAs for analog > phones and AudioCodes PRI gateway. Our "smartphones" are mainly Polycoms for > secretaries and AudioCodes HD320 for "regular" users. > ? > We have to do some move soon as our Nortels are almost 20 years old. We are > now in the process of estimating how much an open source solution would cost > us (needs more man power which we have to estimate the amount of) so we can > compare it to a commecrical solution offerings. > ? > If you need more information don't hesitate to ask. > ? > ?????????????????????? Regards, __Yehavi: > > 2012/1/10 Gabriel Kuri >> Does anyone know of any Universities running FreeSWITCH, or other open >> source VoIP for that matter, for local call routing between handsets? >> >> We're looking at replacing our old Avaya system or upgrade it, and the >> forklift upgrade from Avaya is ridiculously expensive (no surprise). >> >> We'd like to replace our Avaya system with a combination of OpenSIPS >> and FreeSWITCH and some Cisco routers for external PSTN access, but >> it's going to be a tough sell to our CIO, unless we can show someone >> else has done it already. >> >> Any pointers to other Universities would be great. >> >> Cheers, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120111/b4dd8478/attachment.html From yehavi.bourvine at gmail.com Thu Jan 12 09:02:31 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Jan 2012 08:02:31 +0200 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: No, sorry - Hadn't the time yet to write something... __Yehavi: 2012/1/12 Ken Rice > Yahavi, This is great to hear about. Have you buy any chance written a > white paper on your installation? > > K > > > > On 1/11/12 11:28 PM, "Yehavi Bourvine" wrote: > > Hello Gabriel, > > The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving > around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX > connected aside it with about 200 extensions (we do not put more extensions > due to potential problems it might create on the Nortel). > > We started with Asterisk and quite quickly understood that it lacks some > functionality that an organization like us needs, thus we moved to > Freeswitch, and we are happy with this decision. We also have AudioCodes > ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are > mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. > > We have to do some move soon as our Nortels are almost 20 years old. We > are now in the process of estimating how much an open source solution would > cost us (needs more man power which we have to estimate the amount of) so > we can compare it to a commecrical solution offerings. > > If you need more information don't hesitate to ask. > > Regards, __Yehavi: > > 2012/1/10 Gabriel Kuri > > Does anyone know of any Universities running FreeSWITCH, or other open > source VoIP for that matter, for local call routing between handsets? > > We're looking at replacing our old Avaya system or upgrade it, and the > forklift upgrade from Avaya is ridiculously expensive (no surprise). > > We'd like to replace our Avaya system with a combination of OpenSIPS > and FreeSWITCH and some Cisco routers for external PSTN access, but > it's going to be a tough sell to our CIO, unless we can show someone > else has done it already. > > Any pointers to other Universities would be great. > > Cheers, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/db02dabe/attachment-0001.html From anton.jugatsu at gmail.com Thu Jan 12 08:53:17 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 12 Jan 2012 09:53:17 +0400 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: Which version of freeswitch do you use? 2012/1/12 Yehavi Bourvine > Hello Gabriel, > > The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving > around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX > connected aside it with about 200 extensions (we do not put more extensions > due to potential problems it might create on the Nortel). > > We started with Asterisk and quite quickly understood that it lacks some > functionality that an organization like us needs, thus we moved to > Freeswitch, and we are happy with this decision. We also have AudioCodes > ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are > mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. > > We have to do some move soon as our Nortels are almost 20 years old. We > are now in the process of estimating how much an open source solution would > cost us (needs more man power which we have to estimate the amount of) so > we can compare it to a commecrical solution offerings. > > If you need more information don't hesitate to ask. > > Regards, __Yehavi: > > 2012/1/10 Gabriel Kuri > >> Does anyone know of any Universities running FreeSWITCH, or other open >> source VoIP for that matter, for local call routing between handsets? >> >> We're looking at replacing our old Avaya system or upgrade it, and the >> forklift upgrade from Avaya is ridiculously expensive (no surprise). >> >> We'd like to replace our Avaya system with a combination of OpenSIPS >> and FreeSWITCH and some Cisco routers for external PSTN access, but >> it's going to be a tough sell to our CIO, unless we can show someone >> else has done it already. >> >> Any pointers to other Universities would be great. >> >> Cheers, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/fc0b5b9d/attachment.html From yehavi.bourvine at gmail.com Thu Jan 12 10:01:12 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Jan 2012 09:01:12 +0200 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: I am using a version from mid-October due to the BLF's problem. When it is fixed I will move to the latest GIT (I am doing upgrade to the latest GIT once every 2-4 weeks). __Yehavi: 2012/1/12 Anton Kvashenkin > Which version of freeswitch do you use? > > > 2012/1/12 Yehavi Bourvine > >> Hello Gabriel, >> >> The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving >> around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX >> connected aside it with about 200 extensions (we do not put more extensions >> due to potential problems it might create on the Nortel). >> >> We started with Asterisk and quite quickly understood that it lacks >> some functionality that an organization like us needs, thus we moved to >> Freeswitch, and we are happy with this decision. We also have AudioCodes >> ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are >> mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. >> >> We have to do some move soon as our Nortels are almost 20 years old. We >> are now in the process of estimating how much an open source solution would >> cost us (needs more man power which we have to estimate the amount of) so >> we can compare it to a commecrical solution offerings. >> >> If you need more information don't hesitate to ask. >> >> Regards, __Yehavi: >> >> 2012/1/10 Gabriel Kuri >> >>> Does anyone know of any Universities running FreeSWITCH, or other open >>> source VoIP for that matter, for local call routing between handsets? >>> >>> We're looking at replacing our old Avaya system or upgrade it, and the >>> forklift upgrade from Avaya is ridiculously expensive (no surprise). >>> >>> We'd like to replace our Avaya system with a combination of OpenSIPS >>> and FreeSWITCH and some Cisco routers for external PSTN access, but >>> it's going to be a tough sell to our CIO, unless we can show someone >>> else has done it already. >>> >>> Any pointers to other Universities would be great. >>> >>> Cheers, >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/8c5356cb/attachment.html From miha at softnet.si Thu Jan 12 10:16:42 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 12 Jan 2012 08:16:42 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: References: <4F0DA232.2080703@softnet.si> Message-ID: <4F0E88DA.8050103@softnet.si> Hi, i know that looks weird:) First how the phone should register on multi domains freeswitch? If you crate usr 1000 on domain domain.si (on freeswitch), how the user is registering on FS (domain of FS for exp is test.com)? Thanks! miha On 1/11/2012 4:00 PM, Michel Daggelinckx wrote: > 0123456 at fs1.test.si@fs1.xxxx.si <--- looks weird > to me > > On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek > wrote: > > Hi, > > plase help me out why I am getting: > > 2012-01-11 15 :42:56.746075 [WARNING] > sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile > 'internal' for [0123456 at fs1.test.si@fs1.xxxx.si > ] from ip xxx.xxx.xxx.xxx > > I do not know how to fix this or what I am doing wrong. > $${domian} I chaged in fs1.test.si > > http://pastebin.freeswitch.org/18117 > > In user auth for phone I put 0123456 at fs1.test.si , > > > Thanks! > > Br, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/20e020a8/attachment-0001.html From amit.nakum2009 at gmail.com Thu Jan 12 10:18:39 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Thu, 12 Jan 2012 12:48:39 +0530 Subject: [Freeswitch-users] ivr menu problem in no digit is press Message-ID: > > Dear All, > Please Help........... in case of hang up how call travel to next ivr menu if max-failuers=3 is over,and also suggest how i can play some file while no digit is press or enter in ivrs. Thanks in advance. amit nakum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/63f465be/attachment.html From bdfoster at endigotech.com Thu Jan 12 10:34:31 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 12 Jan 2012 02:34:31 -0500 Subject: [Freeswitch-users] ivr menu problem in no digit is press In-Reply-To: References: Message-ID: On Jan 12, 2012 2:19 AM, "amit nakum" wrote: >> >> Dear All, > > > Please Help........... > > in case of hang up how call travel to next ivr menu if max-failuers=3 is over,and also suggest how i can play some file while no digit is press or enter in ivrs. > > Thanks in advance. > amit nakum > > >From the wiki: ...except in your case you would do a playback instead of transferring to extension 1000. -BDF _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/331905ea/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 11:46:37 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 12 Jan 2012 08:46:37 +0000 (GMT) Subject: [Freeswitch-users] Help needing translating Asterisk syntax to Freeswitch Message-ID: <1326357997.76619.YahooMailNeo@web29406.mail.ird.yahoo.com> Problem is Michael that I don't have an Asterisk box, that config line was from some instructions on their website where they document an Asterisk setup. I'd rather not download Asterisk and try to get to grips with something that I'll never have a need to use again, since getting to grips with FreeSwitch is keeping me busy enough ! >I would do is get a siptrace from it working on your Asterisk box and compare that with the sip trace from the freeswitch box. From thomas at chaschperli.ch Thu Jan 12 11:59:58 2012 From: thomas at chaschperli.ch (Thomas Mueller) Date: Thu, 12 Jan 2012 09:59:58 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released Message-ID: <4F0EA10E.7050301@chaschperli.ch> hi all maybe interesting for german / swiss users: Gemeinschaft 4.0, which uses FreeSWITCH, was released yesterday. http://groups.google.com/group/gemeinschaft-announce/browse_thread/thread/9cded3dc8bb261f7 (german only). - Thomas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/0975ae75/attachment.html From miha at softnet.si Thu Jan 12 12:46:53 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 12 Jan 2012 10:46:53 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: <4F0E88DA.8050103@softnet.si> References: <4F0DA232.2080703@softnet.si> <4F0E88DA.8050103@softnet.si> Message-ID: <4F0EAC0D.5090400@softnet.si> Hi, just another question. My FS has fully qualified doman (exp fs1.xxxxx.si). Domains created on FS (for multitenant)are for exp test.si and btest.si. What is the dns for phones to register? test.si.fs1.xxxx.si? If the user for test.si is 1000, what should be the username for phone to register to domain test.si? Regards, Miha On 1/12/2012 8:16 AM, Miha Zoubek wrote: > Hi, > > i know that looks weird:) First how the phone should register on multi > domains freeswitch? > If you crate usr 1000 on domain domain.si (on freeswitch), how the > user is registering on FS (domain of FS for exp is test.com)? > > Thanks! > > miha > > > On 1/11/2012 4:00 PM, Michel Daggelinckx wrote: >> 0123456 at fs1.test.si@fs1.xxxx.si <--- looks weird >> to me >> >> On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek > > wrote: >> >> Hi, >> >> plase help me out why I am getting: >> >> 2012-01-11 15 :42:56.746075 [WARNING] >> sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia profile >> 'internal' for [0123456 at fs1.test.si@fs1.xxxx.si >> ] from ip xxx.xxx.xxx.xxx >> >> I do not know how to fix this or what I am doing wrong. >> $${domian} I chaged in fs1.test.si >> >> http://pastebin.freeswitch.org/18117 >> >> In user auth for phone I put 0123456 at fs1.test.si >> , >> >> >> Thanks! >> >> Br, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/702be065/attachment-0001.html From gmaruzz at gmail.com Thu Jan 12 13:29:45 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 12 Jan 2012 11:29:45 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F0EA10E.7050301@chaschperli.ch> References: <4F0EA10E.7050301@chaschperli.ch> Message-ID: Hi Thomas, thanks a lot for this effort and for letting us know. An English version would be very interesting! BTW, have a look at the subtle but substantial errors that google translate gives on the announcement (always funny). http://translate.google.com/#de|en Congratulations, -giovanni On Thu, Jan 12, 2012 at 9:59 AM, Thomas Mueller wrote: > hi all > > maybe interesting for german / swiss users: Gemeinschaft 4.0, which uses > FreeSWITCH, was released yesterday. > > http://groups.google.com/group/gemeinschaft-announce/browse_thread/thread/9cded3dc8bb261f7 > (german only). > > - Thomas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From shaheryarkh at googlemail.com Thu Jan 12 13:55:29 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 12 Jan 2012 15:55:29 +0500 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: Can you give bit more details on scope, e.g. how many extensions / users you need with what features etc. etc. You can contact me off list (shaheryarkh at gmail.com) if you don't want to post these details to whole mailing list. Thank you. On Thu, Jan 12, 2012 at 12:01 PM, Yehavi Bourvine wrote: > I am using a version from mid-October due to the BLF's problem. When it is > fixed I will move to the latest GIT (I am doing upgrade to the latest GIT > once every 2-4 weeks). > > __Yehavi: > > 2012/1/12 Anton Kvashenkin > >> Which version of freeswitch do you use? >> >> >> 2012/1/12 Yehavi Bourvine >> >>> Hello Gabriel, >>> >>> The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving >>> around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX >>> connected aside it with about 200 extensions (we do not put more extensions >>> due to potential problems it might create on the Nortel). >>> >>> We started with Asterisk and quite quickly understood that it lacks >>> some functionality that an organization like us needs, thus we moved to >>> Freeswitch, and we are happy with this decision. We also have AudioCodes >>> ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are >>> mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. >>> >>> We have to do some move soon as our Nortels are almost 20 years old. We >>> are now in the process of estimating how much an open source solution would >>> cost us (needs more man power which we have to estimate the amount of) so >>> we can compare it to a commecrical solution offerings. >>> >>> If you need more information don't hesitate to ask. >>> >>> Regards, __Yehavi: >>> >>> 2012/1/10 Gabriel Kuri >>> >>>> Does anyone know of any Universities running FreeSWITCH, or other open >>>> source VoIP for that matter, for local call routing between handsets? >>>> >>>> We're looking at replacing our old Avaya system or upgrade it, and the >>>> forklift upgrade from Avaya is ridiculously expensive (no surprise). >>>> >>>> We'd like to replace our Avaya system with a combination of OpenSIPS >>>> and FreeSWITCH and some Cisco routers for external PSTN access, but >>>> it's going to be a tough sell to our CIO, unless we can show someone >>>> else has done it already. >>>> >>>> Any pointers to other Universities would be great. >>>> >>>> Cheers, >>>> Gabe >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/60e14adc/attachment.html From yehavi.bourvine at gmail.com Thu Jan 12 14:19:33 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 12 Jan 2012 13:19:33 +0200 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: Hello Muhammad, We tried to move to FreeSWITCH a representative sample of our users. Thus, it includes a mix of: - Users with analog phone who just want a CallerID (in our current exchange it is not available). We connect them using a 24 ports ATA (MP124 from Audiocodes; had others, junked them...). - Secretaries who need extra functionality: BLF, SLA, Intercoms and multi-lines phones. The connection to the university's main exchange and to an external supplier is done via SIP<->PRI gateway from AudioCodes (who did some software changes upon my request to have more compatibility with FreeSWITCH). In addition we wrote some WEB interface so users can manage issues related to the phone: - List, hear and delete voicemail messages. - Change some characteristics, like whether to have call waiting, whether to send copy of voicemail to Email, etc. - Manage speed dials (people like it...). - Manage the phone book of Polycom and Audiocodes phones. - System managers can do slightly more. - We are now building a simple call center with WEB interface to follow its usage. All our users' database is on MySQL. We have only around 200 extensions as we try to keep the user's old number and have a "follow me" from the old exchnage to the new one. This "Follow me" is somewhat problematic, and we have to re-enable it manually for some extensions from time to time. Regards, __Yehavi: 2012/1/12 Muhammad Shahzad > Can you give bit more details on scope, e.g. how many extensions / users > you need with what features etc. etc. > > You can contact me off list (shaheryarkh at gmail.com) if you don't want > to post these details to whole mailing list. > > Thank you. > > > On Thu, Jan 12, 2012 at 12:01 PM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> wrote: > >> I am using a version from mid-October due to the BLF's problem. When it >> is fixed I will move to the latest GIT (I am doing upgrade to the latest >> GIT once every 2-4 weeks). >> >> __Yehavi: >> >> 2012/1/12 Anton Kvashenkin >> >>> Which version of freeswitch do you use? >>> >>> >>> 2012/1/12 Yehavi Bourvine >>> >>>> Hello Gabriel, >>>> >>>> The Hebrew University of Jerusalem has 5 old Nortel exhcnages serving >>>> around 8,000 extensions. For about 3 years now we have a FreeSWITCH PBX >>>> connected aside it with about 200 extensions (we do not put more extensions >>>> due to potential problems it might create on the Nortel). >>>> >>>> We started with Asterisk and quite quickly understood that it lacks >>>> some functionality that an organization like us needs, thus we moved to >>>> Freeswitch, and we are happy with this decision. We also have AudioCodes >>>> ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are >>>> mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. >>>> >>>> We have to do some move soon as our Nortels are almost 20 years old. We >>>> are now in the process of estimating how much an open source solution would >>>> cost us (needs more man power which we have to estimate the amount of) so >>>> we can compare it to a commecrical solution offerings. >>>> >>>> If you need more information don't hesitate to ask. >>>> >>>> Regards, __Yehavi: >>>> >>>> 2012/1/10 Gabriel Kuri >>>> >>>>> Does anyone know of any Universities running FreeSWITCH, or other open >>>>> source VoIP for that matter, for local call routing between handsets? >>>>> >>>>> We're looking at replacing our old Avaya system or upgrade it, and the >>>>> forklift upgrade from Avaya is ridiculously expensive (no surprise). >>>>> >>>>> We'd like to replace our Avaya system with a combination of OpenSIPS >>>>> and FreeSWITCH and some Cisco routers for external PSTN access, but >>>>> it's going to be a tough sell to our CIO, unless we can show someone >>>>> else has done it already. >>>>> >>>>> Any pointers to other Universities would be great. >>>>> >>>>> Cheers, >>>>> Gabe >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/87469c6c/attachment-0001.html From thomas at chaschperli.ch Thu Jan 12 14:59:00 2012 From: thomas at chaschperli.ch (Thomas Mueller) Date: Thu, 12 Jan 2012 12:59:00 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> Message-ID: <4F0ECB04.20007@chaschperli.ch> hi giovanni here a translation by me: Hi 5 years by now the first version of Gemeinschaft was brought to live. I'm proud to present version 4.0 of Gemeinschaft today => http://www.amooma.de/gemeinschaft On this occation i'd like to thank to the federal office of it security (BSI) for the development contract for Gemeinschaft 4.0 whithin the SiVoIP project. In addition special thanks to Sascha Daniels, Philipp Kempgen, Peter Kozak, Klaus Knopper und Ralf Spenneberg. Gemeinschaft 4.0 is a huge step towards a secure VoIP PBX for everyone. Main differences to Gemeinschaft 3.x ========================= - Asterisk was replaced by FreeSWITCH and Kmailio - PHP was replaced in favor of Ruby 3 - Test-driven development - easy install with with from bootable ISO - Security, stability and performance - all data can be saved encrypted on disk (including fax and voicemails) FAQ === Q: I'm happy with Gemeinschaft 3.x. Shall I upgrade? A: No, never change a running system! But read https://www.heise.de/artikel-archiv/ix/2011/11/131 and decide Q: How many lines of the Gemeinschaft 3.1 code were recycled in Gemeinschaft 4.0? A: 0 lines Q: Why no more Asterisk? A: FreeSWITCH is better, more stable, more secure and gives higher performance. In the past all people used sendmail and now postfix is the best choice. That's the way it is. Q: Why no more PHP? A: Ruby on rails is much better http://www.ruby-auf-schienen.de Q: More FAQ's? A: Yes, have a look at https://github.com/amooma/GS4/wiki/FAQ ... On 12.01.2012 11:29, Giovanni Maruzzelli wrote: > Hi Thomas, > > thanks a lot for this effort and for letting us know. > > An English version would be very interesting! > > BTW, have a look at the subtle but substantial errors that google > translate gives on the announcement (always funny). > > http://translate.google.com/#de|en > > Congratulations, > > -giovanni > > On Thu, Jan 12, 2012 at 9:59 AM, Thomas Mueller wrote: >> hi all >> >> maybe interesting for german / swiss users: Gemeinschaft 4.0, which uses >> FreeSWITCH, was released yesterday. >> >> http://groups.google.com/group/gemeinschaft-announce/browse_thread/thread/9cded3dc8bb261f7 >> (german only). >> >> - Thomas >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -- m?ller it gmbh, kanzleistrasse 126, 8004 z?rich +41 (0)44 515 20 22 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/f2f1bc14/attachment.html From govoiper at gmail.com Thu Jan 12 15:22:26 2012 From: govoiper at gmail.com (Sammy Govind) Date: Thu, 12 Jan 2012 17:22:26 +0500 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: Hey, you should really look into Whistle ! I have a feeling that almost 90%+ of your requirements will be served by that. Regards, Sammy On Thu, Jan 12, 2012 at 4:19 PM, Yehavi Bourvine wrote: > Hello Muhammad, > > We tried to move to FreeSWITCH a representative sample of our users. > Thus, it includes a mix of: > > > - Users with analog phone who just want a CallerID (in our current > exchange it is not available). We connect them using a 24 ports ATA (MP124 > from Audiocodes; had others, junked them...). > - Secretaries who need extra functionality: BLF, SLA, Intercoms and > multi-lines phones. > > The connection to the university's main exchange and to an external > supplier is done via SIP<->PRI gateway from AudioCodes (who did some > software changes upon my request to have more compatibility with > FreeSWITCH). > > In addition we wrote some WEB interface so users can manage issues related > to the phone: > > - List, hear and delete voicemail messages. > - Change some characteristics, like whether to have call waiting, > whether to send copy of voicemail to Email, etc. > - Manage speed dials (people like it...). > - Manage the phone book of Polycom and Audiocodes phones. > - System managers can do slightly more. > - We are now building a simple call center with WEB interface to > follow its usage. > > All our users' database is on MySQL. > > We have only around 200 extensions as we try to keep the user's old number > and have a "follow me" from the old exchnage to the new one. This "Follow > me" is somewhat problematic, and we have to re-enable it manually for some > extensions from time to time. > > Regards, __Yehavi: > > 2012/1/12 Muhammad Shahzad > >> Can you give bit more details on scope, e.g. how many extensions / users >> you need with what features etc. etc. >> >> You can contact me off list (shaheryarkh at gmail.com) if you don't want >> to post these details to whole mailing list. >> >> Thank you. >> >> >> On Thu, Jan 12, 2012 at 12:01 PM, Yehavi Bourvine < >> yehavi.bourvine at gmail.com> wrote: >> >>> I am using a version from mid-October due to the BLF's problem. When it >>> is fixed I will move to the latest GIT (I am doing upgrade to the latest >>> GIT once every 2-4 weeks). >>> >>> __Yehavi: >>> >>> 2012/1/12 Anton Kvashenkin >>> >>>> Which version of freeswitch do you use? >>>> >>>> >>>> 2012/1/12 Yehavi Bourvine >>>> >>>>> Hello Gabriel, >>>>> >>>>> The Hebrew University of Jerusalem has 5 old Nortel exhcnages >>>>> serving around 8,000 extensions. For about 3 years now we have a FreeSWITCH >>>>> PBX connected aside it with about 200 extensions (we do not put more >>>>> extensions due to potential problems it might create on the Nortel). >>>>> >>>>> We started with Asterisk and quite quickly understood that it lacks >>>>> some functionality that an organization like us needs, thus we moved to >>>>> Freeswitch, and we are happy with this decision. We also have AudioCodes >>>>> ATAs for analog phones and AudioCodes PRI gateway. Our "smartphones" are >>>>> mainly Polycoms for secretaries and AudioCodes HD320 for "regular" users. >>>>> >>>>> We have to do some move soon as our Nortels are almost 20 years old. >>>>> We are now in the process of estimating how much an open source solution >>>>> would cost us (needs more man power which we have to estimate the amount >>>>> of) so we can compare it to a commecrical solution offerings. >>>>> >>>>> If you need more information don't hesitate to ask. >>>>> >>>>> Regards, __Yehavi: >>>>> >>>>> 2012/1/10 Gabriel Kuri >>>>> >>>>>> Does anyone know of any Universities running FreeSWITCH, or other open >>>>>> source VoIP for that matter, for local call routing between handsets? >>>>>> >>>>>> We're looking at replacing our old Avaya system or upgrade it, and the >>>>>> forklift upgrade from Avaya is ridiculously expensive (no surprise). >>>>>> >>>>>> We'd like to replace our Avaya system with a combination of OpenSIPS >>>>>> and FreeSWITCH and some Cisco routers for external PSTN access, but >>>>>> it's going to be a tough sell to our CIO, unless we can show someone >>>>>> else has done it already. >>>>>> >>>>>> Any pointers to other Universities would be great. >>>>>> >>>>>> Cheers, >>>>>> Gabe >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/8372f088/attachment-0001.html From Claudio.Cavalera at italtel.it Thu Jan 12 16:47:01 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 12 Jan 2012 14:47:01 +0100 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: By the way, in ancient times FS devs used to bring every unpredictable library into the FS SVN trunk to mantain their own stable version :-) However if gnutls is not used by FS (I'm not sure if its just mod_dingaling or the whole FS now using openssh) maybe someone should update remove it from the wiki (it still appears here and there) http://wiki.freeswitch.org/wiki/Special:Search?search=gnutls&go=Go Kind Regards, Claudio From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: Wednesday, January 11, 2012 4:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] crash-protection documented on the wiki but no more implemented Unfortunately libgnutls ranges from being relatively stable to just unpredictable. As Avi did mention Tony made a few changes back in mid November and switched dingaling to use OpenSSL instead of gnutls. If libgnutls is a dependency for an older version of mod_dingaling... the newer version uses openssh afaik. It's probably a good idea to recompile from latest. -Avi 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. ----------------------------------------------------------------------------------------------------- From daggelinckxmichel at gmail.com Thu Jan 12 16:52:53 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Thu, 12 Jan 2012 14:52:53 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: <4F0EAC0D.5090400@softnet.si> References: <4F0DA232.2080703@softnet.si> <4F0E88DA.8050103@softnet.si> <4F0EAC0D.5090400@softnet.si> Message-ID: they should register to test.si or btest.si On Thu, Jan 12, 2012 at 10:46 AM, Miha Zoubek wrote: > Hi, > > just another question. My FS has fully qualified doman (exp fs1.xxxxx.si). > Domains created on FS (for multitenant)are for exp test.si and btest.si. > > What is the dns for phones to register? test.si.fs1.xxxx.si? > If the user for test.si is 1000, what should be the username for phone to > register to domain test.si? > > Regards, > Miha > > On 1/12/2012 8:16 AM, Miha Zoubek wrote: > > Hi, > > i know that looks weird:) First how the phone should register on multi > domains freeswitch? > If you crate usr 1000 on domain domain.si (on freeswitch), how the user > is registering on FS (domain of FS for exp is test.com)? > > Thanks! > > miha > > > On 1/11/2012 4:00 PM, Michel Daggelinckx wrote: > > 0123456 at fs1.test.si@fs1.xxxx.si <--- looks weird to me > > On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek wrote: > >> Hi, >> >> plase help me out why I am getting: >> >> 2012-01-11 15:42:56.746075 [WARNING] sofia_reg.c:1359 SIP auth failure >> (REGISTER) on sofia profile 'internal' for [ >> 0123456 at fs1.test.si@fs1.xxxx.si] from ip xxx.xxx.xxx.xxx >> >> I do not know how to fix this or what I am doing wrong. >> $${domian} I chaged in fs1.test.si >> >> http://pastebin.freeswitch.org/18117 >> >> In user auth for phone I put 0123456 at fs1.test.si, >> >> >> Thanks! >> >> Br, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/de105c5a/attachment.html From avi at avimarcus.net Thu Jan 12 17:31:21 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 12 Jan 2012 16:31:21 +0200 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Can someone confirm this before deleting all the wiki references? And the list of related dependencies by distribution needs to be updated.. the install instructions are still quite spread out.. Sorry, I meant openssl, not openssh.. -Avi On Thu, Jan 12, 2012 at 3:47 PM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > By the way, > in ancient times FS devs used to bring every unpredictable library into > the FS SVN trunk to mantain their own stable version :-) > > However if gnutls is not used by FS (I'm not sure if its just > mod_dingaling or the whole FS now using openssh) maybe someone should > update remove it from the wiki (it still appears here and there) > http://wiki.freeswitch.org/wiki/Special:Search?search=gnutls&go=Go > > Kind Regards, > Claudio > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 > Sent: Wednesday, January 11, 2012 4:26 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] crash-protection documented on the wiki > but no more implemented > > Unfortunately libgnutls ranges from being relatively stable to just > unpredictable. As Avi did mention Tony made a few changes back in mid > November and switched dingaling to use OpenSSL instead of gnutls. > If > libgnutls is a dependency for an older version of mod_dingaling... the > newer version uses openssh afaik. It's probably a good idea to recompile > from latest. > > > -Avi > > > 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. > > ----------------------------------------------------------------------------------------------------- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/1505e700/attachment-0001.html From miha at softnet.si Thu Jan 12 17:39:58 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 12 Jan 2012 15:39:58 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: References: <4F0DA232.2080703@softnet.si> <4F0E88DA.8050103@softnet.si> <4F0EAC0D.5090400@softnet.si> Message-ID: <4F0EF0BE.2070608@softnet.si> Hi @Michel, i figure it out what was the problems. I did know that every domain crated of FS should be also in domain sever. Thanks! Regards, Miha On 1/12/2012 2:52 PM, Michel Daggelinckx wrote: > they should register to test.si or btest.si > > > > On Thu, Jan 12, 2012 at 10:46 AM, Miha Zoubek > wrote: > > Hi, > > just another question. My FS has fully qualified doman (exp > fs1.xxxxx.si ). Domains created on FS (for > multitenant)are for exp test.si and btest.si > . > > What is the dns for phones to register? test.si.fs1.xxxx.si > ? > If the user for test.si is 1000, what should be > the username for phone to register to domain test.si ? > > Regards, > Miha > > On 1/12/2012 8:16 AM, Miha Zoubek wrote: >> Hi, >> >> i know that looks weird:) First how the phone should register on >> multi domains freeswitch? >> If you crate usr 1000 on domain domain.si (on >> freeswitch), how the user is registering on FS (domain of FS for >> exp is test.com )? >> >> Thanks! >> >> miha >> >> >> On 1/11/2012 4:00 PM, Michel Daggelinckx wrote: >>> 0123456 at fs1.test.si@ fs1.xxxx.si >>> <--- looks weird to me >>> >>> On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek >> > wrote: >>> >>> Hi, >>> >>> plase help me out why I am getting: >>> >>> 2012-01-11 15 :42:56.746075 [WARNING] >>> sofia_reg.c:1359 SIP auth failure (REGISTER) on sofia >>> profile 'internal' for [0123456 at fs1.test.si@fs1.xxxx.si >>> ] from ip >>> xxx.xxx.xxx.xxx >>> >>> I do not know how to fix this or what I am doing wrong. >>> $${domian} I chaged in fs1.test.si >>> >>> http://pastebin.freeswitch.org/18117 >>> >>> In user auth for phone I put 0123456 at fs1.test.si >>> , >>> >>> >>> Thanks! >>> >>> Br, >>> Miha >>> >>> -- >>> Best regards / Lep Pozdrav >>> Miha Zoubek >>> Softnet d.o.o. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/760999da/attachment.html From Claudio.Cavalera at italtel.it Thu Jan 12 17:57:50 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 12 Jan 2012 15:57:50 +0100 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Avi, I've not checked if it is still seggy... however I'm on the latest git and gnutls seems to be still used: ldd mod_dingaling.so | grep tls libgnutls.so.26 => /usr/lib64/libgnutls.so.26 (0x00007f4e24529000) is ldd telling the truth? Can you please cross-check on your FS boxes? Let's wait a minute before erasing all the gnutls pointers from the wiki ^__^ Kind regards, Claudio From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Thursday, January 12, 2012 3:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] crash-protection documented on the wiki but no more implemented Can someone confirm this before deleting all the wiki references? And the list of related dependencies by distribution needs to be updated.. the install instructions are still quite spread out.. Sorry, I meant openssl, not openssh.. -Avi 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. ----------------------------------------------------------------------------------------------------- From curriegrad2004 at gmail.com Thu Jan 12 18:10:22 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 12 Jan 2012 07:10:22 -0800 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Rebootstrap the source code and try again On Thu, Jan 12, 2012 at 6:57 AM, Cavalera Claudio Luigi wrote: > Avi, > I've not checked if it is still seggy... > > however I'm on the latest git and gnutls seems to be still used: ldd > mod_dingaling.so | grep tls > ? ? ? ?libgnutls.so.26 => /usr/lib64/libgnutls.so.26 > (0x00007f4e24529000) > is ldd telling the truth? Can you please cross-check on your FS boxes? > > Let's wait a minute before erasing all the gnutls pointers from the wiki > ^__^ > > Kind regards, > Claudio > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Thursday, January 12, 2012 3:31 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] crash-protection documented on the wiki > but no more implemented > > Can someone confirm this before deleting all the wiki references? > And the list of related dependencies by distribution needs to be > updated.. the install instructions are still quite spread out.. > Sorry, I meant openssl, not openssh.. > > > -Avi > > > 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. > ----------------------------------------------------------------------------------------------------- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From Claudio.Cavalera at italtel.it Thu Jan 12 18:15:01 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 12 Jan 2012 16:15:01 +0100 Subject: [Freeswitch-users] crash-protection documented on the wiki but no more implemented In-Reply-To: References: Message-ID: Hello, what do you mean exactly? $ git pull Current branch master is up to date. I'm sure I did ./bootstrap.sh about half an hour ago. Did someone change anything in the code? Ciao, Claudio > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch- > users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 > Sent: Thursday, January 12, 2012 4:10 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] crash-protection documented on the wiki > but no more implemented > > Rebootstrap the source code and try again > > On Thu, Jan 12, 2012 at 6:57 AM, Cavalera Claudio Luigi > wrote: > > Avi, > > I've not checked if it is still seggy... > > > > however I'm on the latest git and gnutls seems to be still used: ldd > > mod_dingaling.so | grep tls > > ? ? ? ?libgnutls.so.26 => /usr/lib64/libgnutls.so.26 > > (0x00007f4e24529000) > > is ldd telling the truth? Can you please cross-check on your FS > boxes? > > > > Let's wait a minute before erasing all the gnutls pointers from the > wiki > > ^__^ > > > > Kind regards, > > Claudio > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Avi > > Marcus > > Sent: Thursday, January 12, 2012 3:31 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] crash-protection documented on the > wiki > > but no more implemented > > > > Can someone confirm this before deleting all the wiki references? > > And the list of related dependencies by distribution needs to be > > updated.. the install instructions are still quite spread out.. > > Sorry, I meant openssl, not openssh.. > > > > > > -Avi > > > > > > 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. > > --------------------------------------------------------------------- > -------------------------------- > > > > > > > _______________________________________________________________________ > __ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > > http://www.freeswitch.org > > _______________________________________________________________________ > __ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From adam.kelloway at newpace.ca Thu Jan 12 18:59:21 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Thu, 12 Jan 2012 11:59:21 -0400 Subject: [Freeswitch-users] Display Name in redirect dialplan tool Message-ID: <4F0F0359.6040504@newpace.ca> Hi there, Anyone know if there is a way to set the display name when using the redirect dialplan tool? It seems to just default to "unknown". Thanks, Adam From mytemike72 at gmail.com Thu Jan 12 18:02:05 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 12 Jan 2012 16:02:05 +0100 Subject: [Freeswitch-users] ESL Managed IVR and CallControl Message-ID: Hi, I have been playing with mod_managed for a while now but nobody realy seems to be able to help me out. So now I am trying to go to ESL. (which most of the people suggest anyway). What I need is control from .net over the complete call! (so even when bridged and returns). I need to build a main library where all my generic functions will be put in so I can use them in my different scripts. What I''ve figured out is problably the best way is to write a server which connects to my FS an listens to all events. When a call comes in I need to fire a stored procedure to my sqlserver and based on the reply I will decide if the call needsto be answerred, early media needs to be played, or the call needs to be rejected. In any case, at the end of every call (answered or not) I need to finalize my call with another stored procedure. When teh call is answered based on stored procedure result I need to run a script (also managed code) run a script from another dll so I have all my scripts separate but they share the corelib I create. I have been searching the web for a long time now to see if I can find something simular, but I only see generic examples wich don't actually show how to control multiple calls in a scenario like this. Who can help me out point me to the right direction or has experience in setting something up like this? Thanks in advance, Michael Lutz. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/3c1725cd/attachment.html From Hector.Geraldino at ip-soft.net Thu Jan 12 19:42:25 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 12 Jan 2012 11:42:25 -0500 Subject: [Freeswitch-users] ESL Managed IVR and CallControl In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507A23@NY1-EXMB-01.ip-soft.net> I have experience setting up something like that, but in Java. You should use ESL in outbound mode. For every new incoming call, FS will open a socket connection to an ip:port where your application should be listening to, and you will have complete control over the incoming call. So, in theory, you should only need to have a SocketServer listening to an specific port, and send commands/receive events from FS to your application (you can play a little bit with this in linux using the netcat (nc) utility, or maybe in windows by installing Cygwin and using nc.exe) I'm happy using the Java ESL library, a pure socket-based java application (jboss/netty based) with no dependencies on system libs. Maybe (and this is a BIG MAYBE) you can reuse this library from .NET, relying on Mono's IKVM tool, that allows Java-to-NET interoperability by converting Java's JVM bytecode to .NET CLR. I've taken this approach before, when I found a terrific tool in Java (jPOS, for example) with no counterparts in .NET. This is only one option. There's always the option of building your own application listening on a socket, or using the .NET libraries listed on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket#.NET_Client_library Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Thursday, January 12, 2012 10:02 AM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] ESL Managed IVR and CallControl Hi, I have been playing with mod_managed for a while now but nobody realy seems to be able to help me out. So now I am trying to go to ESL. (which most of the people suggest anyway). What I need is control from .net over the complete call! (so even when bridged and returns). I need to build a main library where all my generic functions will be put in so I can use them in my different scripts. What I''ve figured out is problably the best way is to write a server which connects to my FS an listens to all events. When a call comes in I need to fire a stored procedure to my sqlserver and based on the reply I will decide if the call needsto be answerred, early media needs to be played, or the call needs to be rejected. In any case, at the end of every call (answered or not) I need to finalize my call with another stored procedure. When teh call is answered based on stored procedure result I need to run a script (also managed code) run a script from another dll so I have all my scripts separate but they share the corelib I create. I have been searching the web for a long time now to see if I can find something simular, but I only see generic examples wich don't actually show how to control multiple calls in a scenario like this. Who can help me out point me to the right direction or has experience in setting something up like this? Thanks in advance, Michael Lutz. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/1b9c0d61/attachment-0001.html From msc at freeswitch.org Thu Jan 12 19:45:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Jan 2012 08:45:57 -0800 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F0ECB04.20007@chaschperli.ch> References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> Message-ID: Thomas, Thanks for this! I will take this information and work up a story for freeswitch.org. Thanks, MC On Thu, Jan 12, 2012 at 3:59 AM, Thomas Mueller wrote: > hi giovanni > > here a translation by me: > > Hi > > 5 years by now the first version of Gemeinschaft was brought to live. I'm > proud to present version 4.0 of Gemeinschaft today => > http://www.amooma.de/gemeinschaft > > On this occation i'd like to thank to the federal office of it security > (BSI) for the development contract for Gemeinschaft 4.0 whithin the SiVoIP > project. In addition special thanks to Sascha Daniels, Philipp Kempgen, > Peter Kozak, Klaus Knopper und Ralf Spenneberg. > > Gemeinschaft 4.0 is a huge step towards a secure VoIP PBX for everyone. > > Main differences to Gemeinschaft 3.x > ========================= > - Asterisk was replaced by FreeSWITCH and Kmailio > - PHP was replaced in favor of Ruby 3 > - Test-driven development > - easy install with with from bootable ISO > - Security, stability and performance > - all data can be saved encrypted on disk (including fax and voicemails) > > FAQ > === > Q: I'm happy with Gemeinschaft 3.x. Shall I upgrade? > A: No, never change a running system! But read > https://www.heise.de/artikel-archiv/ix/2011/11/131 and decide > > Q: How many lines of the Gemeinschaft 3.1 code were recycled in > Gemeinschaft 4.0? > A: 0 lines > > Q: Why no more Asterisk? > A: FreeSWITCH is better, more stable, more secure and gives higher > performance. In the past all people used sendmail and now postfix is the > best choice. That's the way it is. > > Q: Why no more PHP? > A: Ruby on rails is much better http://www.ruby-auf-schienen.de > > Q: More FAQ's? > A: Yes, have a look at https://github.com/amooma/GS4/wiki/FAQ > > ... > > > On 12.01.2012 11:29, Giovanni Maruzzelli wrote: > > Hi Thomas, > > thanks a lot for this effort and for letting us know. > > An English version would be very interesting! > > BTW, have a look at the subtle but substantial errors that google > translate gives on the announcement (always funny). > http://translate.google.com/#de|en > > Congratulations, > > -giovanni > > On Thu, Jan 12, 2012 at 9:59 AM, Thomas Mueller wrote: > > hi all > > maybe interesting for german / swiss users: Gemeinschaft 4.0, which uses > FreeSWITCH, was released yesterday. > http://groups.google.com/group/gemeinschaft-announce/browse_thread/thread/9cded3dc8bb261f7 > (german only). > > - Thomas > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > m?ller it gmbh, kanzleistrasse 126, 8004 z?rich+41 (0)44 515 20 22 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/0d5aab4e/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 19:51:42 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 12 Jan 2012 16:51:42 +0000 (GMT) Subject: [Freeswitch-users] valet_park timeout Message-ID: <1326387102.43177.YahooMailNeo@web29405.mail.ird.yahoo.com> Hello All, I found an old thread from October 2010 entitled "valet_park timeout and spot announcement" (http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064143.html) I also found this tread from March 2011 entitled "valet parking improvements" ? (http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-March/004778.html) Could anyone tell me whether : 1/ Timeouts have made it into the tree ? ?As far as I can tell they have not ? 2/ Are there perhaps now better way to achieve the same goal ? (e.g. looking at the console, I see a variable generated called?valet_ticket ?) Basically I don't want to run into the same problems Jeremy did, i.e : "If the call is transferred somewhere else first or picked up from the first park on a different extension and then parked again, it doesn't recall the second parker." Thanks Bob From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 19:57:33 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 12 Jan 2012 16:57:33 +0000 (GMT) Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? Message-ID: <1326387453.28139.YahooMailNeo@web29402.mail.ird.yahoo.com> My freeswitch base directory contains the following directories : bin conf db grammar htdocs include lib log mod recordin run script scripts sounds storage What's the difference between "script" and "scripts" ? ? Seems a bit silly to me to have such confusingly similar names in the same parent directory ! From th982a at googlemail.com Thu Jan 12 20:17:44 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Thu, 12 Jan 2012 18:17:44 +0100 Subject: [Freeswitch-users] FreeSWITCH in a University Environment In-Reply-To: References: Message-ID: <4F0F15B8.9010207@googlemail.com> dear Yehavi, I heavily advise you making a cluster including a load balancer at the front of the server. All configurations, and dialplan futures would be through SQL, or write an python application which you embed in the dialplan that does it for you. Freeswitch, seems to be very flexible what I saw. Tamer Am 12.01.2012 12:19, schrieb Yehavi Bourvine: > Hello Muhammad, > > We tried to move to FreeSWITCH a representative sample of our users. > Thus, it includes a mix of: > > > * > Users with analog phone who just want a CallerID (in our current > exchange it is not available). We connect them using a 24 ports ATA > (MP124 from Audiocodes; had others, junked them...). > * > Secretaries who need extra functionality: BLF, SLA, Intercoms and > multi-lines phones. > > The connection to the university's main exchange and to an external > supplier is done via SIP<->PRI gateway from AudioCodes (who did some > software changes upon my request to have more compatibility with > FreeSWITCH). > > In addition we wrote some WEB interface so users can manage issues > related to the phone: > > * > List, hear and delete voicemail messages. > * > Change some characteristics, like whether to have call waiting, > whether to send copy of voicemail to Email, etc. > * > Manage speed dials (people like it...). > * > Manage the phone book of Polycom and Audiocodes phones. > * > System managers can do slightly more. > * > We are now building a simple call center with WEB interface to > follow its usage. > > All our users' database is on MySQL. > > We have only around 200 extensions as we try to keep the user's old > number and have a "follow me" from the old exchnage to the new one. This > "Follow me" is somewhat problematic, and we have to re-enable it > manually for some extensions from time to time. > > Regards, __Yehavi: > > 2012/1/12 Muhammad Shahzad > > > Can you give bit more details on scope, e.g. how many extensions / > users you need with what features etc. etc. > > You can contact me off list (shaheryarkh at gmail.com > ) if you don't want to post these details to whole > mailing list. > > Thank you. > > > On Thu, Jan 12, 2012 at 12:01 PM, Yehavi Bourvine > > wrote: > > I am using a version from mid-October due to the BLF's problem. > When it is fixed I will move to the latest GIT (I am doing > upgrade to the latest GIT once every 2-4 weeks). > > __Yehavi: > > 2012/1/12 Anton Kvashenkin > > > Which version of freeswitch do you use? > > > 2012/1/12 Yehavi Bourvine > > > Hello Gabriel, > > The Hebrew University of Jerusalem has 5 old Nortel > exhcnages serving around 8,000 extensions. For about 3 > years now we have a FreeSWITCH PBX connected aside it > with about 200 extensions (we do not put more extensions > due to potential problems it might create on the Nortel). > > We started with Asterisk and quite quickly understood > that it lacks some functionality that an organization > like us needs, thus we moved to Freeswitch, and we are > happy with this decision. We also have AudioCodes ATAs > for analog phones and AudioCodes PRI gateway. Our > "smartphones" are mainly Polycoms for secretaries and > AudioCodes HD320 for "regular" users. > > We have to do some move soon as our Nortels are almost > 20 years old. We are now in the process of estimating > how much an open source solution would cost us (needs > more man power which we have to estimate the amount of) > so we can compare it to a commecrical solution offerings. > > If you need more information don't hesitate to ask. > > Regards, __Yehavi: > > 2012/1/10 Gabriel Kuri > > > Does anyone know of any Universities running > FreeSWITCH, or other open > source VoIP for that matter, for local call routing > between handsets? > > We're looking at replacing our old Avaya system or > upgrade it, and the > forklift upgrade from Avaya is ridiculously > expensive (no surprise). > > We'd like to replace our Avaya system with a > combination of OpenSIPS > and FreeSWITCH and some Cisco routers for external > PSTN access, but > it's going to be a tough sell to our CIO, unless we > can show someone > else has done it already. > > Any pointers to other Universities would be great. > > Cheers, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Thu Jan 12 20:17:43 2012 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 12 Jan 2012 18:17:43 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F0ECB04.20007@chaschperli.ch> References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> Message-ID: On Thu, Jan 12, 2012 at 12:59 PM, Thomas Mueller wrote: > hi giovanni > > here a translation by me: So much better than Google! Congratulations again, and please consider to make available an English version of it, a very basic and rough translation can do, then the community can do the finesses. I was attending Cluecon 2011, and the presentation about Gemeinweisen was very well received and there was lot of interest in the audience. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From oseslija at gmail.com Thu Jan 12 22:24:42 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 12 Jan 2012 20:24:42 +0100 Subject: [Freeswitch-users] Multi domains issue In-Reply-To: <4F0EF0BE.2070608@softnet.si> References: <4F0DA232.2080703@softnet.si> <4F0E88DA.8050103@softnet.si> <4F0EAC0D.5090400@softnet.si> <4F0EF0BE.2070608@softnet.si> Message-ID: It doesn't if you use outbound proxy. On 12 Jan 2012 15:46, "Miha Zoubek" wrote: > Hi @Michel, > > i figure it out what was the problems. I did know that every domain crated > of FS should be also in domain sever. > > Thanks! > > Regards, > Miha > > On 1/12/2012 2:52 PM, Michel Daggelinckx wrote: > > they should register to test.si or btest.si > > > On Thu, Jan 12, 2012 at 10:46 AM, Miha Zoubek wrote: > >> Hi, >> >> just another question. My FS has fully qualified doman (exp fs1.xxxxx.si). >> Domains created on FS (for multitenant)are for exp test.si and btest.si. >> >> What is the dns for phones to register? test.si.fs1.xxxx.si? >> If the user for test.si is 1000, what should be the username for phone >> to register to domain test.si? >> >> Regards, >> Miha >> >> On 1/12/2012 8:16 AM, Miha Zoubek wrote: >> >> Hi, >> >> i know that looks weird:) First how the phone should register on multi >> domains freeswitch? >> If you crate usr 1000 on domain domain.si (on freeswitch), how the user >> is registering on FS (domain of FS for exp is test.com)? >> >> Thanks! >> >> miha >> >> >> On 1/11/2012 4:00 PM, Michel Daggelinckx wrote: >> >> 0123456 at fs1.test.si@fs1.xxxx.si <--- looks weird to me >> >> On Wed, Jan 11, 2012 at 3:52 PM, Miha Zoubek wrote: >> >>> Hi, >>> >>> plase help me out why I am getting: >>> >>> 2012-01-11 15 <2012-01-11%2015>:42:56.746075 [WARNING] sofia_reg.c:1359 >>> SIP auth failure (REGISTER) on sofia profile 'internal' for [ >>> 0123456 at fs1.test.si@fs1.xxxx.si] from ip xxx.xxx.xxx.xxx >>> >>> I do not know how to fix this or what I am doing wrong. >>> $${domian} I chaged in fs1.test.si >>> >>> http://pastebin.freeswitch.org/18117 >>> >>> In user auth for phone I put 0123456 at fs1.test.si, >>> >>> >>> Thanks! >>> >>> Br, >>> Miha >>> >>> -- >>> Best regards / Lep Pozdrav >>> Miha Zoubek >>> Softnet d.o.o. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/9ee72e01/attachment.html From oz at o-tec.tv Thu Jan 12 20:40:25 2012 From: oz at o-tec.tv (Oz Mortimer) Date: Thu, 12 Jan 2012 17:40:25 -0000 Subject: [Freeswitch-users] mod_curl holding call Message-ID: <016801ccd151$45112750$cf3375f0$@tv> Hi Mailing list, I am having an issue with curl where the app execution doesn't return where there is an issue resolving domain, even if the caller has hungup. The billable duration in the CDRS along with the end time stamp both reflect the script end time rather than call end time. Here is an example to force the issue. Edit /etc/resolve and set a name server to 9.9.9.9 Create a simple lua script session:answer() session:execute("curl","http://www.google.com") When calling this script via softphone and hanging up during the CURL, I see 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing at myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing at myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing at myip [BREAK] 2012-01-12 17:29:23.070105 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/testing at myip [BREAK] And then nothing, until I assume the CURL call ends (including the Callstate Change ACTIVE -> HANGUP) If I then look at the cdrs, they also show billable duration inclusive of the curl timeout even though the call ended some seconds before. I tried setting a hangup hook in the lua but this too doesn't get called when the call terminated. It seems like the state machine for the call gets stuck until curl completes. I've just downloaded the git head and the issue is present there too. Obvious question, shouldn't billable duration and end stamp reflect the actual call? Or is there something I need to do that I've currently missed. Is this issue specific to curl, or could it happen in other modules? Any pointers? Many Thanks Oz. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/aa2576d5/attachment-0001.html From wesleyakio at tuntscorp.com Thu Jan 12 21:13:04 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Thu, 12 Jan 2012 16:13:04 -0200 Subject: [Freeswitch-users] Freeswitch CDR Packet counters Message-ID: Hi all, I'm trying to pinpoint the origin of a call quality problem. The XML CDR has some information on it but I'm having some trouble on finding documentation for some of the fields. 6801 6528 37 0 0 0 273 Most of this fields are self explanatory(I hope to be right on this one), flush_packet_count is the one I cant figure out with certainty. Best, Wesley Akio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/ed0727eb/attachment.html From msc at freeswitch.org Thu Jan 12 22:54:39 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Jan 2012 11:54:39 -0800 Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? In-Reply-To: <1326387453.28139.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326387453.28139.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: I'm pretty sure that "script" is not created by the freeswitch install process. I know for a fact that "scripts" is and it's the default path for lua, perl, js, etc. scripts called from the cmd line or dialplan. -MC On Thu, Jan 12, 2012 at 8:57 AM, Bob Smith wrote: > My freeswitch base directory contains the following directories : > > bin > conf > db > grammar > htdocs > include > lib > log > mod > recordin > run > script > scripts > sounds > storage > > > What's the difference between "script" and "scripts" ? Seems a bit silly > to me to have such confusingly similar names in the same parent directory ! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/64cf0cdf/attachment.html From torstein.knutsen at gmail.com Thu Jan 12 22:59:47 2012 From: torstein.knutsen at gmail.com (Torstein Knutsen) Date: Thu, 12 Jan 2012 20:59:47 +0100 Subject: [Freeswitch-users] Conference Anonymous BBB Message-ID: Hi I am thinkering with FreeSWITCH and BigBlueButton conferences. I have on server running FS, and the BBB server is connected via EventSocket. Now, my challenge is regarding to Private numbers. I can't seem to get the conference to respect privacy. I set (in dialplan) BUT : While monitoring the Event socket on the CUSTOM Actoin:add-member, the caller_id number is back to the original. This cases my setup to leak Privacy marked numbers. I'm not running (FreeSWITCH Version 1.0.head (git-8fde25c 2011-10-20 13-15-58 -0500)) Is there any way around this ? Thank you ! Torstein -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/d93fd892/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 23:13:21 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 12 Jan 2012 20:13:21 +0000 (GMT) Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? Message-ID: <1326399201.87367.YahooMailNeo@web29402.mail.ird.yahoo.com> Seems weird, as "script" has a creation date/time exactly the same as my grammar,htdocs,recordings directories (Dec 16 23:04) Either way it's empty.... so the only thing I can think of is it might have been something to that specific checkout. Seeing as it's empty I might as well rm' it then. Thanks Bob >I'm pretty sure that "script" is not created by the freeswitch install >process. I know for a fact that "scripts" is and it's the default path for >lua, perl, js, etc. scripts called from the cmd line or dialplan. From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 12 23:20:25 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 12 Jan 2012 20:20:25 +0000 (GMT) Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? (found it !.... 023818bc) Message-ID: <1326399625.14157.YahooMailNeo@web29406.mail.ird.yahoo.com> * Marc Olivier Chouinard * 023818bc??on?master * Diff * Annotation * -1 * +1 more FS-302?Error in initial patch.. default scripts folder was missing the s at the end From daggelinckxmichel at gmail.com Thu Jan 12 23:42:56 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Thu, 12 Jan 2012 21:42:56 +0100 Subject: [Freeswitch-users] speech to text engine Message-ID: found this one on thepiaf forum. mightbe usefull for a freeswitch module http://nerd.bz/xqYO2D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/cbf0c7b7/attachment.html From roland at haenel.me Thu Jan 12 23:48:17 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Thu, 12 Jan 2012 21:48:17 +0100 Subject: [Freeswitch-users] Improving mod_conference for H.264 video // bounty? Message-ID: Dear all, We're looking for a way to improve the user experience with FreeSwitch's mod_conference when video (H.264) is involved. After some investigation in mod_26x's source code it seems that the creator actually had the idea to switch video (based on who has the floor) only on H.26x keyframes, but this was not fully implemented (i.e. it doesn't really work well). Maybe someone from the FreeSwitch guys could advise me about the way how to put up a bounty for improvements in this area. I found some boundy lists in the wiki, other things in Jira, and I'm not sure where to post my whishes (and maybe also a hint what amount of $'s seems appropriate here). Basically, that's what we need: - switch video source only on H.264 keyframes to avoid artifacts when the stream switches to a new floor member - feature to throttle floor changes in conference (i.e. floor is only changed after some time of activity, to prevent very frequent video stream changes) - investigation whether the codec parameter negotiation can be influenced/improved for H.264 passthru, e.g. to allow for specific profiles in SDP a=fmtp parameters Greetings, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/2d4cb577/attachment.html From krice at freeswitch.org Thu Jan 12 23:52:17 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Jan 2012 14:52:17 -0600 Subject: [Freeswitch-users] Improving mod_conference for H.264 video // bounty? In-Reply-To: Message-ID: Roland, Contact consulting at freeswitch.org this will be the fastest way to get a response on this. Ken On 1/12/12 2:48 PM, "Roland H?nel" wrote: > Dear all, > > We're looking for a way to improve the user experience with FreeSwitch's > mod_conference when video (H.264) is involved. After some investigation in > mod_26x's source code it seems that the creator actually had the idea to > switch video (based on who has the floor) only on H.26x keyframes, but this > was not fully implemented (i.e. it doesn't really work well). > > Maybe someone from the FreeSwitch guys could advise me about the way how to > put up a bounty for improvements in this area. > > I found some boundy lists in the wiki, other things in Jira, and I'm not sure > where to post my whishes (and maybe also a hint what amount of $'s seems > appropriate here). > > Basically, that's what we need: > ? ?- switch video source only on H.264 keyframes to avoid artifacts when the > stream switches to a new floor member > ? ?- feature to throttle floor changes in conference (i.e. floor is only > changed after some time of activity, to prevent very frequent video stream > changes) > ? ?- investigation whether the codec parameter negotiation can be > influenced/improved for H.264 passthru, e.g. to allow for specific profiles in > SDP a=fmtp parameters > > Greetings, > Roland > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/3021fa2e/attachment-0001.html From jeff at jefflenk.com Thu Jan 12 23:54:06 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 12 Jan 2012 12:54:06 -0800 (PST) Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? (found it !.... 023818bc) In-Reply-To: <1326399625.14157.YahooMailNeo@web29406.mail.ird.yahoo.com> References: <1326399625.14157.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: <1326401646911-7181691.post@n2.nabble.com> Thanks does this mean that you have identified an error in Git? Please open Jira and followup here with the details. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-script-vs-scripts-directories-what-s-the-difference-found-it-023818bc-tp7181590p7181691.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Hector.Geraldino at ip-soft.net Fri Jan 13 00:02:40 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 12 Jan 2012 16:02:40 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507A4D@NY1-EXMB-01.ip-soft.net> I think Avi worked on something similar: http://wiki.fusionpbx.com/index.php?title=Voicemail#Transcribing_Voicemails_with_google_speech_API From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michel Daggelinckx Sent: Thursday, January 12, 2012 3:43 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] speech to text engine found this one on thepiaf forum. mightbe usefull for a freeswitch module http://nerd.bz/xqYO2D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/bb70a7de/attachment.html From msc at freeswitch.org Fri Jan 13 00:18:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Jan 2012 13:18:31 -0800 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: On Thu, Jan 12, 2012 at 12:42 PM, Michel Daggelinckx < daggelinckxmichel at gmail.com> wrote: > found this one on thepiaf forum. > mightbe usefull for a freeswitch module > > http://nerd.bz/xqYO2D > > Yes, this works moderately well. I tinkered with it. One thing I personally found is that I needed to pad my sound file with at least 50ms of silence at the beginning, otherwise Google came up with some really goofy translations on the first couple of words. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/bbab4393/attachment.html From andrew.keil at askinteractive.net Fri Jan 13 00:44:55 2012 From: andrew.keil at askinteractive.net (Andrew Keil) Date: Fri, 13 Jan 2012 08:44:55 +1100 Subject: [Freeswitch-users] Does Freeswitch support Debian 6.0 (squeeze version) Linux on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? Message-ID: To Freeswitch Users, I am looking into purchasing a PIKA WARP Appliance (which currently ships with Asterisk) and try to get Freeswitch to run on it. Quick question, does Freeswitch support Debian 6.0 (squeeze version) Linux on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? If the answer is yes, can you provide a link to what I should download in order for me to build the latest Freeswitch and get it running on this platform. Any comments are also welcome. Thanks in advance for your response. Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/7e83d6fd/attachment.html From avi at avimarcus.net Fri Jan 13 01:14:41 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 13 Jan 2012 00:14:41 +0200 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: If that helps.. can we get sox to easily add that 50ms at the beginning while it converts the file..? Someone know the code for that? Currently, the command is just "sox $file_name.wav $file_name.flac rate 16k" That said, I've had a few good/great transcriptions, and whole bunch of worthless ones. -Avi On Thu, Jan 12, 2012 at 11:18 PM, Michael Collins wrote: > > > On Thu, Jan 12, 2012 at 12:42 PM, Michel Daggelinckx < > daggelinckxmichel at gmail.com> wrote: > >> found this one on thepiaf forum. >> mightbe usefull for a freeswitch module >> >> http://nerd.bz/xqYO2D >> >> > Yes, this works moderately well. I tinkered with it. One thing I > personally found is that I needed to pad my sound file with at least 50ms > of silence at the beginning, otherwise Google came up with some really > goofy translations on the first couple of words. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/cd563931/attachment.html From rhuddleston at gmail.com Fri Jan 13 01:26:34 2012 From: rhuddleston at gmail.com (Robert-IPhone) Date: Thu, 12 Jan 2012 17:26:34 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> Anyone considered looking at SIRI's processing facility? :) Sent from my iPhone 4S On Jan 12, 2012, at 5:14 PM, Avi Marcus wrote: > If that helps.. can we get sox to easily add that 50ms at the beginning while it converts the file..? Someone know the code for that? > Currently, the command is just "sox $file_name.wav $file_name.flac rate 16k" > > That said, I've had a few good/great transcriptions, and whole bunch of worthless ones. > > -Avi > > > On Thu, Jan 12, 2012 at 11:18 PM, Michael Collins wrote: > > > On Thu, Jan 12, 2012 at 12:42 PM, Michel Daggelinckx wrote: > found this one on thepiaf forum. > mightbe usefull for a freeswitch module > > http://nerd.bz/xqYO2D > > > Yes, this works moderately well. I tinkered with it. One thing I personally found is that I needed to pad my sound file with at least 50ms of silence at the beginning, otherwise Google came up with some really goofy translations on the first couple of words. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/1ecc0b95/attachment-0001.html From Hector.Geraldino at ip-soft.net Fri Jan 13 01:43:52 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 12 Jan 2012 17:43:52 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Not that hard (I?m somewhat familiar with the technology), however you need to have a contract with Nuance Corp (which is not precisely a cheap option, btw) From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert-IPhone Sent: Thursday, January 12, 2012 5:27 PM To: FreeSWITCH Users Help Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] speech to text engine Anyone considered looking at SIRI's processing facility? :) Sent from my iPhone 4S On Jan 12, 2012, at 5:14 PM, Avi Marcus > wrote: If that helps.. can we get sox to easily add that 50ms at the beginning while it converts the file..? Someone know the code for that? Currently, the command is just "sox $file_name.wav $file_name.flac rate 16k" That said, I've had a few good/great transcriptions, and whole bunch of worthless ones. -Avi On Thu, Jan 12, 2012 at 11:18 PM, Michael Collins > wrote: On Thu, Jan 12, 2012 at 12:42 PM, Michel Daggelinckx > wrote: found this one on thepiaf forum. mightbe usefull for a freeswitch module http://nerd.bz/xqYO2D Yes, this works moderately well. I tinkered with it. One thing I personally found is that I needed to pad my sound file with at least 50ms of silence at the beginning, otherwise Google came up with some really goofy translations on the first couple of words. -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/cf834af3/attachment.html From msc at freeswitch.org Fri Jan 13 02:46:29 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Jan 2012 15:46:29 -0800 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: On Thu, Jan 12, 2012 at 2:14 PM, Avi Marcus wrote: > If that helps.. can we get sox to easily add that 50ms at the beginning > while it converts the file..? Someone know the code for that? > Currently, the command is just "sox $file_name.wav $file_name.flac rate > 16k" > > AAMOF, I already have the example on the page *YOU* created. :P http://wiki.freeswitch.org/wiki/Transcribing_Voicemail Note the "pad .1 0" args to sox - that adds 100ms to the beginning and 0ms to the end of the sound file being processed. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/030f2437/attachment.html From avi at avimarcus.net Fri Jan 13 02:54:22 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 13 Jan 2012 01:54:22 +0200 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: Message-ID: Great. I'll give that a try! -Avi On Fri, Jan 13, 2012 at 1:46 AM, Michael Collins wrote: > > > On Thu, Jan 12, 2012 at 2:14 PM, Avi Marcus wrote: > >> If that helps.. can we get sox to easily add that 50ms at the beginning >> while it converts the file..? Someone know the code for that? >> Currently, the command is just "sox $file_name.wav $file_name.flac rate >> 16k" >> >> > AAMOF, I already have the example on the page *YOU* created. :P > http://wiki.freeswitch.org/wiki/Transcribing_Voicemail > > Note the "pad .1 0" args to sox - that adds 100ms to the beginning and 0ms > to the end of the sound file being processed. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/11c27b27/attachment.html From rhuddleston at gmail.com Fri Jan 13 03:05:57 2012 From: rhuddleston at gmail.com (Robert-IPhone) Date: Thu, 12 Jan 2012 19:05:57 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Message-ID: Well my point wasn't cost or legal as do we know if Google is open for all use. There are some articles about people using SIRI on Android etc Sent from my iPhone 4S On Jan 12, 2012, at 5:43 PM, Hector Geraldino wrote: > Not that hard (I?m somewhat familiar with the technology), however you need to have a contract with Nuance Corp (which is not precisely a cheap option, btw) > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert-IPhone > Sent: Thursday, January 12, 2012 5:27 PM > To: FreeSWITCH Users Help > Cc: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] speech to text engine > > Anyone considered looking at SIRI's processing facility? :) > > > Sent from my iPhone 4S > > On Jan 12, 2012, at 5:14 PM, Avi Marcus wrote: > > If that helps.. can we get sox to easily add that 50ms at the beginning while it converts the file..? Someone know the code for that? > Currently, the command is just "sox $file_name.wav $file_name.flac rate 16k" > > That said, I've had a few good/great transcriptions, and whole bunch of worthless ones. > > -Avi > > > On Thu, Jan 12, 2012 at 11:18 PM, Michael Collins wrote: > > > On Thu, Jan 12, 2012 at 12:42 PM, Michel Daggelinckx wrote: > found this one on thepiaf forum. > mightbe usefull for a freeswitch module > > http://nerd.bz/xqYO2D > > > Yes, this works moderately well. I tinkered with it. One thing I personally found is that I needed to pad my sound file with at least 50ms of silence at the beginning, otherwise Google came up with some really goofy translations on the first couple of words. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/6698d2f7/attachment-0001.html From brian at freeswitch.org Fri Jan 13 03:23:29 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Jan 2012 18:23:29 -0600 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <1326387102.43177.YahooMailNeo@web29405.mail.ird.yahoo.com> References: <1326387102.43177.YahooMailNeo@web29405.mail.ird.yahoo.com> Message-ID: <5E4B5F77-239A-4376-99D2-C1681D708B74@freeswitch.org> LOOK CLOSER! They are in there! ;) /b On Jan 12, 2012, at 10:51 AM, Bob Smith wrote: > 1/ Timeouts have made it into the tree ? As far as I can tell they have not ? > 2/ Are there perhaps now better way to achieve the same goal ? (e.g. looking at the console, I see a variable generated called valet_ticket ?) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/a53b3f7c/attachment.html From brian at freeswitch.org Fri Jan 13 03:25:23 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Jan 2012 18:25:23 -0600 Subject: [Freeswitch-users] Dial by IP In-Reply-To: References: Message-ID: Use the info app you can tell if the invite domain doesn't equal your hostname and fordward it to the original target URI. /b On Jan 10, 2012, at 10:14 PM, Brian Foster wrote: > I have some phones on my network that are capable of dialing a SIP URI. It > seems by default FreeSWITCH doesn't allow this, as it just picks up > everything before the @ and puts it as the destination number. Is there a > way to allow this? What would be needed in the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/94d4c9e1/attachment.html From brian at freeswitch.org Fri Jan 13 03:29:33 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Jan 2012 18:29:33 -0600 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <5E4B5F77-239A-4376-99D2-C1681D708B74@freeswitch.org> References: <1326387102.43177.YahooMailNeo@web29405.mail.ird.yahoo.com> <5E4B5F77-239A-4376-99D2-C1681D708B74@freeswitch.org> Message-ID: <08E136D4-AC2B-4F69-B3B3-FB7B985A994B@freeswitch.org> Btw here are the variables valet_parking_timeout - Timeout in Seconds valet_parking_orbit_exten - Extensions to transfer to when timeout happens. valet_parking_orbit_dialplan - What dialplan (defaults to current sessions dialplan) valet_parking_orbit_context - What context (defaults to current sessions context) /b On Jan 12, 2012, at 6:23 PM, Brian West wrote: > LOOK CLOSER! They are in there! ;) > > /b > > On Jan 12, 2012, at 10:51 AM, Bob Smith wrote: > >> 1/ Timeouts have made it into the tree ? As far as I can tell they have not ? >> 2/ Are there perhaps now better way to achieve the same goal ? (e.g. looking at the console, I see a variable generated called valet_ticket ?) > From ga at steadfasttelecom.com Fri Jan 13 03:44:21 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Thu, 12 Jan 2012 19:44:21 -0500 Subject: [Freeswitch-users] Voicemail and fax names Message-ID: <6654172324595017257@unknownmsgid> Hi Does anyone know what names the vm and fax files? (DB or FS)? Thanks Sent from my mobile device. From msc at freeswitch.org Fri Jan 13 03:53:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 12 Jan 2012 16:53:24 -0800 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Message-ID: On Thu, Jan 12, 2012 at 4:05 PM, Robert-IPhone wrote: > Well my point wasn't cost or legal as do we know if Google is open for all > use. > > There are some articles about people using SIRI on Android etc > I'm curious to know if there is a generic way to make a request of SIRI to do speech to text... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/156588af/attachment.html From freeswitch at earthspike.net Fri Jan 13 05:06:58 2012 From: freeswitch at earthspike.net (John) Date: Fri, 13 Jan 2012 02:06:58 +0000 Subject: [Freeswitch-users] Voicemail and fax names In-Reply-To: <6654172324595017257@unknownmsgid> References: <6654172324595017257@unknownmsgid> Message-ID: <4F0F91C2.7080500@earthspike.net> On 13/01/12 00:44, Gilad Abada wrote: > Hi > > Does anyone know what names the vm and fax files? (DB or FS)? > > Thanks > FS. John From ga at steadfasttelecom.com Fri Jan 13 05:38:43 2012 From: ga at steadfasttelecom.com (Gilad Abada) Date: Thu, 12 Jan 2012 21:38:43 -0500 Subject: [Freeswitch-users] Voicemail and fax names In-Reply-To: <4F0F91C2.7080500@earthspike.net> References: <6654172324595017257@unknownmsgid> <4F0F91C2.7080500@earthspike.net> Message-ID: <-9042044334272622947@unknownmsgid> Thanks for the response John. Anyway to get the db to name them? Are they named by UUID? Sent from my mobile device. On Jan 12, 2012, at 9:34 PM, John wrote: > On 13/01/12 00:44, Gilad Abada wrote: >> Hi >> >> Does anyone know what names the vm and fax files? (DB or FS)? >> >> Thanks >> > FS. > > John > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rhuddleston at gmail.com Fri Jan 13 06:19:13 2012 From: rhuddleston at gmail.com (Robert-IPhone) Date: Thu, 12 Jan 2012 22:19:13 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Message-ID: I'll do some research - course SIRI is only happy with American English reliably ;) Sent from my iPhone 4S On Jan 12, 2012, at 7:53 PM, Michael Collins wrote: > > > On Thu, Jan 12, 2012 at 4:05 PM, Robert-IPhone wrote: > Well my point wasn't cost or legal as do we know if Google is open for all use. > > There are some articles about people using SIRI on Android etc > > I'm curious to know if there is a generic way to make a request of SIRI to do speech to text... > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/07e5c5e6/attachment.html From bdfoster at endigotech.com Fri Jan 13 07:15:24 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 12 Jan 2012 23:15:24 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Message-ID: Siri itself is not available on Android. There was an app that claimed to be Siri for Android, published by 'Official App' but it was taken down by Google. It was reported to be using Google's API. Google's API for transcription from what I can se is supposed to be used for Android OS only. It's not really clear if it can be used for other purposes. Sirius doesn't have an external API for us to use (no surprise there) and it most likely is using Nuance for transcription. I've tried out the script on a few vm files I have. It doesn't really work well for mobile calls. It worked extremely well on some pre recorded messages I made (the number you have reached is not in service...) and it worked really well. Maybe Google likes me? -BD On Jan 12, 2012 10:25 PM, "Robert-IPhone" wrote: > I'll do some research - course SIRI is only happy with American English > reliably > ;) > > Sent from my iPhone 4S > > On Jan 12, 2012, at 7:53 PM, Michael Collins wrote: > > > > On Thu, Jan 12, 2012 at 4:05 PM, Robert-IPhone wrote: > >> Well my point wasn't cost or legal as do we know if Google is open for >> all use. >> >> There are some articles about people using SIRI on Android etc >> > > I'm curious to know if there is a generic way to make a request of SIRI to > do speech to text... > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/febea805/attachment-0001.html From yehavi.bourvine at gmail.com Fri Jan 13 07:39:56 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 13 Jan 2012 06:39:56 +0200 Subject: [Freeswitch-users] script vs scripts directories, what's the difference ? In-Reply-To: <1326399201.87367.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326399201.87367.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: I think it was for a specific checkout. A while ago I did checkout, built it, and then had problems running Freeswitch. I noticed in the logs that it does not find my scripts as it looks for them at script and not scripts directory. A checkout from a few hours later was ok. I didn't open a JIRA as it has been fixed already... __Yehavi: 2012/1/12 Bob Smith > Seems weird, as "script" has a creation date/time exactly the same as my > grammar,htdocs,recordings directories (Dec 16 23:04) > Either way it's empty.... so the only thing I can think of is it might > have been something to that specific checkout. > Seeing as it's empty I might as well rm' it then. > > Thanks > > Bob > > >I'm pretty sure that "script" is not created by the freeswitch install > >process. I know for a fact that "scripts" is and it's the default path for > >lua, perl, js, etc. scripts called from the cmd line or dialplan. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/75756cac/attachment.html From jmesquita at freeswitch.org Fri Jan 13 07:41:54 2012 From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org) Date: Fri, 13 Jan 2012 01:41:54 -0300 Subject: [Freeswitch-users] Does Freeswitch support Debian 6.0 (squeeze version) Linux on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? In-Reply-To: References: Message-ID: <0CC3D3AC-D1B2-4E7F-B605-6D96F083881A@freeswitch.org> I managed to compile and run freeswitch on an old warp device I have with me a long time ago. I would imagine that you wouldn't have too much problem compiling it again even more if it is a newer version of the device. What I was not able to do is to get openzap(freetdm did not exist at the time) to work with pika boards. They don't seem to care either. Regards, Jo?o Mesquita On 12/01/2012, at 06:44 p.m., Andrew Keil wrote: > To Freeswitch Users, > > I am looking into purchasing a PIKA WARP Appliance (which currently ships with Asterisk) and try to get Freeswitch to run on it. > > Quick question, does Freeswitch support Debian 6.0 (squeeze version) Linux on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? If the answer is yes, can you provide a link to what I should download in order for me to build the latest Freeswitch and get it running on this platform. > > Any comments are also welcome. > > Thanks in advance for your response. > > Andrew Keil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/22c79c05/attachment.html From lakindia89 at gmail.com Fri Jan 13 08:09:54 2012 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 13 Jan 2012 10:39:54 +0530 Subject: [Freeswitch-users] Does Freeswitch support Debian 6.0 (squeeze version) Linux on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? In-Reply-To: References: Message-ID: We use Debian Squeeze only. FreeSWITCH runs on it without any issues. But we are using sangoma hardware. On Fri, Jan 13, 2012 at 3:14 AM, Andrew Keil wrote: > To Freeswitch Users,**** > > ** ** > > I am looking into purchasing a PIKA WARP Appliance (which currently ships > with Asterisk) and try to get Freeswitch to run on it.**** > > ** ** > > Quick question, does Freeswitch support Debian 6.0 (squeeze version) Linux > on the PowerPC (AMCC Power PC 440EP Embedded 533 MHz Processor)? If the > answer is yes, can you provide a link to what I should download in order > for me to build the latest Freeswitch and get it running on this platform. > **** > > ** ** > > Any comments are also welcome.**** > > ** ** > > Thanks in advance for your response.**** > > ** ** > > Andrew Keil**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e466de49/attachment.html From mayamatakeshi at gmail.com Fri Jan 13 09:07:26 2012 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 13 Jan 2012 15:07:26 +0900 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: <4F0582AA.9060708@gmx.net> References: <4F0582AA.9060708@gmx.net> Message-ID: On Thu, Jan 5, 2012 at 7:59 PM, Peter P GMX wrote: > Hello, > > I have a strange phenomen: > > When a target UA is busy, it returns "486 Busy" to Freeswitch. But > Freeswitch then returns "480 Temporarily Unavailable" to the called party. > Where does this come from and how can I change this behaviour? > > See (anonymized) SIP trace with ngrep: > > UA to Freeswitch: > ======================== > U 2012/01/04 13:59:44.928775 :5060 -> :5080 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP > :5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. > From: "026xxxxxxxx" >;tag=py094Kv7vr03a. > To: > ;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. > Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. > CSeq: 22504928 INVITE. > User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). > Content-Length: 0. > > Freeswitch to Caller: > ======================== > U 2012/01/04 13:59:44.930387 :5060 -> :5060 > SIP/2.0 480 Temporarily Unavailable. > Via: SIP/2.0/UDP :5060;branch=z9hG4bK-4896-2830DFA. > From: > ;user=phone>;tag=13517-HB-08a98588-2622da197. > To: ;user=phone>;tag=XQtc5US24QgDa. > Call-ID: 13517-SG-08a98587-0a352e121 at sip.provider.de. > CSeq: 134781549 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 > 13-52-28 -0600. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: precondition, path, replaces. > Allow-Events: talk, hold, refer. > Content-Length: 0. > P-Asserted-Identity: "069xxxxxxxx" >. > > Best regards > Peter > I believe this is a bug. I have opened a jira ticket: http://jira.freeswitch.org/browse/FS-3810 regards, Takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/8baa15cf/attachment.html From hynek.cihlar at gmail.com Fri Jan 13 09:48:11 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 13 Jan 2012 07:48:11 +0100 Subject: [Freeswitch-users] Empty WAV file recorded Message-ID: Hi all, an attempt to uuid_record ends with wav file with no data only header. ct The scenario is following: 0. FS <---> ESL app 1. Bridge is established on one incoming and one originated channel. One channel hangs up. 2. uuid_record start is issued on the active channel. 3. uuid_record stop is issued on the active channel. 4. The result is a wav file with always the same length of 44 bytes. When recording to mp3, the file is always correctly created. Any idea how to diagnose the problem? Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/185032ca/attachment-0001.html From hynek.cihlar at gmail.com Fri Jan 13 09:53:00 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 13 Jan 2012 07:53:00 +0100 Subject: [Freeswitch-users] MP3 playback sounds bad in Chrome, encoding issue? In-Reply-To: References: Message-ID: Hello Eli, did you get any further with this issue? I get the same behavior on my system. Could this be maybe related to the versions of the core libs? Hynek On Tue, Jan 3, 2012 at 3:16 PM, Eli Finkelman wrote: > Hi There, > > Whenever we playback an MP3 recording using the native Chrome MP3 player, > it sounds really horrible, it jumps and crackles. When listening to it in > Firefox/Safari it sounds just fine. Could there be an issue with the way FS > is encoding the files? Is this just an issue with Chrome? > > We're using the native record_session method to record the MP3. > > switch_event.c:1521 Parsing variable [execute_on_answer]=[record_session > /tmp/recordings/RE91fec223f65143e2aa5246b1f3a071a8.mp3] > > Here's a sample of what the audio sounds like, try playing this in Chrome > in Firefox > > > http://recordings.telapi.com/RB1cd90a0383114455830be5755c1891e6/RE91fec223f65143e2aa5246b1f3a071a8.mp3 > > Best Regards > Eli > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e9fd65b0/attachment.html From hynek.cihlar at gmail.com Fri Jan 13 10:04:33 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 13 Jan 2012 08:04:33 +0100 Subject: [Freeswitch-users] Empty WAV file recorded In-Reply-To: References: Message-ID: I was wrong, the same problem is when recording to mp3. Tested on HEAD. It worked ok on version 4-or-so months back. Hynek On Fri, Jan 13, 2012 at 7:48 AM, Hynek Cihlar wrote: > Hi all, > > an attempt to uuid_record ends with wav file with no data only header. > ct > The scenario is following: > 0. FS <---> ESL app > 1. Bridge is established on one incoming and one originated channel. One > channel hangs up. > 2. uuid_record start is issued on the active channel. > 3. uuid_record stop is issued on the active channel. > 4. The result is a wav file with always the same length of 44 bytes. > > When recording to mp3, the file is always correctly created. > > Any idea how to diagnose the problem? > > Hynek > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/0e2739df/attachment.html From lists at redbonez.net Fri Jan 13 04:51:59 2012 From: lists at redbonez.net (Adam Ford) Date: Thu, 12 Jan 2012 18:51:59 -0700 Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] In-Reply-To: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> References: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> Message-ID: <017001ccd195$f1d9b210$d58d1630$@redbonez.net> I am still having issues with MANDATORY_IE_MISSING on incoming calls when using FreeTDM + libpri + DAHDI + foneBridge2. Can anyone help me figure out if this is a configuration issue or a bug/incompatibility? As stated before, I am running the latest git trunk with all default/stock settings with the exception of FreeTDM configuration and minor modification to the dialplan to pass my DID to the default extension 1001. Outgoing calls are working great. Below is the debug log from FreeSWITCH with libpri debugging enabled (highlighted in RED when the call appears to start failing) - Dialplan: FreeTDM/1:1/5530 Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: FreeTDM/1:1/5530 Action export(dialed_extension=1001) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strft ime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: FreeTDM/1:1/5530 Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: FreeTDM/1:1/5530 Action set(ringback=${us-ring}) Dialplan: FreeTDM/1:1/5530 Action set(transfer_ringback=local_stream://moh) Dialplan: FreeTDM/1:1/5530 Action set(call_timeout=30) Dialplan: FreeTDM/1:1/5530 Action set(hangup_after_bridge=true) Dialplan: FreeTDM/1:1/5530 Action set(continue_on_fail=true) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: FreeTDM/1:1/5530 Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: FreeTDM/1:1/5530 Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: FreeTDM/1:1/5530 Action answer() Dialplan: FreeTDM/1:1/5530 Action sleep(1000) Dialplan: FreeTDM/1:1/5530 Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:154 (FreeTDM/1:1/5530) State Change CS_ROUTING -> CS_EXECUTE 2012-01-09 18:31:23.487064 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:410 (FreeTDM/1:1/5530) State ROUTING going to sleep 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_EXECUTE 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE 2012-01-09 18:31:23.487064 [DEBUG] mod_freetdm.c:478 FreeTDM/1:1/5530 CHANNEL EXECUTE 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:192 FreeTDM/1:1/5530 Standard EXECUTE EXECUTE FreeTDM/1:1/5530 set(open=true) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [open]=[true] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-spymap/**********/ce58b396-3b2a-11e1-b7d4-9d287cd8d dec) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/**********/1001) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/global/ce58b396-3b2a-11e1-b7d4-9d287cd8dd ec) EXECUTE FreeTDM/1:1/5530 set(RFC2822_DATE=Mon, 09 Jan 2012 18:31:23 -0700) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [RFC2822_DATE]=[Mon, 09 Jan 2012 18:31:23 -0700] EXECUTE FreeTDM/1:1/5530 export(dialed_extension=1001) 2012-01-09 18:31:23.487064 [DEBUG] switch_channel.c:1091 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE FreeTDM/1:1/5530 bind_meta_app(1 b s execute_extension::dx XML features) 2012-01-09 18:31:23.487064 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/**********.2012-01-09-18-31 -23.wav) 2012-01-09 18:31:23.487064 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/**********.2012-01-09-18-31 -23.wav EXECUTE FreeTDM/1:1/5530 bind_meta_app(3 b s execute_extension::cf XML features) 2012-01-09 18:31:23.487064 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE FreeTDM/1:1/5530 bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-01-09 18:31:23.487064 [INFO] switch_ivr_async.c:3179 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE FreeTDM/1:1/5530 set(ringback=%(2000,4000,440,480)) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [ringback]=[%(2000,4000,440,480)] EXECUTE FreeTDM/1:1/5530 set(transfer_ringback=local_stream://moh) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [transfer_ringback]=[local_stream://moh] EXECUTE FreeTDM/1:1/5530 set(call_timeout=30) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [call_timeout]=[30] EXECUTE FreeTDM/1:1/5530 set(hangup_after_bridge=true) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [hangup_after_bridge]=[true] EXECUTE FreeTDM/1:1/5530 set(continue_on_fail=true) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [continue_on_fail]=[true] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-call_return/1001/**********) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/1001/ce58b396-3b2a-11e1-b7d4-9d287cd8 ddec) EXECUTE FreeTDM/1:1/5530 set(called_party_callgroup=techsupport) 2012-01-09 18:31:23.487064 [DEBUG] mod_dptools.c:1281 FreeTDM/1:1/5530 SET [called_party_callgroup]=[techsupport] EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/techsupport/ce58b396-3b2a-11e1-b7d4-9 d287cd8ddec) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial_ext/global/ce58b396-3b2a-11e1-b7d4-9d287c d8ddec) EXECUTE FreeTDM/1:1/5530 hash(insert/xx.xx.xx.xxx-last_dial/techsupport/ce58b396-3b2a-11e1-b7d4-9d287 cd8ddec) EXECUTE FreeTDM/1:1/5530 bridge(user/1001 at xx.xx.xx.xxx) 2012-01-09 18:31:23.487064 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-09 18:31:23.487064 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-09 18:31:23.487064 [DEBUG] switch_channel.c:1045 FreeTDM/1:1/5530 EXPORTING[export_vars] [dialed_extension]=[1001] to event 2012-01-09 18:31:23.487064 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-09 18:31:23.487064 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[xx.xx.xx.xxx] 2012-01-09 18:31:23.487064 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at xx.xx.xx.xxx] 2012-01-09 18:31:23.487064 [NOTICE] switch_channel.c:924 New Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [ce61ab90-3b2a-11e1-b7db-9d287cd8ddec] 2012-01-09 18:31:23.487064 [DEBUG] mod_sofia.c:4674 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State Change CS_NEW -> CS_INIT 2012-01-09 18:31:23.487064 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_INIT 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State INIT 2012-01-09 18:31:23.487064 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 SOFIA INIT 2012-01-09 18:31:23.487064 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State Change CS_INIT -> CS_ROUTING 2012-01-09 18:31:23.487064 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State INIT going to sleep 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_ROUTING 2012-01-09 18:31:23.487064 [DEBUG] switch_channel.c:1884 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Callstate Change DOWN -> RINGING 2012-01-09 18:31:23.487064 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State ROUTING 2012-01-09 18:31:23.487064 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 SOFIA ROUTING 2012-01-09 18:31:23.487064 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-01-09 18:31:23.487064 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State ROUTING going to sleep 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_CONSUME_MEDIA 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State CONSUME_MEDIA 2012-01-09 18:31:23.487064 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State CONSUME_MEDIA going to sleep 2012-01-09 18:31:23.487064 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 entering state [calling][0] 2012-01-09 18:31:23.566904 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.566904 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.586909 [DEBUG] sofia.c:5482 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 entering state [proceeding][180] 2012-01-09 18:31:23.586909 [NOTICE] sofia.c:5574 Ring-Ready sofia/internal/sip:1001 at xx.xx.xx.xxx:60440! 2012-01-09 18:31:23.586909 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Indicating PROGRESS_MEDIA in state PROCEED 2012-01-09 18:31:23.586909 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROCEED to PROGRESS 2012-01-09 18:31:23.586909 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS] 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROCEED to PROGRESS in 13ms 2012-01-09 18:31:23.586909 [ERR] ftmod_libpri.c:132 XXX Progress message requested but no information is provided 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > DL-DATA request 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=5 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > Message Type: PROGRESS (3) 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=5 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.586909 [DEBUG] ftmod_libpri.c:150 > Message Type: PROGRESS (3) 2012-01-09 18:31:23.586909 [DEBUG] mod_freetdm.c:970 [s1c1][1:1] Changed state from PROGRESS to PROGRESS_MEDIA 2012-01-09 18:31:23.646919 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for PROGRESS_MEDIA 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [PROGRESS_MEDIA] 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS to PROGRESS_MEDIA in 55ms 2012-01-09 18:31:23.646919 [INFO] ftmod_zt.c:656 Setting echo cancel to 64 taps for 1:1 2012-01-09 18:31:23.646919 [WARNING] ftmod_zt.c:661 Echo cancel not available for 1:1 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > DL-DATA request 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=9 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Message Type: PROGRESS (3) 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=9 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Message Type: PROGRESS (3) 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > [1e 02 81 88] 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) 2012-01-09 18:31:23.646919 [DEBUG] ftmod_libpri.c:150 > Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] 2012-01-09 18:31:23.646919 [DEBUG] switch_core_session.c:729 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-09 18:31:23.646919 [NOTICE] switch_ivr_originate.c:1115 Pre-Answer FreeTDM/1:1/5530! 2012-01-09 18:31:23.646919 [DEBUG] switch_channel.c:2930 (FreeTDM/1:1/5530) Callstate Change RINGING -> EARLY 2012-01-09 18:31:23.646919 [DEBUG] switch_ivr_originate.c:1164 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2012-01-09 18:31:23.646919 [DEBUG] switch_core_codec.c:116 FreeTDM/1:1/5530 Push codec L16:70 2012-01-09 18:31:23.646919 [DEBUG] switch_ivr_originate.c:1227 Play Ringback Tone [%(2000,4000,440,480)] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < Protocol Discriminator: Q.931 (8) len=10 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent from originator) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < Message Type: DISCONNECT (69) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < [08 03 80 e0 1e] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 < Cause data 1: 1e (30) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 Received message for call 0x9ff3790 on link 0x9f8403c TEI/SAPI 0/0 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 -- Processing IE 8 (cs0, Cause) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 -- Found active call: 0x9ff3790 cref:5073 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 q931.c:8707 post_handle_q931_message: Call 5073 enters state 12 (Disconnect Indication). Hold state: Idle 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:1065 -- Hangup REQ on channel 1:1 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 q931.c:6837 q931_hangup: Hangup other cref:5073 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 q931.c:6594 __q931_hangup: ourstate Disconnect Indication, peerstate Disconnect Request, hold-state Idle 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 q931.c:5703 q931_release: Call 5073 enters state 19 (Release Request). Hold state: Idle 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > DL-DATA request 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=9 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Message Type: RELEASE (77) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Protocol Discriminator: Q.931 (8) len=9 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent to originator) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Message Type: RELEASE (77) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > [08 02 81 e0] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:150 > Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:1078 [s1c1][1:1] Changed state from PROGRESS_MEDIA to TERMINATING 2012-01-09 18:31:23.726904 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for TERMINATING 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [TERMINATING] 2012-01-09 18:31:23.726904 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from PROGRESS_MEDIA to TERMINATING in 0ms 2012-01-09 18:31:23.726904 [DEBUG] ftdm_io.c:5565 [s1c1][1:1] Scheduling safety hangup timer 2012-01-09 18:31:23.726904 [DEBUG] mod_freetdm.c:2416 got clear channel sig [STOP] 2012-01-09 18:31:23.726904 [DEBUG] switch_channel.c:2846 (FreeTDM/1:1/5530) Callstate Change EARLY -> HANGUP 2012-01-09 18:31:23.726904 [NOTICE] mod_freetdm.c:2441 Hangup FreeTDM/1:1/5530 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2012-01-09 18:31:23.726904 [DEBUG] switch_channel.c:2869 Send signal FreeTDM/1:1/5530 [KILL] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_codec.c:141 FreeTDM/1:1/5530 Restore previous codec PCMU:0. 2012-01-09 18:31:23.726904 [DEBUG] switch_channel.c:2846 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Callstate Change RINGING -> HANGUP 2012-01-09 18:31:23.726904 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-01-09 18:31:23.726904 [DEBUG] switch_channel.c:2869 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [KILL] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_HANGUP 2012-01-09 18:31:23.726904 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-01-09 18:31:23.726904 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2012-01-09 18:31:23.726904 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2012-01-09 18:31:23.726904 [INFO] mod_dptools.c:2900 Originate Failed. Cause: ORIGINATOR_CANCEL 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:2285 FreeTDM/1:1/5530 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:417 (FreeTDM/1:1/5530) State EXECUTE going to sleep 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_HANGUP 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State HANGUP 2012-01-09 18:31:23.726904 [DEBUG] mod_sofia.c:469 Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 hanging up, cause: ORIGINATOR_CANCEL 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP 2012-01-09 18:31:23.726904 [DEBUG] mod_freetdm.c:530 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP ENTER 2012-01-09 18:31:23.726904 [DEBUG] mod_freetdm.c:605 [s1c1][1:1] Changed state from TERMINATING to HANGUP 2012-01-09 18:31:23.726904 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State HANGUP going to sleep 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State Change CS_HANGUP -> CS_REPORTING 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_REPORTING 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State REPORTING 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State REPORTING going to sleep 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State Change CS_REPORTING -> CS_DESTROY 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [BREAK] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_session.c:1380 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Locked, Waiting on external entities 2012-01-09 18:31:23.726904 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Ended 2012-01-09 18:31:23.726904 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 [CS_DESTROY] 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Callstate Change HANGUP -> DOWN 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) Running State Change CS_DESTROY 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State DESTROY 2012-01-09 18:31:23.726904 [DEBUG] mod_sofia.c:374 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 SOFIA DESTROY 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at xx.xx.xx.xxx:60440 Standard DESTROY 2012-01-09 18:31:23.726904 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at xx.xx.xx.xxx:60440) State DESTROY going to sleep 2012-01-09 18:31:23.786903 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP] 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from TERMINATING to HANGUP in 52ms 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:150 q931.c:6837 q931_hangup: Hangup other cref:5073 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:150 q931.c:6594 __q931_hangup: ourstate Release Request, peerstate Disconnect Request, hold-state Idle 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:929 [s1c1][1:1] Changed state from HANGUP to HANGUP_COMPLETE 2012-01-09 18:31:23.786903 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for HANGUP_COMPLETE 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [HANGUP_COMPLETE] 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP to HANGUP_COMPLETE in 0ms 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:939 [s1c1][1:1] Changed state from HANGUP_COMPLETE to DOWN 2012-01-09 18:31:23.786903 [DEBUG] ftdm_state.c:511 [s1c1][1:1] Executing state processor for DOWN 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:679 -- 1:1 STATE [DOWN] 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:687 [s1c1][1:1] Completed state change from HANGUP_COMPLETE to DOWN in 0ms 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:150 Destroying call 0x9ff3790, ourstate Release Request, peerstate Disconnect Request, hold-state Idle 2012-01-09 18:31:23.786903 [DEBUG] ftdm_io.c:2930 [s1c1][1:1] DTMF debug is already disabled 2012-01-09 18:31:23.786903 [DEBUG] ftdm_io.c:2962 [s1c1][1:1] No need to disable input dump 2012-01-09 18:31:23.786903 [DEBUG] ftdm_io.c:2993 [s1c1][1:1] No need to disable output dump 2012-01-09 18:31:23.786903 [DEBUG] mod_freetdm.c:2416 got clear channel sig [RELEASED] 2012-01-09 18:31:23.786903 [DEBUG] ftdm_io.c:6185 Cleared call with id 1 2012-01-09 18:31:23.786903 [DEBUG] ftdm_io.c:2735 [s1c1][1:1] channel done 2012-01-09 18:31:23.786903 [DEBUG] ftmod_libpri.c:704 -- Closed channel 1:1 2012-01-09 18:31:23.786903 [DEBUG] mod_freetdm.c:624 [1:1] FreeTDM/1:1/5530 CHANNEL HANGUP EXIT 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:47 FreeTDM/1:1/5530 Standard HANGUP, cause: MANDATORY_IE_MISSING 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:602 (FreeTDM/1:1/5530) State HANGUP going to sleep 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:393 (FreeTDM/1:1/5530) State Change CS_HANGUP -> CS_REPORTING 2012-01-09 18:31:23.786903 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/1:1/5530) Running State Change CS_REPORTING 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:79 FreeTDM/1:1/5530 Standard REPORTING, cause: MANDATORY_IE_MISSING 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:662 (FreeTDM/1:1/5530) State REPORTING going to sleep 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:387 (FreeTDM/1:1/5530) State Change CS_REPORTING -> CS_DESTROY 2012-01-09 18:31:23.786903 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/5530 [BREAK] 2012-01-09 18:31:23.786903 [DEBUG] switch_core_session.c:1380 Session 1 (FreeTDM/1:1/5530) Locked, Waiting on external entities 2012-01-09 18:31:23.786903 [NOTICE] switch_core_session.c:1398 Session 1 (FreeTDM/1:1/5530) Ended 2012-01-09 18:31:23.786903 [NOTICE] switch_core_session.c:1400 Close Channel FreeTDM/1:1/5530 [CS_DESTROY] 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:491 (FreeTDM/1:1/5530) Callstate Change HANGUP -> DOWN 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:494 (FreeTDM/1:1/5530) Running State Change CS_DESTROY 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:86 FreeTDM/1:1/5530 Standard DESTROY 2012-01-09 18:31:23.786903 [DEBUG] switch_core_state_machine.c:504 (FreeTDM/1:1/5530) State DESTROY going to sleep 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 < Protocol Discriminator: Q.931 (8) len=5 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 < TEI=0 Call Ref: len= 2 (reference 5073/0x13D1) (Sent from originator) 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 < Message Type: RELEASE COMPLETE (90) 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 -- Making new call for cref 5073 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 Received message for call 0xb6a10b50 on link 0x9f8403c TEI/SAPI 0/0 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 q931.c:6837 q931_hangup: Hangup other cref:5073 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 q931.c:6594 __q931_hangup: ourstate Null, peerstate Null, hold-state Idle 2012-01-09 18:31:23.826906 [DEBUG] ftmod_libpri.c:150 Destroying call 0xb6a10b50, ourstate Null, peerstate Null, hold-state Idle I've posted the full log (from starting FreeSWITCH to incoming call failure) at - http://pastebin.com/QaBqbCgU You can see my configuration files in the original e-mail below. Thanks in advance if anyone is able to help. This is my last attempt before settling with FreeSWITCH 1.0.6 + OpenZAP(already working in production) instead of lastest + FreeTDM. -Adam From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adam Ford Sent: Sunday, January 08, 2012 6:33 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] I can't seem to find any info on the problem I am having by searching the archives, so I apologize if this has been answered in the past (found several about MANDATORY_IE_MISSING but they were different situations). I am trying to setup FreeSWITCH using a FreeTDM + libpri + DAHDI + foneBridge2 stack. Outgoing calls work great, but I am running into the 'MANDATORY_IE_MISSING' problem with incoming calls. I am running the latest git version as of this morning, and completely default configuration with the exception of FreeTDM/DAHDI configuration and a modification of the default inbound_did dialplan to pass my DID 5530 to the default extension 1001. Below is what I get in the log, I highlighted in RED as soon as the call appears to start failing - freeswitch/conf/freetdm.conf - [span zt PRI] trunk_type => T1 b-channel=1-23 d-channel=24 freeswitch/conf/autoload_confg/freetdm.conf.xml - /etc/dahdi/system.conf - loadzone = us defaultzone=us dynamic=ethmf,eth0/00:50:c2:65:d7:59/0,24,1 bchan=1-23 dchan=24 I am guessing it is a configuration issue, though this same config is currently working in production with FreeSWITCH 1.0.6 + OpenZAP + Libpri + DAHDI + foneBridge2. Any help is greatly appreciated. -Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120112/079cea99/attachment-0001.html From gcd at i.ph Fri Jan 13 10:16:11 2012 From: gcd at i.ph (Nandy Dagondon) Date: Fri, 13 Jan 2012 15:16:11 +0800 Subject: [Freeswitch-users] Help needing translating Asterisk syntax to Freeswitch In-Reply-To: <1326357997.76619.YahooMailNeo@web29406.mail.ird.yahoo.com> References: <1326357997.76619.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: hi bob, i found in http://www.voip-info.org/wiki/view/Asterisk+SIP+register this sip.conf directive: register => user[:secret[:authuser]]@host[:port][/extension] if i'm correct, there's NO equivalent of "extension" in FS. so i would try: hope it helps. On Thu, Jan 12, 2012 at 4:46 PM, Bob Smith wrote: > Problem is Michael that I don't have an Asterisk box, that config line was > from some instructions on their website where they document an Asterisk > setup. > > I'd rather not download Asterisk and try to get to grips with something > that I'll never have a need to use again, since getting to grips with > FreeSwitch is keeping me busy enough ! > > > >I would do is get a siptrace from it working on your Asterisk box and > compare that with the sip trace from the freeswitch box. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e2724a56/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Fri Jan 13 11:17:21 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Fri, 13 Jan 2012 08:17:21 +0000 (GMT) Subject: [Freeswitch-users] valet_park timeout Message-ID: <1326442641.87954.YahooMailNeo@web29401.mail.ird.yahoo.com> Thank you Brian. Although I appreciate the constant development of FreeSwitch, I don't think it's too much to ask that the developers should keep the Wiki pages up to date in respect of features that have been added and/or removed. ?There is no mention of those parameters on?http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park. ?;-( From avi at avimarcus.net Fri Jan 13 11:42:39 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 13 Jan 2012 10:42:39 +0200 Subject: [Freeswitch-users] Help needing translating Asterisk syntax to Freeswitch In-Reply-To: References: <1326357997.76619.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: "If no extension is given, the 's' extension is used. The extension needs to be defined in extensions.conf to be able to accept calls from this SIP provider." That sounds like "when we get an incoming call, which extension should it go to?" Looks like the equivalent would be: -Avi On Fri, Jan 13, 2012 at 9:16 AM, Nandy Dagondon wrote: > hi bob, > > i found in http://www.voip-info.org/wiki/view/Asterisk+SIP+register this > sip.conf directive: > > register => user[:secret[:authuser]]@host[:port][/extension] > > if i'm correct, there's NO equivalent of "extension" in FS. so i would try: > > > > > > > > > > > > > > hope it helps. > > > On Thu, Jan 12, 2012 at 4:46 PM, Bob Smith < > gb10hkzo-freeswitch at yahoo.co.uk> wrote: > >> Problem is Michael that I don't have an Asterisk box, that config line >> was from some instructions on their website where they document an Asterisk >> setup. >> >> I'd rather not download Asterisk and try to get to grips with something >> that I'll never have a need to use again, since getting to grips with >> FreeSwitch is keeping me busy enough ! >> >> >> >I would do is get a siptrace from it working on your Asterisk box and >> compare that with the sip trace from the freeswitch box. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/b8ded772/attachment.html From thomas at chaschperli.ch Fri Jan 13 11:47:41 2012 From: thomas at chaschperli.ch (Thomas Mueller) Date: Fri, 13 Jan 2012 09:47:41 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> Message-ID: <4F0FEFAD.3070307@chaschperli.ch> On 12.01.2012 18:17, Giovanni Maruzzelli wrote: > On Thu, Jan 12, 2012 at 12:59 PM, Thomas Mueller wrote: >> hi giovanni >> >> here a translation by me: > So much better than Google! > > Congratulations again, and please consider to make available an > English version of it, a very basic and rough translation can do, then > the community can do the finesses. > > I was attending Cluecon 2011, and the presentation about Gemeinweisen > was very well received and there was lot of interest in the audience. seems an english version of Gemeinschaft is not far away. they twittered that the english release is expected in March ( http://twitter.com/GemeinschaftPBX) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e8537615/attachment.html From gmaruzz at gmail.com Fri Jan 13 12:15:31 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 13 Jan 2012 10:15:31 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F0FEFAD.3070307@chaschperli.ch> References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: On Fri, Jan 13, 2012 at 9:47 AM, Thomas Mueller wrote: > > seems an english version of Gemeinschaft is not far away. they twittered > that the english release is expected in March ( > http://twitter.com/GemeinschaftPBX) Super! I'll be back from vacation just in time to test it ;). -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From freeswitch at earthspike.net Fri Jan 13 12:38:56 2012 From: freeswitch at earthspike.net (John) Date: Fri, 13 Jan 2012 09:38:56 +0000 Subject: [Freeswitch-users] Voicemail and fax names In-Reply-To: <-9042044334272622947@unknownmsgid> References: <6654172324595017257@unknownmsgid> <4F0F91C2.7080500@earthspike.net> <-9042044334272622947@unknownmsgid> Message-ID: <4F0FFBB0.9010409@earthspike.net> On 13/01/12 02:38, Gilad Abada wrote: > Thanks for the response John. Anyway to get the db to name them? Are > they named by UUID? > > Sent from my mobile device. > > On Jan 12, 2012, at 9:34 PM, John wrote: > >> On 13/01/12 00:44, Gilad Abada wrote: >>> Hi >>> >>> Does anyone know what names the vm and fax files? (DB or FS)? >>> >>> Thanks >>> >> FS. >> Not sure what your point or purpose is. Filenames are chosen by the devs in the FS code, and vary depending on context. VM files are named by UUID, as seen in the src for mod_voicemail.c (this instance is for the emailed VM file): new_file_path = switch_core_sprintf(pool, "%s%smsg_%s.wav", SWITCH_GLOBAL_dirs.temp_dir, SWITCH_PATH_SEPARATOR, tmp_uuid_str); The DB, presumably, (I haven't looked) links the filename with the date/time of the call. The files are in /db/ and you can examine their structure yourself with the sqlite3 command line tool, or similar. John From anton.vazir at gmail.com Fri Jan 13 14:14:54 2012 From: anton.vazir at gmail.com (Anton VG) Date: Fri, 13 Jan 2012 16:14:54 +0500 Subject: [Freeswitch-users] Unable to get rtp and pre_answer working while looping back via sip (Sofia Bug or I miss something?) Message-ID: Hi Gentlemen! Broke my brain trying to force my FS, controlled by ESL inbound to give me early media in indirect calling scheme. Please help: When call comes, in some cases, I do not just originate destination, but need to dial on behalf of another user (not the user which originally dialed in). For this I dial back my FS box via sip, setting apropriate SIP variables, so FS originates another call, as if the user I need originated a call, and so so I have extra CDR and so on. BUT there is a problem:If I dial like this, - EARLY MEDIA does not work. 180 and 183 are send correctly and in time. however, all works if I dial directly (dialing schemes are given below) tcpdumping shows that there is no RTP coming to my FS box when I dial doing one loop. BUT - RTP appears immediately after answering the inbound channel: (regardless if dialed destination answered or not) - so dialing without the loop and doing pre_answer - I get RTP OK, if after a looped call i do pre_answer - I do not get RTP Also, RTP appears immediately if instead pre_answer I do answer on call 2! My dialing schemes: **** Direct mode ********** Incoming call (UUID1)-> PARK make UUID2 originate (using uuid2) sofia/gateway/some_gateway &park Listen for CHANNEL_PROGRESS_MEDIA and do pre_answer(UUID1) uuid_bridge(UUID1,UUID2) ******** in a looped mode ******* Thread 1 _INCOMING CALL_ (call1) Incoming call -> PARK make UUID2 originate (using uuid2) sofia/gateway/MY_IP_ADDRESS/another_user &park Listen for CHANNEL_PROGRESS_MEDIA and do pre_answer(UUID1) uuid_bridge(UUID1,UUID2) Thread 2 _OUTGOING_ (call 2) incoming call 2 (UUID3) (comes from call above) -> PARK make UUID4 originate (using uuid4) sofia/gateway/some_gateway &park Listen for CHANNEL_PROGRESS_MEDIA and do pre_answer(UUID3) uuid_bridge(UUID3,UUID4) Any clue? Regards, Anton From vipkilla at gmail.com Fri Jan 13 16:05:55 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 13 Jan 2012 08:05:55 -0500 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: What exactly is GemeinschaftPBX? From mytemike72 at gmail.com Fri Jan 13 16:31:21 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Fri, 13 Jan 2012 14:31:21 +0100 Subject: [Freeswitch-users] ESL Managed IVR and CallControl In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225507A23@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0225507A23@NY1-EXMB-01.ip-soft.net> Message-ID: Hi Hector, Thanks for your reply. I am trying the provided ManagedEsl test project now. It seems to do what I want but having much trouble trying to add functionality. I hope someone here can help me with a better server example with esl in outbound mode. I am also wondering how to set eventhandlers for dtmf or channel hangup events? Is there anyone here that can help me with some examples? For example I need to be able to catch the hangup or be able to respond to dtmfs. Thanks, Mike. 2012/1/12 Hector Geraldino > I have experience setting up something like that, but in Java.**** > > ** ** > > You should use ESL in outbound mode. For every new incoming call, FS will > open a socket connection to an ip:port where your application should be > listening to, and you will have complete control over the incoming call. > So, in theory, you should only need to have a SocketServer listening to an > specific port, and send commands/receive events from FS to your application > (you can play a little bit with this in linux using the netcat (nc) > utility, or maybe in windows by installing Cygwin and using nc.exe)**** > > ** ** > > I?m happy using the Java ESL library, a pure socket-based java application > (jboss/netty based) with no dependencies on system libs. Maybe (and this > is a BIG MAYBE) you can reuse this library from .NET, relying on Mono?s > IKVM tool, that allows Java-to-NET interoperability by converting Java?s > JVM bytecode to .NET CLR. I?ve taken this approach before, when I found a > terrific tool in Java (jPOS, for example) with no counterparts in .NET.*** > * > > ** ** > > This is only one option. There?s always the option of building your own > application listening on a socket, or using the .NET libraries listed on > the wiki:**** > > ** ** > > http://wiki.freeswitch.org/wiki/Mod_event_socket#.NET_Client_library**** > > ** ** > > Good luck! **** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Lutz > *Sent:* Thursday, January 12, 2012 10:02 AM > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] ESL Managed IVR and CallControl**** > > ** ** > > Hi,**** > > **** > > I have been playing with mod_managed for a while now but nobody realy > seems to be able to help me out. So now I am trying to go to ESL. (which > most of the people suggest anyway).**** > > **** > > What I need is control from .net over the complete call! (so even when > bridged and returns). **** > > **** > > I need to build a main library where all my generic functions will be put > in so I can use them in my different scripts.**** > > **** > > What I''ve figured out is problably the best way is to write a server > which connects to my FS an listens to all events.**** > > When a call comes in I need to fire a stored procedure to my sqlserver and > based on the reply I will decide if the call needsto be answerred, early > media needs to be played, or the call needs to be rejected.**** > > **** > > In any case, at the end of every call (answered or not) I need to finalize > my call with another stored procedure.**** > > **** > > When teh call is answered based on stored procedure result I need to run a > script (also managed code) **** > > run a script from another dll so I have all my scripts separate but they > share the corelib I create.**** > > **** > > I have been searching the web for a long time now to see if I can find > something simular, but I only see generic examples wich don't actually show > how to control multiple calls in a scenario like this.**** > > **** > > Who can help me out point me to the right direction or has experience in > setting something up like this?**** > > **** > > **** > > Thanks in advance,**** > > Michael Lutz.**** > > **** > > **** > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e5702947/attachment.html From mgg at giagnocavo.net Fri Jan 13 16:41:07 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 13 Jan 2012 08:41:07 -0500 Subject: [Freeswitch-users] ESL Managed IVR and CallControl In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD0225507A23@NY1-EXMB-01.ip-soft.net> Message-ID: <83FF8D7C9F526E44B77C97DD2891652A12C04533@mse17be1.mse17.exchange.ms> What do you need to catch hangup for? If it's for accounting, I'll repeat my general lesson that I painfully learned. Don't do it. Antony and I spent quite some time trying to sort out problems related to doing accounting "in-call", all the while they were telling me to not do it. I ended up with a lot of hacky code that sorta worked most of the time. A much better approach is to set vars on the channel and collect them when the XML CDR is written. If you need real-time accounting, just log the call once it opens to a database, and calculate outstanding balance from there. If it's another thing you're trying to do on hangup, excuse my interjection here :) -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Friday, January 13, 2012 6:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ESL Managed IVR and CallControl Hi Hector, Thanks for your reply. I am trying the provided ManagedEsl test project now. It seems to do what I want but having much trouble trying to add functionality. I hope someone here can help me with a better server example with esl in outbound mode. I am also wondering how to set eventhandlers for dtmf or channel hangup events? Is there anyone here that can help me with some examples? For example I need to be able to catch the hangup or be able to respond to dtmfs. Thanks, Mike. 2012/1/12 Hector Geraldino > I have experience setting up something like that, but in Java. You should use ESL in outbound mode. For every new incoming call, FS will open a socket connection to an ip:port where your application should be listening to, and you will have complete control over the incoming call. So, in theory, you should only need to have a SocketServer listening to an specific port, and send commands/receive events from FS to your application (you can play a little bit with this in linux using the netcat (nc) utility, or maybe in windows by installing Cygwin and using nc.exe) I'm happy using the Java ESL library, a pure socket-based java application (jboss/netty based) with no dependencies on system libs. Maybe (and this is a BIG MAYBE) you can reuse this library from .NET, relying on Mono's IKVM tool, that allows Java-to-NET interoperability by converting Java's JVM bytecode to .NET CLR. I've taken this approach before, when I found a terrific tool in Java (jPOS, for example) with no counterparts in .NET. This is only one option. There's always the option of building your own application listening on a socket, or using the .NET libraries listed on the wiki: http://wiki.freeswitch.org/wiki/Mod_event_socket#.NET_Client_library Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Lutz Sent: Thursday, January 12, 2012 10:02 AM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] ESL Managed IVR and CallControl Hi, I have been playing with mod_managed for a while now but nobody realy seems to be able to help me out. So now I am trying to go to ESL. (which most of the people suggest anyway). What I need is control from .net over the complete call! (so even when bridged and returns). I need to build a main library where all my generic functions will be put in so I can use them in my different scripts. What I''ve figured out is problably the best way is to write a server which connects to my FS an listens to all events. When a call comes in I need to fire a stored procedure to my sqlserver and based on the reply I will decide if the call needsto be answerred, early media needs to be played, or the call needs to be rejected. In any case, at the end of every call (answered or not) I need to finalize my call with another stored procedure. When teh call is answered based on stored procedure result I need to run a script (also managed code) run a script from another dll so I have all my scripts separate but they share the corelib I create. I have been searching the web for a long time now to see if I can find something simular, but I only see generic examples wich don't actually show how to control multiple calls in a scenario like this. Who can help me out point me to the right direction or has experience in setting something up like this? Thanks in advance, Michael Lutz. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/575da464/attachment-0001.html From mytemike72 at gmail.com Fri Jan 13 16:54:12 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Fri, 13 Jan 2012 14:54:12 +0100 Subject: [Freeswitch-users] ESL Managed IVR and CallControl In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12C04533@mse17be1.mse17.exchange.ms> References: <6A6B4C284AD15042B429EB9D904544AD0225507A23@NY1-EXMB-01.ip-soft.net> <83FF8D7C9F526E44B77C97DD2891652A12C04533@mse17be1.mse17.exchange.ms> Message-ID: Hi Michael, Thanks for the interjection :) it is not specially for billing, though I need the hangup to cleanup a lot of database stuff (using stored procedures). My plan was to finalize my data using the hangup and using some other mechanism to finalize my billing data (for example using the mod_xml_cdr to trigger the end of the call). But as I am also going to dialout and bridge calls I also need it to be able to detect the aleg hung up and disconnect the bleg? Or are there other ways to do that? For example I want to do something like: (example code written for mod_managed wich I did't get to work: (leg_a and leg_b.bridged return false after connecting)) route = "3112345678" // Number to dial string obCause = "ERROR"; FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); // Creating the new session... ManagedSession leg_b = new ManagedSession("{ignore_early_media=true,origination_caller_id_number=" + orig + ",originate_timeout=20}sofia/external/" + route); if (leg_b.Ready()) { if (leg_b.answered()) { MCX.AnswerCall(leg_a); // answer the incomming call as it was in 'ring_ready' leg_a.sleep(1000, 0); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); obCause = "SUCCESS"; Debug("As long as both legs are Ready(), we wait......"); while (leg_a.bridged() && leg_b.bridged()) // Returns false right away (bug?) { leg_a.sleep(1000, 0); // Slow it down... } Debug("There is one leg no longer ready..."); if (leg_a.bridged()) { Debug("Session A still ready..."); } else { Debug("Session A NO LONGER ready..."); } if (leg_b.bridged()) { 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 { obCause = leg_b.hangupCause(); } Debug("simpleCall() ended with: " + obCause); return obCause; } Can you help me on how to do this in managedESL ? Thanks, Mike 2012/1/13 Michael Giagnocavo > What do you need to catch hangup for? If it?s for accounting, I?ll repeat > my general lesson that I painfully learned. Don?t do it. Antony and I spent > quite some time trying to sort out problems related to doing accounting > ?in-call?, all the while they were telling me to not do it. I ended up with > a lot of hacky code that sorta worked most of the time. A much better > approach is to set vars on the channel and collect them when the XML CDR is > written. If you need real-time accounting, just log the call once it opens > to a database, and calculate outstanding balance from there.**** > > ** ** > > If it?s another thing you?re trying to do on hangup, excuse my > interjection here J**** > > -Michael**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Lutz > *Sent:* Friday, January 13, 2012 6:31 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] ESL Managed IVR and CallControl**** > > ** ** > > Hi Hector,**** > > **** > > Thanks for your reply. I am trying the provided ManagedEsl test project > now. It seems to do what I want but having much trouble trying to add > functionality. I hope someone here can help me with a better server example > with esl in outbound mode.**** > > **** > > I am also wondering how to set eventhandlers for dtmf or channel hangup > events? **** > > Is there anyone here that can help me with some examples?**** > > **** > > For example I need to be able to catch the hangup or be able to respond to > dtmfs.**** > > **** > > Thanks,**** > > Mike.**** > > 2012/1/12 Hector Geraldino **** > > I have experience setting up something like that, but in Java.**** > > **** > > You should use ESL in outbound mode. For every new incoming call, FS will > open a socket connection to an ip:port where your application should be > listening to, and you will have complete control over the incoming call. > So, in theory, you should only need to have a SocketServer listening to an > specific port, and send commands/receive events from FS to your application > (you can play a little bit with this in linux using the netcat (nc) > utility, or maybe in windows by installing Cygwin and using nc.exe)**** > > **** > > I?m happy using the Java ESL library, a pure socket-based java application > (jboss/netty based) with no dependencies on system libs. Maybe (and this > is a BIG MAYBE) you can reuse this library from .NET, relying on Mono?s > IKVM tool, that allows Java-to-NET interoperability by converting Java?s > JVM bytecode to .NET CLR. I?ve taken this approach before, when I found a > terrific tool in Java (jPOS, for example) with no counterparts in .NET.*** > * > > **** > > This is only one option. There?s always the option of building your own > application listening on a socket, or using the .NET libraries listed on > the wiki:**** > > **** > > http://wiki.freeswitch.org/wiki/Mod_event_socket#.NET_Client_library**** > > **** > > Good luck! **** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael Lutz > *Sent:* Thursday, January 12, 2012 10:02 AM > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] ESL Managed IVR and CallControl**** > > **** > > Hi,**** > > **** > > I have been playing with mod_managed for a while now but nobody realy > seems to be able to help me out. So now I am trying to go to ESL. (which > most of the people suggest anyway).**** > > **** > > What I need is control from .net over the complete call! (so even when > bridged and returns). **** > > **** > > I need to build a main library where all my generic functions will be put > in so I can use them in my different scripts.**** > > **** > > What I''ve figured out is problably the best way is to write a server > which connects to my FS an listens to all events.**** > > When a call comes in I need to fire a stored procedure to my sqlserver and > based on the reply I will decide if the call needsto be answerred, early > media needs to be played, or the call needs to be rejected.**** > > **** > > In any case, at the end of every call (answered or not) I need to finalize > my call with another stored procedure.**** > > **** > > When teh call is answered based on stored procedure result I need to run a > script (also managed code) **** > > run a script from another dll so I have all my scripts separate but they > share the corelib I create.**** > > **** > > I have been searching the web for a long time now to see if I can find > something simular, but I only see generic examples wich don't actually show > how to control multiple calls in a scenario like this.**** > > **** > > Who can help me out point me to the right direction or has experience in > setting something up like this?**** > > **** > > **** > > Thanks in advance,**** > > Michael Lutz.**** > > **** > > **** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/882f19fd/attachment.html From miha at softnet.si Fri Jan 13 16:59:52 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 13 Jan 2012 14:59:52 +0100 Subject: [Freeswitch-users] help about diaplan Message-ID: <4F1038D8.5090901@softnet.si> Hi, to explain what I would like to achieve. Users in different geographic location needs to call a certain special numbers different. If someone in location a is calling number 112, FS need to add prefix 454 before 112 (so 454112). Someone in location b when call to 112, FS needs to add 324 (so 324112). What is the best way to achieve that? Is it possible to add users in location A in group a and users on location B in group B and that mach in dial if it is group A than do this, if it is group B that do this? I noticed do not work? To mention I need to put this condition before my dialplan, so if the condition for dialed number it not matched, that proceed with my default dialplan. Thank you! regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From bdfoster at endigotech.com Fri Jan 13 17:15:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 13 Jan 2012 09:15:58 -0500 Subject: [Freeswitch-users] help about diaplan In-Reply-To: <4F1038D8.5090901@softnet.si> References: <4F1038D8.5090901@softnet.si> Message-ID: Hola, Take a look at the toll_allow variable, and set an outbound route for each location that requires 112. You need to have a condition: ...then set this for each user in the directory: (you can set more than one value in this field, for example you could restrict certain users from dialing internationally). -BD On Fri, Jan 13, 2012 at 8:59 AM, Miha Zoubek wrote: > Hi, > > to explain what I would like to achieve. > > > Users in different geographic location needs to call a certain special > numbers different. If someone in location a is calling number 112, FS > need to add prefix 454 before 112 (so 454112). Someone in location b > when call to 112, FS needs to add 324 (so 324112). > > What is the best way to achieve that? > > > Is it possible to add users in location A in group a and users on > location B in group B and that mach in dial if it is group A than do > this, if it is group B that do this? > > I noticed do not work? > > To mention I need to put this condition before my dialplan, so if the > condition for dialed number it not matched, that proceed with my default > dialplan. > > Thank you! > > regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/558460fd/attachment.html From mgg at giagnocavo.net Fri Jan 13 17:22:08 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 13 Jan 2012 09:22:08 -0500 Subject: [Freeswitch-users] Effect of sync_clock on calls? Message-ID: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> Should there be any audio-quality effects on calls by running fsctl sync_clock? To try to get my timestamps as close as possible to system time, I plan on running this every 5 minutes or so. Is there a fundamental issue with this? Or is there a way to disable the monotonic clock completely? I understand what it's trying to achieve, but I cannot see how it'll ever be any more accurate than the system time, which is sync'd via NTP. I'm more concerned about my timestamps being accurate than protecting against someone turning off NTP and changing the time to some invalid value. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/af6fa44c/attachment.html From bdfoster at endigotech.com Fri Jan 13 17:23:16 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 13 Jan 2012 09:23:16 -0500 Subject: [Freeswitch-users] help about diaplan In-Reply-To: References: <4F1038D8.5090901@softnet.si> Message-ID: Hola again, Here's a great guide on using the toll_allow variable: http://wiki.fusionpbx.com/index.php?title=Extensions#Notes_on_Toll_Allow ... is the proper way to do the condition check on your outbound route. Sorry! If you are running FusionPBX, you can set the toll_allow variable in the extension itself. -BD On Fri, Jan 13, 2012 at 9:15 AM, Brian Foster wrote: > Hola, > > Take a look at the toll_allow variable, and set an outbound route for each > location that requires 112. You need to have a condition: > > > > ...then set this for each user in the directory: > > > (you can set more than one value in this field, for example you could > restrict certain users from dialing internationally). > > -BD > On Fri, Jan 13, 2012 at 8:59 AM, Miha Zoubek wrote: > >> Hi, >> >> to explain what I would like to achieve. >> >> >> Users in different geographic location needs to call a certain special >> numbers different. If someone in location a is calling number 112, FS >> need to add prefix 454 before 112 (so 454112). Someone in location b >> when call to 112, FS needs to add 324 (so 324112). >> >> What is the best way to achieve that? >> >> >> Is it possible to add users in location A in group a and users on >> location B in group B and that mach in dial if it is group A than do >> this, if it is group B that do this? >> >> I noticed do not work? >> >> To mention I need to put this condition before my dialplan, so if the >> condition for dialed number it not matched, that proceed with my default >> dialplan. >> >> Thank you! >> >> regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-429-1069 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/72aac02c/attachment.html From bdfoster at endigotech.com Fri Jan 13 17:35:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 13 Jan 2012 09:35:35 -0500 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> Message-ID: Hola, http://wiki.freeswitch.org/wiki/Clock The general concensus says that you really should not be messing around too much with using sync_clock. If you still plan to go down this road, I suggest only syncing the clock when there are no calls. Syncing the clock could cause some issues if you are doing this when there are calls up. -BD On Fri, Jan 13, 2012 at 9:22 AM, Michael Giagnocavo wrote: > Should there be any audio-quality effects on calls by running fsctl > sync_clock? To try to get my timestamps as close as possible to system > time, I plan on running this every 5 minutes or so. Is there a fundamental > issue with this?**** > > ** ** > > Or is there a way to disable the monotonic clock completely? I understand > what it?s trying to achieve, but I cannot see how it?ll ever be any more > accurate than the system time, which is sync?d via NTP. I?m more concerned > about my timestamps being accurate than protecting against someone turning > off NTP and changing the time to some invalid value.**** > > ** ** > > -Michael**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/c3dc7841/attachment-0001.html From peter.olsson at visionutveckling.se Fri Jan 13 18:08:35 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 13 Jan 2012 15:08:35 +0000 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> Message-ID: <1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> I think in later FS versions (git head) the actual sync is not done until there are no calls. Either that is the case, or there was a new fsctl command added to handle this. Anthm, I think you implemented this a few weeks ago? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian Foster Skickat: den 13 januari 2012 15:36 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Effect of sync_clock on calls? Hola, http://wiki.freeswitch.org/wiki/Clock The general concensus says that you really should not be messing around too much with using sync_clock. If you still plan to go down this road, I suggest only syncing the clock when there are no calls. Syncing the clock could cause some issues if you are doing this when there are calls up. -BD On Fri, Jan 13, 2012 at 9:22 AM, Michael Giagnocavo > wrote: Should there be any audio-quality effects on calls by running fsctl sync_clock? To try to get my timestamps as close as possible to system time, I plan on running this every 5 minutes or so. Is there a fundamental issue with this? Or is there a way to disable the monotonic clock completely? I understand what it's trying to achieve, but I cannot see how it'll ever be any more accurate than the system time, which is sync'd via NTP. I'm more concerned about my timestamps being accurate than protecting against someone turning off NTP and changing the time to some invalid value. -Michael _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. !DSPAM:4f10410632761818622984! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/3947a258/attachment.html From anthony.minessale at gmail.com Fri Jan 13 18:31:59 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Jan 2012 09:31:59 -0600 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: <1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> <1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> Message-ID: I forgot the command but yes. It's an extra arg to the existing sync clock. Again I don't think this even happens on win unless you ask it to. On Jan 13, 2012 9:09 AM, "Peter Olsson" wrote: > I think in later FS versions (git head) the actual sync is not done > until there are no calls. Either that is the case, or there was a new fsctl > command added to handle this.**** > > ** ** > > Anthm, I think you implemented this a few weeks ago?**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Brian Foster > *Skickat:* den 13 januari 2012 15:36 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Effect of sync_clock on calls?**** > > ** ** > > Hola,**** > > **** > > http://wiki.freeswitch.org/wiki/Clock**** > > **** > > The general concensus says that you really should not be messing around > too much with using sync_clock. If you still plan to go down this road, I > suggest only syncing the clock when there are no calls. Syncing the clock > could cause some issues if you are doing this when there are calls up.**** > > **** > > -BD**** > > **** > > On Fri, Jan 13, 2012 at 9:22 AM, Michael Giagnocavo > wrote:**** > > Should there be any audio-quality effects on calls by running fsctl > sync_clock? To try to get my timestamps as close as possible to system > time, I plan on running this every 5 minutes or so. Is there a fundamental > issue with this?**** > > **** > > Or is there a way to disable the monotonic clock completely? I understand > what it?s trying to achieve, but I cannot see how it?ll ever be any more > accurate than the system time, which is sync?d via NTP. I?m more concerned > about my timestamps being accurate than protecting against someone turning > off NTP and changing the time to some invalid value.**** > > **** > > -Michael**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-429-1069 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > !DSPAM:4f10410632761818622984! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/2f0951fe/attachment.html From stkn at freeswitch.org Fri Jan 13 19:06:19 2012 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 13 Jan 2012 17:06:19 +0100 Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] In-Reply-To: <017001ccd195$f1d9b210$d58d1630$@redbonez.net> References: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> <017001ccd195$f1d9b210$d58d1630$@redbonez.net> Message-ID: <4F10567B.4060601@freeswitch.org> On 13.01.2012 02:51, Adam Ford wrote: > I am still having issues with MANDATORY_IE_MISSING on incoming calls > when using FreeTDM + libpri + DAHDI + foneBridge2. Can anyone help me > figure out if this is a configuration issue or a bug/incompatibility? > > As stated before, I am running the latest git trunk with all > default/stock settings with the exception of FreeTDM configuration and > minor modification to the dialplan to pass my DID to the default > extension 1001. Outgoing calls are working great. > Try the attached patch, that should fix it (and hopefully doesn't have any side effects). -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net -------------- next part -------------- A non-text attachment was scrubbed... Name: ftmod_libpri-progress-20120113.patch Type: text/x-patch Size: 663 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/27477890/attachment-0001.bin From mgg at giagnocavo.net Fri Jan 13 19:21:34 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 13 Jan 2012 11:21:34 -0500 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> <1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> Message-ID: <83FF8D7C9F526E44B77C97DD2891652A12C045CB@mse17be1.mse17.exchange.ms> Yea, I found the one that waits until after it finishes calls. What would the actual effect be on a call? Suppose my clock has drifted 5ms and I sync; would that make timers expire sooner/later and cause an audio distortion? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, January 13, 2012 8:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Effect of sync_clock on calls? I forgot the command but yes. It's an extra arg to the existing sync clock. Again I don't think this even happens on win unless you ask it to. On Jan 13, 2012 9:09 AM, "Peter Olsson" > wrote: I think in later FS versions (git head) the actual sync is not done until there are no calls. Either that is the case, or there was a new fsctl command added to handle this. Anthm, I think you implemented this a few weeks ago? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian Foster Skickat: den 13 januari 2012 15:36 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Effect of sync_clock on calls? Hola, http://wiki.freeswitch.org/wiki/Clock The general concensus says that you really should not be messing around too much with using sync_clock. If you still plan to go down this road, I suggest only syncing the clock when there are no calls. Syncing the clock could cause some issues if you are doing this when there are calls up. -BD On Fri, Jan 13, 2012 at 9:22 AM, Michael Giagnocavo > wrote: Should there be any audio-quality effects on calls by running fsctl sync_clock? To try to get my timestamps as close as possible to system time, I plan on running this every 5 minutes or so. Is there a fundamental issue with this? Or is there a way to disable the monotonic clock completely? I understand what it's trying to achieve, but I cannot see how it'll ever be any more accurate than the system time, which is sync'd via NTP. I'm more concerned about my timestamps being accurate than protecting against someone turning off NTP and changing the time to some invalid value. -Michael _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. !DSPAM:4f10410632761818622984! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/ad6e0ead/attachment.html From peter.olsson at visionutveckling.se Fri Jan 13 20:08:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 13 Jan 2012 17:08:20 +0000 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12C045CB@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms> <1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> , <83FF8D7C9F526E44B77C97DD2891652A12C045CB@mse17be1.mse17.exchange.ms> Message-ID: <1FFF97C269757C458224B7C895F35F1502EDBB@cantor.std.visionutv.se> I'm not really sure. But I guess there could be a short gap in audio. 5ms I'm not sure you would even notice. FS probably handles it quite well :) /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Michael Giagnocavo [mgg at giagnocavo.net] Skickat: den 13 januari 2012 17:21 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Effect of sync_clock on calls? Yea, I found the one that waits until after it finishes calls. What would the actual effect be on a call? Suppose my clock has drifted 5ms and I sync; would that make timers expire sooner/later and cause an audio distortion? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, January 13, 2012 8:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Effect of sync_clock on calls? I forgot the command but yes. It's an extra arg to the existing sync clock. Again I don't think this even happens on win unless you ask it to. On Jan 13, 2012 9:09 AM, "Peter Olsson" > wrote: I think in later FS versions (git head) the actual sync is not done until there are no calls. Either that is the case, or there was a new fsctl command added to handle this. Anthm, I think you implemented this a few weeks ago? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Brian Foster Skickat: den 13 januari 2012 15:36 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Effect of sync_clock on calls? Hola, http://wiki.freeswitch.org/wiki/Clock The general concensus says that you really should not be messing around too much with using sync_clock. If you still plan to go down this road, I suggest only syncing the clock when there are no calls. Syncing the clock could cause some issues if you are doing this when there are calls up. -BD On Fri, Jan 13, 2012 at 9:22 AM, Michael Giagnocavo > wrote: Should there be any audio-quality effects on calls by running fsctl sync_clock? To try to get my timestamps as close as possible to system time, I plan on running this every 5 minutes or so. Is there a fundamental issue with this? Or is there a way to disable the monotonic clock completely? I understand what it?s trying to achieve, but I cannot see how it?ll ever be any more accurate than the system time, which is sync?d via NTP. I?m more concerned about my timestamps being accurate than protecting against someone turning off NTP and changing the time to some invalid value. -Michael _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4f1059da32761824717460! From nmenoni at glotweb.com Fri Jan 13 20:53:14 2012 From: nmenoni at glotweb.com (=?ISO-8859-1?Q?Nicol=E1s_Menoni?=) Date: Fri, 13 Jan 2012 15:53:14 -0200 Subject: [Freeswitch-users] Measuring the call quality Message-ID: Hello, Is there a tool for measure the call quality and take the statistics? The idea is take statistics in packet loss, jitter, delay, etc., per each call and save these in a DB. Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/923b7478/attachment-0001.html From msc at freeswitch.org Fri Jan 13 20:59:03 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 09:59:03 -0800 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: On Fri, Jan 13, 2012 at 5:05 AM, Vik Killa wrote: > What exactly is GemeinschaftPBX? > Gemeinschaft is a "German-engineered open source PBX." Previously it was based on Asterisk, however the German version of the FBI needs a "paranoia-compliant" VoIP PBX that is open source. The Amooma guys built that with FreeSWITCH, Kamailio, RoR, etc. in version 4. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/c7b8c4f6/attachment.html From msc at freeswitch.org Fri Jan 13 21:12:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 10:12:00 -0800 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <1326442641.87954.YahooMailNeo@web29401.mail.ird.yahoo.com> References: <1326442641.87954.YahooMailNeo@web29401.mail.ird.yahoo.com> Message-ID: On Fri, Jan 13, 2012 at 12:17 AM, Bob Smith wrote: > Thank you Brian. > > Although I appreciate the constant development of FreeSwitch, I don't > think it's too much to ask that the developers should keep the Wiki pages > up to date in respect of features that have been added and/or removed. > There is no mention of those parameters on > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_valet_park. ;-( > > Actually, it is too much to ask of the developers. However, the wiki still needs maintenance. It's not the developers who maintain the wiki - it's unpaid community volunteers. There is an unwritten agreement between the FreeSWITCH devs and the community: they produce and maintain one of the most advanced soft-switches in the world, for *free*, and in return we do our best to keep up with answering questions on the mailing lists, IRC, and wiki. We also agree to run the latest version when testing for bugs. Any limitations on the wiki are the responsibility of us, the community. So, if you find something in the wiki that is wrong/missing/outdated then you have three choices: 1. fix it 2. ask someone if they can help fix it 3. do nothing (with optional complaining) If you learn something that isn't on the wiki then please do #1 or #2 so that the next person doesn't have to go through what you went through. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/f1c43455/attachment.html From vipkilla at gmail.com Fri Jan 13 21:14:20 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 13 Jan 2012 13:14:20 -0500 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: If it's German-engineered im sure it's quality. But is there a list of features, things it can do? On Fri, Jan 13, 2012 at 12:59 PM, Michael Collins wrote: > > > On Fri, Jan 13, 2012 at 5:05 AM, Vik Killa wrote: >> >> What exactly is GemeinschaftPBX? > > > Gemeinschaft is a "German-engineered open source PBX." Previously it was > based on Asterisk, however the German version of the FBI needs a > "paranoia-compliant" VoIP PBX that is open source. The Amooma guys built > that with FreeSWITCH, Kamailio, RoR, etc. in version 4. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Jan 13 21:19:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 10:19:07 -0800 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: On Fri, Jan 13, 2012 at 10:14 AM, Vik Killa wrote: > If it's German-engineered im sure it's quality. But is there a list of > features, things it can do? > Yes. Please refer to the story on freeswitch.org. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/cdcb8390/attachment.html From msc at freeswitch.org Fri Jan 13 21:26:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 10:26:30 -0800 Subject: [Freeswitch-users] Measuring the call quality In-Reply-To: References: Message-ID: XML CDRs give you lots of packet statistics. If you are looking for a professional tool then check out something like http://www.voipmonitor.org/ -MC 2012/1/13 Nicol?s Menoni > Hello, > > Is there a tool for measure the call quality and take the statistics? The > idea is take statistics in packet loss, jitter, delay, etc., per each call > and save these in a DB. > > Thank you! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/256447d6/attachment.html From msc at freeswitch.org Fri Jan 13 21:29:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 10:29:06 -0800 Subject: [Freeswitch-users] Empty WAV file recorded In-Reply-To: References: Message-ID: get a complete sip trace and call log and put it on pastebin. -MC On Thu, Jan 12, 2012 at 11:04 PM, Hynek Cihlar wrote: > I was wrong, the same problem is when recording to mp3. > > Tested on HEAD. It worked ok on version 4-or-so months back. > > Hynek > > > > > On Fri, Jan 13, 2012 at 7:48 AM, Hynek Cihlar wrote: > >> Hi all, >> >> an attempt to uuid_record ends with wav file with no data only header. >> ct >> The scenario is following: >> 0. FS <---> ESL app >> 1. Bridge is established on one incoming and one originated channel. One >> channel hangs up. >> 2. uuid_record start is issued on the active channel. >> 3. uuid_record stop is issued on the active channel. >> 4. The result is a wav file with always the same length of 44 bytes. >> >> When recording to mp3, the file is always correctly created. >> >> Any idea how to diagnose the problem? >> >> Hynek >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/8f944b44/attachment.html From Jacob.E.Miles at L-3Com.com Fri Jan 13 21:56:49 2012 From: Jacob.E.Miles at L-3Com.com (Jacob.E.Miles at L-3Com.com) Date: Fri, 13 Jan 2012 12:56:49 -0600 Subject: [Freeswitch-users] AS-SIP Message-ID: Is there support for AS-SIP (Assured Services Session Initiation Protocol)? I have only spent maybe 2 minutes googleing this so far with no results yet, so I thought my quickest way of finding out would be to ask you guys. I know AS-SIP uses the SIP Communications Resource Priority (RFC 4412) and the SIP Reason Header for Preemption Events (RFC 441) to allow MLPP (Multi Level Precedence and Preemption). Jacob Miles Software Engineer L-3 Communications - Integrated Systems Greenville Jacob.E.Miles at L-3Com.com 903.457.4422 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/3f51e51a/attachment-0001.html From thomas at chaschperli.ch Fri Jan 13 22:54:08 2012 From: thomas at chaschperli.ch (Thomas Mueller) Date: Fri, 13 Jan 2012 20:54:08 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: <4F108BE0.6060208@chaschperli.ch> On 13.01.2012 19:14, Vik Killa wrote: > If it's German-engineered im sure it's quality. But is there a list of > features, things it can do? IMHO the company is more technician driven than marketing so there is the product (www.amooma.de/gemeinschaft) for download but no fancy homepage yet. you can try it for yourself but it's only available in german by now. an english release is due in march. - Thomas From vipkilla at gmail.com Fri Jan 13 22:57:54 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 13 Jan 2012 14:57:54 -0500 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F108BE0.6060208@chaschperli.ch> References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> <4F108BE0.6060208@chaschperli.ch> Message-ID: Yeah, i couldn't find any info on what is really does.. i guess i'l just install the ISO and find out > > you can try it for yourself but it's only available in german by now. an > english release is due in march. > > - Thomas > From Hector.Geraldino at ip-soft.net Fri Jan 13 23:23:13 2012 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Fri, 13 Jan 2012 15:23:13 -0500 Subject: [Freeswitch-users] speech to text engine In-Reply-To: References: <7E60A092-556D-4519-BB03-456CF6F66C14@gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507A5D@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507AEB@NY1-EXMB-01.ip-soft.net> It think it's because of the acoustic model they used. Somebody explained me that, for example, Siri's recognition accuracy is not optimal when you try to recognize the speech from a phone conversation (8khz audio), and that is because of the acoustic model they've used to train the grammar. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: Thursday, January 12, 2012 11:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] speech to text engine Siri itself is not available on Android. There was an app that claimed to be Siri for Android, published by 'Official App' but it was taken down by Google. It was reported to be using Google's API. Google's API for transcription from what I can se is supposed to be used for Android OS only. It's not really clear if it can be used for other purposes. Sirius doesn't have an external API for us to use (no surprise there) and it most likely is using Nuance for transcription. I've tried out the script on a few vm files I have. It doesn't really work well for mobile calls. It worked extremely well on some pre recorded messages I made (the number you have reached is not in service...) and it worked really well. Maybe Google likes me? -BD On Jan 12, 2012 10:25 PM, "Robert-IPhone" > wrote: I'll do some research - course SIRI is only happy with American English reliably ;) Sent from my iPhone 4S On Jan 12, 2012, at 7:53 PM, Michael Collins > wrote: On Thu, Jan 12, 2012 at 4:05 PM, Robert-IPhone > wrote: Well my point wasn't cost or legal as do we know if Google is open for all use. There are some articles about people using SIRI on Android etc I'm curious to know if there is a generic way to make a request of SIRI to do speech to text... -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/1fd4c72e/attachment.html From anthony.minessale at gmail.com Fri Jan 13 23:45:15 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Jan 2012 14:45:15 -0600 Subject: [Freeswitch-users] Empty WAV file recorded In-Reply-To: References: Message-ID: DO NOT REPORT BUGS ON THIS LIST....... On Fri, Jan 13, 2012 at 12:29 PM, Michael Collins wrote: > get a complete sip trace and call log and put it on pastebin. > -MC > > On Thu, Jan 12, 2012 at 11:04 PM, Hynek Cihlar wrote: > >> I was wrong, the same problem is when recording to mp3. >> >> Tested on HEAD. It worked ok on version 4-or-so months back. >> >> Hynek >> >> >> >> >> On Fri, Jan 13, 2012 at 7:48 AM, Hynek Cihlar wrote: >> >>> Hi all, >>> >>> an attempt to uuid_record ends with wav file with no data only header. >>> ct >>> The scenario is following: >>> 0. FS <---> ESL app >>> 1. Bridge is established on one incoming and one originated channel. One >>> channel hangs up. >>> 2. uuid_record start is issued on the active channel. >>> 3. uuid_record stop is issued on the active channel. >>> 4. The result is a wav file with always the same length of 44 bytes. >>> >>> When recording to mp3, the file is always correctly created. >>> >>> Any idea how to diagnose the problem? >>> >>> Hynek >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/60886cea/attachment-0001.html From anthony.minessale at gmail.com Fri Jan 13 23:48:12 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 13 Jan 2012 14:48:12 -0600 Subject: [Freeswitch-users] Empty WAV file recorded In-Reply-To: References: Message-ID: confirmed to work on GIT HEAD 3bb26bbdc02406eb103a42207d726089b4d51214 On Fri, Jan 13, 2012 at 2:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > DO NOT REPORT BUGS ON THIS LIST....... > > > On Fri, Jan 13, 2012 at 12:29 PM, Michael Collins wrote: > >> get a complete sip trace and call log and put it on pastebin. >> -MC >> >> On Thu, Jan 12, 2012 at 11:04 PM, Hynek Cihlar wrote: >> >>> I was wrong, the same problem is when recording to mp3. >>> >>> Tested on HEAD. It worked ok on version 4-or-so months back. >>> >>> Hynek >>> >>> >>> >>> >>> On Fri, Jan 13, 2012 at 7:48 AM, Hynek Cihlar wrote: >>> >>>> Hi all, >>>> >>>> an attempt to uuid_record ends with wav file with no data only header. >>>> ct >>>> The scenario is following: >>>> 0. FS <---> ESL app >>>> 1. Bridge is established on one incoming and one originated channel. >>>> One channel hangs up. >>>> 2. uuid_record start is issued on the active channel. >>>> 3. uuid_record stop is issued on the active channel. >>>> 4. The result is a wav file with always the same length of 44 bytes. >>>> >>>> When recording to mp3, the file is always correctly created. >>>> >>>> Any idea how to diagnose the problem? >>>> >>>> Hynek >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/1361ca0b/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Sat Jan 14 02:02:53 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Fri, 13 Jan 2012 23:02:53 +0000 (GMT) Subject: [Freeswitch-users] valet_park timeout Message-ID: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> Michael, Permit me to perhaps expand a little on why I said what I did : 1. As a newbie to Freeswitch, it is right and proper that I take the time to review the available documentation (including purchasing the eBook in my case). ?The wiki documentation is severely lacking... some of it is out of date, a lot of it is very sparse, and there many un-necessary and unprofessional comments embedded in it (e.g. "???What is it acknowledging???", "What does this do ???" etc.). ?I have been spending many frustrating hours trying to educate myself in the mysterious FreeSwitch through the wiki and eBooks. ?Why ? Because I consider it the correct approach as opposed to littering the "freeswitch-users" list with a ton of newbie questions. 2. Building on the point above, surely you should be encouraging ease of adoption by a new community of users. ?Uses who will no doubt eventually end up contributing to the project as their experience with Freeswitch grows, and their time permits. ?It is not exactly fair or reasonable to expect newbies to dig around the source code in order to uncover?useful functions which are?undocumented, but have long been stable. ?Particularly, if like me, their background is not in C programming. 3. I was not asking the developers to write a book each time they code a new feature. ?Even just a two column table (function/variable/etc + one or two phrase description of function/variable/etc.) would be better than the old, out of date Wiki of today. ? The developers are the ones best placed to know when a feature is (a) introduced and (b) becomes stable enough in the code tree that they are happy with it. ?As you well know, half of a coders job is documenting .... uncommented code is sloppy, undocumented software is unusable. ? As developers, you should be proud of your work, proud of all the new features you're introducing, and the existing features you are enhancing. ?So surely you should be blowing your trumpet by even modestly documenting your new work by a new phrase or two in the wiki ? Take my example of valet_park timeouts ?. I spent much time trying to search through the lists, more time digging fruitlessly through the wiki and the eBook in search of the faintest trace of documentation of the new variables introduced. ?I tried poking through the source code, but being unfamiliar with (a) C and (b) the FreeSwitch tree, I couldn't really comprehend what was going on. ?So I ask on Freeswitch-Users and get a stupid answer that says "LOOK CLOSELY" (little did the poster know just how much time I had expended "looking closely" !!!) ?? I spent days looking closely, days which could have been avoided if the developers took 2 minutes to add 4 lines to the Wiki just listing and ever so briefly describing the new variables ! Everyone is entitled to their own opinion, of course, but I suspect this is one area where ours may vary ! B From bdfoster at endigotech.com Sat Jan 14 02:14:27 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 13 Jan 2012 18:14:27 -0500 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: The problem is this: if developers had to write out documentation for every feature/variable/trick/etc. they came up with, they wouldn't have time to maintain the software (which of course is free, open source, etc.) Keep in mind that you don't need to be a developer to write documentation. The best thing to do is if something is undocumented, ask. Then, turn around and wikify it. I wholeheartedly believe that this project does quite well with documentation. There has been extensive work done by many to keep up with the new features/variables/tricks/etc. We have and always will have room to grow. Anyway, that's my 2?. Take it or leave it. -BD On Jan 13, 2012 6:04 PM, "Bob Smith" wrote: > Michael, > > > Permit me to perhaps expand a little on why I said what I did : > > > 1. As a newbie to Freeswitch, it is right and proper that I take the time > to review the available documentation (including purchasing the eBook in my > case). The wiki documentation is severely lacking... some of it is out of > date, a lot of it is very sparse, and there many un-necessary and > unprofessional comments embedded in it (e.g. "???What is it > acknowledging???", "What does this do ???" etc.). I have been spending > many frustrating hours trying to educate myself in the mysterious > FreeSwitch through the wiki and eBooks. Why ? Because I consider it the > correct approach as opposed to littering the "freeswitch-users" list with a > ton of newbie questions. > > > 2. Building on the point above, surely you should be encouraging ease of > adoption by a new community of users. Uses who will no doubt eventually > end up contributing to the project as their experience with Freeswitch > grows, and their time permits. It is not exactly fair or reasonable to > expect newbies to dig around the source code in order to uncover useful > functions which are undocumented, but have long been stable. Particularly, > if like me, their background is not in C programming. > > > 3. I was not asking the developers to write a book each time they code a > new feature. Even just a two column table (function/variable/etc + one or > two phrase description of function/variable/etc.) would be better than the > old, out of date Wiki of today. The developers are the ones best placed > to know when a feature is (a) introduced and (b) becomes stable enough in > the code tree that they are happy with it. As you well know, half of a > coders job is documenting .... uncommented code is sloppy, undocumented > software is unusable. As developers, you should be proud of your work, > proud of all the new features you're introducing, and the existing features > you are enhancing. So surely you should be blowing your trumpet by even > modestly documenting your new work by a new phrase or two in the wiki ? > > > Take my example of valet_park timeouts ?. I spent much time trying to > search through the lists, more time digging fruitlessly through the wiki > and the eBook in search of the faintest trace of documentation of the new > variables introduced. I tried poking through the source code, but being > unfamiliar with (a) C and (b) the FreeSwitch tree, I couldn't really > comprehend what was going on. So I ask on Freeswitch-Users and get a > stupid answer that says "LOOK CLOSELY" (little did the poster know just how > much time I had expended "looking closely" !!!) ? I spent days looking > closely, days which could have been avoided if the developers took 2 > minutes to add 4 lines to the Wiki just listing and ever so briefly > describing the new variables ! > > > Everyone is entitled to their own opinion, of course, but I suspect this > is one area where ours may vary ! > > > B > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/e7dc4cab/attachment-0001.html From brian at freeswitch.org Sat Jan 14 02:19:22 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 13 Jan 2012 17:19:22 -0600 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: <47872C6E-6442-43B8-B811-27AC24464100@freeswitch.org> The Valet Parking Timeout bits are about two weeks old... I see someone took my email and put the details of the variables in the wiki.. /b On Jan 13, 2012, at 5:02 PM, Bob Smith wrote: > > > Everyone is entitled to their own opinion, of course, but I suspect this is one area where ours may vary ! > > > B > > ____ -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/be1bb520/attachment.html From freeswitch at earthspike.net Sat Jan 14 02:33:16 2012 From: freeswitch at earthspike.net (John) Date: Fri, 13 Jan 2012 23:33:16 +0000 Subject: [Freeswitch-users] audo sync issues with record_session to mp3 In-Reply-To: References: <6AEC73649FA6431CA9EF84A54F378594@ws4> <4F0C4F2F.3050901@earthspike.net> Message-ID: <4F10BF3C.1070601@earthspike.net> I ran a test this evening with a 15 minute phone call from PSTN via ISDN BRI (FreeTDM) into a SIP handset, all at G.711, recorded to MP3. Previously, this was out of sync. Now it's rock solid in sync. Thanks, Anthony! On 12/01/12 01:00, Anthony Minessale wrote: > I really wish things like this would go to jira it's very hard to > track things from the mailing list...... > > > commit a365fb636ad9e2f4bb5dee43eddc305560699114 > Author: Anthony Minessale > > Date: Wed Jan 11 17:49:35 2012 -0600 > > mailing list > 36bc584d980ce80fe6a6f6e7d7383db9.squirrel at my.tomp.co.uk > > [Freeswitch-users] audo sync issues with record_session to mp3 I redid > the stream recording with timestamps and headers to try to keep it > more synced > > > > On Tue, Jan 10, 2012 at 8:46 AM, John > wrote: > > I've just enabled session recording using MP3 encoding and have > the same > symptoms. In my case it is between a BRI ISDN connection and a SIP > phone, both running with G.711. I haven't tried recording between > extensions which would be a purer test, nor have I attempted using > WAV. > Are there any suggestions for how to fix this? Should I register a bug > on Jira? > > John > > > On 02/01/12 15:01, Frank @ Impact wrote: > > We have the same problem. We are running git from 12/30/11. > our aleg is a > > sip channel coming to FS and the bleg is a sip channels leaving FS. > > > > I noticed this problem really when we started using mp3 instead of > > wav. With wav, it really was not noticeable for us in a > 10-15minute call. > > But with mp3, we notice it after just 2-3 minutes. By 10 > minutes, it is so > > far out of sync, it sounds like 2 different calls. > > > > The relevant dialplan is > > > > > > > >
> > > > > > > > > > > > > > > > > > > data="/mnt/rd/recordfile.mp3"/> > > > data="[park_after_bridge=true,park_timeout=3]${enum_auto_route}"/> > > > > > > > > > >
> > > > > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf > Of Gabriel > > Gunderson > > Sent: Thursday, October 20, 2011 3:07 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] audo sync issues with > record_session to mp3 > > > > On Tue, Oct 18, 2011 at 4:34 AM, Tom Parrott > wrote: > >> Longer calls, after about 10 minutes start to introduce sync issues > >> between the A-leg and the B-leg. > >> > >> We are running record_session on the A-leg, and it seems to get > ahead of > >> the B-leg. > >> > >> For example the caller on the A-leg will be heard to answer a > question > >> whilst the person on the B-leg is asking it. > > What's on the other end of each leg? That might help us figure > this out. > > > > Gabe > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/df1f65a3/attachment-0001.html From msc at freeswitch.org Sat Jan 14 02:44:27 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 15:44:27 -0800 Subject: [Freeswitch-users] valet_park timeout In-Reply-To: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1326495773.33763.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: Bob, I respect your viewpoint. However, nothing you said is new to us. We've been doing this for over five years. Learning a new software platform is usually a challenging task. Learning a full-featured soft-switch is even more daunting. I guarantee that you'd be frustrated by something else, even if this brand new valet parking timeout feature had already been documented. That's just the nature of telecom software. BTW, there is usually a 2-6 week window between when a new feature gets added to FreeSWITCH and when it gets "properly wikified." The reason for this is simple - there is a lot involved: 1. Watch git commits regularly 2. Notice new features, tweaks, changes 3. Spend time learning those new changes, including how they might affect existing features 4. Update a server to the latest git 5. Implement the feature and test test test 6. Update the wiki The list of people who have both the time and skills to do all of the above - not to mention the energy and inclination - is quite limited. Usually what happens is a community member knows a particular subset of FS features and keeps an eye out for changes and then updates them as soon as he reasonably can. Also, we have a community conference call each Wednesday (GMT 1700) wherein we discuss documentation needs, among other things. You are welcome to join us and talk directly to FS power users and even the core developers on occasion. So, in short: we feel your pain. Most of us have felt it at one point or another as we were all once newbies. We invite you to assist us with easing the pain of the scores of new users who download FreeSWITCH each day by helping to improve the wiki. Sign up for a wiki account and I'll get you confirmed ASAP. Thanks, MC On Fri, Jan 13, 2012 at 3:02 PM, Bob Smith wrote: > Michael, > > > Permit me to perhaps expand a little on why I said what I did : > > > 1. As a newbie to Freeswitch, it is right and proper that I take the time > to review the available documentation (including purchasing the eBook in my > case). The wiki documentation is severely lacking... some of it is out of > date, a lot of it is very sparse, and there many un-necessary and > unprofessional comments embedded in it (e.g. "???What is it > acknowledging???", "What does this do ???" etc.). I have been spending > many frustrating hours trying to educate myself in the mysterious > FreeSwitch through the wiki and eBooks. Why ? Because I consider it the > correct approach as opposed to littering the "freeswitch-users" list with a > ton of newbie questions. > > > 2. Building on the point above, surely you should be encouraging ease of > adoption by a new community of users. Uses who will no doubt eventually > end up contributing to the project as their experience with Freeswitch > grows, and their time permits. It is not exactly fair or reasonable to > expect newbies to dig around the source code in order to uncover useful > functions which are undocumented, but have long been stable. Particularly, > if like me, their background is not in C programming. > > > 3. I was not asking the developers to write a book each time they code a > new feature. Even just a two column table (function/variable/etc + one or > two phrase description of function/variable/etc.) would be better than the > old, out of date Wiki of today. The developers are the ones best placed > to know when a feature is (a) introduced and (b) becomes stable enough in > the code tree that they are happy with it. As you well know, half of a > coders job is documenting .... uncommented code is sloppy, undocumented > software is unusable. As developers, you should be proud of your work, > proud of all the new features you're introducing, and the existing features > you are enhancing. So surely you should be blowing your trumpet by even > modestly documenting your new work by a new phrase or two in the wiki ? > > > Take my example of valet_park timeouts ?. I spent much time trying to > search through the lists, more time digging fruitlessly through the wiki > and the eBook in search of the faintest trace of documentation of the new > variables introduced. I tried poking through the source code, but being > unfamiliar with (a) C and (b) the FreeSwitch tree, I couldn't really > comprehend what was going on. So I ask on Freeswitch-Users and get a > stupid answer that says "LOOK CLOSELY" (little did the poster know just how > much time I had expended "looking closely" !!!) ? I spent days looking > closely, days which could have been avoided if the developers took 2 > minutes to add 4 lines to the Wiki just listing and ever so briefly > describing the new variables ! > > > Everyone is entitled to their own opinion, of course, but I suspect this > is one area where ours may vary ! > > > B > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/99659bc6/attachment.html From msc at freeswitch.org Sat Jan 14 04:35:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 13 Jan 2012 17:35:06 -0800 Subject: [Freeswitch-users] Are you using mod_httapi Message-ID: Hello all! I know that many of you already know about mod_httapi. I am collecting examples and experiences from people who have had a chance to try it out. If you are using mod_httapi, even just for testing/tinkering, please contact me off list. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120113/4dc6d336/attachment.html From freeswitch at earthspike.net Sat Jan 14 04:52:38 2012 From: freeswitch at earthspike.net (John) Date: Sat, 14 Jan 2012 01:52:38 +0000 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> Message-ID: <4F10DFE6.3030105@earthspike.net> The German BSI is better compared to the US NSA or UK CESG than the FBI, I believe. But the 'paranoia compliance' is comparable! On 13/01/12 17:59, Michael Collins wrote: > > > On Fri, Jan 13, 2012 at 5:05 AM, Vik Killa > wrote: > > What exactly is GemeinschaftPBX? > > > Gemeinschaft is a "German-engineered open source PBX." Previously it > was based on Asterisk, however the German version of the FBI needs a > "paranoia-compliant" VoIP PBX that is open source. The Amooma guys > built that with FreeSWITCH, Kamailio, RoR, etc. in version 4. > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/31e218cb/attachment.html From georg at riseup.net Sat Jan 14 05:30:04 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 14 Jan 2012 03:30:04 +0100 Subject: [Freeswitch-users] Gemeinschaft 4.0 (FreeSWITCH based) released In-Reply-To: <4F10DFE6.3030105@earthspike.net> References: <4F0EA10E.7050301@chaschperli.ch> <4F0ECB04.20007@chaschperli.ch> <4F0FEFAD.3070307@chaschperli.ch> <4F10DFE6.3030105@earthspike.net> Message-ID: <4287159dbf7c0125772c86798990f6cc.squirrel@fulvetta.riseup.net> > The German BSI is better compared to the US NSA or UK CESG than the FBI, > I believe. But the 'paranoia compliance' is comparable! Actually they are not a secret service. They "take care" of the security of the governmental it systems, mostly at least. Something like a gov-CERT. Regards, Georg From bdfoster at endigotech.com Sat Jan 14 08:08:18 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 14 Jan 2012 00:08:18 -0500 Subject: [Freeswitch-users] stun.freeswitch.org stun service down Message-ID: Stun server is down at stun.freeswitch.org. I don't know who runs it but probably needs a service restart. Thanks! -BD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/1347d76e/attachment.html From anton.vazir at gmail.com Sat Jan 14 09:09:37 2012 From: anton.vazir at gmail.com (Anton VG) Date: Sat, 14 Jan 2012 11:09:37 +0500 Subject: [Freeswitch-users] Unable to get rtp and pre_answer working while looping back via sip (Sofia Bug or I miss something?) In-Reply-To: References: Message-ID: Just a little addition, tried with Septembers git and the latest. Also, if caller is a registered SIP endpoint - it get's EARLY MEDIA ok. 2012/1/13 Anton VG : > Hi Gentlemen! > > Broke my brain trying to force my FS, controlled by ESL inbound to > give me early media in indirect calling scheme. Please help: > > When call comes, in some cases, I do not just originate destination, > but need to dial on behalf of another user (not the user which > originally dialed in). > For this I dial back my FS box via sip, setting apropriate SIP > variables, so FS originates another call, as if the user I need > originated a call, and so so I have extra CDR and so on. > > BUT there is a problem:If I dial like this, - EARLY MEDIA does not > work. 180 and 183 are send correctly and in time. > however, all works if I dial directly (dialing schemes are given below) > > tcpdumping shows that there is no RTP coming to my FS box when I dial > doing one loop. BUT - RTP appears immediately after answering the > inbound channel: > (regardless if dialed destination answered or not) - so dialing > without the loop and ?doing pre_answer - I get RTP OK, > > if after a looped call i do pre_answer - I do not get RTP > Also, RTP appears immediately if instead pre_answer I do answer on call 2! > > My dialing schemes: > > **** Direct mode ********** > > Incoming call (UUID1)-> PARK > make UUID2 > originate (using uuid2) sofia/gateway/some_gateway &park > > Listen for CHANNEL_PROGRESS_MEDIA and do > pre_answer(UUID1) > uuid_bridge(UUID1,UUID2) > > ******** in a looped mode ******* > Thread 1 > _INCOMING CALL_ (call1) > Incoming call -> PARK > make UUID2 > originate (using uuid2) sofia/gateway/MY_IP_ADDRESS/another_user &park > > Listen for CHANNEL_PROGRESS_MEDIA and do > pre_answer(UUID1) > uuid_bridge(UUID1,UUID2) > > Thread 2 > _OUTGOING_ (call 2) > incoming call 2 (UUID3) (comes from call above) -> PARK > make UUID4 > originate (using uuid4) sofia/gateway/some_gateway &park > > Listen for CHANNEL_PROGRESS_MEDIA and do > pre_answer(UUID3) > uuid_bridge(UUID3,UUID4) > > Any clue? > > Regards, > Anton From psullivan at concerttelecom.com Sat Jan 14 08:15:37 2012 From: psullivan at concerttelecom.com (Paige Sullivan) Date: Sat, 14 Jan 2012 00:15:37 -0500 Subject: [Freeswitch-users] T.38 Faxing - at a loss... Message-ID: <4F110F79.1000902@concerttelecom.com> I've been trying in vein to get T.38 faxing to work properly. My provider is VOIP Innovations and my ATA is an EdgeWater 200EW (also tried a Grandstream HT-502) all of which support T.38. I have read the wiki about a thousand times and tried every combination of parameters in my dialplan that I can think of. I have the latest version of Freeswitch (built from git repo just today) and everything else works like a charm. What info should I post so maybe someone out there can give me some ideas? Like I said, I've tried about everything I can think of - even enabling t38_passthru = true on each individual call. Any ideas at all? Anyone using this provider and having T.38 work? I'm pretty desperate here and would even be willing to pay for an answer... -------------- next part -------------- A non-text attachment was scrubbed... Name: psullivan.vcf Type: text/x-vcard Size: 303 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/c2d98dff/attachment.vcf From anton.vazir at gmail.com Sat Jan 14 12:41:47 2012 From: anton.vazir at gmail.com (Anton VG) Date: Sat, 14 Jan 2012 14:41:47 +0500 Subject: [Freeswitch-users] Unable to get rtp and pre_answer working while looping back via sip (Sofia Bug or I miss something?) In-Reply-To: References: Message-ID: Please disregard the questions, the problem was in the originating ITSP - which did not accept the early media from my side. 2012/1/14 Anton VG : > Just a little addition, tried with Septembers git and the latest. > Also, if caller is a registered SIP endpoint - it get's EARLY MEDIA > ok. > > 2012/1/13 Anton VG : >> Hi Gentlemen! >> >> Broke my brain trying to force my FS, controlled by ESL inbound to >> give me early media in indirect calling scheme. Please help: >> >> When call comes, in some cases, I do not just originate destination, >> but need to dial on behalf of another user (not the user which >> originally dialed in). >> For this I dial back my FS box via sip, setting apropriate SIP >> variables, so FS originates another call, as if the user I need >> originated a call, and so so I have extra CDR and so on. >> >> BUT there is a problem:If I dial like this, - EARLY MEDIA does not >> work. 180 and 183 are send correctly and in time. >> however, all works if I dial directly (dialing schemes are given below) >> >> tcpdumping shows that there is no RTP coming to my FS box when I dial >> doing one loop. BUT - RTP appears immediately after answering the >> inbound channel: >> (regardless if dialed destination answered or not) - so dialing >> without the loop and ?doing pre_answer - I get RTP OK, >> >> if after a looped call i do pre_answer - I do not get RTP >> Also, RTP appears immediately if instead pre_answer I do answer on call 2! >> >> My dialing schemes: >> >> **** Direct mode ********** >> >> Incoming call (UUID1)-> PARK >> make UUID2 >> originate (using uuid2) sofia/gateway/some_gateway &park >> >> Listen for CHANNEL_PROGRESS_MEDIA and do >> pre_answer(UUID1) >> uuid_bridge(UUID1,UUID2) >> >> ******** in a looped mode ******* >> Thread 1 >> _INCOMING CALL_ (call1) >> Incoming call -> PARK >> make UUID2 >> originate (using uuid2) sofia/gateway/MY_IP_ADDRESS/another_user &park >> >> Listen for CHANNEL_PROGRESS_MEDIA and do >> pre_answer(UUID1) >> uuid_bridge(UUID1,UUID2) >> >> Thread 2 >> _OUTGOING_ (call 2) >> incoming call 2 (UUID3) (comes from call above) -> PARK >> make UUID4 >> originate (using uuid4) sofia/gateway/some_gateway &park >> >> Listen for CHANNEL_PROGRESS_MEDIA and do >> pre_answer(UUID3) >> uuid_bridge(UUID3,UUID4) >> >> Any clue? >> >> Regards, >> Anton From samia.razaq at gmail.com Sat Jan 14 13:53:21 2012 From: samia.razaq at gmail.com (sam) Date: Sat, 14 Jan 2012 10:53:21 +0000 (UTC) Subject: [Freeswitch-users] how to manipulate sip headers in Freeswitch Message-ID: hi, i am new to free switch and am trying to connect it to a call out system. But whenever i make the out call, the remote server says "too many Allow events" and the support representative says i need to remove the Allow-Events. does any one know how to manipulate or remove the "allow-event" part of the sip header ? My current allow events are Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer and the server has problem with Allow-Events. they only want talk and hold. thanks. -sam From brian at freeswitch.org Sat Jan 14 14:39:50 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 14 Jan 2012 05:39:50 -0600 Subject: [Freeswitch-users] how to manipulate sip headers in Freeswitch In-Reply-To: References: Message-ID: <44CED05F-C403-42C0-8CBE-0D67FE0E98FC@freeswitch.org> First what device are you talking to that requires this change? /b On Jan 14, 2012, at 4:53 AM, sam wrote: > hi, > i am new to free switch and am trying to connect it to a call out system. But > whenever i make the out call, the remote server says "too many Allow events" and > the support representative says i need to remove the Allow-Events. > > does any one know how to manipulate or remove the "allow-event" part of the sip > header ? > > My current allow events are > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > > and the server has problem with Allow-Events. they only want talk and hold. > > thanks. > -sam > -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/fe55f23b/attachment.html From samia.razaq at gmail.com Sat Jan 14 15:30:57 2012 From: samia.razaq at gmail.com (sam) Date: Sat, 14 Jan 2012 12:30:57 +0000 (UTC) Subject: [Freeswitch-users] how to manipulate sip headers in Freeswitch References: <44CED05F-C403-42C0-8CBE-0D67FE0E98FC@freeswitch.org> Message-ID: Manipulating Sip header in Freeswitch - "Status 500: Too many Allow-Event" I am using a local company's support for call out, and they would not tell me the device name nor the details at their end, only the error and the reason for it being too many allow-events. however, i had tested the same system on my local setup with spa 3102 and it worked fine. i dont know how to change the sip header part, and the error remains "Status 500: Too many Allow-Event". is there a conf file i need to change, because i dont want to go into the packet making problems. -sam From samia.razaq at gmail.com Sat Jan 14 15:53:09 2012 From: samia.razaq at gmail.com (sam) Date: Sat, 14 Jan 2012 12:53:09 +0000 (UTC) Subject: [Freeswitch-users] how to manipulate sip headers in Freeswitch References: <44CED05F-C403-42C0-8CBE-0D67FE0E98FC@freeswitch.org> Message-ID: Brian West writes: > > First what device are you talking to that requires this change? > /b > I am using another call out setup and i am only forwarding my calls via sip,rtp to them. i have no information about their devices and setup. only a voice-ip to which i send and from which i receive my calls. the error i get is "status 500 : too many Allow-Event". -- sam > > On Jan 14, 2012, at 4:53 AM, sam wrote: > hi,i am new to free switch and am trying to connect it to a call out system. Butwhenever i make the out call, the remote server says "too many Allow events" andthe support representative says i need to remove the Allow-Events.does any one know how to manipulate or remove the "allow-event" part of the sipheader ?My current allow events are?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,REFER, NOTIFY, PUBLISH, SUBSCRIBEAllow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,include-session-description, presence.winfo, message-summary, referand the server has problem with Allow-Events. they only want talk and hold.?thanks.-sam > > > > --? > Brian West? > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266? > Fax: ? +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From woodydickson at gmail.com Sat Jan 14 18:37:03 2012 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 14 Jan 2012 23:37:03 +0800 Subject: [Freeswitch-users] Measuring the call quality In-Reply-To: References: Message-ID: Hi, Please contact me offlist and I may have a solution for you. -------------------- Woody Dickson US LRN NPANXX 6/6 billing starting at 0.0008 US LRN Short Duration NPANXX 6/6 billing starting at 0.0009 supporting 2000 CPS US Offnet 60+ ASR at 0.019 Canada CC at 0.0034 China CC at 0.017 2012/1/14 Nicol?s Menoni > Hello, > > Is there a tool for measure the call quality and take the statistics? The > idea is take statistics in packet loss, jitter, delay, etc., per each call > and save these in a DB. > > Thank you! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Woody Dickson woodydickson at gmail.com US and Worldwide Termination ============ Contact me for the following offering ============ USA Onnet - 0.0049/min USA Offnet - 0.011/min USA Mobile starting from 0.001/min USA All NPANXX starting from 0.001/min USA Conference - 0.011/min India mobile - 0.0105/min China CLI - 0.0099/min China Non-CLI - 0.0057/min UK Mobile - 0.0565/min US Dialer - 0.007/min Canada Dialer - 0.0035/min UK Dialer - 0.0067/min -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/d35c44ef/attachment-0001.html From sherifomran2000 at yahoo.com Sat Jan 14 20:39:01 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 14 Jan 2012 09:39:01 -0800 (PST) Subject: [Freeswitch-users] Measuring the call quality In-Reply-To: Message-ID: <1326562741.7615.YahooMailClassic@web110801.mail.gq1.yahoo.com> for statistics see this http://www.cdr-stats.org/documentation/installation/ --- On Sat, 1/14/12, Woody Dickson wrote: From: Woody Dickson Subject: Re: [Freeswitch-users] Measuring the call quality To: "FreeSWITCH Users Help" Date: Saturday, January 14, 2012, 5:37 PM Hi, Please contact me offlist and I may have a solution for you. -------------------- Woody DicksonUS LRN NPANXX 6/6 billing starting at 0.0008 US LRN Short Duration NPANXX 6/6 billing starting at 0.0009 supporting 2000 CPS? US Offnet 60+ ASR at 0.019Canada CC at 0.0034 China CC at 0.017 2012/1/14 Nicol?s Menoni Hello, Is there a tool for measure the call quality and take the statistics? The idea is take statistics in packet loss, jitter, delay, etc., per each call and save these in a DB. Thank you! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Woody Dickson woodydickson at gmail.com US and Worldwide Termination ============ Contact me for the following offering?============ USA Onnet - 0.0049/min? USA Offnet - 0.011/min USA Mobile starting from 0.001/min USA All NPANXX starting from 0.001/min USA Conference - 0.011/min India mobile - 0.0105/min China CLI - 0.0099/min China Non-CLI - 0.0057/min UK Mobile - 0.0565/min US Dialer - 0.007/min Canada Dialer - 0.0035/min UK Dialer - 0.0067/min -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/02d5459f/attachment.html From herman.griffin at gmail.com Sat Jan 14 21:16:22 2012 From: herman.griffin at gmail.com (Herman Griffin) Date: Sat, 14 Jan 2012 10:16:22 -0800 Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND Message-ID: Hello Freeswitchers, I'm using stacked condition that are supposed to behave line logical AND operators. The call trace has been separated in to two section. The first section corresponds to the first set of stacked conditions in my dialplan and the second section corresponds to my second set of stacked conditions. The second condition in the second set of stack condition evaluates to FAIL (false). However the 'action' that are wrapped are still being exacuted. I expect the 'anti-action' to execute when any of the AND operands evaluates to FAIL. Note: I'm using break=never because I don't want a failed condition to break me out of the extension. There are multiple condition blocks in the extension. Regardless of whether break being set to never, it seems that the wrapped actions should not execute unless all conditions evalute to PASS (true) Can someone help point out my mistake? Thank you, Herman Griffin ------------------------------------ This is my call trace: Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) [emergency] ${open}(true) =~ /^true$/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) [emergency] ${emergency_call}() =~ /^true$/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) [emergency] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) [emergency] ${emergency_call}() =~ /^true$/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) [emergency] ${db(select/emergency/autoanswer)}(1) =~ /^1$/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(call_timeout=60) Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}) Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(autoanswered=true) Dialplan: sofia/external/Unknown at 72.37.252.18 Action answer() Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav) Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(group_confirm_key=1) Dialplan: sofia/external/Unknown at 72.37.252.18 Action bridge(user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}) ------------------------------------ This is the corresponding dialplan: From gb10hkzo-freeswitch at yahoo.co.uk Sat Jan 14 21:51:45 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sat, 14 Jan 2012 18:51:45 +0000 (GMT) Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? Message-ID: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> Hi, I'm trying to get a basic CURL XML going that will dynamically manage DDI numbers as part of a database-driven follow me function. My database generates the following output : ? ? ? ? ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
 When the action is "transfer", everything works great. ?However when "transfer" is the commented one and bridge is active nothing works. I have tried adding ? when bridging, but that has no effect either. Here's what it looks like when trying to bridge : 2012-01-14 15:19:24.136494 [INFO] mod_dialplan_xml.c:481 Processing 12345678 <12345678>->87654321 in context public Dialplan: sofia/internal/12345678 at 192.168.1.134 parsing [public->curl_dialplan] continue=false Dialplan: sofia/internal/12345678 at 192.168.1.134 Regex (PASS) [curl_dialplan] destination_number(87654321) =~ /^(87654321)$/ break=on-false Dialplan: sofia/internal/12345678 at 192.168.1.134 Action set(domain_name=my.domain.name)? Dialplan: sofia/internal/12345678 at 192.168.1.134 Action bridge(sofia/gateway/mygate/12345678)? 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/12345678 at 192.168.1.134) State Change CS_ROUTING -> CS_EXECUTE 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/12345678 at 192.168.1.134 [BREAK] 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/12345678 at 192.168.1.134) State ROUTING going to sleep 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/12345678 at 192.168.1.134) Running State Change CS_EXECUTE 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/12345678 at 192.168.1.134) State EXECUTE 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:241 sofia/internal/12345678 at 192.168.1.134 SOFIA EXECUTE 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:192 sofia/internal/12345678 at 192.168.1.134 Standard EXECUTE EXECUTE sofia/internal/12345678 at 192.168.1.134 set(domain_name=my.domain.name) 2012-01-14 15:19:24.176505 [DEBUG] mod_dptools.c:1281 sofia/internal/12345678 at 192.168.1.134 SET [domain_name]=[my.domain.name] EXECUTE sofia/internal/12345678 at 192.168.1.134 bridge(sofia/gateway/mygate/12345678) 2012-01-14 15:19:24.176505 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-01-14 15:19:24.176505 [NOTICE] switch_channel.c:930 New Channel sofia/external/12345678 [2408b0f2-3ec3-11e1-afaf-89a010d7e66c] 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:4674 (sofia/external/12345678) State Change CS_NEW -> CS_INIT 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 (sofia/external/12345678) Running State Change CS_INIT 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:401 (sofia/external/12345678) State INIT 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:85 sofia/external/12345678 SOFIA INIT 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:125 (sofia/external/12345678) State Change CS_INIT -> CS_ROUTING 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:401 (sofia/external/12345678) State INIT going to sleep 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 (sofia/external/12345678) Running State Change CS_ROUTING 2012-01-14 15:19:24.176505 [DEBUG] switch_channel.c:1890 (sofia/external/12345678) Callstate Change DOWN -> RINGING 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 (sofia/external/12345678) State ROUTING 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:148 sofia/external/12345678 SOFIA ROUTING 2012-01-14 15:19:24.176505 [DEBUG] switch_ivr_originate.c:66 (sofia/external/12345678) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 (sofia/external/12345678) State ROUTING going to sleep 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 (sofia/external/12345678) Running State Change CS_CONSUME_MEDIA 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:429 (sofia/external/12345678) State CONSUME_MEDIA 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:429 (sofia/external/12345678) State CONSUME_MEDIA going to sleep 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:875 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.176505 [DEBUG] sofia.c:5494 Channel sofia/external/12345678 entering state [calling][0] 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal sofia/external/12345678 [BREAK] 2012-01-14 15:19:24.316487 [DEBUG] sofia.c:5494 Channel sofia/external/12345678 entering state [terminated][503] 2012-01-14 15:19:24.316487 [DEBUG] switch_channel.c:2852 (sofia/external/12345678) Callstate Change RINGING -> HANGUP 2012-01-14 15:19:24.316487 [NOTICE] sofia.c:6258 Hangup sofia/external/12345678 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] From avi at avimarcus.net Sat Jan 14 21:59:08 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 14 Jan 2012 20:59:08 +0200 Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? In-Reply-To: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: The logs show you are getting a "NORMAL_TEMPORARY_FAILURE" from the endpoint. Have you tried a real number..? -Avi On Sat, Jan 14, 2012 at 8:51 PM, Bob Smith wrote: > Hi, > > I'm trying to get a basic CURL XML going that will dynamically manage DDI > numbers as part of a database-driven follow me function. > > > My database generates the following output : > > > > >
> > field="destination_number" expression="^(87654321)$"> > > > > > > >
> > > > > When the action is "transfer", everything works great. However when > "transfer" is the commented one and bridge is active nothing works. > > I have tried adding data="hangup_after_bridge=true"/> when bridging, but that has no effect > either. > > > Here's what it looks like when trying to bridge : > > > 2012-01-14 15:19:24.136494 [INFO] mod_dialplan_xml.c:481 Processing > 12345678 <12345678>->87654321 in context public > Dialplan: sofia/internal/12345678 at 192.168.1.134 parsing > [public->curl_dialplan] continue=false > Dialplan: sofia/internal/12345678 at 192.168.1.134 Regex (PASS) > [curl_dialplan] destination_number(87654321) =~ /^(87654321)$/ > break=on-false > Dialplan: sofia/internal/12345678 at 192.168.1.134 Action set(domain_name= > my.domain.name) > Dialplan: sofia/internal/12345678 at 192.168.1.134 Action > bridge(sofia/gateway/mygate/12345678) > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/12345678 at 192.168.1.134) State Change CS_ROUTING -> > CS_EXECUTE > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal > sofia/internal/12345678 at 192.168.1.134 [BREAK] > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/12345678 at 192.168.1.134) State ROUTING going to sleep > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/12345678 at 192.168.1.134) Running State Change CS_EXECUTE > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/12345678 at 192.168.1.134) State EXECUTE > 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:241 sofia/internal/ > 12345678 at 192.168.1.134 SOFIA EXECUTE > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/12345678 at 192.168.1.134 Standard EXECUTE > EXECUTE sofia/internal/12345678 at 192.168.1.134 set(domain_name= > my.domain.name) > 2012-01-14 15:19:24.176505 [DEBUG] mod_dptools.c:1281 sofia/internal/ > 12345678 at 192.168.1.134 SET [domain_name]=[my.domain.name] > EXECUTE sofia/internal/12345678 at 192.168.1.134bridge(sofia/gateway/mygate/12345678) > 2012-01-14 15:19:24.176505 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-01-14 15:19:24.176505 [NOTICE] switch_channel.c:930 New Channel > sofia/external/12345678 [2408b0f2-3ec3-11e1-afaf-89a010d7e66c] > 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:4674 > (sofia/external/12345678) State Change CS_NEW -> CS_INIT > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/12345678) Running State Change CS_INIT > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:401 > (sofia/external/12345678) State INIT > 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:85 sofia/external/12345678 > SOFIA INIT > 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:125 > (sofia/external/12345678) State Change CS_INIT -> CS_ROUTING > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:401 > (sofia/external/12345678) State INIT going to sleep > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/12345678) Running State Change CS_ROUTING > 2012-01-14 15:19:24.176505 [DEBUG] switch_channel.c:1890 > (sofia/external/12345678) Callstate Change DOWN -> RINGING > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 > (sofia/external/12345678) State ROUTING > 2012-01-14 15:19:24.176505 [DEBUG] mod_sofia.c:148 > sofia/external/12345678 SOFIA ROUTING > 2012-01-14 15:19:24.176505 [DEBUG] switch_ivr_originate.c:66 > (sofia/external/12345678) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:410 > (sofia/external/12345678) State ROUTING going to sleep > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:362 > (sofia/external/12345678) Running State Change CS_CONSUME_MEDIA > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:429 > (sofia/external/12345678) State CONSUME_MEDIA > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_state_machine.c:429 > (sofia/external/12345678) State CONSUME_MEDIA going to sleep > 2012-01-14 15:19:24.176505 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.176505 [DEBUG] sofia.c:5494 Channel > sofia/external/12345678 entering state [calling][0] > 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.316487 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/12345678 [BREAK] > 2012-01-14 15:19:24.316487 [DEBUG] sofia.c:5494 Channel > sofia/external/12345678 entering state [terminated][503] > 2012-01-14 15:19:24.316487 [DEBUG] switch_channel.c:2852 > (sofia/external/12345678) Callstate Change RINGING -> HANGUP > 2012-01-14 15:19:24.316487 [NOTICE] sofia.c:6258 Hangup > sofia/external/12345678 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/4b5ba01a/attachment-0001.html From paul at cupis.co.uk Sat Jan 14 22:07:35 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 14 Jan 2012 19:07:35 +0000 Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? In-Reply-To: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: <4F11D277.5000604@cupis.co.uk> On 14/01/12 18:51, Bob Smith wrote: > 2012-01-14 15:19:24.176505 [DEBUG] sofia.c:5494 Channel sofia/external/12345678 entering state [calling][0] > 2012-01-14 15:19:24.316487 [DEBUG] sofia.c:5494 Channel sofia/external/12345678 entering state [terminated][503] > 2012-01-14 15:19:24.316487 [DEBUG] switch_channel.c:2852 (sofia/external/12345678) Callstate Change RINGING -> HANGUP > 2012-01-14 15:19:24.316487 [NOTICE] sofia.c:6258 Hangup sofia/external/12345678 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] Looks like you are getting a 503 somewhere... try doing a SIP trace for a call, but you might be getting rejected by the far end or something (so an issue with your gateway configuration or bridge command, not with xml_curl specifically). Regards, From gb10hkzo-freeswitch at yahoo.co.uk Sat Jan 14 22:23:48 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sat, 14 Jan 2012 19:23:48 +0000 (GMT) Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? In-Reply-To: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: <1326569028.55647.YahooMailNeo@web29402.mail.ird.yahoo.com> Has it occurred to you that I might be obfuscating my numbers when posting in public to protect the innocent ? ?;-) >?Have you tried a real number..? Bridging works just fine thank you. Have been using the same PSTN destination number via the same gateway for testing stuff all last week. > but you might be getting rejected by the far end or something From avi at avimarcus.net Sat Jan 14 22:32:15 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 14 Jan 2012 21:32:15 +0200 Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? In-Reply-To: <1326569028.55647.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1326567105.38510.YahooMailNeo@web29402.mail.ird.yahoo.com> <1326569028.55647.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: So doing the bridge to the same number with static XML worked, but when bridging to the exact same endpoint with xml_curl you're getting a 503? Either way, it seems to be an actual endpoint response. A sip trace as Paul suggested might show something.. (BTW, I have been noticing several carriers sending a 503 instead of a busy..) -Avi On Sat, Jan 14, 2012 at 9:23 PM, Bob Smith wrote: > Has it occurred to you that I might be obfuscating my numbers when posting > in public to protect the innocent ? ;-) > > > Have you tried a real number..? > > > Bridging works just fine thank you. Have been using the same PSTN > destination number via the same gateway > for testing stuff all last week. > > > but you might be getting rejected by the far end or something > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/d0559464/attachment.html From jeff at jefflenk.com Sat Jan 14 22:42:45 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sat, 14 Jan 2012 11:42:45 -0800 (PST) Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND In-Reply-To: References: Message-ID: <1326570165847-7188271.post@n2.nabble.com> By putting the break=never you are defeating the "and" processing. Remove that. Move the second group into its own extension. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stacked-conditions-are-not-acting-like-logical-AND-tp7188082p7188271.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kris at kriskinc.com Sat Jan 14 23:47:24 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Sat, 14 Jan 2012 15:47:24 -0500 Subject: [Freeswitch-users] T.38 Faxing - at a loss... In-Reply-To: <4F110F79.1000902@concerttelecom.com> References: <4F110F79.1000902@concerttelecom.com> Message-ID: Hi Paige, As you've discovered, faxing is a beast! Let's start with a couple of questions: 1) What are you trying to accomplish? I'm guessing you're using FreeSWITCH as a PBX/SBC between your provider and your T.38 capable endpoints. 2) Does VoIP Innovations claim T.38 support for every call, every route, and every DID? In addition to answering these questions, could you post console output for the type(s) of calls you're attempting? You can post them to http://pastebin.freeswitch.org AFTER you enable "sofia global siptrace on". That should help us get started. On Sat, Jan 14, 2012 at 12:15 AM, Paige Sullivan wrote: > I've been trying in vein to get T.38 faxing to work properly. > > My provider is VOIP Innovations and my ATA is an EdgeWater 200EW (also tried > a Grandstream HT-502) all of which support T.38. ?I have read the wiki about > a thousand times and tried every combination of parameters in my dialplan > that I can think of. ?I have the latest version of Freeswitch (built from > git repo just today) and everything else works like a charm. > > What info should I post so maybe someone out there can give me some ideas? > ?Like I said, I've tried about everything I can think of - even enabling > t38_passthru = true on each individual call. > > Any ideas at all? ?Anyone using this provider and having T.38 work? ?I'm > pretty desperate here and would even be willing to pay for an answer... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From avi at avimarcus.net Sun Jan 15 00:04:08 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 14 Jan 2012 23:04:08 +0200 Subject: [Freeswitch-users] T.38 Faxing - at a loss... In-Reply-To: References: <4F110F79.1000902@concerttelecom.com> Message-ID: iirc, voip innovations has a special t.38 route. -Avi On Sat, Jan 14, 2012 at 10:47 PM, Kristian Kielhofner wrote: > Hi Paige, > > As you've discovered, faxing is a beast! Let's start with a couple > of questions: > > 1) What are you trying to accomplish? I'm guessing you're using > FreeSWITCH as a PBX/SBC between your provider and your T.38 capable > endpoints. > > 2) Does VoIP Innovations claim T.38 support for every call, every > route, and every DID? > > In addition to answering these questions, could you post console > output for the type(s) of calls you're attempting? You can post them > to http://pastebin.freeswitch.org AFTER you enable "sofia global > siptrace on". > > That should help us get started. > > On Sat, Jan 14, 2012 at 12:15 AM, Paige Sullivan > wrote: > > I've been trying in vein to get T.38 faxing to work properly. > > > > My provider is VOIP Innovations and my ATA is an EdgeWater 200EW (also > tried > > a Grandstream HT-502) all of which support T.38. I have read the wiki > about > > a thousand times and tried every combination of parameters in my dialplan > > that I can think of. I have the latest version of Freeswitch (built from > > git repo just today) and everything else works like a charm. > > > > What info should I post so maybe someone out there can give me some > ideas? > > Like I said, I've tried about everything I can think of - even enabling > > t38_passthru = true on each individual call. > > > > Any ideas at all? Anyone using this provider and having T.38 work? I'm > > pretty desperate here and would even be willing to pay for an answer... > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/2b40a98f/attachment-0001.html From notlikeme75 at yahoo.com Sun Jan 15 01:41:02 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 14 Jan 2012 14:41:02 -0800 (PST) Subject: [Freeswitch-users] cdr minutes of use per did Message-ID: <1326580862.62555.YahooMailNeo@web65311.mail.ac2.yahoo.com> does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. Calls by Day,?DNIS?-> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) Daily Month to Date (Start of Month = Day 21) Day DNIS DNIS Group Total Calls Minutes of Use Average Hold Time (Minutes) Total Calls Minutes of Use Average Hold Time (Minutes) ? No rows ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/d9cbc72f/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Sun Jan 15 01:59:03 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sat, 14 Jan 2012 22:59:03 +0000 (GMT) Subject: [Freeswitch-users] Problems with CURL_XML and bridging ? Message-ID: <1326581943.46678.YahooMailNeo@web29403.mail.ird.yahoo.com> Actually, you're quite right. ?I've gone back and double checked. ?Does not work with static XML. ?Same behaviour... transfer works, bridge fails. However I can definitely originate a call from the CLI over the same route ..... "originate sofia/gateway/mygate/12345678 &park" works great. That's what's really bugging me, if I can CLI originate to that number, if I can call that PSTN number from my FreeSwitch softphone, why on earth can't i do this simple bridge ! ;-( >So doing the bridge to the same number with static XML worked, but when > bridging to the exact same endpoint with xml_curl you're getting a 503? From avi at avimarcus.net Sun Jan 15 02:03:22 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 15 Jan 2012 01:03:22 +0200 Subject: [Freeswitch-users] cdr minutes of use per did In-Reply-To: <1326580862.62555.YahooMailNeo@web65311.mail.ac2.yahoo.com> References: <1326580862.62555.YahooMailNeo@web65311.mail.ac2.yahoo.com> Message-ID: It depends on how your cdr statistics are stored.. CSV file, sqlite, pgsql, mysql.. for how easy it is to run a query. And if you want just inbound or outbound "on" that DID. Then just add that SQL query to a script and have a cron job run it.. e.g. if you use mysql in fusionpbx to store your cdrs (via xml_cdr posts) I can run: SELECT SUM (`billsec`)/60 FROM `v_xml_cdr ` WHERE `sip_to_user` = '$number' AND `start_stamp` LIKE '2011-11%' = 831.4500 minutes for all of November 2011. (pgsql doesn't like using a like like that on a time stamp field, it seems..) Your query, and source info, will vary.. -Avi On Sun, Jan 15, 2012 at 12:41 AM, Rodney wrote: > does anyone have a cdr viewer or method that can help me determine the > total minutes for all calls on a specified DID for the hour/day/month to > date? I would like a method to generate such a report and maybe have it > emailed on a 24 hour basis to the person in charge of the DID. the > following I have seen in the past with any method to find the original > program that would give an output like this. > Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard > Time) > Daily Month to Date > (Start of Month = Day 21) Day DNIS DNIS > Group Total > Calls Minutes > of Use Average > Hold Time > (Minutes) Total > Calls Minutes > of Use Average > Hold Time > (Minutes) ? No rows ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/5aecc643/attachment-0001.html From notlikeme75 at yahoo.com Sun Jan 15 02:07:21 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 14 Jan 2012 15:07:21 -0800 (PST) Subject: [Freeswitch-users] cdr records /destinations total minutes Message-ID: <1326582441.95290.YahooMailNeo@web65316.mail.ac2.yahoo.com> I have a situation now where transfer to conferences etc. are not part of the total per call, it shows up as a total for a different destination. (the one they hung up from). is there a way to get total from the start of the call no matter what destination they hang up from? ie. keep destination DID the same no matter what destination the end up at so the call total will be intact? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/475861da/attachment.html From notlikeme75 at yahoo.com Sun Jan 15 02:13:26 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 14 Jan 2012 15:13:26 -0800 (PST) Subject: [Freeswitch-users] cdr minutes of use per did In-Reply-To: References: Message-ID: <1326582806.6814.YahooMailNeo@web65315.mail.ac2.yahoo.com> Avi, thank you for quick response, I am using sqlite with fusionpbx but it is in windows, so I don't believe I can run cron jobs in windows.? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Saturday, January 14, 2012 6:03 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 137 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. cdr minutes of use per did (Rodney) ? 2. Re: Problems with CURL_XML and bridging ? (Bob Smith) ? 3. Re: cdr minutes of use per did (Avi Marcus) does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. Calls by Day,?DNIS?-> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) Daily Month to Date (Start of Month = Day 21) Day DNIS DNIS Group Total Calls Minutes of Use Average Hold Time (Minutes) Total Calls Minutes of Use Average Hold Time (Minutes) ? No rows ? Actually, you're quite right. ?I've gone back and double checked. ?Does not work with static XML. ?Same behaviour... transfer works, bridge fails. However I can definitely originate a call from the CLI over the same route ..... "originate sofia/gateway/mygate/12345678 &park" works great. That's what's really bugging me, if I can CLI originate to that number, if I can call that PSTN number from my FreeSwitch softphone, why on earth can't i do this simple bridge ! ;-( >So doing the bridge to the same number with static XML worked, but when > bridging to the exact same endpoint with xml_curl you're getting a 503? It depends on how your cdr?statistics?are stored.. CSV file, sqlite, pgsql, mysql.. for how easy it is to run a query. And if you want just inbound or outbound "on" that DID. Then just add that SQL query to a script and have a cron job run it.. e.g. if you use mysql in fusionpbx to store your cdrs (via xml_cdr posts) I can run: SELECT?SUM(`billsec`)/60?FROM?`v_xml_cdr`?WHERE?`sip_to_user`?=?'$number'?AND?`start_stamp`?LIKE?'2011-11%' =?831.4500 minutes for all of November 2011. (pgsql doesn't like using a like like that on a time stamp field, it seems..) Your query, and source info, will vary.. -Avi On Sun, Jan 15, 2012 at 12:41 AM, Rodney wrote: does anyone have a cdr viewer or method that can help me determine the total minutes for all calls on a specified DID for the hour/day/month to date? I would like a method to generate such a report and maybe have it emailed on a 24 hour basis to the person in charge of the DID. the following I have seen in the past with any method to find the original program that would give an output like this. >Calls by Day,?DNIS?-> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard Time) > >Daily >Month to Date >(Start of Month = Day 21) >Day >DNIS >DNIS >Group >Total >Calls >Minutes >of Use >Average >Hold Time >(Minutes) >Total >Calls >Minutes >of Use >Average >Hold Time >(Minutes) >? No rows ? >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/a6ca00c7/attachment-0001.html From bdfoster at endigotech.com Sun Jan 15 06:06:32 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 14 Jan 2012 22:06:32 -0500 Subject: [Freeswitch-users] Using Puppet on large scale setups Message-ID: Has anyone had experience with Puppet and if so, have you used it in conjunction with Freeswitch? Just trying to see if this is a good avenue for server management. -BDF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/ce4a32ee/attachment.html From herman.griffin at gmail.com Sun Jan 15 06:39:32 2012 From: herman.griffin at gmail.com (Herman Griffin) Date: Sat, 14 Jan 2012 19:39:32 -0800 Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND In-Reply-To: <1326570165847-7188271.post@n2.nabble.com> References: <1326570165847-7188271.post@n2.nabble.com> Message-ID: Hello everyone, In irc SwK said that freeswitch AND operator is lazy. To me this means that if a condition is set to break=true and it evaluates to FAIL, then the that break=never will invert the meaning of FAIL; Change it to a PASS . But why doesn't break=false behave in this same manner when there is only a single condition? Take this call trace and dialplan for example. I use a single condition with break=false. However, unlike the stacked conditions, the action is not executed when the single condition evaluates to FAIL. Why doesn't break=never cause the action to be executed in a non stacked condition? Call trace: Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) [emergency_set_variables] break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) [emergency_set_variables] break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) INLINE EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) Dialplan: ------------------------------------------------- I also getting confusing behavior with anti-action and stacked condition. I expect the anti-action to be executed if any of the stacked conditions evaluates to FAIL. However, in the case below, the extension simply breaks if any of the condition evaluates to FAIL. Call trace: Dialplan: sofia/external/Unknown at 72.37.252.18 parsing [public->emergency_bridge] continue=true Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) [emergency_bridge] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ break=on-false Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) [emergency_bridge] ${emergency_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/Unknown at 72.37.252.18 parsing [public->public_did] continue=false <-- [[MOVES ON TO ANOTHER EXTENSION WITH EXECUTING THE anti-action]] Dialplan: SwK suggested that I use a script language, which is probably what I'll end up doing. However, I'm very interested in understanding why the scenarios above. Are these bugs or are they just counter intuitive rules that we must simply memorize? Thanks for your input. Herman Griffin On Sat, Jan 14, 2012 at 11:42 AM, Jeff Lenk wrote: > By putting the break=never you are defeating the "and" processing. Remove > that. > > Move the second group into its own extension. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stacked-conditions-are-not-acting-like-logical-AND-tp7188082p7188271.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From valery.kalinin at gmail.com Sun Jan 15 07:12:33 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Sun, 15 Jan 2012 10:12:33 +0600 Subject: [Freeswitch-users] Critical error buffer size Message-ID: What is error: 2012-01-15 10:12:36.123103 [CRIT] switch_core_codec.c:660 Buffer size sanity check failed! What should I do? FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) OS: CentOS 5.5 From vetali100 at gmail.com Sun Jan 15 08:12:11 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sat, 14 Jan 2012 21:12:11 -0800 Subject: [Freeswitch-users] Commercial G729 codec - more calls than available licenses Message-ID: Hi, If we buy 5 commercial licenses of G729, and suddenly we get more users dialing, would the 6th, etc call fail, or FS will just propose one of the free codecs from the outgoing codecs list "G729,PCMA,PCMU"? Thank you, Vitalie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120114/081792db/attachment.html From shaheryarkh at googlemail.com Sun Jan 15 13:58:16 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 15 Jan 2012 15:58:16 +0500 Subject: [Freeswitch-users] how to manipulate sip headers in Freeswitch In-Reply-To: References: <44CED05F-C403-42C0-8CBE-0D67FE0E98FC@freeswitch.org> Message-ID: Interesting.. You can try stripping this sip header in your dial plan like this, and try again. If the termination carrier says Allow-Event header is mandatory then try this, Hope this helps. Thank you. On Sat, Jan 14, 2012 at 5:30 PM, sam wrote: > Manipulating Sip header in Freeswitch - "Status 500: Too many Allow-Event" > > I am using a local company's support for call out, and they would not tell > me > the device name nor the details at their end, only the error and the > reason for > it being too many allow-events. > > however, i had tested the same system on my local setup with spa 3102 and > it > worked fine. i dont know how to change the sip header part, and the error > remains "Status 500: Too many Allow-Event". is there a conf file i need to > change, because i dont want to go into the packet making problems. > > -sam > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/40dd9f67/attachment.html From isis at isisdesign.com Sun Jan 15 11:17:28 2012 From: isis at isisdesign.com (Daniel Melnechuk) Date: Sun, 15 Jan 2012 03:17:28 -0500 Subject: [Freeswitch-users] Freeswitch with Bluebox on Mac OS X 10.6.8 Message-ID: Dear Freeswitch Users, I am just setting up for the first time, a freeswitch with bluebox on a mac mini running 10.6.8 snow leopard. I am trying to move off of a callweaver setup i have had going for a couple of years. Everything seemed to be going right along. Started with one "device" and could get and receive calls. Then added a second device and could call from one to the other. Then after doing something which at this point i confess i don't remember, one of the devices won't get a call anymore, but can still place one. What i keep getting in the log file is: 2012-01-15 00:47:28.529909 [DEBUG] switch_channel.c:1097 EXPORT (export_vars) [sip_invite_req_uri]=[sip:2001error/user_not_registered] But in fact the device (iSoftPhone on another Mac) says it is connected and Bluebox sip registrations shows it as registered. The device is on the LAN. And it can call out. I tried XLite instead of iSoftphone and the same thing happens. So now i'm starting to think the config has gotten messed up under the hood. In case it was a config issue for the device that wasn't receiving calls i created a new device and configured iSoftphone to use that device auth but same thing persists. No firewalls on. The device that works both directions is Bria for iPhone. Since i'm working with two new packages, freeswitch and bluebox, not sure where to go next. Any clues as to what i could look for would be most appreciated. I know i haven't given much config info but main thing to know that both devices are "Internal" numbers, they are configed the same, only one has a problem receiving calls. I haven't done much so far so i'm thinking to wipe clean the config and start from scratch if nothing shakes out soon. Peace, Dan From anton.jugatsu at gmail.com Sun Jan 15 14:13:29 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sun, 15 Jan 2012 15:13:29 +0400 Subject: [Freeswitch-users] T.38 Faxing - at a loss... In-Reply-To: <4F110F79.1000902@concerttelecom.com> References: <4F110F79.1000902@concerttelecom.com> Message-ID: First, try to catche RE-INVITE with, for example, ngrep -d eth0 -q -W byline -i t38 host . End debug udptl flow with tshark -R t38 or tcpdump -s0 and then analyze with wireshark. 2012/1/14 Paige Sullivan > I've been trying in vein to get T.38 faxing to work properly. > > My provider is VOIP Innovations and my ATA is an EdgeWater 200EW (also > tried a Grandstream HT-502) all of which support T.38. I have read the > wiki about a thousand times and tried every combination of parameters in my > dialplan that I can think of. I have the latest version of Freeswitch > (built from git repo just today) and everything else works like a charm. > > What info should I post so maybe someone out there can give me some ideas? > Like I said, I've tried about everything I can think of - even enabling > t38_passthru = true on each individual call. > > Any ideas at all? Anyone using this provider and having T.38 work? I'm > pretty desperate here and would even be willing to pay for an answer... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/2ce655a3/attachment-0001.html From woodydickson at gmail.com Sun Jan 15 17:16:38 2012 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 15 Jan 2012 22:16:38 +0800 Subject: [Freeswitch-users] cdr records /destinations total minutes In-Reply-To: <1326582441.95290.YahooMailNeo@web65316.mail.ac2.yahoo.com> References: <1326582441.95290.YahooMailNeo@web65316.mail.ac2.yahoo.com> Message-ID: You may need to store the first destination number and then export it to the CDR. They can be done in dialplan. On Sun, Jan 15, 2012 at 7:07 AM, Rodney wrote: > I have a situation now where transfer to conferences etc. are not part of > the total per call, it shows up as a total for a different destination. > (the one they hung up from). is there a way to get total from the start of > the call no matter what destination they hang up from? ie. keep destination > DID the same no matter what destination the end up at so the call total > will be intact? > -- Woody Dickson woodydickson at gmail.com US and Worldwide Termination US LRN NPANXX 6/6 billing starting at 0.0008 US LRN Short Duration NPANXX 6/6 billing starting at 0.0009 supporting 2000 CPS US Offnet 60+ ASR at 0.019 Canada CC at 0.0034 China CC at 0.017 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/d49e3738/attachment.html From justlikeef at gmail.com Sun Jan 15 22:48:55 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Sun, 15 Jan 2012 14:48:55 -0500 Subject: [Freeswitch-users] Freeswitch with Bluebox on Mac OS X 10.6.8 In-Reply-To: References: Message-ID: <201201151448.55926.justlikeef@gmail.com> Most of the time, this is because you have changed and saved the Domain field under the Location definition and resaved the number without changing the auth user or resaving the user. What you end up is a user that can register using the old domain, but when you try to bridge a call to them, you are actually attempting to bridge to the new domain. Freeswitch is by default multitenant, and your auth user must match the user at domain format presented in bluebox when you create the user and the number. In other words, the location has to be correct before you do anything else, and if you change the domain, you have to start over. Without seeing your directory and dialplan, this is a guess based on your symptoms. There is a dedicated Blue.Box mailing list (http://groups.google.com/group/2600hz-users) and IRC channel (irc://irc.freenode.net/2600hz) to ask Blue.Box related questions on that you may get better answers from. Thanks, Rob On Sunday 15 January 2012 03:17:28 Daniel Melnechuk wrote: > Dear Freeswitch Users, > > I am just setting up for the first time, a freeswitch with bluebox on a mac mini running 10.6.8 snow leopard. I am trying to move off of a callweaver setup i have had going for a couple of years. > > Everything seemed to be going right along. Started with one "device" and could get and receive calls. Then added a second device and could call from one to the other. > > Then after doing something which at this point i confess i don't remember, one of the devices won't get a call anymore, but can still place one. What i keep getting in the log file is: > > 2012-01-15 00:47:28.529909 [DEBUG] switch_channel.c:1097 EXPORT (export_vars) [sip_invite_req_uri]=[sip:2001error/user_not_registered] > > But in fact the device (iSoftPhone on another Mac) says it is connected and Bluebox sip registrations shows it as registered. The device is on the LAN. And it can call out. > > I tried XLite instead of iSoftphone and the same thing happens. So now i'm starting to think the config has gotten messed up under the hood. In case it was a config issue for the device that wasn't receiving calls i created a new device and configured iSoftphone to use that device auth but same thing persists. No firewalls on. > > The device that works both directions is Bria for iPhone. > > Since i'm working with two new packages, freeswitch and bluebox, not sure where to go next. > > Any clues as to what i could look for would be most appreciated. I know i haven't given much config info but main thing to know that both devices are "Internal" numbers, they are configed the same, only one has a problem receiving calls. > > I haven't done much so far so i'm thinking to wipe clean the config and start from scratch if nothing shakes out soon. > > Peace, > Dan > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/a33a390e/attachment.html From freeswitch at earthspike.net Mon Jan 16 01:17:03 2012 From: freeswitch at earthspike.net (John) Date: Sun, 15 Jan 2012 22:17:03 +0000 Subject: [Freeswitch-users] Critical error buffer size In-Reply-To: References: Message-ID: <4F13505F.4070801@earthspike.net> Raise a bug on jira.freeswitch.org, I guess. You'll need to pastebin your dialplan and the rest of the debug output leading up to the error for it to be meaningful. John On 15/01/12 04:12, Valery Kalinin wrote: > What is error: > 2012-01-15 10:12:36.123103 [CRIT] switch_core_codec.c:660 Buffer size > sanity check failed! > > What should I do? > > FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) > OS: CentOS 5.5 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From frank at rosengart.de Mon Jan 16 01:18:14 2012 From: frank at rosengart.de (Frank Rosengart) Date: Sun, 15 Jan 2012 23:18:14 +0100 Subject: [Freeswitch-users] undefined symbol: snd_config in mod_portaudio.so Message-ID: <4F1350A6.7040000@rosengart.de> Hi, after updating and compiling FS and mod_portaudio, I can not load the module anymore: Error Loading module /home/frank/fs_celt/mod/mod_portaudio.so **/home/frank/fs_celt/mod/mod_portaudio.so: undefined symbol: snd_config** It's the latest git checkout, and I did upgrade from Ubuntu natty to oneiric since it was working last time. I found a similar issue on a mailinglist in 2009, which seems to be a libtool update problem. My mod_portaudio.so is not linked to the libasound2, which could be the reason for the missing snd_config I did ran bootstrap. Any help appreciated Frank From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 01:24:02 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sun, 15 Jan 2012 22:24:02 +0000 (GMT) Subject: [Freeswitch-users] Voicemail overriding leg_delay_start ? Message-ID: <1326666242.59402.YahooMailNeo@web29402.mail.ird.yahoo.com> Hi, I am trying to implement a follow-me using the XML syntax below.? However? instead of the second number ringing after 12 seconds, the voicemail picks up instead ? (For those of your reading this thread and remembering my earlier bridging problems, I've fixed those, so that's not the issue here ;-) ?? ?????
???????? ??????????? ???????? ????? ??
From notlikeme75 at yahoo.com Mon Jan 16 01:35:55 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 15 Jan 2012 14:35:55 -0800 (PST) Subject: [Freeswitch-users] sched_hangup with playback Message-ID: <1326666955.63456.YahooMailNeo@web65315.mail.ac2.yahoo.com> I would like to schedule a hangup at a set call time max but play a recording letting the person know the max time is up. is this possible in the dialplan? I have read?http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup?and see a reason code to set but no mention of playback warning the party. In addition, I would also like a 5 minute or so warning, like on a calling card platform. like "this call will end in ____ minutes" kind of thing. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/09a5c939/attachment.html From avi at avimarcus.net Mon Jan 16 01:41:59 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 16 Jan 2012 00:41:59 +0200 Subject: [Freeswitch-users] sched_hangup with playback In-Reply-To: <1326666955.63456.YahooMailNeo@web65315.mail.ac2.yahoo.com> References: <1326666955.63456.YahooMailNeo@web65315.mail.ac2.yahoo.com> Message-ID: Try: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_broadcast and schedule both the warning and the actual hangup. -Avi On Mon, Jan 16, 2012 at 12:35 AM, Rodney wrote: > I would like to schedule a hangup at a set call time max but play a > recording letting the person know the max time is up. is this possible in > the dialplan? > I have read > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup and see > a reason code to set but no mention of playback warning the party. In > addition, I would also like a 5 minute or so warning, like on a calling > card platform. like "this call will end in ____ minutes" kind of thing. > thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/7c165041/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 01:43:03 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Sun, 15 Jan 2012 22:43:03 +0000 (GMT) Subject: [Freeswitch-users] Voicemail overriding leg_delay_start ? (more info) Message-ID: <1326667383.12727.YahooMailNeo@web29404.mail.ird.yahoo.com> Should add that I also tried the following syntax, in this instance, the second bridge number rings, but then on expiry, it never gets to voicemail, it just appears to drop: ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
Has anyone got this sort of setup running ? From avi at avimarcus.net Mon Jan 16 02:38:00 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 16 Jan 2012 01:38:00 +0200 Subject: [Freeswitch-users] Bsuy, Wait, Callback - lua script..? Message-ID: Hi, I'm looking to implement a callback functionality. I get an incoming invite which has the user who is making the call and his destination. I'm supposed to 486 that and then after 2 second call them back, and connect the call. Should I do a hangup hook to a lua script which sleeps 2 seconds and does an originate? or zombie exec? And if I do the originate within the lua script, won't that leave the lua script running for the whole calls? What's the best way.. or even just a good way to do this - I don't expect more than 10 simultaneous calls as of now on this particular method. Thanks, -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/9dfd387d/attachment.html From rhow at exemail.com.au Mon Jan 16 07:05:25 2012 From: rhow at exemail.com.au (Ryan How) Date: Mon, 16 Jan 2012 12:05:25 +0800 Subject: [Freeswitch-users] voicemail prompts are very quiet Message-ID: <4F13A205.7050102@exemail.com.au> Hi, I've got a basic freeswitch install, just minor tweaks from the default (adding extensions, a gateway and DID). I find the volume of the voicemail prompts are very quiet. Recordings and recorded greetings are great. Is there a way to turn the volume of them up? I had a look through the wiki, but I can't seem to find anything with gain or volume. Maybe I am just looking for the wrong thing. Thanks, Ryan From bdfoster at endigotech.com Mon Jan 16 07:17:04 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 15 Jan 2012 23:17:04 -0500 Subject: [Freeswitch-users] voicemail prompts are very quiet In-Reply-To: <4F13A205.7050102@exemail.com.au> References: <4F13A205.7050102@exemail.com.au> Message-ID: Turn up your phone. -BDF On Sun, Jan 15, 2012 at 11:05 PM, Ryan How wrote: > Hi, > > I've got a basic freeswitch install, just minor tweaks from the default > (adding extensions, a gateway and DID). I find the volume of the > voicemail prompts are very quiet. Recordings and recorded greetings are > great. Is there a way to turn the volume of them up? > > I had a look through the wiki, but I can't seem to find anything with > gain or volume. Maybe I am just looking for the wrong thing. > > Thanks, Ryan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120115/23accccc/attachment.html From ocset at the800group.com Mon Jan 16 07:49:35 2012 From: ocset at the800group.com (ocset) Date: Mon, 16 Jan 2012 12:49:35 +0800 Subject: [Freeswitch-users] Cordless IP Phone Message-ID: <4F13AC5F.4030400@the800group.com> Hi I have had great success using Yealink phones with Freeswitch and a customer has asked me for a cordless phone for their office. I want to ensure that they can keep as much functionality with a cordless phone as they have with the Yealink T28. I am looking at the Siemens C610IP phone but don't know how well it plays with Freeswitch. Can someone please shed some light on the Siemens phone or any alternative that you have successfully implemented. Thanks in advance From jaybinks at gmail.com Mon Jan 16 08:20:43 2012 From: jaybinks at gmail.com (Jay Binks) Date: Mon, 16 Jan 2012 15:20:43 +1000 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F13AC5F.4030400@the800group.com> References: <4F13AC5F.4030400@the800group.com> Message-ID: <3628F761-EEC3-4651-B9E4-A3FD606C44E4@gmail.com> I have great success with my Siemens sip sect handset , love it , it works great !!! On 16/01/2012, at 2:49 PM, ocset wrote: > Siemens From paul at cupis.co.uk Mon Jan 16 10:46:38 2012 From: paul at cupis.co.uk (=?utf-8?B?UGF1bCBDdXBpcw==?=) Date: Mon, 16 Jan 2012 07:46:38 +0000 Subject: [Freeswitch-users] =?utf-8?q?Voicemail_overriding_leg=5Fdelay=5Fs?= =?utf-8?b?dGFydCA/IChtb3JlCWluZm8p?= Message-ID: <0MNLsv-1RkDE83eIr-006vFX@mrelayeu.kundenserver.de> Try setting continue_on_fail to true before the bridge command. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/822d4a3d/attachment.html From Claudio.Cavalera at italtel.it Mon Jan 16 10:55:47 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 16 Jan 2012 08:55:47 +0100 Subject: [Freeswitch-users] Effect of sync_clock on calls? In-Reply-To: <83FF8D7C9F526E44B77C97DD2891652A12C045CB@mse17be1.mse17.exchange.ms> References: <83FF8D7C9F526E44B77C97DD2891652A12C04551@mse17be1.mse17.exchange.ms><1FFF97C269757C458224B7C895F35F1502ECC0@cantor.std.visionutv.se> <83FF8D7C9F526E44B77C97DD2891652A12C045CB@mse17be1.mse17.exchange.ms> Message-ID: Hello, just for the record the command is: fsctl sync_clock_when_idle I've added it to http://wiki.freeswitch.org/wiki/Clock Kind Regards, Claudio 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. ----------------------------------------------------------------------------------------------------- From freeswitch at earthspike.net Mon Jan 16 11:39:21 2012 From: freeswitch at earthspike.net (John) Date: Mon, 16 Jan 2012 08:39:21 +0000 Subject: [Freeswitch-users] voicemail prompts are very quiet In-Reply-To: <4F13A205.7050102@exemail.com.au> References: <4F13A205.7050102@exemail.com.au> Message-ID: <4F13E239.9040502@earthspike.net> Ryan, There is a long thread on this topic from 26/6/11 onwards, subject "Proper prompt gain/level". I cannot remember the outcome, but the discussion should provide your answers or reinforce your point, one or the other. John On 16/01/12 04:05, Ryan How wrote: > Hi, > > I've got a basic freeswitch install, just minor tweaks from the default > (adding extensions, a gateway and DID). I find the volume of the > voicemail prompts are very quiet. Recordings and recorded greetings are > great. Is there a way to turn the volume of them up? > > I had a look through the wiki, but I can't seem to find anything with > gain or volume. Maybe I am just looking for the wrong thing. > > Thanks, Ryan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From avi at avimarcus.net Mon Jan 16 11:49:58 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 16 Jan 2012 10:49:58 +0200 Subject: [Freeswitch-users] cdr minutes of use per did In-Reply-To: <1326582806.6814.YahooMailNeo@web65315.mail.ac2.yahoo.com> References: <1326582806.6814.YahooMailNeo@web65315.mail.ac2.yahoo.com> Message-ID: Windows has cron-like things. Google has several options, including the built-in windows scheduler. -Avi On Sun, Jan 15, 2012 at 1:13 AM, Rodney wrote: > Avi, > > thank you for quick response, I am using sqlite with fusionpbx but it is > in windows, so I don't believe I can run cron jobs in windows. > > ------------------------------ > *From:* "freeswitch-users-request at lists.freeswitch.org" < > freeswitch-users-request at lists.freeswitch.org> > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Saturday, January 14, 2012 6:03 PM > *Subject:* FreeSWITCH-users Digest, Vol 67, Issue 137 > > ----- Forwarded Message ----- > > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. cdr minutes of use per did (Rodney) > 2. Re: Problems with CURL_XML and bridging ? (Bob Smith) > 3. Re: cdr minutes of use per did (Avi Marcus) > does anyone have a cdr viewer or method that can help me determine the > total minutes for all calls on a specified DID for the hour/day/month to > date? I would like a method to generate such a report and maybe have it > emailed on a 24 hour basis to the person in charge of the DID. the > following I have seen in the past with any method to find the original > program that would give an output like this. > Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard > Time) > Daily Month to Date > (Start of Month = Day 21) Day DNIS DNIS > Group Total > Calls Minutes > of Use Average > Hold Time > (Minutes) Total > Calls Minutes > of Use Average > Hold Time > (Minutes) ? No rows ? > Actually, you're quite right. I've gone back and double checked. Does > not work with static XML. Same behaviour... transfer works, bridge fails.. > > However I can definitely originate a call from the CLI over the same route > ..... "originate sofia/gateway/mygate/12345678 &park" works great. > > That's what's really bugging me, if I can CLI originate to that number, if > I can call that PSTN number from my FreeSwitch softphone, why on earth > can't i do this simple bridge ! ;-( > > > >So doing the bridge to the same number with static XML worked, but when > > bridging to the exact same endpoint with xml_curl you're getting a 503? > > > It depends on how your cdr statistics are stored.. CSV file, sqlite, > pgsql, mysql.. for how easy it is to run a query. > And if you want just inbound or outbound "on" that DID. > > Then just add that SQL query to a script and have a cron job run it.. > e.g. if you use mysql in fusionpbx to store your cdrs (via xml_cdr posts) > I can run: > SELECT SUM > (`billsec`)/60 FROM `v_xml_cdr > ` WHERE `sip_to_user` = '$number' AND > `start_stamp` LIKE > '2011-11%' > = 831.4500 minutes for all of November 2011. (pgsql doesn't like using a > like like that on a time stamp field, it seems..) > > Your query, and source info, will vary.. > > -Avi > > > > On Sun, Jan 15, 2012 at 12:41 AM, Rodney wrote: > > does anyone have a cdr viewer or method that can help me determine the > total minutes for all calls on a specified DID for the hour/day/month to > date? I would like a method to generate such a report and maybe have it > emailed on a 24 hour basis to the person in charge of the DID. the > following I have seen in the past with any method to find the original > program that would give an output like this. > Calls by Day, DNIS -> DID ACCOUNT-> Friday, 8/19/11 (Eastern Standard > Time) > Daily Month to Date > (Start of Month = Day 21) Day DNIS DNIS > Group Total > Calls Minutes > of Use Average > Hold Time > (Minutes) Total > Calls Minutes > of Use Average > Hold Time > (Minutes) ? No rows ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/49e3bdac/attachment-0001.html From tha_tux at hotmail.com Mon Jan 16 16:34:47 2012 From: tha_tux at hotmail.com (Tux Tux) Date: Mon, 16 Jan 2012 14:34:47 +0100 Subject: [Freeswitch-users] Caller-Caller-ID-Number on originate using ESL Message-ID: originate {sip_callee_id_numberorigination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML amteam') -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/f51ad506/attachment.html From tha_tux at hotmail.com Mon Jan 16 16:40:04 2012 From: tha_tux at hotmail.com (Tux Tux) Date: Mon, 16 Jan 2012 14:40:04 +0100 Subject: [Freeswitch-users] Caller-Caller-ID-Number on originate Message-ID: Hi, When trying this: originate {sip_callee_id_numberorigination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML default') Caller-Caller-ID-Number will eventually be set to: internal/sip:102 at 192.168.57.209:32710;rinstance=44018f313d6dd1aa While when using two softphones, the Caller-Caller-ID-Number will stay 102 instead of internal/sip:102 at 192.168.57.209:32710;rinstance=44018f313d6dd1aa Can I set a channel variable to keep the Caller-Caller-ID-Number the same? Discard my last email please. Thanks, Nico -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/87bc2e86/attachment.html From tha_tux at hotmail.com Mon Jan 16 16:48:28 2012 From: tha_tux at hotmail.com (Tux Tux) Date: Mon, 16 Jan 2012 14:48:28 +0100 Subject: [Freeswitch-users] Caller-Caller-ID-Number on originate In-Reply-To: References: Message-ID: Sorry for wasting you time it works now, I use this to make a call from the fs_cli and let it go through the dialplan: originate {sip_callee_id_number=102,origination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML default') Had to set sip_callee_id_number... From: tha_tux at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Mon, 16 Jan 2012 14:40:04 +0100 Subject: [Freeswitch-users] Caller-Caller-ID-Number on originate Hi, When trying this: originate {sip_callee_id_numberorigination_caller_id_name=103,origination_caller_id_number=103,instant_ringback=true,ignore_early_media=true,ringback=\'%(1000,4000,425.0,0.0)\'}user/102 &transfer('103 XML default') Caller-Caller-ID-Number will eventually be set to: internal/sip:102 at 192.168.57.209:32710;rinstance=44018f313d6dd1aa While when using two softphones, the Caller-Caller-ID-Number will stay 102 instead of internal/sip:102 at 192.168.57.209:32710;rinstance=44018f313d6dd1aa Can I set a channel variable to keep the Caller-Caller-ID-Number the same? Discard my last email please. Thanks, Nico _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/9fa08a89/attachment.html From mustafa.pk at gmail.com Mon Jan 16 16:59:54 2012 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 16 Jan 2012 18:59:54 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. Message-ID: Hi, i have upgraded FS from previously running vesion 1.0.head (git-472ab0c 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we are connected with our carrier's Huawei SoftX3000, outgoing calls are working ok but as soon as an incoming call arrives SoftX3000 drops it with the message "SSF00159L00681 Session Timer Check Message Failed" Here i have pasted both working (before upgrade) and current version sip traces. http://pastebin.freeswitch.org/18136 i am sure it's not a bug and i might be missing some configuration parameter, any clue? thanks -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/342eedd2/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 17:54:55 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 16 Jan 2012 14:54:55 +0000 (GMT) Subject: [Freeswitch-users] Voicemail overriding leg_delay_start ? Message-ID: <1326725695.78718.YahooMailNeo@web29403.mail.ird.yahoo.com> Thanks Paul. ?Will try. >?Try setting continue_on_fail to true before the bridge command -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/3bf53f44/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 17:57:21 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 16 Jan 2012 14:57:21 +0000 (GMT) Subject: [Freeswitch-users] Javascript switch_ivr_originate.c:177 Invalid Application ? Message-ID: <1326725841.23854.YahooMailNeo@web29401.mail.ird.yahoo.com> Hi, I'm trying to get an authenticated bridge going along the lines of the one featured on this website (http://blog.shimaore.net/2009/03/better-followme-for-freeswitch.html). However I'm getting the following error in the console : 2012-01-16 14:52:43.174602 [ERR] switch_ivr_originate.c:177 Invalid Application /usr/local/freeswitch/scripts/confirm.js Is there any special magic required that the blog poster forgot to mention ? (e.g. some includes or similar in the Javascript ?).? Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/5e1e30a7/attachment.html From avi at avimarcus.net Mon Jan 16 18:06:54 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 16 Jan 2012 17:06:54 +0200 Subject: [Freeswitch-users] Javascript switch_ivr_originate.c:177 Invalid Application ? In-Reply-To: <1326725841.23854.YahooMailNeo@web29401.mail.ird.yahoo.com> References: <1326725841.23854.YahooMailNeo@web29401.mail.ird.yahoo.com> Message-ID: The example looks right. Can you post your code exactly? I think you missed a "javascript" reference.. e.g instead of: maybe you did ? -Avi On Mon, Jan 16, 2012 at 4:57 PM, Bob Smith wrote: > Hi, > > I'm trying to get an authenticated bridge going along the lines of the one > featured on this website ( > http://blog.shimaore.net/2009/03/better-followme-for-freeswitch.html). > > However I'm getting the following error in the console : > > 2012-01-16 14:52:43.174602 [ERR] switch_ivr_originate.c:177 Invalid > Application /usr/local/freeswitch/scripts/confirm.js > > Is there any special magic required that the blog poster forgot to mention > ? (e.g. some includes or similar in the Javascript ?). > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/8099a03b/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 18:32:05 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 16 Jan 2012 15:32:05 +0000 (GMT) Subject: [Freeswitch-users] Javascript switch_ivr_originate.c:177 Invalid Application ? Message-ID: <1326727925.65747.YahooMailNeo@web29404.mail.ird.yahoo.com> Hi Avi, The pertinent lines are : So I'm a step further now, I get: 2012-01-16 15:26:20.954602 [ERR] confirm.js:61 near ?SyntaxError: invalid returnreturn true; The same occurs if I use brackets around the "true". I've adapted the script slightly to use a pin pulled from a database (but I'm no Javascript Guru, so I'm guessing I missed a glaring bug !). Thanks for your help. ============================================================= // confirm.js - Freeswitch Call PIN Check // // console_log("info", "Destination: "+ session.destination + "\n"); // if(!session.getVariable('my_legconf')) { ? console_log("info", "No B-Leg Confirmation Requested \n"); ? exit(); } // // // Variables... // var v_continue = false; var conf_pin = session.getVariable('my_legpin'); var attempts = 3; var sound_file = "ivr-please_enter_pin_followed_by_pound.wav"; var pin_lastdigit=""; var pin = new Object(); pin.digits = ""; function parseInput (session,type,data,arg) { if ( type == "dtmf" ) { pin_lastdigit = data.digit; pin.digits += pin_lastdigit; console.log("info","Read digit " + pin_lastdigit + "\n"); return false; } } // // if (session.ready()) { session.answer(); session.flushDigits(); console.log("info","Starting PIN Collection\n"); var cnt = attempts; while (session.ready() && ! v_continue && cnt-- > 0) { session.execute("sleep","200"); session.streamFile(sound_file,parseInput); while (pin_lastdigit != "#" && sesson.ready()) { session.collectInput(parseInput,pin,5000); } console.log("info","Collected PIN: " + pin.digits + "\n"); if ( pin.digits == conf_pin ) { v_continue = true; console.log("info","PIN OK !.\n"); } else { pin_lastdigit = ""; pin.digits = ""; session.flushDigits(); } } } else { console.log("info","Session not ready.\n"); } // // if(v_continue) { return true; } else { return false; } ============================================================= -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/9fc764e7/attachment.html From rhow at exemail.com.au Mon Jan 16 19:11:30 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 17 Jan 2012 00:11:30 +0800 Subject: [Freeswitch-users] voicemail prompts are very quiet In-Reply-To: <4F13E239.9040502@earthspike.net> References: <4F13A205.7050102@exemail.com.au> <4F13E239.9040502@earthspike.net> Message-ID: <4F144C32.2070308@exemail.com.au> Thanks for the info. I read the thread. I don't really understand the technical stuff to that sort of level. It seems there were no changes to the system. All I noticed was that the voice prompts are very quiet when compared to a recorded message. I keep turning the volume up on the phone but it doesn't go no louder :). If I can't live with it I'll boost the levels and see what happens :) On 16/01/2012 4:39 PM, John wrote: > Ryan, > > There is a long thread on this topic from 26/6/11 onwards, subject > "Proper prompt gain/level". I cannot remember the outcome, but the > discussion should provide your answers or reinforce your point, one or > the other. > > John > > On 16/01/12 04:05, Ryan How wrote: >> Hi, >> >> I've got a basic freeswitch install, just minor tweaks from the default >> (adding extensions, a gateway and DID). I find the volume of the >> voicemail prompts are very quiet. Recordings and recorded greetings are >> great. Is there a way to turn the volume of them up? >> >> I had a look through the wiki, but I can't seem to find anything with >> gain or volume. Maybe I am just looking for the wrong thing. >> >> Thanks, Ryan >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gb10hkzo-freeswitch at yahoo.co.uk Mon Jan 16 19:14:48 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 16 Jan 2012 16:14:48 +0000 (GMT) Subject: [Freeswitch-users] Need Less CPU Intensive Way to Detect Voicemail and Answering Machines Message-ID: <1326730488.92018.YahooMailNeo@web29404.mail.ird.yahoo.com> How about something along these lines (http://blog.shimaore.net/2009/03/better-followme-for-freeswitch.html). If a user fails to confirm then assume it's a robot and move along your dialplan ? B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/f4be22c4/attachment.html From krice at freeswitch.org Mon Jan 16 19:15:04 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 16 Jan 2012 10:15:04 -0600 Subject: [Freeswitch-users] voicemail prompts are very quiet In-Reply-To: <4F144C32.2070308@exemail.com.au> Message-ID: The sound files should all be mastered at the same level, keep in mind when using a softphone for testing there are 2 gain controls... 1 for your mic and 1 for you speakers... Do you have a hard phone to test with? K On 1/16/12 10:11 AM, "Ryan How" wrote: > Thanks for the info. I read the thread. I don't really understand the > technical stuff to that sort of level. It seems there were no changes to > the system. All I noticed was that the voice prompts are very quiet when > compared to a recorded message. I keep turning the volume up on the > phone but it doesn't go no louder :). If I can't live with it I'll boost > the levels and see what happens :) > > > > On 16/01/2012 4:39 PM, John wrote: >> Ryan, >> >> There is a long thread on this topic from 26/6/11 onwards, subject >> "Proper prompt gain/level". I cannot remember the outcome, but the >> discussion should provide your answers or reinforce your point, one or >> the other. >> >> John >> >> On 16/01/12 04:05, Ryan How wrote: >>> Hi, >>> >>> I've got a basic freeswitch install, just minor tweaks from the default >>> (adding extensions, a gateway and DID). I find the volume of the >>> voicemail prompts are very quiet. Recordings and recorded greetings are >>> great. Is there a way to turn the volume of them up? >>> >>> I had a look through the wiki, but I can't seem to find anything with >>> gain or volume. Maybe I am just looking for the wrong thing. >>> >>> Thanks, Ryan >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rhow at exemail.com.au Mon Jan 16 20:29:58 2012 From: rhow at exemail.com.au (Ryan How) Date: Tue, 17 Jan 2012 01:29:58 +0800 Subject: [Freeswitch-users] voicemail prompts are very quiet In-Reply-To: References: Message-ID: <4F145E96.8000300@exemail.com.au> I'm using a Nokia e51 with the nokia sip client. it only has volume control, no mic gain control. perhaps its inbuilt gain is too high? On 17/01/2012 12:15 AM, Ken Rice wrote: > The sound files should all be mastered at the same level, keep in mind when > using a softphone for testing there are 2 gain controls... 1 for your mic > and 1 for you speakers... > > Do you have a hard phone to test with? > > K > > > On 1/16/12 10:11 AM, "Ryan How" wrote: > >> Thanks for the info. I read the thread. I don't really understand the >> technical stuff to that sort of level. It seems there were no changes to >> the system. All I noticed was that the voice prompts are very quiet when >> compared to a recorded message. I keep turning the volume up on the >> phone but it doesn't go no louder :). If I can't live with it I'll boost >> the levels and see what happens :) >> >> >> >> On 16/01/2012 4:39 PM, John wrote: >>> Ryan, >>> >>> There is a long thread on this topic from 26/6/11 onwards, subject >>> "Proper prompt gain/level". I cannot remember the outcome, but the >>> discussion should provide your answers or reinforce your point, one or >>> the other. >>> >>> John >>> >>> On 16/01/12 04:05, Ryan How wrote: >>>> Hi, >>>> >>>> I've got a basic freeswitch install, just minor tweaks from the default >>>> (adding extensions, a gateway and DID). I find the volume of the >>>> voicemail prompts are very quiet. Recordings and recorded greetings are >>>> great. Is there a way to turn the volume of them up? >>>> >>>> I had a look through the wiki, but I can't seem to find anything with >>>> gain or volume. Maybe I am just looking for the wrong thing. >>>> >>>> Thanks, Ryan >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dan at subformat.net Mon Jan 16 17:29:27 2012 From: dan at subformat.net (dan at subformat.net) Date: Mon, 16 Jan 2012 14:29:27 -0000 Subject: [Freeswitch-users] Play media dialplan Message-ID: Hi guys, I'm new to Freeswitch and am currently looking to setup a dialplan that basically does the following: Ring an extension for 10 secs if not answered, play media file. After that has played, go to voice mail. This doesn't seem to work though, it goes straight from ringing the extension to voicemail and misses out the media file. If you setup the dialplan to just play the media file it plays fine, just not when its in the middle of a dialplan. If you could point me in the right direction, that would be great. Thanks From gerald.weber at besharp.at Mon Jan 16 18:39:18 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Mon, 16 Jan 2012 15:39:18 +0000 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice Message-ID: Hello, i'm trying to send custom events using PHP ESL Library. (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 21-35-19 -0600)) My PHP Code: addHeader("Agents","2022"); $sock->sendEvent($e); ?> When I connect to fs using telnet and run the php script from another xterm, I can see the following output: Content-Length: 657 Content-Type: text/event-json { "Event-Subclass": "CONFIG::AGENT_LIST", "Event-Name": "SOCKET_DATA", "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea", "FreeSWITCH-Hostname": "freeswitch.local", "FreeSWITCH-Switchname": "freeswitch.local", "FreeSWITCH-IPv4": "192.168.20.73", "FreeSWITCH-IPv6": "::1", "Event-Date-Local": "2012-01-16 15:20:23", "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT", "Event-Date-Timestamp": "1326723623776184", "Event-Calling-File": "mod_event_socket.c", "Event-Calling-Function": "read_packet", "Event-Calling-Line-Number": "1188", "Command": "sendevent CUSTOM", "Event-Name": "CUSTOM", "Agents": "2022", "ZMQ-Msg-Cnt": "3508" } Looks ok, but i don't get why "Event-Name" is sent twice. Shouldn't there be an Event-Name: CUSTOM only ? Is there a way to avoid this or am i missing something ? thx & regards, gw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/184934c0/attachment.html From djbinter at gmail.com Mon Jan 16 21:21:09 2012 From: djbinter at gmail.com (DJB International) Date: Mon, 16 Jan 2012 10:21:09 -0800 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: I am just curious whether there was a 200 OK with CSeq: 2 BYE responding back to the BYE message from Huawei since I did not see it in your pastebin? -djbinter On Mon, Jan 16, 2012 at 5:59 AM, Ghulam Mustafa wrote: > Hi, > > i have upgraded FS from previously running vesion 1.0.head (git-472ab0c > 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we > are connected with our carrier's Huawei SoftX3000, outgoing calls are > working ok but as soon as an incoming call arrives SoftX3000 drops it with > the message "SSF00159L00681 Session Timer Check Message Failed" > > Here i have pasted both working (before upgrade) and current version sip > traces. http://pastebin.freeswitch.org/18136 > > i am sure it's not a bug and i might be missing some configuration > parameter, any clue? > > thanks > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/51571666/attachment.html From anthony.minessale at gmail.com Mon Jan 16 21:32:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 12:32:01 -0600 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: Try this from the build root. git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 libs/sofia-sip/libsofia-sip-ua | patch -p1 -R touch libs/sofia-sip/.update make mod_sofia-clean make mod_sofia-install This commit took away the Require: timer option which is an optional field and clearly the pedantic switch is worrying about it. On Mon, Jan 16, 2012 at 7:59 AM, Ghulam Mustafa wrote: > Hi, > > i have upgraded FS from previously running vesion 1.0.head (git-472ab0c > 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we > are connected with our carrier's Huawei SoftX3000, outgoing calls are > working ok but as soon as an incoming call arrives SoftX3000 drops it with > the message "SSF00159L00681 Session Timer Check Message Failed" > > Here i have pasted both working (before upgrade) and current version sip > traces. http://pastebin.freeswitch.org/18136 > > i am sure it's not a bug and i might be missing some configuration > parameter, any clue? > > thanks > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/ee2ba9e1/attachment-0001.html From mustafa.pk at gmail.com Mon Jan 16 21:33:38 2012 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 16 Jan 2012 23:33:38 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: The version numbers mentioned in my email are vice versa, its a copy paste mistake. On Jan 16, 2012 6:59 PM, "Ghulam Mustafa" wrote: > Hi, > > i have upgraded FS from previously running vesion 1.0.head (git-472ab0c > 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we > are connected with our carrier's Huawei SoftX3000, outgoing calls are > working ok but as soon as an incoming call arrives SoftX3000 drops it with > the message "SSF00159L00681 Session Timer Check Message Failed" > > Here i have pasted both working (before upgrade) and current version sip > traces. http://pastebin.freeswitch.org/18136 > > i am sure it's not a bug and i might be missing some configuration > parameter, any clue? > > thanks > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/fd2e43de/attachment.html From shaheryarkh at googlemail.com Mon Jan 16 21:54:46 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 16 Jan 2012 23:54:46 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: Hi, couldn't see your sip trace in pastebin, getting authentication error, perhaps i forgot my password. Anyways, send me trace as attachment off-list please. Thank you. On Mon, Jan 16, 2012 at 11:33 PM, Ghulam Mustafa wrote: > The version numbers mentioned in my email are vice versa, its a copy > paste mistake. > On Jan 16, 2012 6:59 PM, "Ghulam Mustafa" wrote: > >> Hi, >> >> i have upgraded FS from previously running vesion 1.0.head (git-472ab0c >> 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we >> are connected with our carrier's Huawei SoftX3000, outgoing calls are >> working ok but as soon as an incoming call arrives SoftX3000 drops it with >> the message "SSF00159L00681 Session Timer Check Message Failed" >> >> Here i have pasted both working (before upgrade) and current version sip >> traces. http://pastebin.freeswitch.org/18136 >> >> i am sure it's not a bug and i might be missing some configuration >> parameter, any clue? >> >> thanks >> >> -- >> Ghulam Mustafa >> cell: +92 333.611.7681 >> sip: cyrenity at ekiga.net >> mail: mustafa.pk at gmail.com >> web: cyrenity.wordpress.com >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/1afdc35c/attachment.html From dujinfang at gmail.com Mon Jan 16 22:15:02 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 17 Jan 2012 03:15:02 +0800 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg Message-ID: Hi, I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. I manually comment the following lines and it seems to work. // switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); } switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/2288bf57/attachment.html From anthony.minessale at gmail.com Mon Jan 16 22:21:11 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 13:21:11 -0600 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: I'm pretty sure I just explained the problem. On Mon, Jan 16, 2012 at 12:54 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Hi, > > couldn't see your sip trace in pastebin, getting authentication error, > perhaps i forgot my password. Anyways, send me trace as attachment off-list > please. > > Thank you. > > > On Mon, Jan 16, 2012 at 11:33 PM, Ghulam Mustafa wrote: > >> The version numbers mentioned in my email are vice versa, its a copy >> paste mistake. >> On Jan 16, 2012 6:59 PM, "Ghulam Mustafa" wrote: >> >>> Hi, >>> >>> i have upgraded FS from previously running vesion 1.0.head (git-472ab0c >>> 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we >>> are connected with our carrier's Huawei SoftX3000, outgoing calls are >>> working ok but as soon as an incoming call arrives SoftX3000 drops it with >>> the message "SSF00159L00681 Session Timer Check Message Failed" >>> >>> Here i have pasted both working (before upgrade) and current version sip >>> traces. http://pastebin.freeswitch.org/18136 >>> >>> i am sure it's not a bug and i might be missing some configuration >>> parameter, any clue? >>> >>> thanks >>> >>> -- >>> Ghulam Mustafa >>> cell: +92 333.611.7681 >>> sip: cyrenity at ekiga.net >>> mail: mustafa.pk at gmail.com >>> web: cyrenity.wordpress.com >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/423de3e4/attachment.html From paul at cupis.co.uk Mon Jan 16 22:21:37 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Mon, 16 Jan 2012 19:21:37 +0000 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: Message-ID: <4F1478C1.2000909@cupis.co.uk> On 16/01/12 14:29, dan at subformat.net wrote: > If you could point me in the right direction, that would be great. Can you provide a log of the call from FreeSWITCH, on http://pastebin.freeswitch.org/ for us to look at, please? Regards, From aksrini at hotmail.com Mon Jan 16 22:49:06 2012 From: aksrini at hotmail.com (Srini K) Date: Mon, 16 Jan 2012 11:49:06 -0800 Subject: [Freeswitch-users] Event date timestamp in different time zone Message-ID: Hi,Event Date timestamp is in GMT. For example in CHANNEL_CALLSTATE event, Caller-Channel-Created-Time is in GMT (unixTimestamp). Is there any way to get it to user defined time zone (Setting it in config file). CHANNEL_CALLSTATE 4dead845-6636-4ace-ae9a-c257f9b179cb 2012-01-16%2010%3A49%3A12 Mon,%2016%20Jan%202012%2018%3A49%3A12%20GMT 1326739752973123 1326739752973123 Any help would be greatly appreciatedRegards Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/a71071e9/attachment.html From tahir at ictinnovations.com Mon Jan 16 22:46:00 2012 From: tahir at ictinnovations.com (tahir almas) Date: Tue, 17 Jan 2012 00:46:00 +0500 Subject: [Freeswitch-users] Open Source unified autodialer software released Message-ID: Pleased to announce the release of open source Fax , SMS and Voice broadcasting software solution ICTDialer http://www.ictdialer.org developed over reknown Drupal Conent Mnagment System and powerfull Plivo Communication framework , Your contribution and suggestions are welcome Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/dbfb5490/attachment.html From shaheryarkh at googlemail.com Mon Jan 16 23:21:14 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 17 Jan 2012 01:21:14 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: Thanks, i guess i missed your earlier reply. Thank you. On Tue, Jan 17, 2012 at 12:21 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I'm pretty sure I just explained the problem. > > > On Mon, Jan 16, 2012 at 12:54 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> Hi, >> >> couldn't see your sip trace in pastebin, getting authentication error, >> perhaps i forgot my password. Anyways, send me trace as attachment off-list >> please. >> >> Thank you. >> >> >> On Mon, Jan 16, 2012 at 11:33 PM, Ghulam Mustafa wrote: >> >>> The version numbers mentioned in my email are vice versa, its a copy >>> paste mistake. >>> On Jan 16, 2012 6:59 PM, "Ghulam Mustafa" wrote: >>> >>>> Hi, >>>> >>>> i have upgraded FS from previously running vesion 1.0.head (git-472ab0c >>>> 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we >>>> are connected with our carrier's Huawei SoftX3000, outgoing calls are >>>> working ok but as soon as an incoming call arrives SoftX3000 drops it with >>>> the message "SSF00159L00681 Session Timer Check Message Failed" >>>> >>>> Here i have pasted both working (before upgrade) and current version >>>> sip traces. http://pastebin.freeswitch.org/18136 >>>> >>>> i am sure it's not a bug and i might be missing some configuration >>>> parameter, any clue? >>>> >>>> thanks >>>> >>>> -- >>>> Ghulam Mustafa >>>> cell: +92 333.611.7681 >>>> sip: cyrenity at ekiga.net >>>> mail: mustafa.pk at gmail.com >>>> web: cyrenity.wordpress.com >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/fd9eef7f/attachment.html From shaheryarkh at googlemail.com Tue Jan 17 01:01:50 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 17 Jan 2012 03:01:50 +0500 Subject: [Freeswitch-users] Annoying error while compiling FreeSWITCH Message-ID: I am getting build failure repeatedly due to this annoying -Werror. Each time any warning appears in code, the compilation fails treating it as error, e.g. /usr/src/svn-src/freeswitch/src/mod/applications/mod_voicemail_ivr/mod_voicemail_ivr.c:57:6: error: variable 'argc' set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors Can you guy help me how to get rid of it? I tried finding and replacing -Werror with -Wall but there are so many make files to edit.. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/1000c5fd/attachment.html From alexis.mailinglist at de-bruyn.fr Tue Jan 17 01:37:20 2012 From: alexis.mailinglist at de-bruyn.fr (Alexis de BRUYN [Mailinglists]) Date: Mon, 16 Jan 2012 23:37:20 +0100 Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Message-ID: <4F14A6A0.5080409@de-bruyn.fr> Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia at 192.168.1.6:5080 BIND-URL sip:mod_sofia at 192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001 at X.Y.Z.T Contact: "Freeswitch" Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001 at X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000 at X.Y.Z.T Contact: "freeswitch" Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000 at X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr From msc at freeswitch.org Tue Jan 17 01:54:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 14:54:21 -0800 Subject: [Freeswitch-users] Nous avons besoin de votre assistance avec les traductions! Message-ID: Salut! If you can read the subject line then you already know what I need: help with some French translations. As you know, our very own Marc Olivier Chouinard (Moc) has produced a set of French-Canadian sound prompts for everyone's benefit. The FreeSWITCH team and Barracuda Networks recently added some prompts to that set. We are looking to expand that sound set even more with some of our newest sound prompts. I have a grand total of about 220 prompts that I would like to get translated into French. Many of these are less-common prompts that a lot of people won't need, but I would like to get the translations done anyway so that we can get the most useful ones recorded as soon as possible. Also, some of these prompts are silly ones that may not translate too well into French. (I've heard that the French don't have a sense of humor but I cannot confirm that report at this time. :) If you'd like to help with this translation project please email me off list and I will send you a spreadsheet with what we have to do. Merci beaucoup pour tout votre soutien! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/5fe2a1ff/attachment.html From anthony.minessale at gmail.com Tue Jan 17 02:11:41 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 17:11:41 -0600 Subject: [Freeswitch-users] Annoying error while compiling FreeSWITCH In-Reply-To: References: Message-ID: no, The solution is to fix the code and report the compilation errors as bugs. Warnings ARE errors. We have a strict policy. On Mon, Jan 16, 2012 at 4:01 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I am getting build failure repeatedly due to this annoying -Werror. Each > time any warning appears in code, the compilation fails treating it as > error, e.g. > > /usr/src/svn-src/freeswitch/src/mod/applications/mod_voicemail_ivr/mod_voicemail_ivr.c:57:6: > error: variable 'argc' set but not used [-Werror=unused-but-set-variable] > cc1: all warnings being treated as errors > > Can you guy help me how to get rid of it? I tried finding and replacing > -Werror with -Wall but there are so many make files to edit.. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/801d481d/attachment.html From msc at freeswitch.org Tue Jan 17 02:24:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 15:24:33 -0800 Subject: [Freeswitch-users] Annoying error while compiling FreeSWITCH In-Reply-To: References: Message-ID: On Mon, Jan 16, 2012 at 2:01 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I am getting build failure repeatedly due to this annoying -Werror. Each > time any warning appears in code, the compilation fails treating it as > error, e.g. > > /usr/src/svn-src/freeswitch/src/mod/applications/mod_voicemail_ivr/mod_voicemail_ivr.c:57:6: > error: variable 'argc' set but not used [-Werror=unused-but-set-variable] > cc1: all warnings being treated as errors > In addition to what Tony said, this is a good reason to report bugs to jira.freeswitch.org. Also, just for reference, this particular module is not critical for FreeSWITCH to work - it was contributed by a community member (IRC: moc). Moc has A LOT of practice fixing his errors, so assign the bug to him. ;) In the meantime you can exclude this module from being built just by editing modules.conf in your freeswitch source directory. In fact, it might be worth it for you to review which modules you are building by default. There may be modules you don't want or need that you can skip. This will make your builds go faster and you won't get tripped up by annoying errors on modules that you don't actually use. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/28566ad9/attachment.html From anthony.minessale at gmail.com Tue Jan 17 02:26:54 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 17:26:54 -0600 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: I am not seeing that, perhaps you are on an older version? On Mon, Jan 16, 2012 at 9:39 AM, Gerald Weber wrote: > Hello,**** > > ** ** > > i?m trying to send custom events using PHP ESL Library.**** > > (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 21-35-19 > -0600))**** > > My PHP Code:**** > > ** ** > > > require_once('ESL.php');**** > > ** ** > > $sock = new ESLconnection('192.168.20.73', '8021', 'ClueCon');**** > > ** ** > > $e = new ESLevent("CUSTOM","CONFIG::AGENT_LIST");**** > > $e->addHeader("Agents","2022");**** > > ** ** > > $sock->sendEvent($e);**** > > ?>**** > > ** ** > > When I connect to fs using telnet and run the php script from another > xterm, I can see the following output:**** > > ** ** > > Content-Length: 657**** > > Content-Type: text/event-json**** > > ** ** > > {**** > > "Event-Subclass": "CONFIG::AGENT_LIST",**** > > "Event-Name": "SOCKET_DATA",**** > > "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea",**** > > "FreeSWITCH-Hostname": "freeswitch.local",**** > > "FreeSWITCH-Switchname": "freeswitch.local",**** > > "FreeSWITCH-IPv4": "192.168.20.73",**** > > "FreeSWITCH-IPv6": "::1",**** > > "Event-Date-Local": "2012-01-16 15:20:23",**** > > "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT",**** > > "Event-Date-Timestamp": "1326723623776184",**** > > "Event-Calling-File": "mod_event_socket.c",**** > > "Event-Calling-Function": "read_packet",**** > > "Event-Calling-Line-Number": "1188",**** > > "Command": "sendevent CUSTOM",**** > > "Event-Name": "CUSTOM",**** > > "Agents": "2022",**** > > "ZMQ-Msg-Cnt": "3508"**** > > }**** > > ** ** > > Looks ok, but i don?t get why ?Event-Name? is sent twice.**** > > Shouldn?t there be an Event-Name: CUSTOM only ?**** > > Is there a way to avoid this or am i missing something ?**** > > ** ** > > thx & regards,**** > > gw**** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/cff85e3e/attachment-0001.html From msc at freeswitch.org Tue Jan 17 03:29:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 16:29:00 -0800 Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND In-Reply-To: References: <1326570165847-7188271.post@n2.nabble.com> Message-ID: Hi Herman, These are common questions when dealing with the dialplan. I highly recommend that you acquire the FreeSWITCH "bridge book" (see link near top of wiki.freeswitch.org) and read chapters 5 and 8. (Full disclosure: I wrote chapter 5 and Darren Schreiber wrote chapter 8.) Both of these chapters cover the break attribute in detail. I think you might be confusing the purpose of the break attribute. Consider this dp fragment: When the dialplan parser gets to this extension, the first thing it does is test the "foo" field. If the test fails then the parser *does not even look at the rest of this extension*. Why not? Because all conditions have an implied break="on-false". The following two conditions are identical in function: The dialplan parser logged this: Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) [emergency_set_variables] break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) [emergency_set_variables] break=never Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) INLINE EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) Note: I added the color so that you could see which dialplan lines correspond to the log output. The first condition (purple) fails. ({$caller_id_name} has the value "Unknown" which fails the regex test against /^Emerg_/) However, since you have break="never", the parser continues on to the next condition inside this extension. Had you done break="on-false" (or no break attribute) then the parser would have moved on to the next extension in the dialplan. The second condition (orange) also fails. Again, you have break="never", so even though it fails, the parser moves on to the next condition. The third condition (green) passes, so the parser adds the to the task list. (Since you have inline="true" the action gets executed immediately during dialplan parsing instead later on during the "execution phase".) In other words, the log output is exactly what I would expect it to be. As to your other question about anti-actions in a stack: this also is a common question. It looks like you are trying to do something like this: IF (cond1 AND cond2 AND cond3) THEN do actions ELSE do other actions ENDIF You cannot do this particular construct just with conditions and anti-actions. Instead you'll need the brand spanking new "regex" syntax mentioned here: http://wiki.freeswitch.org/wiki/Dialplan_XML#Multiple_Conditions_.28Logical_OR.2C_XOR.29 So, to do what you wanted to do in the second example (in your second post) you could try this: The tells the parser, "Hey, execute the 's only if all regexes PASS, otherwise execute any 's". That should give you what you need. Hope this helps! -Michael On Sat, Jan 14, 2012 at 7:39 PM, Herman Griffin wrote: > Hello everyone, > > In irc SwK said that freeswitch AND operator is lazy. To me this means > that if a condition is set to break=true and it evaluates to FAIL, > then the that break=never will invert the meaning of FAIL; Change it > to a PASS . But why doesn't break=false behave in this same manner > when there is only a single condition? > > Take this call trace and dialplan for example. I use a single > condition with break=false. However, unlike the stacked conditions, > the action is not executed when the single condition evaluates to > FAIL. Why doesn't break=never cause the action to be executed in a non > stacked condition? > > Call trace: > > Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ > break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) > [emergency_set_variables] break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) > [emergency_set_variables] break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) INLINE > EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) > > Dialplan: > > > > > > > > > > > > > > ------------------------------------------------- > > I also getting confusing behavior with anti-action and stacked > condition. I expect the anti-action to be executed if any of the > stacked conditions evaluates to FAIL. However, in the case below, the > extension simply breaks if any of the condition evaluates to FAIL. > > Call trace: > > Dialplan: sofia/external/Unknown at 72.37.252.18 parsing > [public->emergency_bridge] continue=true > Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) > [emergency_bridge] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ > break=on-false > Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > [emergency_bridge] ${emergency_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/Unknown at 72.37.252.18 parsing > [public->public_did] continue=false <-- [[MOVES ON TO ANOTHER > EXTENSION WITH EXECUTING THE anti-action]] > > Dialplan: > > > > > > > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> > > > > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > > data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > > > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> > > > > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > > data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > > > > > SwK suggested that I use a script language, which is probably what > I'll end up doing. However, I'm very interested in understanding why > the scenarios above. Are these bugs or are they just counter intuitive > rules that we must simply memorize? > > Thanks for your input. > Herman Griffin > > > > On Sat, Jan 14, 2012 at 11:42 AM, Jeff Lenk wrote: > > By putting the break=never you are defeating the "and" processing. Remove > > that. > > > > Move the second group into its own extension. > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Stacked-conditions-are-not-acting-like-logical-AND-tp7188082p7188271.html > > Sent from the freeswitch-users mailing list archive at Nabble.com. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/cb9ddf6e/attachment-0001.html From msc at freeswitch.org Tue Jan 17 03:35:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 16:35:45 -0800 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: Just for testing, try this at the fs_cli and see if you receive two different "Event-Name" headers: /log 0 /event plain custom Run your php script in another terminal window and see what shows up on the console. This is a straight ESL connection between fs_cli and the FS server so whatever you see displayed on the console is "real". Let us know what happens. -MC P.S. - Just curious, what's up with the ZMQ stuff? On Mon, Jan 16, 2012 at 7:39 AM, Gerald Weber wrote: > Hello,**** > > ** ** > > i?m trying to send custom events using PHP ESL Library.**** > > (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 21-35-19 > -0600))**** > > My PHP Code:**** > > ** ** > > > require_once('ESL.php');**** > > ** ** > > $sock = new ESLconnection('192.168.20.73', '8021', 'ClueCon');**** > > ** ** > > $e = new ESLevent("CUSTOM","CONFIG::AGENT_LIST");**** > > $e->addHeader("Agents","2022");**** > > ** ** > > $sock->sendEvent($e);**** > > ?>**** > > ** ** > > When I connect to fs using telnet and run the php script from another > xterm, I can see the following output:**** > > ** ** > > Content-Length: 657**** > > Content-Type: text/event-json**** > > ** ** > > {**** > > "Event-Subclass": "CONFIG::AGENT_LIST",**** > > "Event-Name": "SOCKET_DATA",**** > > "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea",**** > > "FreeSWITCH-Hostname": "freeswitch.local",**** > > "FreeSWITCH-Switchname": "freeswitch.local",**** > > "FreeSWITCH-IPv4": "192.168.20.73",**** > > "FreeSWITCH-IPv6": "::1",**** > > "Event-Date-Local": "2012-01-16 15:20:23",**** > > "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT",**** > > "Event-Date-Timestamp": "1326723623776184",**** > > "Event-Calling-File": "mod_event_socket.c",**** > > "Event-Calling-Function": "read_packet",**** > > "Event-Calling-Line-Number": "1188",**** > > "Command": "sendevent CUSTOM",**** > > "Event-Name": "CUSTOM",**** > > "Agents": "2022",**** > > "ZMQ-Msg-Cnt": "3508"**** > > }**** > > ** ** > > Looks ok, but i don?t get why ?Event-Name? is sent twice.**** > > Shouldn?t there be an Event-Name: CUSTOM only ?**** > > Is there a way to avoid this or am i missing something ?**** > > ** ** > > thx & regards,**** > > gw**** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/2bd50085/attachment.html From msc at freeswitch.org Tue Jan 17 03:37:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 16:37:59 -0800 Subject: [Freeswitch-users] Event date timestamp in different time zone In-Reply-To: References: Message-ID: Not at this time. You'll need to calculate the offset in your program. -MC On Mon, Jan 16, 2012 at 11:49 AM, Srini K wrote: > Hi, > Event Date timestamp is in GMT. For example in CHANNEL_CALLSTATE event, > Caller-Channel-Created-Time is in GMT (unixTimestamp). Is there any way to > get it to user defined time zone (Setting it in config file). > > CHANNEL_CALLSTATE > 4dead845-6636-4ace-ae9a-c257f9b179cb > 2012-01-16%2010%3A49%3A12 > > Mon,%2016%20Jan%202012%2018%3A49%3A12%20GMT > > 1326739752973123 > > 1326739752973123 > > Any help would be greatly appreciated > Regards > Srini > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/75e430e9/attachment.html From msc at freeswitch.org Tue Jan 17 03:40:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 16:40:31 -0800 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: <4F1478C1.2000909@cupis.co.uk> References: <4F1478C1.2000909@cupis.co.uk> Message-ID: On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis wrote: > On 16/01/12 14:29, dan at subformat.net wrote: > > If you could point me in the right direction, that would be great. > > Can you provide a log of the call from FreeSWITCH, on > http://pastebin.freeswitch.org/ for us to look at, please? > > Also, you might want to add an action right after the answer: Sometimes it takes a moment for media to "come up" and your playback could occur before media is happily flowing. If you sleep for a short period of time usually that helps. Try making the sleep duration longer or shorter to see what happens. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/de72cecd/attachment.html From msc at freeswitch.org Tue Jan 17 03:42:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jan 2012 16:42:08 -0800 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg In-Reply-To: References: Message-ID: What about explicitly setting it to false on the b leg dialstring? originate {sip_auto_answer}user/1000 &bridge({sip_auto_answer=false}user/1001) Just curious to see what would happen. -MC On Mon, Jan 16, 2012 at 11:15 AM, Seven Du wrote: > Hi, > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want > 1000 to be auto answered but not 1001, but it seems FS automatically copy > to b-leg. I manually comment the following lines and it seems to work. > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", > "intercom=true"); > } > > switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); > // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); > > would it be good to default to not copy but leave the "export" or > equivalent to do that? Or should I patch to add a var to disable that? > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/af6a7c28/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 17 03:44:35 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 18:44:35 -0600 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg In-Reply-To: References: Message-ID: originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001) On Mon, Jan 16, 2012 at 1:15 PM, Seven Du wrote: > Hi, > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want > 1000 to be auto answered but not 1001, but it seems FS automatically copy > to b-leg. I manually comment the following lines and it seems to work. > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", > "intercom=true"); > } > > switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); > // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); > > would it be good to default to not copy but leave the "export" or > equivalent to do that? Or should I patch to add a var to disable that? > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/a7302200/attachment.html From mayamatakeshi at gmail.com Tue Jan 17 04:07:28 2012 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Tue, 17 Jan 2012 10:07:28 +0900 Subject: [Freeswitch-users] 486 Busy turns to 480 Temporarily Unavailable In-Reply-To: References: <4F0582AA.9060708@gmx.net> Message-ID: On Fri, Jan 13, 2012 at 3:07 PM, mayamatakeshi wrote: > > > On Thu, Jan 5, 2012 at 7:59 PM, Peter P GMX wrote: > >> Hello, >> >> I have a strange phenomen: >> >> When a target UA is busy, it returns "486 Busy" to Freeswitch. But >> Freeswitch then returns "480 Temporarily Unavailable" to the called party. >> Where does this come from and how can I change this behaviour? >> >> See (anonymized) SIP trace with ngrep: >> >> UA to Freeswitch: >> ======================== >> U 2012/01/04 13:59:44.928775 :5060 -> :5080 >> SIP/2.0 486 Busy Here. >> Via: SIP/2.0/UDP >> :5080;rport=5080;branch=z9hG4bKNZZDv0Syp4eyr. >> From: "026xxxxxxxx" >;tag=py094Kv7vr03a. >> To: >> > ;uniq=B05FE4881A55AEEB69361EFA327DB>;tag=E1C3374B97DAB2DE. >> Call-ID: d0d0d057-b176-122f-1f8d-001ec9b9da3c. >> CSeq: 22504928 INVITE. >> User-Agent: AVM FRITZ!Box 6360 Cable 85.05.07 (Sep 14 2011). >> Content-Length: 0. >> >> Freeswitch to Caller: >> ======================== >> U 2012/01/04 13:59:44.930387 :5060 -> >> :5060 >> SIP/2.0 480 Temporarily Unavailable. >> Via: SIP/2.0/UDP :5060;branch=z9hG4bK-4896-2830DFA. >> From: >> > ;user=phone>;tag=13517-HB-08a98588-2622da197. >> To: ;user=phone>;tag=XQtc5US24QgDa. >> Call-ID: 13517-SG-08a98587-0a352e121 at sip.provider.de. >> CSeq: 134781549 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-68627e8 2011-11-21 >> 13-52-28 -0600. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY. >> Supported: precondition, path, replaces. >> Allow-Events: talk, hold, refer. >> Content-Length: 0. >> P-Asserted-Identity: "069xxxxxxxx" >. >> >> Best regards >> Peter >> > > I believe this is a bug. > I have opened a jira ticket: > http://jira.freeswitch.org/browse/FS-3810 > Solved. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/6c578f83/attachment.html From anthony.minessale at gmail.com Tue Jan 17 04:10:14 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jan 2012 19:10:14 -0600 Subject: [Freeswitch-users] Open Source unified autodialer software released In-Reply-To: References: Message-ID: One piece of advice is to not release it under AGPL which is the one that triggers the copyleft over a socket and will turn away 99% of your perspective testers. On Mon, Jan 16, 2012 at 1:46 PM, tahir almas wrote: > Pleased to announce the release of open source Fax , SMS and Voice > broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped over reknown Drupal Conent Mnagment System and powerfull Plivo > Communication framework , Your contribution and suggestions are welcome > > Regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > **************************************************************************************************************** > NOTICE OF CONFIDENTIALITY > This communication including any information transmitted with it is > intended only for the use of the addressees and is confidential and may > be protected by legal privilege . If you are not an intended recipient, be > aware that any disclosure, copying, distribution or use of this e-mail or > any attachment is prohibited. If you have received this e-mail in error, > please notify us immediately by returning it to the sender and delete this > copy from your system. Thank you for your cooperation. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120116/5822ed96/attachment.html From fs-list at communicatefreely.net Tue Jan 17 06:20:01 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 16 Jan 2012 22:20:01 -0500 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: References: <4EE12D98.8090100@communicatefreely.net> Message-ID: <4F14E8E1.6060700@communicatefreely.net> Okay, so I got another machine, built the update, (which wouldn't compile until I hacked mod_spandsp), etc. etc. I now have a fairly recent version of FreeSWITCH and got a lab together and can figure things out better. I did notice some improvements in BLFs, and they handle multiple calls better, but I'm still having the core issue that started this to begin with, and makes my customers complain. Here's how to duplicate: Phone A calls phone B, phone B answers The lamp for phone A lights, and the lamp for phone B flashes. When phone B answers, they are both steady. Phone C calls phone B. Phone B's lamp flashes (it should, it is ringing). Phone C hangs up before phone B answers the second call. Phone B's light goes out <---- This is the problem here If phone B does something, ie. puts a call on hold and picks it up again, all the lamps do what they are supposed to. In fact, they will no correct themselves if phone B does something. What I'm trying to solve is the fact that phone B's lamp went out even though they are still on an active call with phone A. This is fairly common in a group reception environment. Not everyone can get all the calls all the time, but the other people in the group want to know who has an active call. Any ideas? Should I submit this as a bug? Thanks! -Tim Anthony Minessale wrote: > You are running a version from August. Too bad for you that you are > missing a whole autumn worth of updates including a whole bunch of > work on presence. > > while(!current) update(); > > > > On Thu, Dec 8, 2011 at 3:35 PM, Tim St. Pierre > > > wrote: > > Hello, > > I'm having all sorts of problems with BLFs not being in the correct > state. It works fine in some cases, but others are wrong. Here's > what > I can see: > > Single endpoint, single call, everything works fine. Light flashes on > ring, goes steady on answer, goes out on hangup. > > Here's where it gets tricky: > > If there are two phones registered to the extension, it flashes when > they ring, but then goes out when one of them answers. > If either phone places an outgoing call, the lamp comes on. > > If a single phone gets a second call, their lamp flashes again, > but then > goes out when they answer the second call. > > I'm constantly getting complaints from users where the lamps are stuck > on. It happens more often when they have a lot of phones in ring > groups. The lamps work fine for ring groups - they all flash, and > whomever picks up the call stays steady while the rest go out. > > Is there anything I can do to get freeswitch to base the state on > whether or not that user has any active calls, rather than just > what the > last thing that the phone did was? > > This happens on every model of Aastra phone, and I have all of > them. I > haven't had a chance to try it yet on Polycom. > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 > 11-27-20 > -0500) > > Any ideas would be more than welcome. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fs-list at communicatefreely.net Tue Jan 17 06:23:25 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 16 Jan 2012 22:23:25 -0500 Subject: [Freeswitch-users] Nous avons besoin de votre assistance avec les traductions! In-Reply-To: References: Message-ID: <4F14E9AD.2050007@communicatefreely.net> Hi Michael, I may be able to arrange for some of those translations. Send along a spreadsheet and I'll pass it to our resident bi-lingual operator. Thanks! -Tim Michael Collins wrote: > Salut! > > If you can read the subject line then you already know what I need: > help with some French translations. As you know, our very own Marc > Olivier Chouinard (Moc) has produced a set of French-Canadian sound > prompts for everyone's benefit. The FreeSWITCH team and Barracuda > Networks recently added some prompts to that set. We are looking to > expand that sound set even more with some of our newest sound prompts. > > I have a grand total of about 220 prompts that I would like to get > translated into French. Many of these are less-common prompts that a > lot of people won't need, but I would like to get the translations > done anyway so that we can get the most useful ones recorded as soon > as possible. Also, some of these prompts are silly ones that may not > translate too well into French. (I've heard that the French don't have > a sense of humor but I cannot confirm that report at this time. :) > > If you'd like to help with this translation project please email me > off list and I will send you a spreadsheet with what we have to do. > > Merci beaucoup pour tout votre soutien! > > -MC > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fs-list at communicatefreely.net Tue Jan 17 06:25:39 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 16 Jan 2012 22:25:39 -0500 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F13AC5F.4030400@the800group.com> References: <4F13AC5F.4030400@the800group.com> Message-ID: <4F14EA33.8040505@communicatefreely.net> Hello, I have been using the Gigaset S675IP and the A580IP with good success. They do G.722, can handle SIP SRV records, and the BLFs work (more than I can say for the Aastra MBU400). They are also very inexpensive and sound great! They only caveats: SIP passwords have to be fairly short No easy provisioning mechanism. Other than that, they work great. Not all models can transfer though - I believe the S675 can. I haven't played with the C610 Good luck! -Tim ocset wrote: > Hi > > I have had great success using Yealink phones with Freeswitch and a > customer has asked me for a cordless phone for their office. I want to > ensure that they can keep as much functionality with a cordless phone as > they have with the Yealink T28. I am looking at the Siemens C610IP phone > but don't know how well it plays with Freeswitch. > > Can someone please shed some light on the Siemens phone or any > alternative that you have successfully implemented. > > Thanks in advance > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mustafa.pk at gmail.com Tue Jan 17 07:38:44 2012 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 17 Jan 2012 09:38:44 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: @Anthm: Thank you, will give it a try today. @DJb: yes i think i missed this one in pastebin. ------------------------------------------------------------------------ send 486 bytes to udp/[xx.xx.xx.xx]:5060 at 13:16:43.100639: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKcd83be7d98d57e63bec66f99e From: ;tag=02daafe8-CC-25 To: ;tag=aS1U0erU4N8ca Call-ID: SBC44aef04ba24b8ba7cddb30b9d4ea236e at SoftX3000 CSeq: 2 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 On Mon, Jan 16, 2012 at 11:32 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Try this from the build root. > > git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 > libs/sofia-sip/libsofia-sip-ua | patch -p1 -R > touch libs/sofia-sip/.update > > make mod_sofia-clean > make mod_sofia-install > > This commit took away the Require: timer option which is an optional field > and clearly the pedantic switch is worrying about it. > > > On Mon, Jan 16, 2012 at 7:59 AM, Ghulam Mustafa wrote: > >> Hi, >> >> i have upgraded FS from previously running vesion 1.0.head (git-472ab0c >> 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we >> are connected with our carrier's Huawei SoftX3000, outgoing calls are >> working ok but as soon as an incoming call arrives SoftX3000 drops it with >> the message "SSF00159L00681 Session Timer Check Message Failed" >> >> Here i have pasted both working (before upgrade) and current version sip >> traces. http://pastebin.freeswitch.org/18136 >> >> i am sure it's not a bug and i might be missing some configuration >> parameter, any clue? >> >> thanks >> >> -- >> Ghulam Mustafa >> cell: +92 333.611.7681 >> sip: cyrenity at ekiga.net >> mail: mustafa.pk at gmail.com >> web: cyrenity.wordpress.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/a9d42584/attachment.html From piyush.sharma at coraltele.com Tue Jan 17 08:19:26 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 17 Jan 2012 05:19:26 -0000 Subject: [Freeswitch-users] FreeSWITCH MySQL Support In-Reply-To: References: Message-ID: <1334639129.3499.5.camel@localhost.localdomain> Would somebody tell me, How can I change core.db database to MySQL Server instead of Sqlite. Actually I want FreeSwitch to work with MySQL. I Changed in switch.conf.xml, internal.xml, external.xml files but only few tables were created in MySQL ex. nat,aliases. But I think it still use core.db to store live calls. Thank You. With Regards, Piyush Sharma, Coral Telecom Limited. From greg at brilliantecho.com Tue Jan 17 08:38:42 2012 From: greg at brilliantecho.com (Greg Millam) Date: Mon, 16 Jan 2012 21:38:42 -0800 Subject: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? Message-ID: Hi folks - I have a freeswitch dialplan + script that first calls play_fsv to play a greeting, then record_fsv to record incoming video to fsv. It works fine, but there's one issue: Apparently, freeswitch is buffering incoming video during play_fsv. When record_fsv is called, that buffer is dumped into the .fsv file, resulting in several seconds of unneeded video. Is there a way to empty that incoming video buffer before record_fsv begins recording? Thank you! - Greg Millam From tahir at ictinnovations.com Tue Jan 17 09:35:52 2012 From: tahir at ictinnovations.com (tahir almas) Date: Tue, 17 Jan 2012 11:35:52 +0500 Subject: [Freeswitch-users] Open Source unified autodialer software released In-Reply-To: References: Message-ID: Realy thankful for your suggestion, ICTDialer is developed over Drupal 7.0 Licensed as GPL version 2 or later so we have to license ICTDialer as GPL compatible license What open source License you recommend for ICTDialer ? Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > One piece of advice is to not release it under AGPL which is the one that > triggers the copyleft over a socket and will turn away 99% of your > perspective testers. > > > On Mon, Jan 16, 2012 at 1:46 PM, tahir almas wrote: > >> Pleased to announce the release of open source Fax , SMS and Voice >> broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped over reknown Drupal Conent Mnagment System and powerfull Plivo >> Communication framework , Your contribution and suggestions are welcome >> >> Regards >> *Tahir Almas* >> >> Managing Partner >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> >> **************************************************************************************************************** >> NOTICE OF CONFIDENTIALITY >> This communication including any information transmitted with it is >> intended only for the use of the addressees and is confidential and may >> be protected by legal privilege . If you are not an intended recipient, be >> aware that any disclosure, copying, distribution or use of this e-mail or >> any attachment is prohibited. If you have received this e-mail in error, >> please notify us immediately by returning it to the sender and delete this >> copy from your system. Thank you for your cooperation. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/2968d28d/attachment-0001.html From dujinfang at gmail.com Tue Jan 17 09:45:28 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 17 Jan 2012 14:45:28 +0800 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg In-Reply-To: References: Message-ID: seems yealink phone has problem, see http://pastebin.freeswitch.org/18138 there's no intercom in b-leg, but maybe because Call-Info present ? Call-Info: ;answer-after=0 The following confirmed work originate {sip_auto_answer=true}user/1000 &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) INVITE sip:1002 at 192.168.7.105:5062 SIP/2.0 Call-Info: On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: > > originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001) > > > On Mon, Jan 16, 2012 at 1:15 PM, Seven Du wrote: > > Hi, > > > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. I manually comment the following lines and it seems to work. > > > > > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); > > } > > > > switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); > > // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); > > > > > > would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? > > > > Thanks. > > > > -- > > About: http://about.me/dujinfang > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/8c63b79d/attachment.html From krice at freeswitch.org Tue Jan 17 10:09:12 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 17 Jan 2012 01:09:12 -0600 Subject: [Freeswitch-users] FreeSWITCH MySQL Support In-Reply-To: <1334639129.3499.5.camel@localhost.localdomain> Message-ID: This is clearly outlined ont eh wiki via ODBC On 4/16/12 11:05 PM, "Piyush Sharma" wrote: > Would somebody tell me, > How can I change core.db database to MySQL Server instead of Sqlite. > > Actually I want FreeSwitch to work with MySQL. I Changed in > switch.conf.xml, internal.xml, external.xml files but only few tables > were created in MySQL ex. nat,aliases. > But I think it still use core.db to store live calls. > > Thank You. > With Regards, > Piyush Sharma, > Coral Telecom Limited. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaasmailing at gmail.com Tue Jan 17 11:05:40 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Tue, 17 Jan 2012 09:05:40 +0100 Subject: [Freeswitch-users] Question about the future of mod_nibblebill Message-ID: <4F152BD4.9080705@gmail.com> Hi all, I'm thinking about a development of a custom prepaid application based on mod_nibblebill (that is very interesting). I'm wondering about the status and the future of this module in order to understand if this is the better way to achieve the aim of the project. I have seen in the mod_nibblebill source code this notes: * TODO: Fix what happens when the DB is not available * TODO: Fix what happens when the DB queries fail (right now, all are acting like success) * TODO: Add buffering abilities * TODO: Make error handling for database, such that when the database is down (or not installed) we just log to a text file * FUTURE: Possibly make the hooks not tied per-channel, and instead just do this as a supervision style application with one thread that watches all calls Does the developers think to continue supporting the module (not only bugfix but new features)? I'm sorry for the particular question but I wouldn't do errors at this stage of the project. Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/b68d9b78/attachment.html From gerald.weber at besharp.at Tue Jan 17 11:17:16 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Tue, 17 Jan 2012 08:17:16 +0000 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: Hi, thanks for your answers @Anthony: I did a "make current" mins ago, fs starts with version FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600) @MC I connect using ./fs_cli -H 192.168.20.73 -P 8021 /log 0 /event plain custom (additionally I connect to fs using telnet 192.168.20.73 8021 on another terminal and issued "event plain all") Starting my php script, fs_cli shows nothing, telnet shows the event: Content-Length: 583 Content-Type: text/event-plain Event-Subclass: CONFIG%3A%3AAGENT_LIST Event-Name: SOCKET_DATA ... Event-Date-Timestamp: 1326785361276007 ... Event-Name: CUSTOM Agents: 2022 ZMQ-Msg-Cnt: 244 /event plain CUSTOM in fs_cli and restart php script -> no output. (first event subscription except all doesn't work ?) /event plain all in fs_cli and restart php script shows: RECV EVENT Event-Subclass: CONFIG::AGENT_LIST Core-UUID: ca0f2950-40d9-11e1-8145-99e6c2b17445 ... Event-Date-Timestamp: 1326785361276007 ... Command: sendevent CUSTOM Event-Name: CUSTOM Agents: 2022 ZMQ-Msg-Cnt: 244 "/event plain custom" (or /event plain CUSTOM) in fs_cli and restart php script shows the event and other events too (like RE_SCHEDULE and HEARTBEAT) This leads me to the following question: - Why does /event plain custom (/event plain CUSTOM) in fs_cli show other events too ? Then I played a bit in telnet: telnet 192.168.20.73 8021 .. auth stuff... event plain custom (or events plain CUSTOM) and restart php -> no output event plain all and restart php -> event shown filter Event-Name CUSTOM (or custom) and restart php -> no output filter delete Event-Name CUSTOM filter Event-Subclass CONFIG::AGENT_LIST and restart php -> event shown Played more with telnet: Fresh connect event plain CUSTOM -> callcenter_config agent set status .... -> no output event plain ALL -> callcenter_config agent set status .... -> event shown event filter Event-Name CUSTOM -> callcenter_config agent set status .... -> event shown (and only custom events) Seems that sending Events using the esl library adds SOCKET_DATA and event all CUSTOM / filter Event-Name custom can't handle that ? Regarding the ZMQ stuff: I use mod_event_zmq and was missing a couple of events, so I added a sequence to every event zmq publishes. Later I discovered a bug in my code, I was calling the recv function twice and didn't recognize it for a whole day..... Regards, gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Dienstag, 17. J?nner 2012 01:36 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice Just for testing, try this at the fs_cli and see if you receive two different "Event-Name" headers: /log 0 /event plain custom Run your php script in another terminal window and see what shows up on the console. This is a straight ESL connection between fs_cli and the FS server so whatever you see displayed on the console is "real". Let us know what happens. -MC P.S. - Just curious, what's up with the ZMQ stuff? On Mon, Jan 16, 2012 at 7:39 AM, Gerald Weber > wrote: Hello, i'm trying to send custom events using PHP ESL Library. (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 21-35-19 -0600)) My PHP Code: addHeader("Agents","2022"); $sock->sendEvent($e); ?> When I connect to fs using telnet and run the php script from another xterm, I can see the following output: Content-Length: 657 Content-Type: text/event-json { "Event-Subclass": "CONFIG::AGENT_LIST", "Event-Name": "SOCKET_DATA", "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea", "FreeSWITCH-Hostname": "freeswitch.local", "FreeSWITCH-Switchname": "freeswitch.local", "FreeSWITCH-IPv4": "192.168.20.73", "FreeSWITCH-IPv6": "::1", "Event-Date-Local": "2012-01-16 15:20:23", "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT", "Event-Date-Timestamp": "1326723623776184", "Event-Calling-File": "mod_event_socket.c", "Event-Calling-Function": "read_packet", "Event-Calling-Line-Number": "1188", "Command": "sendevent CUSTOM", "Event-Name": "CUSTOM", "Agents": "2022", "ZMQ-Msg-Cnt": "3508" } Looks ok, but i don't get why "Event-Name" is sent twice. Shouldn't there be an Event-Name: CUSTOM only ? Is there a way to avoid this or am i missing something ? thx & regards, gw _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/6d1494e8/attachment-0001.html From benkokakao at gmail.com Tue Jan 17 13:32:02 2012 From: benkokakao at gmail.com (Christian Benke) Date: Tue, 17 Jan 2012 11:32:02 +0100 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F14EA33.8040505@communicatefreely.net> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> Message-ID: We are using Snom M9 - while i can imagine much more functionality(Better phonebook, better haptics), the big plus is that it's possible to provision them with XML via tftp - and they are pretty cheap! Snom M3 sucked though - unstable and unsolid update-mechanism(Bricked after i pressed the reset button while it tried to contact the snom-update-server, which it could not reach as there was no internet-connection). Regards, Christian From mustafa.pk at gmail.com Tue Jan 17 13:34:01 2012 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Tue, 17 Jan 2012 15:34:01 +0500 Subject: [Freeswitch-users] session timer problem after upgrade. In-Reply-To: References: Message-ID: Hi Anthony, Your workaround solved my problem. thank you so much. -m On Tue, Jan 17, 2012 at 9:38 AM, Ghulam Mustafa wrote: > @Anthm: Thank you, will give it a try today. > > @DJb: yes i think i missed this one in pastebin. > > ------------------------------------------------------------------------ > send 486 bytes to udp/[xx.xx.xx.xx]:5060 at 13:16:43.100639: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xx.xx.xx.xx:5060;branch=z9hG4bKcd83be7d98d57e63bec66f99e > From: ;tag=02daafe8-CC-25 > To: ;tag=aS1U0erU4N8ca > Call-ID: SBC44aef04ba24b8ba7cddb30b9d4ea236e at SoftX3000 > CSeq: 2 BYE > User-Agent: FreeSWITCH > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > > > > > On Mon, Jan 16, 2012 at 11:32 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Try this from the build root. >> >> git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 >> libs/sofia-sip/libsofia-sip-ua | patch -p1 -R >> touch libs/sofia-sip/.update >> >> make mod_sofia-clean >> make mod_sofia-install >> >> This commit took away the Require: timer option which is an optional >> field and clearly the pedantic switch is worrying about it. >> >> >> On Mon, Jan 16, 2012 at 7:59 AM, Ghulam Mustafa wrote: >> >>> Hi, >>> >>> i have upgraded FS from previously running vesion 1.0.head (git-472ab0c >>> 2012-01-08 14-20-58 -0600) to (git-2d190b3 2011-02-03 23-46-19 -0600), we >>> are connected with our carrier's Huawei SoftX3000, outgoing calls are >>> working ok but as soon as an incoming call arrives SoftX3000 drops it with >>> the message "SSF00159L00681 Session Timer Check Message Failed" >>> >>> Here i have pasted both working (before upgrade) and current version sip >>> traces. http://pastebin.freeswitch.org/18136 >>> >>> i am sure it's not a bug and i might be missing some configuration >>> parameter, any clue? >>> >>> thanks >>> >>> -- >>> Ghulam Mustafa >>> cell: +92 333.611.7681 >>> sip: cyrenity at ekiga.net >>> mail: mustafa.pk at gmail.com >>> web: cyrenity.wordpress.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Ghulam Mustafa > cell: +92 333.611.7681 > sip: cyrenity at ekiga.net > mail: mustafa.pk at gmail.com > web: cyrenity.wordpress.com > -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/1a0aa9f9/attachment.html From dan at subformat.net Tue Jan 17 13:34:40 2012 From: dan at subformat.net (dan at subformat.net) Date: Tue, 17 Jan 2012 10:34:40 -0000 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: <4F1478C1.2000909@cupis.co.uk> Message-ID: <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> Thanks MC, Tried that but sleep doesn't appear to be having any effect? I have pasted my log here: http://pastebin.freeswitch.org/18143 As you can see in the log the ext is not registered, but the dial plan should continue on fail? to the media file etc. Thanks Dan. > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis wrote: > >> On 16/01/12 14:29, dan at subformat.net wrote: >> > If you could point me in the right direction, that would be great. >> >> Can you provide a log of the call from FreeSWITCH, on >> http://pastebin.freeswitch.org/ for us to look at, please? >> >> > Also, you might want to add an action right after the answer: > > > Sometimes it takes a moment for media to "come up" and your playback could > occur before media is happily flowing. If you sleep for a short period of > time usually that helps. Try making the sleep duration longer or shorter > to > see what happens. > > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Tue Jan 17 14:47:52 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 17 Jan 2012 16:47:52 +0500 Subject: [Freeswitch-users] Annoying error while compiling FreeSWITCH In-Reply-To: References: Message-ID: Thanks both of you. I will practice it in the future. BTW i did get a way around the problem. That is after running bootstrap.sh, i edited configure file and changed all -Werror to -Wall, and then run configure script and everything worked fine. Not a recommended practice but a tip for desperate ones at desperate times, like me last night. Thank you. On Tue, Jan 17, 2012 at 4:24 AM, Michael Collins wrote: > > > On Mon, Jan 16, 2012 at 2:01 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> I am getting build failure repeatedly due to this annoying -Werror. Each >> time any warning appears in code, the compilation fails treating it as >> error, e.g. >> >> /usr/src/svn-src/freeswitch/src/mod/applications/mod_voicemail_ivr/mod_voicemail_ivr.c:57:6: >> error: variable 'argc' set but not used [-Werror=unused-but-set-variable] >> cc1: all warnings being treated as errors >> > > In addition to what Tony said, this is a good reason to report bugs to > jira.freeswitch.org. Also, just for reference, this particular module is > not critical for FreeSWITCH to work - it was contributed by a community > member (IRC: moc). Moc has A LOT of practice fixing his errors, so assign > the bug to him. ;) > > In the meantime you can exclude this module from being built just by > editing modules.conf in your freeswitch source directory. In fact, it might > be worth it for you to review which modules you are building by default. > There may be modules you don't want or need that you can skip. This will > make your builds go faster and you won't get tripped up by annoying errors > on modules that you don't actually use. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/55f98958/attachment-0001.html From dgarcia at anew.com.ve Tue Jan 17 15:21:09 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 17 Jan 2012 07:51:09 -0430 Subject: [Freeswitch-users] Open Source unified autodialer software released In-Reply-To: References: Message-ID: <4F1567B5.50007@anew.com.ve> Hi, Whi not LGPL? take a look to gnu page: http://www.gnu.org/licenses/gpl-faq.html specially: http://www.gnu.org/licenses/gpl-faq.html#compat-matrix-footnote-7 On 1/17/2012 2:05 AM, tahir almas wrote: > Realy thankful for your suggestion, ICTDialer is developed over Drupal > 7.0 Licensed as GPL version 2 or later so we have to license > ICTDialer as GPL compatible license > > What open source License you recommend for ICTDialer ? > > Regards > > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > **************************************************************************************************************** > NOTICE OF CONFIDENTIALITY > This communication including any information transmitted with it is > intended only for the use of the addressees and is confidential and > may be protected by legal privilege . If you are not an intended > recipient, be aware that any disclosure, copying, distribution or use > of this e-mail or any attachment is prohibited. If you have received > this e-mail in error, please notify us immediately by returning it to > the sender and delete this copy from your system. Thank you for your > cooperation. > > > > > On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale > > wrote: > > One piece of advice is to not release it under AGPL which is the > one that triggers the copyleft over a socket and will turn away > 99% of your perspective testers. > > > On Mon, Jan 16, 2012 at 1:46 PM, tahir almas > > wrote: > > Pleased to announce the release of open source Fax , SMS and > Voice broadcasting software solution ICTDialer > http://www.ictdialer.org developed over reknown Drupal Conent > Mnagment System and powerfull Plivo Communication framework , > Your contribution and suggestions are welcome > > Regards > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > **************************************************************************************************************** > NOTICE OF CONFIDENTIALITY > This communication including any information transmitted with > it is intended only for the use of the addressees and is > confidential and may be protected by legal privilege . If you > are not an intended recipient, be aware that any disclosure, > copying, distribution or use of this e-mail or any attachment > is prohibited. If you have received this e-mail in error, > please notify us immediately by returning it to the sender and > delete this copy from your system. Thank you for your cooperation. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4746 - Release Date: 01/16/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/ef650592/attachment.html From piyush.sharma at coraltele.com Tue Jan 17 11:51:58 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 17 Jan 2012 08:51:58 -0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 67, Issue 153 In-Reply-To: References: Message-ID: <1334651875.3499.10.camel@localhost.localdomain> Thanks for your reply, I checked it, it didn't work. Ok I am new to FreeSWITCH, I do it again. Thank You !!! On Tue, 2012-01-17 at 11:17 +0300, freeswitch-users-request at lists.freeswitch.org wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > Today's Topics: > > 1. Re: sip_auto_answer auto copy to bleg (Seven Du) > 2. Re: FreeSWITCH MySQL Support (Ken Rice) > 3. Question about the future of mod_nibblebill (Carlo Dimaggio) > 4. Re: PHP ESL Custom Events Event-Name sent twice (Gerald Weber) > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Seven Du > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] sip_auto_answer auto copy to bleg > > Date: Tue, 17 Jan 2012 14:45:28 +0800 > > > > seems yealink phone has problem, see > > > > > > http://pastebin.freeswitch.org/18138 > > > > > > there's no intercom in b-leg, but maybe because Call-Info present ? > > > > > > Call-Info: ;answer-after=0 > > > > > > > > > > > > > > > > > > > > > > The following confirmed work > > > > > > originate {sip_auto_answer=true}user/1000 > > &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) > > > > > > INVITE sip:1002 at 192.168.7.105:5062 SIP/2.0 > > Call-Info: > > > > > > > > > > > > On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: > > > > > > > > originate [sip_auto_answer=true]user/1000 > > > &bridge([sip_auto_answer=false]user/1001) > > > > > > > > > > > > On Mon, Jan 16, 2012 at 1:15 PM, Seven Du > > > wrote: > > > > Hi, > > > > > > > > > > > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) > > > > and want 1000 to be auto answered but not 1001, but it seems FS > > > > automatically copy to b-leg. I manually comment the following > > > > lines and it seems to work. > > > > > > > > > > > > > > > > > > > > > > > > > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", > > > > "intercom=true"); > > > > } > > > > > > > > > > > > switch_ivr_transfer_variable(session, nsession, > > > > SOFIA_REPLACES_HEADER); > > > > // switch_ivr_transfer_variable(session, nsession, > > > > "sip_auto_answer"); > > > > > > > > > > > > would it be good to default to not copy but leave the "export" > > > > or equivalent to do that? Or should I patch to add a var to > > > > disable that? > > > > > > > > > > > > Thanks. > > > > > > > > > > > > -- > > > > About: http://about.me/dujinfang > > > > Blog: http://www.dujinfang.com > > > > Proj: http://www.freeswitch.org.cn > > > > > > > > > > > > Sent with Sparrow > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > IRC: irc.freenode.net #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org > > > googletalk:conf+888 at conference.freeswitch.org > > > pstn:+19193869900 > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Ken Rice > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FreeSWITCH MySQL Support > > Date: Tue, 17 Jan 2012 01:09:12 -0600 > > > > This is clearly outlined ont eh wiki via ODBC > > > > > > On 4/16/12 11:05 PM, "Piyush Sharma" wrote: > > > > > Would somebody tell me, > > > How can I change core.db database to MySQL Server instead of Sqlite. > > > > > > Actually I want FreeSwitch to work with MySQL. I Changed in > > > switch.conf.xml, internal.xml, external.xml files but only few tables > > > were created in MySQL ex. nat,aliases. > > > But I think it still use core.db to store live calls. > > > > > > Thank You. > > > With Regards, > > > Piyush Sharma, > > > Coral Telecom Limited. > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Carlo Dimaggio > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: [Freeswitch-users] Question about the future of > > mod_nibblebill > > Date: Tue, 17 Jan 2012 09:05:40 +0100 > > > > Hi all, > > > > I'm thinking about a development of a custom prepaid application > > based on mod_nibblebill (that is very interesting). > > I'm wondering about the status and the future of this module in > > order to understand if this is the better way to achieve the aim of > > the project. > > I have seen in the mod_nibblebill source code this notes: > > > > * TODO: Fix what happens when the DB is not available > > * TODO: Fix what happens when the DB queries fail (right now, all > > are acting like success) > > * TODO: Add buffering abilities > > * TODO: Make error handling for database, such that when the > > database is down (or not installed) we just log to a text file > > * FUTURE: Possibly make the hooks not tied per-channel, and instead > > just do this as a supervision style application with one thread that > > watches all calls > > > > Does the developers think to continue supporting the module (not > > only bugfix but new features)? > > > > I'm sorry for the particular question but I wouldn't do errors at > > this stage of the project. > > > > > > Best Regards, > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Gerald Weber > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name > > sent twice > > Date: Tue, 17 Jan 2012 08:17:16 +0000 > > > > Hi, > > > > thanks for your answers > > > > > > > > @Anthony: > > > > I did a ?make current? mins ago, fs starts with version > > > > FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600) > > > > > > > > @MC > > > > I connect using ./fs_cli ?H 192.168.20.73 ?P 8021 > > > > /log 0 > > > > /event plain custom > > > > > > > > (additionally I connect to fs using telnet 192.168.20.73 8021 on > > another terminal and issued ?event plain all?) > > > > > > > > Starting my php script, fs_cli shows nothing, telnet shows the > > event: > > > > > > > > Content-Length: 583 > > > > Content-Type: text/event-plain > > > > > > > > Event-Subclass: CONFIG%3A%3AAGENT_LIST > > > > Event-Name: SOCKET_DATA > > > > ? > > > > Event-Date-Timestamp: 1326785361276007 > > > > ? > > > > Event-Name: CUSTOM > > > > Agents: 2022 > > > > ZMQ-Msg-Cnt: 244 > > > > > > > > /event plain CUSTOM in fs_cli and restart php script -> no output. > > > > (first event subscription except all doesn?t work ?) > > > > /event plain all in fs_cli and restart php script shows: > > > > > > > > RECV EVENT > > > > Event-Subclass: CONFIG::AGENT_LIST > > > > Core-UUID: ca0f2950-40d9-11e1-8145-99e6c2b17445 > > > > ? > > > > Event-Date-Timestamp: 1326785361276007 > > > > ? > > > > Command: sendevent CUSTOM > > > > Event-Name: CUSTOM > > > > Agents: 2022 > > > > ZMQ-Msg-Cnt: 244 > > > > > > > > > > > > ?/event plain custom? (or /event plain CUSTOM) in fs_cli and restart > > php script shows the event and other events too (like RE_SCHEDULE > > and HEARTBEAT) > > > > > > > > This leads me to the following question: > > > > - Why does /event plain custom (/event plain CUSTOM) in > > fs_cli show other events too ? > > > > > > > > Then I played a bit in telnet: > > > > telnet 192.168.20.73 8021 .. auth stuff? > > > > event plain custom (or events plain CUSTOM) and restart php -> no > > output > > > > event plain all and restart php -> event shown > > > > filter Event-Name CUSTOM (or custom) and restart php -> no output > > > > filter delete Event-Name CUSTOM > > > > filter Event-Subclass CONFIG::AGENT_LIST and restart php -> event > > shown > > > > > > > > Played more with telnet: > > > > Fresh connect > > > > event plain CUSTOM -> callcenter_config agent set status ?. -> no > > output > > > > event plain ALL -> callcenter_config agent set status ?. -> event > > shown > > > > event filter Event-Name CUSTOM -> callcenter_config agent set status > > ?. -> event shown (and only custom events) > > > > > > > > > > > > Seems that sending Events using the esl library adds SOCKET_DATA and > > event all CUSTOM / filter Event-Name custom can?t > > > > handle that ? > > > > > > > > Regarding the ZMQ stuff: > > > > I use mod_event_zmq and was missing a couple of events, so I added a > > sequence to every event zmq publishes. > > > > Later I discovered a bug in my code, I was calling the recv function > > twice and didn?t recognize it for a whole day?.. > > > > > > > > Regards, > > > > gw > > > > > > > > Von: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag > > von Michael Collins > > Gesendet: Dienstag, 17. J?nner 2012 01:36 > > An: FreeSWITCH Users Help > > Betreff: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name > > sent twice > > > > > > > > Just for testing, try this at the fs_cli and see if you receive two > > different "Event-Name" headers: > > > > /log 0 > > /event plain custom > > > > Run your php script in another terminal window and see what shows up > > on the console. This is a straight ESL connection between fs_cli and > > the FS server so whatever you see displayed on the console is > > "real". Let us know what happens. > > > > -MC > > > > P.S. - Just curious, what's up with the ZMQ stuff? > > > > On Mon, Jan 16, 2012 at 7:39 AM, Gerald Weber > > wrote: > > > > Hello, > > > > > > > > i?m trying to send custom events using PHP ESL Library. > > > > (running FreeSWITCH Version 1.0.head (git-c37c029 2012-01-11 > > 21-35-19 -0600)) > > > > My PHP Code: > > > > > > > > > > > require_once('ESL.php'); > > > > > > > > $sock = new ESLconnection('192.168.20.73', '8021', > > 'ClueCon'); > > > > > > > > $e = new ESLevent("CUSTOM","CONFIG::AGENT_LIST"); > > > > $e->addHeader("Agents","2022"); > > > > > > > > $sock->sendEvent($e); > > > > ?> > > > > > > > > When I connect to fs using telnet and run the php script from > > another xterm, I can see the following output: > > > > > > > > Content-Length: 657 > > > > Content-Type: text/event-json > > > > > > > > { > > > > "Event-Subclass": "CONFIG::AGENT_LIST", > > > > "Event-Name": "SOCKET_DATA", > > > > "Core-UUID": "b3ac8858-4019-11e1-910a-9b04baf8e5ea", > > > > "FreeSWITCH-Hostname": "freeswitch.local", > > > > "FreeSWITCH-Switchname": "freeswitch.local", > > > > "FreeSWITCH-IPv4": "192.168.20.73", > > > > "FreeSWITCH-IPv6": "::1", > > > > "Event-Date-Local": "2012-01-16 15:20:23", > > > > "Event-Date-GMT": "Mon, 16 Jan 2012 14:20:23 GMT", > > > > "Event-Date-Timestamp": "1326723623776184", > > > > "Event-Calling-File": "mod_event_socket.c", > > > > "Event-Calling-Function": "read_packet", > > > > "Event-Calling-Line-Number": "1188", > > > > "Command": "sendevent CUSTOM", > > > > "Event-Name": "CUSTOM", > > > > "Agents": "2022", > > > > "ZMQ-Msg-Cnt": "3508" > > > > } > > > > > > > > Looks ok, but i don?t get why ?Event-Name? is sent twice. > > > > Shouldn?t there be an Event-Name: CUSTOM only ? > > > > Is there a way to avoid this or am i missing something ? > > > > > > > > thx & regards, > > > > gw > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Piyush Sharma From drazen.blanusa at nth.ch Tue Jan 17 16:54:37 2012 From: drazen.blanusa at nth.ch (Drazen Blanusa) Date: Tue, 17 Jan 2012 14:54:37 +0100 Subject: [Freeswitch-users] eavesdrop problem Message-ID: <00cf01ccd51f$8d4cb9e0$a7e62da0$@blanusa@nth.ch> Hi, I have one FS instance running localy, and two clients connected. I make one call from A(1019), and one call from B(1010). I want to spy channel A from channel B. I'm using application eavesdrop(uuid) and this is correct. But after application is executed, I hear Input from A mic in A speaker. If I use eavesdrop(all), I hear A and B mic in A speaker. I checked and uuid is correct. >From log: switch_ivr.c:589 sofia/internal/1019 at XX.XXX.X.XX Command Execute eavesdrop(45b73c25-a1e7-4435-8214-76fecca7c6cd) I'm not using dialplan, just Also,I'm using ESL . Any suggestions?Why it is inverted? Tnx in advance! Drazen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/ae028c3c/attachment.html From msc at freeswitch.org Tue Jan 17 19:37:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 08:37:45 -0800 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: On Tue, Jan 17, 2012 at 12:17 AM, Gerald Weber wrote: > Hi,**** > > thanks for your answers**** > > ** ** > > @Anthony:**** > > I did a ?make current? mins ago, fs starts with version**** > > FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600)**** > > ** ** > > @MC**** > > I connect using ./fs_cli ?H 192.168.20.73 ?P 8021**** > > /log 0**** > > /event plain custom**** > > ** ** > > (additionally I connect to fs using telnet 192.168.20.73 8021 on another > terminal and issued ?event plain all?)**** > > ** ** > > Starting my php script, fs_cli shows nothing, telnet shows the event:**** > > ** ** > > Content-Length: 583**** > > Content-Type: text/event-plain**** > > ** ** > > Event-Subclass: CONFIG%3A%3AAGENT_LIST**** > > Event-Name: SOCKET_DATA**** > > ?**** > > Event-Date-Timestamp: 1326785361276007**** > > ?**** > > Event-Name: CUSTOM**** > > Agents: 2022**** > > ZMQ-Msg-Cnt: 244**** > > ** ** > > /event plain CUSTOM in fs_cli and restart php script -> no output.**** > > (first event subscription except all doesn?t work ?)**** > > /event plain all in fs_cli and restart php script shows:**** > > Hmm, I guess it does not work. You can do a filter: /event plain all /filter Event-Name CUSTOM In any case I'll have to defer to Tony as to why you're seeing two different header's named "Event-Name" -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/d3269323/attachment.html From msc at freeswitch.org Tue Jan 17 19:41:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 08:41:52 -0800 Subject: [Freeswitch-users] Question about the future of mod_nibblebill In-Reply-To: <4F152BD4.9080705@gmail.com> References: <4F152BD4.9080705@gmail.com> Message-ID: The author of this module (Darren Schreiber) uses it quite a bit, as do a number of community members. I believe it is a "safe" module to use for your project. It is still actively maintained. You can open tickets at jira.freeswitch.org for mod_nibblebill and assign them to Darren and he'll see them and respond. He's a little slow but only because his plate is full. ;) -MC On Tue, Jan 17, 2012 at 12:05 AM, Carlo Dimaggio wrote: > Hi all, > > I'm thinking about a development of a custom prepaid application based on > mod_nibblebill (that is very interesting). > I'm wondering about the status and the future of this module in order to > understand if this is the better way to achieve the aim of the project. > I have seen in the mod_nibblebill source code this notes: > > * TODO: Fix what happens when the DB is not available > * TODO: Fix what happens when the DB queries fail (right now, all are > acting like success) > * TODO: Add buffering abilities > * TODO: Make error handling for database, such that when the database is > down (or not installed) we just log to a text file > * FUTURE: Possibly make the hooks not tied per-channel, and instead just > do this as a supervision style application with one thread that watches all > calls > > Does the developers think to continue supporting the module (not only > bugfix but new features)? > > I'm sorry for the particular question but I wouldn't do errors at this > stage of the project. > > > Best Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/b6b2a30f/attachment.html From msc at freeswitch.org Tue Jan 17 19:44:15 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 08:44:15 -0800 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> Message-ID: Oops, you forgot to enable debug. It looks like you're capturing right from the console instead of using fs_cli. If you're at the console then type this before doing a test call: console loglevel debug Note: if you use fs_cli to connect to FS you'll automatically be at loglevel debug. Anyway, turn on debug level output and pastebin a new log. -MC On Tue, Jan 17, 2012 at 2:34 AM, wrote: > Thanks MC, Tried that but sleep doesn't appear to be having any effect? > > I have pasted my log here: http://pastebin.freeswitch.org/18143 > > As you can see in the log the ext is not registered, but the dial plan > should continue on fail? to the media file etc. > > Thanks > > Dan. > > > > > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis wrote: > > > >> On 16/01/12 14:29, dan at subformat.net wrote: > >> > If you could point me in the right direction, that would be great. > >> > >> Can you provide a log of the call from FreeSWITCH, on > >> http://pastebin.freeswitch.org/ for us to look at, please? > >> > >> > > Also, you might want to add an action right after the answer: > > > > > > Sometimes it takes a moment for media to "come up" and your playback > could > > occur before media is happily flowing. If you sleep for a short period of > > time usually that helps. Try making the sleep duration longer or shorter > > to > > see what happens. > > > > -MC > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/bc1d8f09/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 17 20:10:40 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Jan 2012 11:10:40 -0600 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: you cannot do *all* custom events with /event plain custom that command is expecting also a subclass param /event plain custom CONFIG::AGENT_LIST you have made a mistake if you think this unlocks heartbeat etc, probably you did events plain all and forgot. When I do it from perl I do not see what you reported, perl test.pl [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1381 esl_send() SEND auth ClueCon [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:502 esl_sendevent() SEND EVENT Event-Name: CUSTOM Event-Subclass: CONFIG::AGENT_LIST [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK +OK log level [7] freeswitch at internal> /event plain custom CONFIG::AGENT_LIST +OK event listener enabled plain RECV EVENT Event-Subclass: CONFIG::AGENT_LIST Core-UUID: 23c2fe9b-f686-45e6-b43b-a3431d32d3e2 FreeSWITCH-Hostname: deathstar.freeswitch.org FreeSWITCH-Switchname: DeathSTAR FreeSWITCH-IPv4: 8.19.97.170 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-01-17 11:33:51 Event-Date-GMT: Tue, 17 Jan 2012 17:33:51 GMT Event-Date-Timestamp: 1326821631120687 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1188 Command: sendevent CUSTOM Event-Name: CUSTOM freeswitch at internal> cat test.pl require ESL; ESL::eslSetLogLevel(7); my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $e = new ESL::ESLevent("CUSTOM","CONFIG::AGENT_LIST"); $con->sendEvent($e); On Tue, Jan 17, 2012 at 10:37 AM, Michael Collins wrote: > > > On Tue, Jan 17, 2012 at 12:17 AM, Gerald Weber wrote: > >> Hi,**** >> >> thanks for your answers**** >> >> ** ** >> >> @Anthony:**** >> >> I did a ?make current? mins ago, fs starts with version**** >> >> FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600)**** >> >> ** ** >> >> @MC**** >> >> I connect using ./fs_cli ?H 192.168.20.73 ?P 8021**** >> >> /log 0**** >> >> /event plain custom**** >> >> ** ** >> >> (additionally I connect to fs using telnet 192.168.20.73 8021 on another >> terminal and issued ?event plain all?)**** >> >> ** ** >> >> Starting my php script, fs_cli shows nothing, telnet shows the event:*** >> * >> >> ** ** >> >> Content-Length: 583**** >> >> Content-Type: text/event-plain**** >> >> ** ** >> >> Event-Subclass: CONFIG%3A%3AAGENT_LIST**** >> >> Event-Name: SOCKET_DATA**** >> >> ?**** >> >> Event-Date-Timestamp: 1326785361276007**** >> >> ?**** >> >> Event-Name: CUSTOM**** >> >> Agents: 2022**** >> >> ZMQ-Msg-Cnt: 244**** >> >> ** ** >> >> /event plain CUSTOM in fs_cli and restart php script -> no output.**** >> >> (first event subscription except all doesn?t work ?)**** >> >> /event plain all in fs_cli and restart php script shows:**** >> >> > Hmm, I guess it does not work. You can do a filter: > > /event plain all > /filter Event-Name CUSTOM > > In any case I'll have to defer to Tony as to why you're seeing two > different header's named "Event-Name" > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/7e0f2f51/attachment.html From anthony.minessale at gmail.com Tue Jan 17 20:22:40 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Jan 2012 11:22:40 -0600 Subject: [Freeswitch-users] Open Source unified autodialer software released In-Reply-To: References: Message-ID: LGPL or GPL are ok for your case, AGPL is a little scary. It really depends if you have intent to stop anyone from using it or not, if you don't then just choose MIT or BSD which are both compat with GPL and less restrictive. Remember, GPL doesn't like BSD because it says you can do anything you want (including sell it or distribute it) it's hard to understand at first. GPL wants code to be free like a birdy soaring in the sky not free like it costs no money, so they actually get mad at licenses with no restrictions that are considered free as in contain no restrictions so you have a large political battle over weather you want it to run free in a field be free of rules or be free of charge........AGPL takes it a step further and says if someone downloads your project and runs it on their website as a service, that it will be violating the license unless they provide the code to their entire infrastructure for everyone to look at. I don't know if that will go over too well for most business ppl. In reality debating licenses over code written in plaintext scripting languages makes me chuckle a bit, just the C snob in me I suppose. Really if you choose to write code in stuff like that you may as well expect people to do whatever they want with it. I don't even really like debating licenses at all, that's why we chose BSD and MPL both the (free of restrictions) variety of license. The only obligation in MPL is to show changes to the guy who wrote the code you are changing so he can see if it helps his own personal cause or not. On Tue, Jan 17, 2012 at 12:35 AM, tahir almas wrote: > Realy thankful for your suggestion, ICTDialer is developed over Drupal > 7.0 Licensed as GPL version 2 or later so we have to license ICTDialer as > GPL compatible license > > What open source License you recommend for ICTDialer ? > > Regards > > *Tahir Almas* > > Managing Partner > ICT Innovations > http://www.ictinnovations.com > Leveraging open source in ICT > > > **************************************************************************************************************** > NOTICE OF CONFIDENTIALITY > This communication including any information transmitted with it is > intended only for the use of the addressees and is confidential and may > be protected by legal privilege . If you are not an intended recipient, be > aware that any disclosure, copying, distribution or use of this e-mail or > any attachment is prohibited. If you have received this e-mail in error, > please notify us immediately by returning it to the sender and delete this > copy from your system. Thank you for your cooperation. > > > > > On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> One piece of advice is to not release it under AGPL which is the one that >> triggers the copyleft over a socket and will turn away 99% of your >> perspective testers. >> >> >> On Mon, Jan 16, 2012 at 1:46 PM, tahir almas wrote: >> >>> Pleased to announce the release of open source Fax , SMS and Voice >>> broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped over reknown Drupal Conent Mnagment System and powerfull Plivo >>> Communication framework , Your contribution and suggestions are welcome >>> >>> Regards >>> *Tahir Almas* >>> >>> Managing Partner >>> ICT Innovations >>> http://www.ictinnovations.com >>> Leveraging open source in ICT >>> >>> >>> **************************************************************************************************************** >>> NOTICE OF CONFIDENTIALITY >>> This communication including any information transmitted with it is >>> intended only for the use of the addressees and is confidential and may >>> be protected by legal privilege . If you are not an intended recipient, be >>> aware that any disclosure, copying, distribution or use of this e-mail or >>> any attachment is prohibited. If you have received this e-mail in error, >>> please notify us immediately by returning it to the sender and delete this >>> copy from your system. Thank you for your cooperation. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/94ea98f1/attachment-0001.html From anthony.minessale at gmail.com Tue Jan 17 20:30:48 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Jan 2012 11:30:48 -0600 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg In-Reply-To: References: Message-ID: I heard on the beta firmware for yealink that they support the multicast paging as done in esf_page_group from mod_esf On Tue, Jan 17, 2012 at 12:45 AM, Seven Du wrote: > seems yealink phone has problem, see > > http://pastebin.freeswitch.org/18138 > > there's no intercom in b-leg, but maybe because Call-Info present ? > > Call-Info: ;answer-after=0 > > > > > > The following confirmed work > > originate {sip_auto_answer=true}user/1000 > &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) > > INVITE sip:1002 at 192.168.7.105:5062 SIP/2.0 > Call-Info: > > > On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: > > > originate [sip_auto_answer=true]user/1000 > &bridge([sip_auto_answer=false]user/1001) > > > On Mon, Jan 16, 2012 at 1:15 PM, Seven Du wrote: > > Hi, > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want > 1000 to be auto answered but not 1001, but it seems FS automatically copy > to b-leg. I manually comment the following lines and it seems to work. > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", > "intercom=true"); > } > > switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); > // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); > > would it be good to default to not copy but leave the "export" or > equivalent to do that? Or should I patch to add a var to disable that? > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/05f70af7/attachment.html From anthony.minessale at gmail.com Tue Jan 17 20:52:33 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Jan 2012 11:52:33 -0600 Subject: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? In-Reply-To: References: Message-ID: try this: commit c358f67fe4348b8b5209328660aef02d8ffaf15f Author: Anthony Minessale Date: Tue Jan 17 12:19:23 2012 -0600 eat inbound vid while playing fsv files I didn't have a way to test it setup so i'll need feedback. On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam wrote: > Hi folks - > > I have a freeswitch dialplan + script that first calls play_fsv to > play a greeting, then record_fsv to record incoming video to fsv. > > It works fine, but there's one issue: Apparently, freeswitch is > buffering incoming video during play_fsv. When record_fsv is called, > that buffer is dumped into the .fsv file, resulting in several seconds > of unneeded video. > > Is there a way to empty that incoming video buffer before record_fsv > begins recording? > > Thank you! > > - Greg Millam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/816a8996/attachment.html From anthony.minessale at gmail.com Tue Jan 17 20:59:48 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 17 Jan 2012 11:59:48 -0600 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4F14E8E1.6060700@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> <4F14E8E1.6060700@communicatefreely.net> Message-ID: That problem has already been confirmed to be improved by 10 difft ppl. you need the absolute latest GIT on a daily basis not a fairly recent version, also now I am in the middle of a major rework of presence that may find you with even more problems depending on how right we do things vs how well we tolerate various imperfections. The majority will be served and you may have to adjust to what works for the most people. On Mon, Jan 16, 2012 at 9:20 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Okay, so I got another machine, built the update, (which wouldn't > compile until I hacked mod_spandsp), etc. etc. > > I now have a fairly recent version of FreeSWITCH and got a lab together > and can figure things out better. > > I did notice some improvements in BLFs, and they handle multiple calls > better, but I'm still having the core issue that started this to begin > with, and makes my customers complain. Here's how to duplicate: > > Phone A calls phone B, phone B answers > The lamp for phone A lights, and the lamp for phone B flashes. > When phone B answers, they are both steady. > Phone C calls phone B. Phone B's lamp flashes (it should, it is ringing). > Phone C hangs up before phone B answers the second call. > Phone B's light goes out <---- This is the problem here > > If phone B does something, ie. puts a call on hold and picks it up > again, all the lamps do what they are supposed to. In fact, they will > no correct themselves if phone B does something. What I'm trying to > solve is the fact that phone B's lamp went out even though they are > still on an active call with phone A. > > This is fairly common in a group reception environment. Not everyone > can get all the calls all the time, but the other people in the group > want to know who has an active call. > > Any ideas? Should I submit this as a bug? > > Thanks! > > -Tim > > > Anthony Minessale wrote: > > You are running a version from August. Too bad for you that you are > > missing a whole autumn worth of updates including a whole bunch of > > work on presence. > > > > while(!current) update(); > > > > > > > > On Thu, Dec 8, 2011 at 3:35 PM, Tim St. Pierre > > > > > wrote: > > > > Hello, > > > > I'm having all sorts of problems with BLFs not being in the correct > > state. It works fine in some cases, but others are wrong. Here's > > what > > I can see: > > > > Single endpoint, single call, everything works fine. Light flashes > on > > ring, goes steady on answer, goes out on hangup. > > > > Here's where it gets tricky: > > > > If there are two phones registered to the extension, it flashes when > > they ring, but then goes out when one of them answers. > > If either phone places an outgoing call, the lamp comes on. > > > > If a single phone gets a second call, their lamp flashes again, > > but then > > goes out when they answer the second call. > > > > I'm constantly getting complaints from users where the lamps are > stuck > > on. It happens more often when they have a lot of phones in ring > > groups. The lamps work fine for ring groups - they all flash, and > > whomever picks up the call stays steady while the rest go out. > > > > Is there anything I can do to get freeswitch to base the state on > > whether or not that user has any active calls, rather than just > > what the > > last thing that the phone did was? > > > > This happens on every model of Aastra phone, and I have all of > > them. I > > haven't had a chance to try it yet on Polycom. > > > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 > > 11-27-20 > > -0500) > > > > Any ideas would be more than welcome. > > > > -Tim > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:+19193869900 > > ------------------------------------------------------------------------ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/89f56b20/attachment-0001.html From msc at freeswitch.org Tue Jan 17 22:13:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 11:13:21 -0800 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Tomorrow Message-ID: Hello all! Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18 Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :) Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/88e949c7/attachment.html From notlikeme75 at yahoo.com Tue Jan 17 22:32:22 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 17 Jan 2012 11:32:22 -0800 (PST) Subject: [Freeswitch-users] announce count mod conference Message-ID: <1326828742.9641.YahooMailNeo@web65306.mail.ac2.yahoo.com> the following section of mod conference does not seem to work ?for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that.? I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers.? for announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/c032d587/attachment.html From msc at freeswitch.org Tue Jan 17 22:54:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 11:54:33 -0800 Subject: [Freeswitch-users] announce count mod conference In-Reply-To: <1326828742.9641.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1326828742.9641.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: I'm pretty sure that this is for the whole room. You are probably better off calculating the number of users in the conference just prior to sending them in: http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline HTH, MC On Tue, Jan 17, 2012 at 11:32 AM, Rodney wrote: > the following section of mod conference does not seem to work for me. is > this meant to be played for the whole room or just the user entering the > room. I would like it only to work for the person entering the room to cut > down on the amount of them pressing the dtmf to execute the announce count > extension. it seems this is the first thing they do and I want them to not > have to do that. > > I have set this to 0 and 1 and still no caller count announced. how may I > trouble shoot this problem? thanks. when i go into an empty room it says my > zerocallers.wav but nothing else if i enter a room with 1 or more callers. > > > forannounce-countRequires TTS. The system will speak the total number of > callers in the conference when a new person joins, but only once the > threshold specified in this parameter is reached.5 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/7422b8ad/attachment.html From alexis.mailinglist at de-bruyn.fr Wed Jan 18 00:04:30 2012 From: alexis.mailinglist at de-bruyn.fr (Alexis de BRUYN [Mailinglists]) Date: Tue, 17 Jan 2012 22:04:30 +0100 Subject: [Freeswitch-users] Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) Message-ID: <4F15E25E.8030405@de-bruyn.fr> Hi Everybody, I am trying to get a conversation between two natted clients and a natted freeswitch server. I am playing with a fresh install, the external profile and default directory (1000 & 1001 users). I can see that my clients are registered on the server logs (on the 5080 port). All others needed ports are opened too. I have followed the instructions from the wiki to get a double nat configuration, but my client phone are not ringing (and I cannot hear anything). All is working fine with defaults in a non nat configuration (with internal profile). Is there anything special to set up in the directory / user configuration ? Can anyone give me some hints with his natted configuration ? I am stuck. Thanks for your help. Regards, -------- Original Message -------- Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Date: Mon, 16 Jan 2012 23:37:20 +0100 From: Alexis de BRUYN [Mailinglists] Reply-To: FreeSWITCH Users Help To: freeswitch-users at lists.freeswitch.org Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.1.6 SIP-IP 192.168.1.6 URL sip:mod_sofia at 192.168.1.6:5080 BIND-URL sip:mod_sofia at 192.168.1.6:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 3 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User: 1001 at X.Y.Z.T Contact: "Freeswitch" Agent: Bria iOS 2.0.0 Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host: phone IP: A.B.C.D Port: 50193 Auth-User: 1001 Auth-Realm: X.Y.Z.T MWI-Account: 1001 at X.Y.Z.T Call-ID: 3c27b7424837-bsfi877ujb2n User: 1000 at X.Y.Z.T Contact: "freeswitch" Agent: snom300/8.4.32 Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host: phone IP: A.B.C.D Port: 62061 Auth-User: 1000 Auth-Realm: X.Y.Z.T MWI-Account: 1000 at X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr From dan at subformat.net Wed Jan 18 01:03:03 2012 From: dan at subformat.net (dan at subformat.net) Date: Tue, 17 Jan 2012 22:03:03 -0000 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> Message-ID: <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> Aaah sorry! That shows just how new I am to FS. Heres the rather large log: http://pastebin.freeswitch.org/18151 Thanks again. Dan. > Oops, you forgot to enable debug. It looks like you're capturing right > from > the console instead of using fs_cli. If you're at the console then type > this before doing a test call: > > console loglevel debug > > Note: if you use fs_cli to connect to FS you'll automatically be at > loglevel debug. Anyway, turn on debug level output and pastebin a new log. > > -MC > > On Tue, Jan 17, 2012 at 2:34 AM, wrote: > >> Thanks MC, Tried that but sleep doesn't appear to be having any effect? >> >> I have pasted my log here: http://pastebin.freeswitch.org/18143 >> >> As you can see in the log the ext is not registered, but the dial plan >> should continue on fail? to the media file etc. >> >> Thanks >> >> Dan. >> >> >> >> > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis wrote: >> > >> >> On 16/01/12 14:29, dan at subformat.net wrote: >> >> > If you could point me in the right direction, that would be great. >> >> >> >> Can you provide a log of the call from FreeSWITCH, on >> >> http://pastebin.freeswitch.org/ for us to look at, please? >> >> >> >> >> > Also, you might want to add an action right after the answer: >> > >> > >> > Sometimes it takes a moment for media to "come up" and your playback >> could >> > occur before media is happily flowing. If you sleep for a short period >> of >> > time usually that helps. Try making the sleep duration longer or >> shorter >> > to >> > see what happens. >> > >> > -MC >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Wed Jan 18 04:26:26 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 18 Jan 2012 09:26:26 +0800 Subject: [Freeswitch-users] sip_auto_answer auto copy to bleg In-Reply-To: References: Message-ID: <7BBCD65866504193809DB8B2C37CF2B6@gmail.com> I use 7.60.0.110.rom btw. On Wednesday, January 18, 2012 at 1:30 AM, Anthony Minessale wrote: > I heard on the beta firmware for yealink that they support the multicast paging as done in esf_page_group from mod_esf > > > On Tue, Jan 17, 2012 at 12:45 AM, Seven Du wrote: > > seems yealink phone has problem, see > > > > http://pastebin.freeswitch.org/18138 > > > > there's no intercom in b-leg, but maybe because Call-Info present ? > > > > Call-Info: ;answer-after=0 > > > > > > > > > > > > The following confirmed work > > > > originate {sip_auto_answer=true}user/1000 &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) > > > > INVITE sip:1002 at 192.168.7.105:5062 (http://sip:1002 at 192.168.7.105:5062) SIP/2.0 > > Call-Info: > > > > > > > > > > On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: > > > > > > > > originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001) > > > > > > > > > On Mon, Jan 16, 2012 at 1:15 PM, Seven Du wrote: > > > > Hi, > > > > > > > > I use originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. I manually comment the following lines and it seems to work. > > > > > > > > > > > > > > > > // switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); > > > > } > > > > > > > > switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); > > > > // switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); > > > > > > > > > > > > would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? > > > > > > > > Thanks. > > > > > > > > -- > > > > About: http://about.me/dujinfang > > > > Blog: http://www.dujinfang.com > > > > Proj: http://www.freeswitch.org.cn > > > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > > > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > > > pstn:+19193869900 (tel:%2B19193869900) > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/d4bda6f4/attachment-0001.html From notlikeme75 at yahoo.com Wed Jan 18 04:54:32 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 17 Jan 2012 17:54:32 -0800 (PST) Subject: [Freeswitch-users] announce count mod conference In-Reply-To: References: Message-ID: <1326851672.23930.YahooMailNeo@web65315.mail.ac2.yahoo.com> yes i would love to implement the announce conference count to user before entering but cant figure out how to "calculate" that. also. the announce count should work because i have tts installed. so i am still concerned why the whole room announce does not work. thanks. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, January 17, 2012 8:27 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 159 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. FreeSWITCH Community Conference Call Tomorrow (Michael Collins) ? 2. announce count mod conference (Rodney) ? 3. Re: announce count mod conference (Michael Collins) ? 4. Freeswitch & double NAT Configuration (Was: ? ? ? USER_NOT_REGISTERED with external profile) ? ? ? (Alexis de BRUYN [Mailinglists]) ? 5. Re: Play media dialplan (dan at subformat.net) ? 6. Re: sip_auto_answer auto copy to bleg (Seven Du) Hello all! Just a reminder that we have our FreeSWITCH community conference call tomorrow. We have Moc scheduled to discuss some of the new things he's been cooking up. However, he might have to go play with his cable modem for a bit so we'll see how the schedule pans out. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_01_18 Make a note of the upcoming conference calls. Normally I could say that we are booked solid through February, but since 2012 is a leap year we actually have a call scheduled for February 29. I'd love to hear some suggestions for a special leap-year edition of the FS community conference call! Let me know what you think. :) Thanks, Michael the following section of mod conference does not seem to work ?for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that.? I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers.? for announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5 I'm pretty sure that this is for the whole room. You are probably better off calculating the number of users in the conference just prior to sending them in: http://wiki.freeswitch.org/wiki/Conference_Announce_Count_Inline HTH, MC On Tue, Jan 17, 2012 at 11:32 AM, Rodney wrote: the following section of mod conference does not seem to work ?for me. is this meant to be played for the whole room or just the user entering the room. I would like it only to work for the person entering the room to cut down on the amount of them pressing the dtmf to execute the announce count extension. it seems this is the first thing they do and I want them to not have to do that.? > > >I have set this to 0 and 1 and still no caller count announced. how may I trouble shoot this problem? thanks. when i go into an empty room it says my zerocallers.wav but nothing else if i enter a room with 1 or more callers.? > > > > >for >announce-count Requires TTS. The system will speak the total number of callers in the conference when a new person joins, but only once the threshold specified in this parameter is reached. 5 >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > Hi Everybody, I am trying to get a conversation between two natted clients and a natted freeswitch server. I am playing with a fresh install, the external profile and default directory (1000 & 1001 users). I can see that my clients are registered on the server logs (on the 5080 port). All others needed ports are opened too. I have followed the instructions from the wiki to get a double nat configuration, but my client phone are not ringing (and I cannot hear anything). All is working fine with defaults in a non nat configuration (with internal profile). Is there anything special to set up in the directory / user configuration ? Can anyone give me some hints with his natted configuration ? I am stuck. Thanks for your help. Regards, -------- Original Message -------- Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile Date: Mon, 16 Jan 2012 23:37:20 +0100 From: Alexis de BRUYN [Mailinglists] Reply-To: FreeSWITCH Users Help To: freeswitch-users at lists.freeswitch.org Hi Everybody, I am trying to use FreeSwitch in a (double) NAT Configuration from a fresh snapshot install (Debian Squeeze, server outside from my client LAN) with default directory for users. When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the callee, the phone doesn't ring and automatically hangup, I see in the console : 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] However, 1000 and 1001 are registered (from the same LAN) : sofia status profile external ================================================================================================= Name? ? ? ? ? ? ??? external Domain Name? ? ? ??? N/A Auto-NAT? ? ? ? ??? false DBName? ? ? ? ? ??? sofia_reg_external Pres Hosts? ? ? ??? Dialplan? ? ? ? ??? XML Context? ? ? ? ? ??? public Challenge Realm? ??? auto_to RTP-IP? ? ? ? ? ??? 192.168.1.6 SIP-IP? ? ? ? ? ??? 192.168.1.6 URL? ? ? ? ? ? ? ??? sip:mod_sofia at 192.168.1.6:5080 BIND-URL? ? ? ? ??? sip:mod_sofia at 192.168.1.6:5080 HOLD-MUSIC? ? ? ??? local_stream://moh OUTBOUND-PROXY? ??? N/A CODECS IN? ? ? ? ??? G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT? ? ? ??? PCMU,PCMA,GSM TEL-EVENT? ? ? ? ??? 101 DTMF-MODE? ? ? ? ??? rfc2833 CNG? ? ? ? ? ? ? ??? 13 SESSION-TO? ? ? ??? 0 MAX-DIALOG? ? ? ??? 0 NOMEDIA? ? ? ? ? ??? false LATE-NEG? ? ? ? ??? false PROXY-MEDIA? ? ? ??? false AGGRESSIVENAT? ? ??? false STUN-ENABLED? ? ??? true STUN-AUTO-DISABLE??? false CALLS-IN? ? ? ? ??? 3 FAILED-CALLS-IN? ??? 0 CALLS-OUT? ? ? ? ??? 0 FAILED-CALLS-OUT ??? 0 Registrations: ================================================================================================= Call-ID:? ? ??? aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm User:? ? ? ??? 1001 at X.Y.Z.T Contact:? ? ??? "Freeswitch" Agent:? ? ? ??? Bria iOS 2.0.0 Status:? ? ??? Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) Host:? ? ? ??? phone IP:? ? ? ? ??? A.B.C.D Port:? ? ? ??? 50193 Auth-User:? ??? 1001 Auth-Realm: ??? X.Y.Z.T MWI-Account:??? 1001 at X.Y.Z.T Call-ID:? ? ??? 3c27b7424837-bsfi877ujb2n User:? ? ? ??? 1000 at X.Y.Z.T Contact:? ? ??? "freeswitch" Agent:? ? ? ??? snom300/8.4.32 Status:? ? ??? Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) Host:? ? ? ??? phone IP:? ? ? ? ??? A.B.C.D Port:? ? ? ??? 62061 Auth-User:? ??? 1000 Auth-Realm: ??? X.Y.Z.T MWI-Account:??? 1000 at X.Y.Z.T Total items returned: 2 ================================================================================================= All necessary ports are opened/forwarded on the server. I See on the 1000 configuration that this is the local ip address which is set as contact. Is there any other setups to do in the directory ? Or other parameters in the external profile ? Thanks for your help ! Regards, -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr Aaah sorry! That shows just how new I am to FS. Heres the rather large log: http://pastebin.freeswitch.org/18151 Thanks again. Dan. > Oops, you forgot to enable debug. It looks like you're capturing right > from > the console instead of using fs_cli. If you're at the console then type > this before doing a test call: > > console loglevel debug > > Note: if you use fs_cli to connect to FS you'll automatically be at > loglevel debug. Anyway, turn on debug level output and pastebin a new log. > > -MC > > On Tue, Jan 17, 2012 at 2:34 AM, wrote: > >> Thanks MC, Tried that but sleep doesn't appear to be having any effect? >> >> I have pasted my log here: http://pastebin.freeswitch.org/18143 >> >> As you can see in the log the ext is not registered, but the dial plan >> should continue on fail? to the media file etc. >> >> Thanks >> >> Dan. >> >> >> >> > On Mon, Jan 16, 2012 at 11:21 AM, Paul Cupis wrote: >> > >> >> On 16/01/12 14:29, dan at subformat.net wrote: >> >> > If you could point me in the right direction, that would be great. >> >> >> >> Can you provide a log of the call from FreeSWITCH, on >> >> http://pastebin.freeswitch.org/ for us to look at, please? >> >> >> >> >> > Also, you might want to add an action right after the answer: >> > >> > >> > Sometimes it takes a moment for media to "come up" and your playback >> could >> > occur before media is happily flowing. If you sleep for a short period >> of >> > time usually that helps. Try making the sleep duration longer or >> shorter >> > to >> > see what happens. >> > >> > -MC >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > I use?7.60.0.110.rom btw. On Wednesday, January 18, 2012 at 1:30 AM, Anthony Minessale wrote: I heard on the beta firmware for yealink that they support the multicast paging as done in esf_page_group from mod_esf > > > >On Tue, Jan 17, 2012 at 12:45 AM, Seven Du wrote: > >seems yealink phone has problem, see >> >> >>http://pastebin.freeswitch.org/18138 >> >> >>there's no intercom in b-leg, but maybe because Call-Info present ? >> >> >>? ?Call-Info: ;answer-after=0 >> >> >> >> >> >> >> >> >> >> >>The following confirmed work >> >> >>?originate {sip_auto_answer=true}user/1000 &bridge({sip_auto_answer=false,sip_h_Call-info='x'}user/1002) >> >> >>? ?INVITE sip:1002 at 192.168.7.105:5062 SIP/2.0 >>? ?Call-Info: >> >> >> >> >>On Tuesday, January 17, 2012 at 8:44 AM, Anthony Minessale wrote: >> >>>originate [sip_auto_answer=true]user/1000 &bridge([sip_auto_answer=false]user/1001) >>> >>> >>> >>> >>>On Mon, Jan 16, 2012 at 1:15 PM, Seven Du wrote: >>> >>>Hi, >>>> >>>> >>>>I use ? originate {sip_auto_answer}user/1000 &bridge(user/1001) and want 1000 to be auto answered but not 1001, but it seems FS automatically copy to b-leg. ?I manually comment the following lines and it seems to work. >>>> >>>> >>>> >>>> >>>> >>>> >>>>// switch_channel_set_variable(nchannel, "sip_invite_params", "intercom=true"); >>>>} >>>> >>>> >>>>switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER); >>>>// switch_ivr_transfer_variable(session, nsession, "sip_auto_answer"); >>>> >>>> >>>>would it be good to default to not copy but leave the "export" or equivalent to do that? Or should I patch to add a var to disable that? >>>> >>>> >>>>Thanks. >>>> >>>>--? >>>>About: http://about.me/dujinfang >>>>Blog: http://www.dujinfang.com >>>>Proj: ?http://www.freeswitch.org.cn >>>> >>>>Sent with Sparrow >>>> >>>> >>>> >>>>_________________________________________________________________________ >>>>Professional FreeSWITCH Consulting Services: >>>>consulting at freeswitch.org >>>>http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>>Official FreeSWITCH Sites >>>>http://www.freeswitch.org >>>>http://wiki.freeswitch.org >>>>http://www.cluecon.com >>>> >>>>FreeSWITCH-users mailing list >>>>FreeSWITCH-users at lists.freeswitch.org >>>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>http://www.freeswitch.org >>>> >>>> >>> >>> >>> >>>-- >>>Anthony Minessale II >>> >>>FreeSWITCH http://www.freeswitch.org/ >>>ClueCon http://www.cluecon.com/ >>>Twitter: http://twitter.com/FreeSWITCH_wire >>> >>>AIM: anthm >>>MSN:anthony_minessale at hotmail.com >>>GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>IRC: irc.freenode.net #freeswitch >>> >>>FreeSWITCH Developer Conference >>>sip:888 at conference.freeswitch.org >>>googletalk:conf+888 at conference.freeswitch.org >>>pstn:+19193869900 >>> >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at lists.freeswitch.org >>>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>http://www.freeswitch.org >> >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > > > >-- >Anthony Minessale II > >FreeSWITCH http://www.freeswitch.org/ >ClueCon http://www.cluecon.com/ >Twitter: http://twitter.com/FreeSWITCH_wire > >AIM: anthm >MSN:anthony_minessale at hotmail.com >GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >IRC: irc.freenode.net #freeswitch > >FreeSWITCH Developer Conference >sip:888 at conference.freeswitch.org >googletalk:conf+888 at conference.freeswitch.org >pstn:+19193869900 > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/5421192c/attachment-0001.html From msc at freeswitch.org Wed Jan 18 05:01:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jan 2012 18:01:21 -0800 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> Message-ID: On Tue, Jan 17, 2012 at 2:03 PM, wrote: > Aaah sorry! That shows just how new I am to FS. > > Heres the rather large log: http://pastebin.freeswitch.org/18151 > > Thanks again. > > Dan. > > Well, no new info there. I would confirm that the sound files themselves are not broken. You might want to clean out the sounds directory and then re-run "make cd-sounds-install" -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120117/d5fc514f/attachment.html From rhow at exemail.com.au Wed Jan 18 07:06:09 2012 From: rhow at exemail.com.au (Ryan How) Date: Wed, 18 Jan 2012 12:06:09 +0800 Subject: [Freeswitch-users] Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) In-Reply-To: <4F15E25E.8030405@de-bruyn.fr> References: <4F15E25E.8030405@de-bruyn.fr> Message-ID: <4F164531.3030201@exemail.com.au> Hi, I set mine up for the external phones to log into the internal profile and it just worked (port 5060). But it might depend on the type of NAT or something, I'm by no means an expert on the subject. Probably worth securing it better than the default config once you get it going if it works on the internal profile. Ryan On 18/01/2012 5:04 AM, Alexis de BRUYN [Mailinglists] wrote: > Hi Everybody, > > I am trying to get a conversation between two natted clients and a > natted freeswitch server. I am playing with a fresh install, the > external profile and default directory (1000& 1001 users). > > I can see that my clients are registered on the server logs (on the 5080 > port). All others needed ports are opened too. > > I have followed the instructions from the wiki to get a double nat > configuration, but my client phone are not ringing (and I cannot hear > anything). > > All is working fine with defaults in a non nat configuration (with > internal profile). > > Is there anything special to set up in the directory / user configuration ? > > Can anyone give me some hints with his natted configuration ? I am stuck. > > Thanks for your help. > > Regards, > > -------- Original Message -------- > Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile > Date: Mon, 16 Jan 2012 23:37:20 +0100 > From: Alexis de BRUYN [Mailinglists] > Reply-To: FreeSWITCH Users Help > To: freeswitch-users at lists.freeswitch.org > > Hi Everybody, > > I am trying to use FreeSwitch in a (double) NAT Configuration from a > fresh snapshot install (Debian Squeeze, server outside from my client > LAN) with default directory for users. > > When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the > callee, the phone doesn't ring and automatically hangup, I see in the > console : > > 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot > create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot > create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] > > However, 1000 and 1001 are registered (from the same LAN) : > > sofia status profile external > > ================================================================================================= > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 192.168.1.6 > SIP-IP 192.168.1.6 > URL sip:mod_sofia at 192.168.1.6:5080 > BIND-URL sip:mod_sofia at 192.168.1.6:5080 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 3 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > Registrations: > ================================================================================================= > Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm > User: 1001 at X.Y.Z.T > Contact: "Freeswitch" > Agent: Bria iOS 2.0.0 > Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) > Host: phone > IP: A.B.C.D > Port: 50193 > Auth-User: 1001 > Auth-Realm: X.Y.Z.T > MWI-Account: 1001 at X.Y.Z.T > > Call-ID: 3c27b7424837-bsfi877ujb2n > User: 1000 at X.Y.Z.T > Contact: "freeswitch" > Agent: snom300/8.4.32 > Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) > Host: phone > IP: A.B.C.D > Port: 62061 > Auth-User: 1000 > Auth-Realm: X.Y.Z.T > MWI-Account: 1000 at X.Y.Z.T > > Total items returned: 2 > ================================================================================================= > > All necessary ports are opened/forwarded on the server. > > I See on the 1000 configuration that this is the local ip address which > is set as contact. Is there any other setups to do in the directory ? > Or other parameters in the external profile ? > > Thanks for your help ! > > Regards, > > -- > Alexis de BRUYN > Mail : alexis.mailinglist at de-bruyn.fr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From miha at softnet.si Wed Jan 18 12:33:20 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 18 Jan 2012 10:33:20 +0100 Subject: [Freeswitch-users] Variable in dialplan Message-ID: <4F1691E0.2090403@softnet.si> Hi, I need a little help about passing variable between dialplans. I have created one default diaplan. In this dialplan a have set: In check special numbers XML there is a condition () and than . In special numbers XML I have a condition and if this condition is right the variable $test is set to 112. SO in default dialplan after all this is set must default dialplan execute this where ${test} should be variable from special numbers XML. Why the value of variable is not passed? Thanks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/7df46dc0/attachment.html From herman.griffin at gmail.com Wed Jan 18 12:45:38 2012 From: herman.griffin at gmail.com (Herman Griffin) Date: Wed, 18 Jan 2012 01:45:38 -0800 Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND In-Reply-To: References: <1326570165847-7188271.post@n2.nabble.com> Message-ID: Hello Michael, That response was prefect. I get it know! Thanks for taking the time to respond. Herman On Mon, Jan 16, 2012 at 4:29 PM, Michael Collins wrote: > Hi Herman, > > These are common questions when dealing with the dialplan. I highly > recommend that you acquire the FreeSWITCH "bridge book" (see link near top > of wiki.freeswitch.org) and read chapters 5 and 8. (Full disclosure: I wrote > chapter 5 and Darren Schreiber wrote chapter 8.) Both of these chapters > cover the break attribute in detail. I think you might be confusing the > purpose of the break attribute. > > Consider this dp fragment: > > ? > ? > ??? > ??? > ? > > > When the dialplan parser gets to this extension, the first thing it does is > test the "foo" field. If the test fails then the parser does not even look > at the rest of this extension. Why not? Because all conditions have an > implied break="on-false". The following two conditions are identical in > function: > ? > ? break="never". Why not? Because break="never" tells the dialplan parser, > "Hey, even if this condition fails, go ahead and evaluate the next condition > in this extension." If you have break="on-false" (or no 'break' attribute at > all) then that tells the dialplan parser, "Hey, if this condition fails then > 'break out' of this extension and continue parsing the dialplan." > > Using what we've just discussed, let's look at the example you cited in your > second post. > Dialplan extension "emergency_set_variables": > > > > ? ? > ? > > ? > ? ? > ? > > ? > ? ? > ? > > The dialplan parser logged this: > > > Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ > break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) > [emergency_set_variables] break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) > [emergency_set_variables] break=never > Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) INLINE > EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) > > Note: I added the color so that you could see which dialplan lines > correspond to the log output. > > The first condition (purple) fails. ({$caller_id_name} has the value > "Unknown" which fails the regex test against /^Emerg_/) However, since you > have break="never", the parser continues on to the next condition inside > this extension. Had you done break="on-false" (or no break attribute) then > the parser would have moved on to the next extension in the dialplan. > > The second condition (orange) also fails. Again, you have break="never", so > even though it fails, the parser moves on to the next condition. > > The third condition (green) passes, so the parser adds the to the > task list. (Since you have inline="true" the action gets executed > immediately during dialplan parsing instead later on during the "execution > phase".) > > In other words, the log output is exactly what I would expect it to be. > > As to your other question about anti-actions in a stack: this also is a > common question. It looks like you are trying to do something like this: > > IF (cond1 AND cond2 AND cond3) THEN > do actions > ELSE > do other actions > ENDIF > > You cannot do this particular construct just with conditions and > anti-actions. Instead you'll need the brand spanking new "regex" syntax > mentioned here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#Multiple_Conditions_.28Logical_OR.2C_XOR.29 > > So, to do what you wanted to do in the second example (in your second post) > you could try this: > > > ? > ? > ? > ? > > ?? > ?? data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> > ?? > ?? > ?? data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > ?? > ?? data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > ? > > ?? data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> > ?? > ?? > ?? data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > ?? > ?? data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > > > > The tells the parser, "Hey, execute the 's > only if all regexes PASS, otherwise execute any 's". That > should give you what you need. > > Hope this helps! > > -Michael > > > > > On Sat, Jan 14, 2012 at 7:39 PM, Herman Griffin > wrote: >> >> Hello everyone, >> >> In irc SwK said that freeswitch AND operator is lazy. To me this means >> that if a condition is set to break=true and it evaluates to FAIL, >> then the that break=never will invert the meaning of FAIL; Change it >> to a PASS . But why doesn't break=false behave in this same manner >> when there is only a single condition? >> >> Take this call trace and dialplan for example. I use a single >> condition with break=false. However, unlike the stacked conditions, >> the action is not executed when the single condition evaluates to >> FAIL. Why doesn't break=never cause the action to be executed in a non >> stacked condition? >> >> Call trace: >> >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) >> [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ >> break=never >> Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) >> [emergency_set_variables] break=never >> Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) >> [emergency_set_variables] break=never >> Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) INLINE >> EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) >> >> Dialplan: >> >> >> ? ? >> ? >> >> ? >> ? ? >> ? >> >> ? >> ? ? >> ? >> >> ------------------------------------------------- >> >> I also getting confusing behavior with anti-action and stacked >> condition. I expect the anti-action to be executed if any of the >> stacked conditions evaluates to FAIL. However, in the case below, the >> extension simply breaks if any of the condition evaluates to FAIL. >> >> Call trace: >> >> Dialplan: sofia/external/Unknown at 72.37.252.18 parsing >> [public->emergency_bridge] continue=true >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) >> [emergency_bridge] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ >> break=on-false >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) >> [emergency_bridge] ${emergency_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/Unknown at 72.37.252.18 parsing >> [public->public_did] continue=false <-- [[MOVES ON TO ANOTHER >> EXTENSION WITH EXECUTING THE anti-action]] >> >> Dialplan: >> >> >> ? ? >> ? ? >> ? ? ? >> ? ? ?> >> data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> >> ? ? ? >> ? ? ? >> ? ? ?> >> data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> >> ? ? ? >> ? ? ?> data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> >> >> ? ? ?> >> data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> >> ? ? ? >> ? ? ? >> ? ? ?> >> data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> >> ? ? ? >> ? ? ?> data="user/1000@${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> >> ? ? >> ? >> >> >> SwK suggested that I use a script language, which is probably what >> I'll end up doing. However, I'm very interested in understanding why >> the scenarios above. Are these bugs or are they just counter intuitive >> rules that we must simply memorize? >> >> Thanks for your input. >> Herman Griffin >> >> >> >> On Sat, Jan 14, 2012 at 11:42 AM, Jeff Lenk wrote: >> > By putting the break=never you are defeating the "and" processing. >> > Remove >> > that. >> > >> > Move the second group into its own extension. >> > >> > -- >> > View this message in context: >> > http://freeswitch-users.2379917.n2.nabble.com/Stacked-conditions-are-not-acting-like-logical-AND-tp7188082p7188271.html >> > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gerald.weber at besharp.at Wed Jan 18 12:58:54 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 18 Jan 2012 09:58:54 +0000 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: I get the same results with your script. (maybe I was not clear in my last reply and mixed things up, still new to fs) I was just curious why a plain telnet connect to fs shows "Event-Name CUSTOM" AND "Event-Name SOCKET_DATA" and fs_cli only shows Event-Name CUSTOM Seems there is a difference between fs_cli and a plain telnet connect to fs ? My real problem: When I want to receive all CUSTOM events (not being specific to a Event-Subclass) I don't receive them using $sock->sendRecv("event plain custom"); and $sock->recvEvent() in e.g. get.php when generating this events using ESLevent("CUSTOM","CONFIG::AGENT_LIST"); and sendEvent in send.php. I guess recvEvent is listening for the first Event-Name entry. If it doesn't match -> go ahead. It works using $sock->sendRecv("event plain custom CONFIG::AGENT_LIST"); But this means I have to know where the event is injected. Or am I on the wrong way, again ? Thx & regards, gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Dienstag, 17. J?nner 2012 18:11 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice you cannot do *all* custom events with /event plain custom that command is expecting also a subclass param /event plain custom CONFIG::AGENT_LIST you have made a mistake if you think this unlocks heartbeat etc, probably you did events plain all and forgot. When I do it from perl I do not see what you reported, perl test.pl [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1381 esl_send() SEND auth ClueCon [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:502 esl_sendevent() SEND EVENT Event-Name: CUSTOM Event-Subclass: CONFIG::AGENT_LIST [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK +OK log level [7] freeswitch at internal> /event plain custom CONFIG::AGENT_LIST +OK event listener enabled plain RECV EVENT Event-Subclass: CONFIG::AGENT_LIST Core-UUID: 23c2fe9b-f686-45e6-b43b-a3431d32d3e2 FreeSWITCH-Hostname: deathstar.freeswitch.org FreeSWITCH-Switchname: DeathSTAR FreeSWITCH-IPv4: 8.19.97.170 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-01-17 11:33:51 Event-Date-GMT: Tue, 17 Jan 2012 17:33:51 GMT Event-Date-Timestamp: 1326821631120687 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1188 Command: sendevent CUSTOM Event-Name: CUSTOM freeswitch at internal> cat test.pl require ESL; ESL::eslSetLogLevel(7); my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $e = new ESL::ESLevent("CUSTOM","CONFIG::AGENT_LIST"); $con->sendEvent($e); On Tue, Jan 17, 2012 at 10:37 AM, Michael Collins > wrote: On Tue, Jan 17, 2012 at 12:17 AM, Gerald Weber > wrote: Hi, thanks for your answers @Anthony: I did a "make current" mins ago, fs starts with version FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600) @MC I connect using ./fs_cli -H 192.168.20.73 -P 8021 /log 0 /event plain custom (additionally I connect to fs using telnet 192.168.20.73 8021 on another terminal and issued "event plain all") Starting my php script, fs_cli shows nothing, telnet shows the event: Content-Length: 583 Content-Type: text/event-plain Event-Subclass: CONFIG%3A%3AAGENT_LIST Event-Name: SOCKET_DATA ... Event-Date-Timestamp: 1326785361276007 ... Event-Name: CUSTOM Agents: 2022 ZMQ-Msg-Cnt: 244 /event plain CUSTOM in fs_cli and restart php script -> no output. (first event subscription except all doesn't work ?) /event plain all in fs_cli and restart php script shows: Hmm, I guess it does not work. You can do a filter: /event plain all /filter Event-Name CUSTOM In any case I'll have to defer to Tony as to why you're seeing two different header's named "Event-Name" -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/d2418c37/attachment-0001.html From gopalakrishnan.an at gmail.com Wed Jan 18 13:44:58 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 18 Jan 2012 16:14:58 +0530 Subject: [Freeswitch-users] Building Customized module for Transcoding Card Message-ID: Hi, We are developing a module for freeswitch for transcoding card we have some static libraries with extension (*.a) which has to be linked to get the final module for freeswitch. Can any one please help in letting us know where the static libraries are to be placed? And with what option it has to be included in the Makefile. Thanks in advance. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/5f50076c/attachment.html From gopalakrishnan.an at gmail.com Wed Jan 18 13:46:13 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 18 Jan 2012 16:16:13 +0530 Subject: [Freeswitch-users] Dahdi module load error Message-ID: Hi, We compiled the mod_dahdi_codec it got compiled when we try to load the module from freswitch command line it gives an error as freeswitch at localhost.localdomain> load mod_dahdi_codec -ERR [module load file routine returned an error] 2012-01-18 23:07:56.654211 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_dahdi_codec.so **/usr/local/freeswitch/mod/mod_dahdi_codec.so: invalid ELF header** freeswitch at localhost.localdomain> Can you please help in this matter. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/6a84761d/attachment.html From gopalakrishnan.an at gmail.com Wed Jan 18 14:26:40 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 18 Jan 2012 16:56:40 +0530 Subject: [Freeswitch-users] Building Customized module for Transcoding Card In-Reply-To: References: Message-ID: we found out. Thanks On Wed, Jan 18, 2012 at 4:14 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Hi, > > > We are developing a module for freeswitch for transcoding card we have > some static libraries with extension (*.a) which has to be linked to get > the final module for freeswitch. > > > Can any one please help in letting us know where the static libraries are > to be placed? > > > And with what option it has to be included in the Makefile. > > > Thanks in advance. > > > Regards. > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/52cf77cd/attachment.html From gopalakrishnan.an at gmail.com Wed Jan 18 14:27:03 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 18 Jan 2012 16:57:03 +0530 Subject: [Freeswitch-users] Dahdi module load error In-Reply-To: References: Message-ID: The module is loaded after rebooting. Thank you. On Wed, Jan 18, 2012 at 4:16 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > Hi, > > > We compiled the mod_dahdi_codec it got compiled when we try to load the > module from freswitch command line it gives an error as > > > > freeswitch at localhost.localdomain> load mod_dahdi_codec > > > > -ERR [module load file routine returned an error] > > > > 2012-01-18 23:07:56.654211 [CRIT] switch_loadable_module.c:882 Error > Loading module /usr/local/freeswitch/mod/mod_dahdi_codec.so > > **/usr/local/freeswitch/mod/mod_dahdi_codec.so: invalid ELF header** > > freeswitch at localhost.localdomain> > > > > Can you please help in this matter. > > > Regards. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/0b38cdf2/attachment.html From dan at subformat.net Wed Jan 18 14:35:05 2012 From: dan at subformat.net (dan at subformat.net) Date: Wed, 18 Jan 2012 11:35:05 -0000 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> Message-ID: <3b40b751b8da8b91b0894edf1ea98a98.squirrel@vm1.subformat.net> The sound file does work, as i can just set the dial plan to go straight to play the media and it plays fine. Its only when the play media is in the middle of the dial plan that it doesnt seem to work. Should this work? Or should i be achieving this from a different setup/dialplan? For instance, maybe specifying the voicemail file (wav) to play for a certain extension number? Thanks Dan. > On Tue, Jan 17, 2012 at 2:03 PM, wrote: > >> Aaah sorry! That shows just how new I am to FS. >> >> Heres the rather large log: http://pastebin.freeswitch.org/18151 >> >> Thanks again. >> >> Dan. >> >> > Well, no new info there. I would confirm that the sound files themselves > are not broken. You might want to clean out the sounds directory and then > re-run "make cd-sounds-install" > > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From govoiper at gmail.com Wed Jan 18 14:40:44 2012 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 18 Jan 2012 16:40:44 +0500 Subject: [Freeswitch-users] Stacked conditions are not acting like logical AND In-Reply-To: References: <1326570165847-7188271.post@n2.nabble.com> Message-ID: A Big thanks from me as well for the easiest and simplest explanation. :) On Wed, Jan 18, 2012 at 2:45 PM, Herman Griffin wrote: > Hello Michael, > > That response was prefect. I get it know! Thanks for taking the time to > respond. > > Herman > > On Mon, Jan 16, 2012 at 4:29 PM, Michael Collins > wrote: > > Hi Herman, > > > > These are common questions when dealing with the dialplan. I highly > > recommend that you acquire the FreeSWITCH "bridge book" (see link near > top > > of wiki.freeswitch.org) and read chapters 5 and 8. (Full disclosure: I > wrote > > chapter 5 and Darren Schreiber wrote chapter 8.) Both of these chapters > > cover the break attribute in detail. I think you might be confusing the > > purpose of the break attribute. > > > > Consider this dp fragment: > > > > > > > > > > > > > > > > > > When the dialplan parser gets to this extension, the first thing it does > is > > test the "foo" field. If the test fails then the parser does not even > look > > at the rest of this extension. Why not? Because all conditions have an > > implied break="on-false". The following two conditions are identical in > > function: > > > > > break="never". Why not? Because break="never" tells the dialplan parser, > > "Hey, even if this condition fails, go ahead and evaluate the next > condition > > in this extension." If you have break="on-false" (or no 'break' > attribute at > > all) then that tells the dialplan parser, "Hey, if this condition fails > then > > 'break out' of this extension and continue parsing the dialplan." > > > > Using what we've just discussed, let's look at the example you cited in > your > > second post. > > Dialplan extension "emergency_set_variables": > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > The dialplan parser logged this: > > > > > > Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > > [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ > > break=never > > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) > > [emergency_set_variables] break=never > > Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) > > [emergency_set_variables] break=never > > Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) > INLINE > > EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) > > > > Note: I added the color so that you could see which dialplan lines > > correspond to the log output. > > > > The first condition (purple) fails. ({$caller_id_name} has the value > > "Unknown" which fails the regex test against /^Emerg_/) However, since > you > > have break="never", the parser continues on to the next condition inside > > this extension. Had you done break="on-false" (or no break attribute) > then > > the parser would have moved on to the next extension in the dialplan. > > > > The second condition (orange) also fails. Again, you have break="never", > so > > even though it fails, the parser moves on to the next condition. > > > > The third condition (green) passes, so the parser adds the to > the > > task list. (Since you have inline="true" the action gets executed > > immediately during dialplan parsing instead later on during the > "execution > > phase".) > > > > In other words, the log output is exactly what I would expect it to be. > > > > As to your other question about anti-actions in a stack: this also is a > > common question. It looks like you are trying to do something like this: > > > > IF (cond1 AND cond2 AND cond3) THEN > > do actions > > ELSE > > do other actions > > ENDIF > > > > You cannot do this particular construct just with conditions and > > anti-actions. Instead you'll need the brand spanking new "regex" syntax > > mentioned here: > > > http://wiki.freeswitch.org/wiki/Dialplan_XML#Multiple_Conditions_.28Logical_OR.2C_XOR.29 > > > > So, to do what you wanted to do in the second example (in your second > post) > > you could try this: > > > > > > > > > > > > > > > > > > > > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> > > > > > > > > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > > > > > data="user/1000@ > ${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > > > > > > > > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> > > > > > > > > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > > > > > data="user/1000@ > ${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > > > > > > > > The tells the parser, "Hey, execute the > 's > > only if all regexes PASS, otherwise execute any 's". That > > should give you what you need. > > > > Hope this helps! > > > > -Michael > > > > > > > > > > On Sat, Jan 14, 2012 at 7:39 PM, Herman Griffin < > herman.griffin at gmail.com> > > wrote: > >> > >> Hello everyone, > >> > >> In irc SwK said that freeswitch AND operator is lazy. To me this means > >> that if a condition is set to break=true and it evaluates to FAIL, > >> then the that break=never will invert the meaning of FAIL; Change it > >> to a PASS . But why doesn't break=false behave in this same manner > >> when there is only a single condition? > >> > >> Take this call trace and dialplan for example. I use a single > >> condition with break=false. However, unlike the stacked conditions, > >> the action is not executed when the single condition evaluates to > >> FAIL. Why doesn't break=never cause the action to be executed in a non > >> stacked condition? > >> > >> Call trace: > >> > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > >> [emergency_set_variables] ${caller_id_name}(Unknown) =~ /^Emerg_/ > >> break=never > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Date/TimeMatch (FAIL) > >> [emergency_set_variables] break=never > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Date/Time Match (PASS) > >> [emergency_set_variables] break=never > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Action set(open=true) > INLINE > >> EXECUTE sofia/external/Unknown at 72.37.252.18 set(open=true) > >> > >> Dialplan: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> ------------------------------------------------- > >> > >> I also getting confusing behavior with anti-action and stacked > >> condition. I expect the anti-action to be executed if any of the > >> stacked conditions evaluates to FAIL. However, in the case below, the > >> extension simply breaks if any of the condition evaluates to FAIL. > >> > >> Call trace: > >> > >> Dialplan: sofia/external/Unknown at 72.37.252.18 parsing > >> [public->emergency_bridge] continue=true > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (PASS) > >> [emergency_bridge] ${sip_gateway}(1006_7217) =~ /^1006_7217$/ > >> break=on-false > >> Dialplan: sofia/external/Unknown at 72.37.252.18 Regex (FAIL) > >> [emergency_bridge] ${emergency_call}() =~ /^true$/ break=on-false > >> Dialplan: sofia/external/Unknown at 72.37.252.18 parsing > >> [public->public_did] continue=false <-- [[MOVES ON TO ANOTHER > >> EXTENSION WITH EXECUTING THE anti-action]] > >> > >> Dialplan: > >> > >> > >> > >> expression="^1$"> > >> > >> >> > >> > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|Auto%1)}"/> > >> > >> > >> >> > >> > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > >> > >> >> data="user/1000@ > ${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > >> > >> >> > >> > data="effective_caller_id_name=${regex(${caller_id_name}|^Emerg(_.*)$|NotAuto%1)}"/> > >> > >> > >> >> > >> > data="group_confirm_file=/usr/local/freeswitch/sounds/en/us/callie/emergency/press_to_accept.wav"/> > >> > >> >> data="user/1000@ > ${domain_name},sofia/gateway/1006_7217/${mobile_number}"/> > >> > >> > >> > >> > >> SwK suggested that I use a script language, which is probably what > >> I'll end up doing. However, I'm very interested in understanding why > >> the scenarios above. Are these bugs or are they just counter intuitive > >> rules that we must simply memorize? > >> > >> Thanks for your input. > >> Herman Griffin > >> > >> > >> > >> On Sat, Jan 14, 2012 at 11:42 AM, Jeff Lenk wrote: > >> > By putting the break=never you are defeating the "and" processing. > >> > Remove > >> > that. > >> > > >> > Move the second group into its own extension. > >> > > >> > -- > >> > View this message in context: > >> > > http://freeswitch-users.2379917.n2.nabble.com/Stacked-conditions-are-not-acting-like-logical-AND-tp7188082p7188271.html > >> > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/2ea5be2b/attachment-0001.html From avi at avimarcus.net Wed Jan 18 14:59:54 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 18 Jan 2012 13:59:54 +0200 Subject: [Freeswitch-users] Variable in dialplan In-Reply-To: <4F1691E0.2090403@softnet.si> References: <4F1691E0.2090403@softnet.si> Message-ID: How are you setting test to 112? And can you pastbin your fs_cli log of the call? That often makes it clear what is going on. -Avi On Wed, Jan 18, 2012 at 11:33 AM, Miha Zoubek wrote: > Hi, > > I need a little help about passing variable between dialplans. > > I have created one default diaplan. In this dialplan a have set: > > > In check special numbers XML there is a condition ( field="destination_number" expression="^check special numbers$" >) and > than > . > > In special numbers XML I have a condition and if this condition is right > the variable $test is set to 112. SO in default dialplan after all this is > set must default dialplan execute this "sofia/external/386${test}@xx.xxx.xxx.xxx" > /> > where ${test} should be variable from special numbers XML. > > Why the value of variable is not passed? > > Thanks! > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/338c8046/attachment.html From jaasmailing at gmail.com Wed Jan 18 15:02:01 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 18 Jan 2012 13:02:01 +0100 Subject: [Freeswitch-users] Question about the future of mod_nibblebill In-Reply-To: References: <4F152BD4.9080705@gmail.com> Message-ID: <4F16B4B9.2080604@gmail.com> Good, just wondering if there is a roadmap... Why not include information in the wiki about the developers and status of the modules, similarly to *ser projects? (ex. mantained/unmantained, alpha/beta/stable) (OT) Does anyone have used mod_nibblebill to handle prepaid promotions (first 100 free minutes, than another tariff)? Regards, Il 17/01/12 17.41, Michael Collins ha scritto: > The author of this module (Darren Schreiber) uses it quite a bit, as > do a number of community members. I believe it is a "safe" module to > use for your project. It is still actively maintained. You can open > tickets at jira.freeswitch.org for > mod_nibblebill and assign them to Darren and he'll see them and > respond. He's a little slow but only because his plate is full. ;) > > -MC > > On Tue, Jan 17, 2012 at 12:05 AM, Carlo Dimaggio > > wrote: > > Hi all, > > I'm thinking about a development of a custom prepaid application > based on mod_nibblebill (that is very interesting). > I'm wondering about the status and the future of this module in > order to understand if this is the better way to achieve the aim > of the project. > I have seen in the mod_nibblebill source code this notes: > > * TODO: Fix what happens when the DB is not available > * TODO: Fix what happens when the DB queries fail (right now, all > are acting like success) > * TODO: Add buffering abilities > * TODO: Make error handling for database, such that when the > database is down (or not installed) we just log to a text file > * FUTURE: Possibly make the hooks not tied per-channel, and > instead just do this as a supervision style application with one > thread that watches all calls > > Does the developers think to continue supporting the module (not > only bugfix but new features)? > > I'm sorry for the particular question but I wouldn't do errors at > this stage of the project. > > > Best Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/756c8b6f/attachment.html From nbhatti at gmail.com Wed Jan 18 15:09:01 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 18 Jan 2012 15:09:01 +0300 Subject: [Freeswitch-users] Question about the future of mod_nibblebill In-Reply-To: <4F16B4B9.2080604@gmail.com> References: <4F152BD4.9080705@gmail.com> <4F16B4B9.2080604@gmail.com> Message-ID: vBilling has a mod_nibble equivalent written in PHP. Just < 200 lines. Will discuss in Wednesday's conference call on 25th. Muhammad On Wed, Jan 18, 2012 at 3:02 PM, Carlo Dimaggio wrote: > Good, just wondering if there is a roadmap... > Why not include information in the wiki about the developers and status of > the modules, similarly to *ser projects? (ex. mantained/unmantained, > alpha/beta/stable) > > (OT) Does anyone have used mod_nibblebill to handle prepaid promotions > (first 100 free minutes, than another tariff)? > > Regards, > > > Il 17/01/12 17.41, Michael Collins ha scritto: > > The author of this module (Darren Schreiber) uses it quite a bit, as do a > number of community members. I believe it is a "safe" module to use for your > project. It is still actively maintained. You can open tickets at > jira.freeswitch.org for mod_nibblebill and assign them to Darren and he'll > see them and respond. He's a little slow but only because his plate is full. > ;) > > -MC > > On Tue, Jan 17, 2012 at 12:05 AM, Carlo Dimaggio > wrote: >> >> Hi all, >> >> I'm thinking about a development of a custom prepaid application based on >> mod_nibblebill (that is very interesting). >> I'm wondering about the status and the future of this module in order to >> understand if this is the better way to achieve the aim of the project. >> I have seen in the mod_nibblebill source code this notes: >> >> * TODO: Fix what happens when the DB is not available >> ?* TODO: Fix what happens when the DB queries fail (right now, all are >> acting like success) >> ?* TODO: Add buffering abilities >> ?* TODO: Make error handling for database, such that when the database is >> down (or not installed) we just log to a text file >> ?* FUTURE: Possibly make the hooks not tied per-channel, and instead just >> do this as a supervision style application with one thread that watches all >> calls >> >> Does the developers think to continue supporting the module (not only >> bugfix but new features)? >> >> I'm sorry for the particular question but I wouldn't do errors at this >> stage of the project. >> >> >> Best Regards, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From govoiper at gmail.com Wed Jan 18 15:32:38 2012 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 18 Jan 2012 17:32:38 +0500 Subject: [Freeswitch-users] Question about the future of mod_nibblebill In-Reply-To: References: <4F152BD4.9080705@gmail.com> <4F16B4B9.2080604@gmail.com> Message-ID: What kind of sorcery is this !? I must attend the Wednesday's Conference now. On Wed, Jan 18, 2012 at 5:09 PM, Muhammad Naseer Bhatti wrote: > vBilling has a mod_nibble equivalent written in PHP. Just < 200 lines. > Will discuss in Wednesday's conference call on 25th. > > Muhammad > > On Wed, Jan 18, 2012 at 3:02 PM, Carlo Dimaggio > wrote: > > Good, just wondering if there is a roadmap... > > Why not include information in the wiki about the developers and status > of > > the modules, similarly to *ser projects? (ex. mantained/unmantained, > > alpha/beta/stable) > > > > (OT) Does anyone have used mod_nibblebill to handle prepaid promotions > > (first 100 free minutes, than another tariff)? > > > > Regards, > > > > > > Il 17/01/12 17.41, Michael Collins ha scritto: > > > > The author of this module (Darren Schreiber) uses it quite a bit, as do a > > number of community members. I believe it is a "safe" module to use for > your > > project. It is still actively maintained. You can open tickets at > > jira.freeswitch.org for mod_nibblebill and assign them to Darren and > he'll > > see them and respond. He's a little slow but only because his plate is > full. > > ;) > > > > -MC > > > > On Tue, Jan 17, 2012 at 12:05 AM, Carlo Dimaggio > > wrote: > >> > >> Hi all, > >> > >> I'm thinking about a development of a custom prepaid application based > on > >> mod_nibblebill (that is very interesting). > >> I'm wondering about the status and the future of this module in order to > >> understand if this is the better way to achieve the aim of the project. > >> I have seen in the mod_nibblebill source code this notes: > >> > >> * TODO: Fix what happens when the DB is not available > >> * TODO: Fix what happens when the DB queries fail (right now, all are > >> acting like success) > >> * TODO: Add buffering abilities > >> * TODO: Make error handling for database, such that when the database > is > >> down (or not installed) we just log to a text file > >> * FUTURE: Possibly make the hooks not tied per-channel, and instead > just > >> do this as a supervision style application with one thread that watches > all > >> calls > >> > >> Does the developers think to continue supporting the module (not only > >> bugfix but new features)? > >> > >> I'm sorry for the particular question but I wouldn't do errors at this > >> stage of the project. > >> > >> > >> Best Regards, > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/906abf53/attachment-0001.html From devel at omninet.eu Wed Jan 18 15:47:16 2012 From: devel at omninet.eu (Anestis Mavro) Date: Wed, 18 Jan 2012 14:47:16 +0200 Subject: [Freeswitch-users] choppy sound on voice recording and conference Message-ID: Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729 at 40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/48a2078a/attachment.html From sherifomran2000 at yahoo.com Wed Jan 18 16:35:38 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Wed, 18 Jan 2012 05:35:38 -0800 (PST) Subject: [Freeswitch-users] Question about the future of mod_nibblebill In-Reply-To: Message-ID: <1326893738.92649.YahooMailClassic@web110811.mail.gq1.yahoo.com> I can not use vBilling after long time investment, it is useless. I don't see CDR data and any other details. There is no documentation as well. --- On Wed, 1/18/12, Muhammad Naseer Bhatti wrote: From: Muhammad Naseer Bhatti Subject: Re: [Freeswitch-users] Question about the future of mod_nibblebill To: "FreeSWITCH Users Help" Date: Wednesday, January 18, 2012, 2:09 PM vBilling has a mod_nibble equivalent written in PHP. Just < 200 lines. Will discuss in Wednesday's conference call on 25th. Muhammad On Wed, Jan 18, 2012 at 3:02 PM, Carlo Dimaggio wrote: > Good, just wondering if there is a roadmap... > Why not include information in the wiki about the developers and status of > the modules, similarly to *ser projects? (ex. mantained/unmantained, > alpha/beta/stable) > > (OT) Does anyone have used mod_nibblebill to handle prepaid promotions > (first 100 free minutes, than another tariff)? > > Regards, > > > Il 17/01/12 17.41, Michael Collins ha scritto: > > The author of this module (Darren Schreiber) uses it quite a bit, as do a > number of community members. I believe it is a "safe" module to use for your > project. It is still actively maintained. You can open tickets at > jira.freeswitch.org for mod_nibblebill and assign them to Darren and he'll > see them and respond. He's a little slow but only because his plate is full. > ;) > > -MC > > On Tue, Jan 17, 2012 at 12:05 AM, Carlo Dimaggio > wrote: >> >> Hi all, >> >> I'm thinking about a development of a custom prepaid application based on >> mod_nibblebill (that is very interesting). >> I'm wondering about the status and the future of this module in order to >> understand if this is the better way to achieve the aim of the project. >> I have seen in the mod_nibblebill source code this notes: >> >> * TODO: Fix what happens when the DB is not available >> ?* TODO: Fix what happens when the DB queries fail (right now, all are >> acting like success) >> ?* TODO: Add buffering abilities >> ?* TODO: Make error handling for database, such that when the database is >> down (or not installed) we just log to a text file >> ?* FUTURE: Possibly make the hooks not tied per-channel, and instead just >> do this as a supervision style application with one thread that watches all >> calls >> >> Does the developers think to continue supporting the module (not only >> bugfix but new features)? >> >> I'm sorry for the particular question but I wouldn't do errors at this >> stage of the project. >> >> >> Best Regards, >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/682ce3c3/attachment.html From miha at softnet.si Wed Jan 18 16:56:12 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 18 Jan 2012 14:56:12 +0100 Subject: [Freeswitch-users] Variable in dialplan In-Reply-To: References: <4F1691E0.2090403@softnet.si> Message-ID: <4F16CF7C.3060802@softnet.si> On 1/18/2012 12:59 PM, Avi Marcus wrote: > How are you setting test to 112? > And can you pastbin your fs_cli log of the call? That often makes it > clear what is going on. > > -Avi > > > On Wed, Jan 18, 2012 at 11:33 AM, Miha Zoubek > wrote: > > Hi, > > I need a little help about passing variable between dialplans. > > I have created one default diaplan. In this dialplan a have set: > > > In check special numbers XML there is a condition ( field="destination_number" expression="^check special numbers$" >) > and than > . > > In special numbers XML I have a condition and if this condition is > right the variable $test is set to 112. SO in default dialplan > after all this is set must default dialplan execute this application="bridge" > data="sofia/external/386${test}@xx.xxx.xxx.xxx" > /> > where ${test} should be variable from special numbers XML. > > Why the value of variable is not passed? > > Thanks! > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi @Avi, I found out what was causing the problem. Thanks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/a5f08fb2/attachment-0001.html From tonybecq at yahoo.fr Wed Jan 18 17:51:13 2012 From: tonybecq at yahoo.fr (obbyone) Date: Wed, 18 Jan 2012 06:51:13 -0800 (PST) Subject: [Freeswitch-users] Issues with Linksys ATA and DTMF Message-ID: <1326898273671-7200341.post@n2.nabble.com> Hi, On my freeswitch server, I have created a number 9910 that has to recieve messages in DTMF. As I call that number with a softphone (xlite) on a PC, everything works, but as I try to call and send DTMF from an analog phone plugged into an ATA linksys spa 3000, freeswitch answers, but no DTMF is recieved. The linksys is configured to send DTMF in inband mode. Do you have any idea how I can handle this issue ? Thanks, Tony -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-DTMF-tp7200341p7200341.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tahir at ictinnovations.com Wed Jan 18 20:04:27 2012 From: tahir at ictinnovations.com (tahir almas) Date: Wed, 18 Jan 2012 22:04:27 +0500 Subject: [Freeswitch-users] Open Source unified autodialer software released In-Reply-To: References: Message-ID: Thanks for valuable suggestions , will consider these recommendations Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation. On Tue, Jan 17, 2012 at 10:22 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > LGPL or GPL are ok for your case, AGPL is a little scary. It really > depends if you have intent to stop anyone from using it or not, if you > don't then just choose MIT or BSD which are both compat with GPL and less > restrictive. Remember, GPL doesn't like BSD because it says you can do > anything you want (including sell it or distribute it) it's hard to > understand at first. GPL wants code to be free like a birdy soaring in the > sky not free like it costs no money, so they actually get mad at licenses > with no restrictions that are considered free as in contain no restrictions > so you have a large political battle over weather you want it to run free > in a field be free of rules or be free of charge........AGPL takes it a > step further and says if someone downloads your project and runs it on > their website as a service, that it will be violating the license unless > they provide the code to their entire infrastructure for everyone to look > at. I don't know if that will go over too well for most business ppl. > > > In reality debating licenses over code written in plaintext scripting > languages makes me chuckle a bit, just the C snob in me I suppose. Really > if you choose to write code in stuff like that you may as well expect > people to do whatever they want with it. > > I don't even really like debating licenses at all, that's why we chose BSD > and MPL both the (free of restrictions) variety of license. The only > obligation in MPL is to show changes to the guy who wrote the code you are > changing so he can see if it helps his own personal cause or not. > > > On Tue, Jan 17, 2012 at 12:35 AM, tahir almas wrote: > >> Realy thankful for your suggestion, ICTDialer is developed over Drupal >> 7.0 Licensed as GPL version 2 or later so we have to license ICTDialer as >> GPL compatible license >> >> What open source License you recommend for ICTDialer ? >> >> Regards >> >> *Tahir Almas* >> >> Managing Partner >> ICT Innovations >> http://www.ictinnovations.com >> Leveraging open source in ICT >> >> >> **************************************************************************************************************** >> NOTICE OF CONFIDENTIALITY >> This communication including any information transmitted with it is >> intended only for the use of the addressees and is confidential and may >> be protected by legal privilege . If you are not an intended recipient, be >> aware that any disclosure, copying, distribution or use of this e-mail or >> any attachment is prohibited. If you have received this e-mail in error, >> please notify us immediately by returning it to the sender and delete this >> copy from your system. Thank you for your cooperation. >> >> >> >> >> On Tue, Jan 17, 2012 at 6:10 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> One piece of advice is to not release it under AGPL which is the one >>> that triggers the copyleft over a socket and will turn away 99% of your >>> perspective testers. >>> >>> >>> On Mon, Jan 16, 2012 at 1:46 PM, tahir almas wrote: >>> >>>> Pleased to announce the release of open source Fax , SMS and Voice >>>> broadcasting software solution ICTDialer http://www.ictdialer.orgdeveloped over reknown Drupal Conent Mnagment System and powerfull Plivo >>>> Communication framework , Your contribution and suggestions are welcome >>>> >>>> Regards >>>> *Tahir Almas* >>>> >>>> Managing Partner >>>> ICT Innovations >>>> http://www.ictinnovations.com >>>> Leveraging open source in ICT >>>> >>>> >>>> **************************************************************************************************************** >>>> NOTICE OF CONFIDENTIALITY >>>> This communication including any information transmitted with it is >>>> intended only for the use of the addressees and is confidential and >>>> may be protected by legal privilege . If you are not an intended recipient, >>>> be aware that any disclosure, copying, distribution or use of this e-mail >>>> or any attachment is prohibited. If you have received this e-mail in error, >>>> please notify us immediately by returning it to the sender and delete this >>>> copy from your system. Thank you for your cooperation. >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/57276a19/attachment.html From errotan at elder.hu Wed Jan 18 20:22:54 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Wed, 18 Jan 2012 18:22:54 +0100 Subject: [Freeswitch-users] Issues with Linksys ATA and DTMF In-Reply-To: <1326898273671-7200341.post@n2.nabble.com> References: <1326898273671-7200341.post@n2.nabble.com> Message-ID: <4F16FFEE.2060701@elder.hu> 2012-01-18 15:51 keltez?ssel, obbyone ?rta: > Hi, > > On my freeswitch server, I have created a number 9910 that has to recieve > messages in DTMF. As I call that number with a softphone (xlite) on a PC, > everything works, but as I try to call and send DTMF from an analog phone > plugged into an ATA linksys spa 3000, freeswitch answers, but no DTMF is > recieved. > The linksys is configured to send DTMF in inband mode. > > Do you have any idea how I can handle this issue ? > > Thanks, > > Tony > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-DTMF-tp7200341p7200341.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > For inband DTMF use 'start_dtmf' ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf ) application before the bridge. Why are use inband DTMF? From tonybecq at yahoo.fr Wed Jan 18 20:44:31 2012 From: tonybecq at yahoo.fr (obbyone) Date: Wed, 18 Jan 2012 09:44:31 -0800 (PST) Subject: [Freeswitch-users] Issues with Linksys ATA and DTMF In-Reply-To: <4F16FFEE.2060701@elder.hu> References: <1326898273671-7200341.post@n2.nabble.com> <4F16FFEE.2060701@elder.hu> Message-ID: <1326908671872-7200932.post@n2.nabble.com> Inband is what I've been told to work best. I've tried AVT (rfc2833) and it has the same behaviour. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-DTMF-tp7200341p7200932.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jkomar at jbox.ca Wed Jan 18 21:14:19 2012 From: jkomar at jbox.ca (Komar, Jason) Date: Wed, 18 Jan 2012 11:14:19 -0700 Subject: [Freeswitch-users] GSMopen and Embedded Combined Devices Message-ID: Hi, I'm thinking about giving GSMopen a try and am wondering if anyone has a good source and recommendation for an embedded combined device for use with it? I have done some web searching, but must not be using the best search terms. Thanks, Jason Komar From anthony.minessale at gmail.com Wed Jan 18 21:28:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 12:28:22 -0600 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: You are confusing the envelope event of type socket data which contains the encapsulated custom event. As I said, we do not support blindly subscribing to every custom events, you must supply every subclass name. On Wed, Jan 18, 2012 at 3:58 AM, Gerald Weber wrote: > I get the same results with your script. (maybe I was not clear in my last > reply and mixed things up, still new to fs)**** > > ** ** > > I was just curious why a plain telnet connect to fs shows ?Event-Name > CUSTOM? AND ?Event-Name SOCKET_DATA?**** > > and fs_cli only shows Event-Name CUSTOM **** > > Seems there is a difference between fs_cli and a plain telnet connect to > fs ?**** > > ** ** > > My real problem:**** > > When I want to receive all CUSTOM events (not being specific to a > Event-Subclass) I don?t receive them using**** > > $sock->sendRecv("event plain custom"); and $sock->recvEvent() in e.g. > get.php**** > > when generating this events using > ESLevent("CUSTOM","CONFIG::AGENT_LIST"); and sendEvent in send.php.**** > > ** ** > > I guess recvEvent is listening for the first Event-Name entry. If it > doesn?t match -> go ahead.**** > > ** ** > > It works using $sock->sendRecv("event plain custom CONFIG::AGENT_LIST");** > ** > > ** ** > > But this means I have to know where the event is injected.**** > > Or am I on the wrong way, again ?**** > > ** ** > > Thx & regards,**** > > gw**** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony > Minessale > *Gesendet:* Dienstag, 17. J?nner 2012 18:11 > > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] PHP ESL Custom Events Event-Name sent > twice**** > > ** ** > > you cannot do *all* custom events with /event plain custom**** > > that command is expecting also a subclass param **** > > ** ** > > /event plain custom CONFIG::AGENT_LIST**** > > > you have made a mistake if you think this unlocks heartbeat etc, probably > you did events plain all and forgot.**** > > ** ** > > When I do it from perl I do not see what you reported, **** > > ** ** > > ** ** > > perl test.pl**** > > [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = > [auth/request]**** > > [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE**** > > Event-Name: SOCKET_DATA**** > > Content-Type: auth/request**** > > ** ** > > ** ** > > [DEBUG] esl.c:1381 esl_send() SEND**** > > auth ClueCon**** > > ** ** > > ** ** > > [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = > [command/reply]**** > > [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK > accepted]**** > > [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE**** > > Event-Name: SOCKET_DATA**** > > Content-Type: command/reply**** > > Reply-Text: +OK accepted**** > > ** ** > > ** ** > > [DEBUG] esl.c:502 esl_sendevent() SEND EVENT**** > > Event-Name: CUSTOM**** > > Event-Subclass: CONFIG::AGENT_LIST**** > > ** ** > > ** ** > > [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = > [command/reply]**** > > [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK]**** > > [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE**** > > Event-Name: SOCKET_DATA**** > > Content-Type: command/reply**** > > Reply-Text: +OK**** > > ** ** > > ** ** > > ** ** > > +OK log level [7]**** > > freeswitch at internal> /event plain custom CONFIG::AGENT_LIST**** > > +OK event listener enabled plain**** > > RECV EVENT**** > > Event-Subclass: CONFIG::AGENT_LIST**** > > Core-UUID: 23c2fe9b-f686-45e6-b43b-a3431d32d3e2**** > > FreeSWITCH-Hostname: deathstar.freeswitch.org**** > > FreeSWITCH-Switchname: DeathSTAR**** > > FreeSWITCH-IPv4: 8.19.97.170**** > > FreeSWITCH-IPv6: ::1**** > > Event-Date-Local: 2012-01-17 11:33:51**** > > Event-Date-GMT: Tue, 17 Jan 2012 17:33:51 GMT**** > > Event-Date-Timestamp: 1326821631120687**** > > Event-Calling-File: mod_event_socket.c**** > > Event-Calling-Function: read_packet**** > > Event-Calling-Line-Number: 1188**** > > Command: sendevent CUSTOM**** > > Event-Name: CUSTOM**** > > ** ** > > ** ** > > freeswitch at internal>**** > > ** ** > > ** ** > > ** ** > > ** ** > > cat test.pl**** > > ** ** > > require ESL;**** > > ** ** > > ** ** > > ESL::eslSetLogLevel(7);**** > > ** ** > > my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");**** > > ** ** > > $e = new ESL::ESLevent("CUSTOM","CONFIG::AGENT_LIST");**** > > $con->sendEvent($e);**** > > ** ** > > ** ** > > ** ** > > ** ** > > On Tue, Jan 17, 2012 at 10:37 AM, Michael Collins > wrote:**** > > ** ** > > On Tue, Jan 17, 2012 at 12:17 AM, Gerald Weber > wrote:**** > > Hi,**** > > thanks for your answers**** > > **** > > @Anthony:**** > > I did a ?make current? mins ago, fs starts with version**** > > FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600)**** > > **** > > @MC**** > > I connect using ./fs_cli ?H 192.168.20.73 ?P 8021**** > > /log 0**** > > /event plain custom**** > > **** > > (additionally I connect to fs using telnet 192.168.20.73 8021 on another > terminal and issued ?event plain all?)**** > > **** > > Starting my php script, fs_cli shows nothing, telnet shows the event:**** > > **** > > Content-Length: 583**** > > Content-Type: text/event-plain**** > > **** > > Event-Subclass: CONFIG%3A%3AAGENT_LIST**** > > Event-Name: SOCKET_DATA**** > > ?**** > > Event-Date-Timestamp: 1326785361276007**** > > ?**** > > Event-Name: CUSTOM**** > > Agents: 2022**** > > ZMQ-Msg-Cnt: 244**** > > **** > > /event plain CUSTOM in fs_cli and restart php script -> no output.**** > > (first event subscription except all doesn?t work ?)**** > > /event plain all in fs_cli and restart php script shows:**** > > > Hmm, I guess it does not work. You can do a filter: > > /event plain all > /filter Event-Name CUSTOM > > In any case I'll have to defer to Tony as to why you're seeing two > different header's named "Event-Name" > > -MC **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/00e868af/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 18 21:33:19 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 12:33:19 -0600 Subject: [Freeswitch-users] choppy sound on voice recording and conference In-Reply-To: References: Message-ID: Is it our commercial G.729 module or something else? We only have control over our own code. On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro wrote: > Hello,**** > > ** ** > > I have a problem with one of my sip trunks. The provider uses g729 with a > ptime of 40ms and all incoming calls that are being recorded or go to the > conference have a big delay and choppy sound, like a mismatch of ptime.*** > * > > ** ** > > I have already tried to put rtp-autofix-timing=false into the profiles and > even changed the inbound-codec?. to g729 at 40i, but nothing helps.**** > > We have done a test with the provider switching for a test call to 20ms > and the sound was perfect.**** > > ** ** > > The issue now is that the provider can?t keep this setting only for us, he > wants to send with 40ms.**** > > ** ** > > Is there a way to configure it on Freeswitch?**** > > ** ** > > Thank you**** > > Anestis**** > > ** ** > > ** ** > > ** ** > > ** ** > > ** ** > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/f482d769/attachment.html From bernard.david.murphy at gmail.com Wed Jan 18 10:17:51 2012 From: bernard.david.murphy at gmail.com (Bernard Murphy) Date: Wed, 18 Jan 2012 07:17:51 +0000 Subject: [Freeswitch-users] mod_sofia displaying on Avaya 9630 as caller id contact Message-ID: Hi, I have a soft phone calling an Avaya 9630 both registered on the same freeswitch and have a problem with the Avaya displaying the inbound contact as 'mod_sofia' instead of the originating extension number. The invite from the originating extension and the invite to the Avaya are both pasted below and you will see theat freeswitch includes mod_sofia as the contact uri. Can this be corrected in anyway so that the Avaya phone displays the incoming extension number ? thanks murph* ORIGINAL INVITE* INVITE sip:1000 at 192.168.2.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.31;rport;branch=z9hG4bKc0a8021f000000234f166fd50000386400000044 Content-Length: 337 Contact: Call-ID: BC96029E-D214-4CBF-BCF5-BD0A260757C1 at 192.168.2.31 Content-Type: application/sdp CSeq: 2 INVITE From: "unknown";tag=1383828710312 Max-Forwards: 70 To: User-Agent: SJphone/1.60.289a (SJ Labs) Proxy-Authorization: Digest username="1001",realm="192.168.2.42",nonce="6ff3bebc-b5c5-446a-8625-e89dcd2dd861",uri=" sip:1000 at 192.168.2.42 ",response="1754cba9b8bc72cfac8c69ff0bee8a21",algorithm="MD5",cnonce="1383887920463",qop="auth",nc="00000001" *INVITE FROM SOFIA* INVITE sip:1000 at 192.168.2.30;avaya-sc-enabled;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.42;rport;branch=z9hG4bKjtc19v5754K6c Max-Forwards: 69 From: "Extension 1001" ;tag=QvNQrZya6vHjS To: Call-ID: f73b031a-bc45-122f-388f-000c291b7d13 CSeq: 23099169 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120115T023659Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 427 X-FS-Support: update_display Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/6ce3d9b2/attachment.html From henrikaagaardsorensen at gmail.com Wed Jan 18 19:15:23 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 18 Jan 2012 17:15:23 +0100 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X Message-ID: I've had a server running only FreeSWITCH for about 6 months now. I have had it turned off for some while. Now, when trying to start FreeSWTICH, I get the errors: Error Creating SIP UA for profile: X where X differs between internal, external and others. When starting my CentOS 6 server I run: service iptables stop service ip6tables stop just to make sure that they aren't blocking anything. I have nothing else running on CentOS. I've tried to log everything from console, warning etc. but there's nothing indicating what the problem is. Can someone help me out? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/38873028/attachment.html From greg at brilliantecho.com Wed Jan 18 20:33:02 2012 From: greg at brilliantecho.com (Greg Millam) Date: Wed, 18 Jan 2012 09:33:02 -0800 Subject: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? Message-ID: > commit c358f67fe4348b8b5209328660aef02d8ffaf15f > Author: Anthony Minessale > Date: ? Tue Jan 17 12:19:23 2012 -0600 > > ? ? eat inbound vid while playing fsv files > > > I didn't have a way to test it setup so i'll need feedback. It works, but now since it's missing the first frame, the resulting video is only partial, showing new bits. Is there a way to send a message to the client to "clear and send a full frame of video" ? > On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam wrote: >> >> Hi folks - >> >> I have a freeswitch dialplan + script that first calls play_fsv to >> play a greeting, then record_fsv to record incoming video to fsv. >> >> It works fine, but there's one issue: Apparently, freeswitch is >> buffering incoming video during play_fsv. When record_fsv is called, >> that buffer is dumped into the .fsv file, resulting in several seconds >> of unneeded video. >> >> Is there a way to empty that incoming video buffer before record_fsv >> begins recording? >> >> Thank you! >> >> - Greg Millam From freeswitchlist at gmail.com Wed Jan 18 20:06:37 2012 From: freeswitchlist at gmail.com (bob smith) Date: Wed, 18 Jan 2012 12:06:37 -0500 Subject: [Freeswitch-users] feature list Message-ID: Hi everyone. Does anyone have a feature list for freeswitch? I have not been able to find a good one anywhere ! Pls let me know ! B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/2b1c9459/attachment.html From henrikaagaardsorensen at gmail.com Wed Jan 18 22:28:17 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 18 Jan 2012 20:28:17 +0100 Subject: [Freeswitch-users] feature list In-Reply-To: References: Message-ID: freeswitch.org -> features (in the top menu): http://wiki.freeswitch.org/wiki/Specsheet On Wed, Jan 18, 2012 at 6:06 PM, bob smith wrote: > Hi everyone. Does anyone have a feature list for freeswitch? I have not > been able to find a good one anywhere ! > > Pls let me know ! > > B > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/3dc69581/attachment-0001.html From anthony.minessale at gmail.com Wed Jan 18 22:41:49 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 13:41:49 -0600 Subject: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? In-Reply-To: References: Message-ID: try this: commit 67f559685f97b273dc5f5d23b69cc85cf90e32b0 Author: Anthony Minessale Date: Wed Jan 18 14:08:55 2012 -0600 req vid refresh in fsv On Wed, Jan 18, 2012 at 11:33 AM, Greg Millam wrote: > > commit c358f67fe4348b8b5209328660aef02d8ffaf15f > > Author: Anthony Minessale > > Date: Tue Jan 17 12:19:23 2012 -0600 > > > > eat inbound vid while playing fsv files > > > > > > I didn't have a way to test it setup so i'll need feedback. > > It works, but now since it's missing the first frame, the resulting > video is only partial, showing new bits. Is there a way to send a > message to the client to "clear and send a full frame of video" ? > > > On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam > wrote: > >> > >> Hi folks - > >> > >> I have a freeswitch dialplan + script that first calls play_fsv to > >> play a greeting, then record_fsv to record incoming video to fsv. > >> > >> It works fine, but there's one issue: Apparently, freeswitch is > >> buffering incoming video during play_fsv. When record_fsv is called, > >> that buffer is dumped into the .fsv file, resulting in several seconds > >> of unneeded video. > >> > >> Is there a way to empty that incoming video buffer before record_fsv > >> begins recording? > >> > >> Thank you! > >> > >> - Greg Millam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/ed1422d2/attachment.html From freeswitch at earthspike.net Wed Jan 18 22:49:18 2012 From: freeswitch at earthspike.net (John) Date: Wed, 18 Jan 2012 19:49:18 +0000 Subject: [Freeswitch-users] choppy sound on voice recording and conference In-Reply-To: References: Message-ID: <4F17223E.4020008@earthspike.net> What operating system/platform are you using? Some people have problems with some Ubuntu kernels. John On 18/01/12 12:47, Anestis Mavro wrote: > > Hello, > > I have a problem with one of my sip trunks. The provider uses g729 > with a ptime of 40ms and all incoming calls that are being recorded or > go to the conference have a big delay and choppy sound, like a > mismatch of ptime. > > I have already tried to put rtp-autofix-timing=false into the profiles > and even changed the inbound-codec.... to g729 at 40i, but nothing helps. > > We have done a test with the provider switching for a test call to > 20ms and the sound was perfect. > > The issue now is that the provider can't keep this setting only for > us, he wants to send with 40ms. > > Is there a way to configure it on Freeswitch? > > Thank you > > Anestis > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/49d841d8/attachment.html From brian at freeswitch.org Wed Jan 18 22:58:49 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Jan 2012 13:58:49 -0600 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: Message-ID: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> 'sofia loglevel all 9' 'sofia profile xxx start' watch the error log. /b On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: > I've had a server running only FreeSWITCH for about 6 months now. I have > had it turned off for some while. > > Now, when trying to start FreeSWTICH, I get the errors: > Error Creating SIP UA for profile: X where X differs between internal, > external and others. > > When starting my CentOS 6 server I run: > service iptables stop > service ip6tables stop > just to make sure that they aren't blocking anything. > > I have nothing else running on CentOS. > > I've tried to log everything from console, warning etc. but there's nothing > indicating what the problem is. > > Can someone help me out? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/b4f0f99c/attachment.html From errotan at elder.hu Wed Jan 18 22:59:58 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Wed, 18 Jan 2012 20:59:58 +0100 Subject: [Freeswitch-users] mod_sofia displaying on Avaya 9630 as caller id contact In-Reply-To: References: Message-ID: <4F1724BE.20204@elder.hu> User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120115T023659Z 1.0.6 is very old please try the latest git HEAD. 2012-01-18 08:17 keltez?ssel, Bernard Murphy ?rta: > Hi, > > I have a soft phone calling an Avaya 9630 both registered on the same > freeswitch and have a problem with the Avaya displaying the inbound > contact as 'mod_sofia' instead of the originating extension number. > The invite from the originating extension and the invite to the Avaya > are both pasted below and you will see theat freeswitch includes > mod_sofia as the contact uri. Can this be corrected in anyway so that > the Avaya phone displays the incoming extension number ? > > thanks > murph* > > ORIGINAL INVITE* > INVITE sip:1000 at 192.168.2.42 SIP/2.0 > Via: SIP/2.0/UDP > 192.168.2.31;rport;branch=z9hG4bKc0a8021f000000234f166fd50000386400000044 > Content-Length: 337 > Contact: > > Call-ID: BC96029E-D214-4CBF-BCF5-BD0A260757C1 at 192.168.2.31 > > Content-Type: application/sdp > CSeq: 2 INVITE > From: "unknown" >;tag=1383828710312 > Max-Forwards: 70 > To: > > User-Agent: SJphone/1.60.289a (SJ Labs) > Proxy-Authorization: Digest > username="1001",realm="192.168.2.42",nonce="6ff3bebc-b5c5-446a-8625-e89dcd2dd861",uri="sip:1000 at 192.168.2.42 > ",response="1754cba9b8bc72cfac8c69ff0bee8a21",algorithm="MD5",cnonce="1383887920463",qop="auth",nc="00000001" > > *INVITE FROM SOFIA* > INVITE sip:1000 at 192.168.2.30 > ;avaya-sc-enabled;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.42;rport;branch=z9hG4bKjtc19v5754K6c > Max-Forwards: 69 > From: "Extension 1001" >;tag=QvNQrZya6vHjS > To: ;avaya-sc-enabled;transport=udp> > Call-ID: f73b031a-bc45-122f-388f-000c291b7d13 > CSeq: 23099169 INVITE > Contact: > > User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120115T023659Z > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Allow-Events: talk, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 427 > X-FS-Support: update_display > Remote-Party-ID: "Extension 1001" >;party=calling;screen=yes;privacy=off > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/faf62919/attachment-0001.html From devel at omninet.eu Wed Jan 18 23:15:49 2012 From: devel at omninet.eu (Anestis Mavro) Date: Wed, 18 Jan 2012 22:15:49 +0200 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: <4F17223E.4020008@earthspike.net> References: <4F17223E.4020008@earthspike.net> Message-ID: It is CentOS, fresh installation 6 months ago, runs only FS (CPU at 3% :-) ) thanks _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of John Sent: Wednesday, January 18, 2012 9:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] choppy sound on voice recording andconference What operating system/platform are you using? Some people have problems with some Ubuntu kernels. John On 18/01/12 12:47, Anestis Mavro wrote: Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729 at 40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/cc83a140/attachment.html From freeswitch at earthspike.net Wed Jan 18 23:27:46 2012 From: freeswitch at earthspike.net (John) Date: Wed, 18 Jan 2012 20:27:46 +0000 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: References: <4F17223E.4020008@earthspike.net> Message-ID: <4F172B42.6040701@earthspike.net> CentOS usually comes with the kernel timer set to 1000Hz, but if it has been set to 100Hz (eg because it is on a virtualised platform) then this can cause choppy sound with mismatched ptimes. If it's not that, then I have no better ideas. John On 18/01/12 20:15, Anestis Mavro wrote: > > It is CentOS, fresh installation 6 months ago, runs only FS (CPU at 3% J ) > > thanks > > ------------------------------------------------------------------------ > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *John > *Sent:* Wednesday, January 18, 2012 9:49 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] choppy sound on voice recording > andconference > > What operating system/platform are you using? Some people have > problems with some Ubuntu kernels. > > John > > On 18/01/12 12:47, Anestis Mavro wrote: > > Hello, > > I have a problem with one of my sip trunks. The provider uses g729 > with a ptime of 40ms and all incoming calls that are being recorded or > go to the conference have a big delay and choppy sound, like a > mismatch of ptime. > > I have already tried to put rtp-autofix-timing=false into the profiles > and even changed the inbound-codec.... to g729 at 40i, but nothing helps. > > We have done a test with the provider switching for a test call to > 20ms and the sound was perfect. > > The issue now is that the provider can't keep this setting only for > us, he wants to send with 40ms. > > Is there a way to configure it on Freeswitch? > > Thank you > > Anestis > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/13611c41/attachment-0001.html From anestis at omninet.gr Wed Jan 18 22:58:50 2012 From: anestis at omninet.gr (Anestis Mavrofillidis) Date: Wed, 18 Jan 2012 21:58:50 +0200 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: References: Message-ID: Yes, it is the commercial module. I have bought a few licenses for this server and I am ready to buy a few for a second node, but this problem is now critical. even voicemail is not usable! You hear the announcement, but the recorded message is very bad. (just to remind you: only the incoming calls on this trunk are affected) Thanks Anestis _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, January 18, 2012 8:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] choppy sound on voice recording andconference Is it our commercial G.729 module or something else? We only have control over our own code. On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro wrote: Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729 at 40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/a7168415/attachment.html From anthony.minessale at gmail.com Wed Jan 18 23:41:17 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 14:41:17 -0600 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: References: Message-ID: We can lab it up and see if we can replicate it. On Wed, Jan 18, 2012 at 1:58 PM, Anestis Mavrofillidis wrote: > ** > > ** ** > > Yes, it is the commercial module. I have bought a few licenses for this > server and I am ready to buy a few for a second node, but this problem is > now critical? even voicemail is not usable! You hear the announcement, but > the recorded message is very bad. (just to remind you: only the incoming > calls on this trunk are affected)**** > > ** ** > > Thanks**** > > Anestis**** > > ** ** > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, January 18, 2012 8:33 PM > > *To:* **FreeSWITCH Users Help** > *Subject:* Re: [Freeswitch-users] choppy sound on voice recording > andconference > **** > > ** ** > > Is it our commercial G.729 module or something else? We only have control > over our own code.**** > > On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro wrote:** > ** > > Hello,**** > > **** > > I have a problem with one of my sip trunks. The provider uses g729 with a > ptime of 40ms and all incoming calls that are being recorded or go to the > conference have a big delay and choppy sound, like a mismatch of ptime.*** > * > > **** > > I have already tried to put rtp-autofix-timing=false into the profiles and > even changed the inbound-codec?. to g729 at 40i, but nothing helps.**** > > We have done a test with the provider switching for a test call to 20ms > and the sound was perfect.**** > > **** > > The issue now is that the provider can?t keep this setting only for us, he > wants to send with 40ms.**** > > **** > > Is there a way to configure it on Freeswitch?**** > > **** > > Thank you**** > > Anestis**** > > **** > > **** > > **** > > **** > > **** > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > ** ** > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________**** > > ** ** > > The message was checked by ESET NOD32 Antivirus.**** > > ** ** > > http://www.eset.com**** > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/7a468083/attachment-0001.html From henrikaagaardsorensen at gmail.com Wed Jan 18 23:41:35 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 18 Jan 2012 21:41:35 +0100 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: I've just completely reinstalled CentOS and FreeSwitch and still get the problem. I've tried what you've done and this is my log (freeswitch.log): 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA for profile: external On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: > 'sofia loglevel all 9' > > 'sofia profile xxx start' > > watch the error log. > > /b > > On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: > > I've had a server running only FreeSWITCH for about 6 months now. I have > had it turned off for some while. > > Now, when trying to start FreeSWTICH, I get the errors: > Error Creating SIP UA for profile: X where X differs between internal, > external and others. > > When starting my CentOS 6 server I run: > service iptables stop > service ip6tables stop > just to make sure that they aren't blocking anything. > > I have nothing else running on CentOS. > > I've tried to log everything from console, warning etc. but there's nothing > indicating what the problem is. > > Can someone help me out? > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/0218acf8/attachment.html From anthony.minessale at gmail.com Wed Jan 18 23:47:18 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 14:47:18 -0600 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: Either its trying to bind to an ip that is not actually on the interface, or the address is already in use. 2012/1/18 Henrik Aagaard S?rensen > I've just completely reinstalled CentOS and FreeSwitch and still get the > problem. > > I've tried what you've done and this is my log (freeswitch.log): > 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA for > profile: external > > > On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: > >> 'sofia loglevel all 9' >> >> 'sofia profile xxx start' >> >> watch the error log. >> >> /b >> >> On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: >> >> I've had a server running only FreeSWITCH for about 6 months now. I have >> had it turned off for some while. >> >> Now, when trying to start FreeSWTICH, I get the errors: >> Error Creating SIP UA for profile: X where X differs between internal, >> external and others. >> >> When starting my CentOS 6 server I run: >> service iptables stop >> service ip6tables stop >> just to make sure that they aren't blocking anything. >> >> I have nothing else running on CentOS. >> >> I've tried to log everything from console, warning etc. but there's >> nothing >> indicating what the problem is. >> >> Can someone help me out? >> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/2f4307b0/attachment.html From devel at omninet.eu Thu Jan 19 00:02:52 2012 From: devel at omninet.eu (Anestis Mavro) Date: Wed, 18 Jan 2012 23:02:52 +0200 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: References: Message-ID: <958351C61E3548F7BCA489781FC3EDF3@omni1.local> How can I help on this? I could offer access to the second node, which can receive calls from the same provider. I did not check its behavior, since it has just been installed; and this is a Debian installation. It is not in production. Should we continue talking "off list" ? _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, January 18, 2012 10:41 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] choppy sound on voice recording andconference We can lab it up and see if we can replicate it. On Wed, Jan 18, 2012 at 1:58 PM, Anestis Mavrofillidis wrote: Yes, it is the commercial module. I have bought a few licenses for this server and I am ready to buy a few for a second node, but this problem is now critical. even voicemail is not usable! You hear the announcement, but the recorded message is very bad. (just to remind you: only the incoming calls on this trunk are affected) Thanks Anestis _____ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, January 18, 2012 8:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] choppy sound on voice recording andconference Is it our commercial G.729 module or something else? We only have control over our own code. On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro wrote: Hello, I have a problem with one of my sip trunks. The provider uses g729 with a ptime of 40ms and all incoming calls that are being recorded or go to the conference have a big delay and choppy sound, like a mismatch of ptime. I have already tried to put rtp-autofix-timing=false into the profiles and even changed the inbound-codec.. to g729 at 40i, but nothing helps. We have done a test with the provider switching for a test call to 20ms and the sound was perfect. The issue now is that the provider can't keep this setting only for us, he wants to send with 40ms. Is there a way to configure it on Freeswitch? Thank you Anestis __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/143f9507/attachment-0001.html From henrikaagaardsorensen at gmail.com Thu Jan 19 00:06:19 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 18 Jan 2012 22:06:19 +0100 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: Could it be something with IPv6? On Wed, Jan 18, 2012 at 9:47 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Either its trying to bind to an ip that is not actually on the interface, > or the address is already in use. > > > 2012/1/18 Henrik Aagaard S?rensen > >> I've just completely reinstalled CentOS and FreeSwitch and still get the >> problem. >> >> I've tried what you've done and this is my log (freeswitch.log): >> 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA for >> profile: external >> >> >> On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: >> >>> 'sofia loglevel all 9' >>> >>> 'sofia profile xxx start' >>> >>> watch the error log. >>> >>> /b >>> >>> On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: >>> >>> I've had a server running only FreeSWITCH for about 6 months now. I have >>> had it turned off for some while. >>> >>> Now, when trying to start FreeSWTICH, I get the errors: >>> Error Creating SIP UA for profile: X where X differs between internal, >>> external and others. >>> >>> When starting my CentOS 6 server I run: >>> service iptables stop >>> service ip6tables stop >>> just to make sure that they aren't blocking anything. >>> >>> I have nothing else running on CentOS. >>> >>> I've tried to log everything from console, warning etc. but there's >>> nothing >>> indicating what the problem is. >>> >>> Can someone help me out? >>> >>> >>> -- >>> Brian West >>> FreeSWITCH Solutions, LLC >>> Phone: +1 (918) 420-9266 >>> Fax: +1 (918) 420-9267 >>> brian at freeswitch.org >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/9aab43fd/attachment.html From anthony.minessale at gmail.com Thu Jan 19 00:07:28 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 15:07:28 -0600 Subject: [Freeswitch-users] choppy sound on voice recording andconference In-Reply-To: <958351C61E3548F7BCA489781FC3EDF3@omni1.local> References: <958351C61E3548F7BCA489781FC3EDF3@omni1.local> Message-ID: I'll see if we can reproduce it and get back to you. On Wed, Jan 18, 2012 at 3:02 PM, Anestis Mavro wrote: > ** > > How can I help on this? I could offer access to the second node, which can > receive calls from the same provider. I did not check its behavior, since > it has just been installed; and this is a Debian installation.**** > > It is not in production.**** > > ** ** > > Should we continue talking ?off list? ?**** > > ** ** > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, January 18, 2012 10:41 PM > > *To:* **FreeSWITCH Users Help** > *Subject:* Re: [Freeswitch-users] choppy sound on voice recording > andconference > **** > > ** ** > > We can lab it up and see if we can replicate it.**** > > ** ** > > On Wed, Jan 18, 2012 at 1:58 PM, Anestis Mavrofillidis > wrote:**** > > **** > > Yes, it is the commercial module. I have bought a few licenses for this > server and I am ready to buy a few for a second node, but this problem is > now critical? even voicemail is not usable! You hear the announcement, but > the recorded message is very bad. (just to remind you: only the incoming > calls on this trunk are affected)**** > > **** > > Thanks**** > > Anestis**** > > **** > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, January 18, 2012 8:33 PM**** > > > *To:* **FreeSWITCH Users Help** > *Subject:* Re: [Freeswitch-users] choppy sound on voice recording > andconference**** > > **** > > Is it our commercial G.729 module or something else? We only have control > over our own code.**** > > On Wed, Jan 18, 2012 at 6:47 AM, Anestis Mavro wrote:** > ** > > Hello,**** > > **** > > I have a problem with one of my sip trunks. The provider uses g729 with a > ptime of 40ms and all incoming calls that are being recorded or go to the > conference have a big delay and choppy sound, like a mismatch of ptime.*** > * > > **** > > I have already tried to put rtp-autofix-timing=false into the profiles and > even changed the inbound-codec?. to g729 at 40i, but nothing helps.**** > > We have done a test with the provider switching for a test call to 20ms > and the sound was perfect.**** > > **** > > The issue now is that the provider can?t keep this setting only for us, he > wants to send with 40ms.**** > > **** > > Is there a way to configure it on Freeswitch?**** > > **** > > Thank you**** > > Anestis**** > > **** > > **** > > **** > > **** > > **** > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > **** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > **** > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________**** > > **** > > The message was checked by ESET NOD32 Antivirus.**** > > **** > > http://www.eset.com**** > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > ** ** > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________**** > > ** ** > > The message was checked by ESET NOD32 Antivirus.**** > > ** ** > > http://www.eset.com**** > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/21425ff8/attachment-0001.html From henrikaagaardsorensen at gmail.com Thu Jan 19 00:10:14 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 18 Jan 2012 22:10:14 +0100 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: This is, by the way, my netstat -tunlp: tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN 906/sshd tcp 0 0 127.0.0.1:25 0.0.0.0:* LISTEN 1122/master tcp 0 0 0.0.0.0:3306 0.0.0.0:* LISTEN 1031/mysqld tcp 0 0 :::80 :::* LISTEN 1132/httpd tcp 0 0 :::22 :::* LISTEN 906/sshd tcp 0 0 ::1:25 :::* LISTEN 1122/master 2012/1/18 Henrik Aagaard S?rensen > Could it be something with IPv6? > > > On Wed, Jan 18, 2012 at 9:47 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Either its trying to bind to an ip that is not actually on the interface, >> or the address is already in use. >> >> >> 2012/1/18 Henrik Aagaard S?rensen >> >>> I've just completely reinstalled CentOS and FreeSwitch and still get the >>> problem. >>> >>> I've tried what you've done and this is my log (freeswitch.log): >>> 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA for >>> profile: external >>> >>> >>> On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: >>> >>>> 'sofia loglevel all 9' >>>> >>>> 'sofia profile xxx start' >>>> >>>> watch the error log. >>>> >>>> /b >>>> >>>> On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: >>>> >>>> I've had a server running only FreeSWITCH for about 6 months now. I have >>>> had it turned off for some while. >>>> >>>> Now, when trying to start FreeSWTICH, I get the errors: >>>> Error Creating SIP UA for profile: X where X differs between internal, >>>> external and others. >>>> >>>> When starting my CentOS 6 server I run: >>>> service iptables stop >>>> service ip6tables stop >>>> just to make sure that they aren't blocking anything. >>>> >>>> I have nothing else running on CentOS. >>>> >>>> I've tried to log everything from console, warning etc. but there's >>>> nothing >>>> indicating what the problem is. >>>> >>>> Can someone help me out? >>>> >>>> >>>> -- >>>> Brian West >>>> FreeSWITCH Solutions, LLC >>>> Phone: +1 (918) 420-9266 >>>> Fax: +1 (918) 420-9267 >>>> brian at freeswitch.org >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/8e95b0fa/attachment.html From alexis.mailinglist at de-bruyn.fr Thu Jan 19 00:48:43 2012 From: alexis.mailinglist at de-bruyn.fr (Alexis de BRUYN [Mailinglists]) Date: Wed, 18 Jan 2012 22:48:43 +0100 Subject: [Freeswitch-users] Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) In-Reply-To: <4F164531.3030201@exemail.com.au> References: <4F15E25E.8030405@de-bruyn.fr> <4F164531.3030201@exemail.com.au> Message-ID: <4F173E3B.9020506@de-bruyn.fr> Hi Ryan, Thank you for your answer, your server is behind NAT and your client too ? Regards, On 18.01.2012 05:06, Ryan How wrote: > Hi, > > I set mine up for the external phones to log into the internal profile > and it just worked (port 5060). But it might depend on the type of NAT > or something, I'm by no means an expert on the subject. > > Probably worth securing it better than the default config once you get > it going if it works on the internal profile. > > Ryan > > > On 18/01/2012 5:04 AM, Alexis de BRUYN [Mailinglists] wrote: >> Hi Everybody, >> >> I am trying to get a conversation between two natted clients and a >> natted freeswitch server. I am playing with a fresh install, the >> external profile and default directory (1000& 1001 users). >> >> I can see that my clients are registered on the server logs (on the 5080 >> port). All others needed ports are opened too. >> >> I have followed the instructions from the wiki to get a double nat >> configuration, but my client phone are not ringing (and I cannot hear >> anything). >> >> All is working fine with defaults in a non nat configuration (with >> internal profile). >> >> Is there anything special to set up in the directory / user configuration ? >> >> Can anyone give me some hints with his natted configuration ? I am stuck. >> >> Thanks for your help. >> >> Regards, >> >> -------- Original Message -------- >> Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile >> Date: Mon, 16 Jan 2012 23:37:20 +0100 >> From: Alexis de BRUYN [Mailinglists] >> Reply-To: FreeSWITCH Users Help >> To: freeswitch-users at lists.freeswitch.org >> >> Hi Everybody, >> >> I am trying to use FreeSwitch in a (double) NAT Configuration from a >> fresh snapshot install (Debian Squeeze, server outside from my client >> LAN) with default directory for users. >> >> When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the >> callee, the phone doesn't ring and automatically hangup, I see in the >> console : >> >> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot >> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] >> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot >> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] >> >> However, 1000 and 1001 are registered (from the same LAN) : >> >> sofia status profile external >> >> ================================================================================================= >> Name external >> Domain Name N/A >> Auto-NAT false >> DBName sofia_reg_external >> Pres Hosts >> Dialplan XML >> Context public >> Challenge Realm auto_to >> RTP-IP 192.168.1.6 >> SIP-IP 192.168.1.6 >> URL sip:mod_sofia at 192.168.1.6:5080 >> BIND-URL sip:mod_sofia at 192.168.1.6:5080 >> HOLD-MUSIC local_stream://moh >> OUTBOUND-PROXY N/A >> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM >> CODECS OUT PCMU,PCMA,GSM >> TEL-EVENT 101 >> DTMF-MODE rfc2833 >> CNG 13 >> SESSION-TO 0 >> MAX-DIALOG 0 >> NOMEDIA false >> LATE-NEG false >> PROXY-MEDIA false >> AGGRESSIVENAT false >> STUN-ENABLED true >> STUN-AUTO-DISABLE false >> CALLS-IN 3 >> FAILED-CALLS-IN 0 >> CALLS-OUT 0 >> FAILED-CALLS-OUT 0 >> >> Registrations: >> ================================================================================================= >> Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm >> User: 1001 at X.Y.Z.T >> Contact: "Freeswitch" >> Agent: Bria iOS 2.0.0 >> Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) >> Host: phone >> IP: A.B.C.D >> Port: 50193 >> Auth-User: 1001 >> Auth-Realm: X.Y.Z.T >> MWI-Account: 1001 at X.Y.Z.T >> >> Call-ID: 3c27b7424837-bsfi877ujb2n >> User: 1000 at X.Y.Z.T >> Contact: "freeswitch" >> Agent: snom300/8.4.32 >> Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) >> Host: phone >> IP: A.B.C.D >> Port: 62061 >> Auth-User: 1000 >> Auth-Realm: X.Y.Z.T >> MWI-Account: 1000 at X.Y.Z.T >> >> Total items returned: 2 >> ================================================================================================= >> >> All necessary ports are opened/forwarded on the server. >> >> I See on the 1000 configuration that this is the local ip address which >> is set as contact. Is there any other setups to do in the directory ? >> Or other parameters in the external profile ? >> >> Thanks for your help ! >> >> Regards, >> >> -- >> Alexis de BRUYN >> Mail : alexis.mailinglist at de-bruyn.fr >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- -- Alexis de BRUYN Mail : alexis.mailinglist at de-bruyn.fr From paul at cupis.co.uk Thu Jan 19 00:53:51 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Wed, 18 Jan 2012 21:53:51 +0000 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: <4F173F6F.8090506@cupis.co.uk> On 18/01/12 21:10, Henrik Aagaard S?rensen wrote: > This is, by the way, my netstat -tunlp: Do you have an IP address hardcoded in the profile? Is that IP address active on the box? In the relevant sip_profile XML configuration: for example. Regards, From curriegrad2004 at gmail.com Thu Jan 19 01:48:17 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 18 Jan 2012 14:48:17 -0800 Subject: [Freeswitch-users] Issues with Linksys ATA and DTMF In-Reply-To: <1326908671872-7200932.post@n2.nabble.com> References: <1326898273671-7200341.post@n2.nabble.com> <4F16FFEE.2060701@elder.hu> <1326908671872-7200932.post@n2.nabble.com> Message-ID: RFC2833 has to be configured on the sofia profile wherever your ATA is registered to. The ATA you have just simply converts the inband tones back to RFC2833 complaint DTMF signalling information, so why have FS to do the job when the ATA can do a sufficient job for that type of conversion? On Wed, Jan 18, 2012 at 9:44 AM, obbyone wrote: > Inband is what I've been told to work best. I've tried AVT (rfc2833) and it > has the same behaviour. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-DTMF-tp7200341p7200932.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From yungwei at resolvity.com Thu Jan 19 03:47:37 2012 From: yungwei at resolvity.com (Yungwei Chen) Date: Wed, 18 Jan 2012 19:47:37 -0500 Subject: [Freeswitch-users] Unable to export a variable to b leg (callcenter) Message-ID: <33095823FD21DF429B481B5163264B795E7217729F@VMBX102.ihostexchange.net> Hi, With the extension below, I am able to export caller_id_name to b leg without problems. However, exporting another variable doesn't seem to work because I don't see it in the corresponding CDRs (cdr.variables). It looks like only known variables are exportable. Is there way to export a custom variable in this case? Thanks. From rhow at exemail.com.au Thu Jan 19 04:23:23 2012 From: rhow at exemail.com.au (Ryan How) Date: Thu, 19 Jan 2012 09:23:23 +0800 Subject: [Freeswitch-users] Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) In-Reply-To: <4F173E3B.9020506@de-bruyn.fr> References: <4F15E25E.8030405@de-bruyn.fr> <4F164531.3030201@exemail.com.au> <4F173E3B.9020506@de-bruyn.fr> Message-ID: <7A910C39-F18D-44B9-8220-39B6FD5141E0@exemail.com.au> Yes. Both the server and client are behind a NAT. I haven't tried 2 clients behind the same NAT. On 19/01/2012, at 5:48 AM, "Alexis de BRUYN [Mailinglists]" wrote: > Hi Ryan, > > Thank you for your answer, your server is behind NAT and your client too ? > > Regards, > > On 18.01.2012 05:06, Ryan How wrote: >> Hi, >> >> I set mine up for the external phones to log into the internal profile >> and it just worked (port 5060). But it might depend on the type of NAT >> or something, I'm by no means an expert on the subject. >> >> Probably worth securing it better than the default config once you get >> it going if it works on the internal profile. >> >> Ryan >> >> >> On 18/01/2012 5:04 AM, Alexis de BRUYN [Mailinglists] wrote: >>> Hi Everybody, >>> >>> I am trying to get a conversation between two natted clients and a >>> natted freeswitch server. I am playing with a fresh install, the >>> external profile and default directory (1000& 1001 users). >>> >>> I can see that my clients are registered on the server logs (on the 5080 >>> port). All others needed ports are opened too. >>> >>> I have followed the instructions from the wiki to get a double nat >>> configuration, but my client phone are not ringing (and I cannot hear >>> anything). >>> >>> All is working fine with defaults in a non nat configuration (with >>> internal profile). >>> >>> Is there anything special to set up in the directory / user configuration ? >>> >>> Can anyone give me some hints with his natted configuration ? I am stuck. >>> >>> Thanks for your help. >>> >>> Regards, >>> >>> -------- Original Message -------- >>> Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile >>> Date: Mon, 16 Jan 2012 23:37:20 +0100 >>> From: Alexis de BRUYN [Mailinglists] >>> Reply-To: FreeSWITCH Users Help >>> To: freeswitch-users at lists.freeswitch.org >>> >>> Hi Everybody, >>> >>> I am trying to use FreeSwitch in a (double) NAT Configuration from a >>> fresh snapshot install (Debian Squeeze, server outside from my client >>> LAN) with default directory for users. >>> >>> When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the >>> callee, the phone doesn't ring and automatically hangup, I see in the >>> console : >>> >>> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot >>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] >>> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot >>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] >>> >>> However, 1000 and 1001 are registered (from the same LAN) : >>> >>> sofia status profile external >>> >>> ================================================================================================= >>> Name external >>> Domain Name N/A >>> Auto-NAT false >>> DBName sofia_reg_external >>> Pres Hosts >>> Dialplan XML >>> Context public >>> Challenge Realm auto_to >>> RTP-IP 192.168.1.6 >>> SIP-IP 192.168.1.6 >>> URL sip:mod_sofia at 192.168.1.6:5080 >>> BIND-URL sip:mod_sofia at 192.168.1.6:5080 >>> HOLD-MUSIC local_stream://moh >>> OUTBOUND-PROXY N/A >>> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM >>> CODECS OUT PCMU,PCMA,GSM >>> TEL-EVENT 101 >>> DTMF-MODE rfc2833 >>> CNG 13 >>> SESSION-TO 0 >>> MAX-DIALOG 0 >>> NOMEDIA false >>> LATE-NEG false >>> PROXY-MEDIA false >>> AGGRESSIVENAT false >>> STUN-ENABLED true >>> STUN-AUTO-DISABLE false >>> CALLS-IN 3 >>> FAILED-CALLS-IN 0 >>> CALLS-OUT 0 >>> FAILED-CALLS-OUT 0 >>> >>> Registrations: >>> ================================================================================================= >>> Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm >>> User: 1001 at X.Y.Z.T >>> Contact: "Freeswitch" >>> Agent: Bria iOS 2.0.0 >>> Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) EXPSECS(934) >>> Host: phone >>> IP: A.B.C.D >>> Port: 50193 >>> Auth-User: 1001 >>> Auth-Realm: X.Y.Z.T >>> MWI-Account: 1001 at X.Y.Z.T >>> >>> Call-ID: 3c27b7424837-bsfi877ujb2n >>> User: 1000 at X.Y.Z.T >>> Contact: "freeswitch" >>> Agent: snom300/8.4.32 >>> Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) EXPSECS(3658) >>> Host: phone >>> IP: A.B.C.D >>> Port: 62061 >>> Auth-User: 1000 >>> Auth-Realm: X.Y.Z.T >>> MWI-Account: 1000 at X.Y.Z.T >>> >>> Total items returned: 2 >>> ================================================================================================= >>> >>> All necessary ports are opened/forwarded on the server. >>> >>> I See on the 1000 configuration that this is the local ip address which >>> is set as contact. Is there any other setups to do in the directory ? >>> Or other parameters in the external profile ? >>> >>> Thanks for your help ! >>> >>> Regards, >>> >>> -- >>> Alexis de BRUYN >>> Mail : alexis.mailinglist at de-bruyn.fr >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > -- > Alexis de BRUYN > Mail : alexis.mailinglist at de-bruyn.fr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Thu Jan 19 04:53:08 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 18 Jan 2012 19:53:08 -0600 Subject: [Freeswitch-users] feature list In-Reply-To: References: Message-ID: we should probably update that =p there is also the changelog file in the build root which is pretty big but sort of up to date. 2012/1/18 Henrik Aagaard S?rensen > freeswitch.org -> features (in the top menu): > http://wiki.freeswitch.org/wiki/Specsheet > > > On Wed, Jan 18, 2012 at 6:06 PM, bob smith wrote: > >> Hi everyone. Does anyone have a feature list for freeswitch? I have not >> been able to find a good one anywhere ! >> >> Pls let me know ! >> >> B >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120118/4731a0d9/attachment.html From anton.jugatsu at gmail.com Thu Jan 19 07:49:56 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 19 Jan 2012 08:49:56 +0400 Subject: [Freeswitch-users] Freeswitch & double NAT Configuration (Was: USER_NOT_REGISTERED with external profile) In-Reply-To: <7A910C39-F18D-44B9-8220-39B6FD5141E0@exemail.com.au> References: <4F15E25E.8030405@de-bruyn.fr> <4F164531.3030201@exemail.com.au> <4F173E3B.9020506@de-bruyn.fr> <7A910C39-F18D-44B9-8220-39B6FD5141E0@exemail.com.au> Message-ID: Try to dump with tcpdump or ngrep and analyse using wireshark. 2012/1/19 Ryan How > Yes. Both the server and client are behind a NAT. I haven't tried 2 > clients behind the same NAT. > > > > On 19/01/2012, at 5:48 AM, "Alexis de BRUYN [Mailinglists]" < > alexis.mailinglist at de-bruyn.fr> wrote: > > > Hi Ryan, > > > > Thank you for your answer, your server is behind NAT and your client too > ? > > > > Regards, > > > > On 18.01.2012 05:06, Ryan How wrote: > >> Hi, > >> > >> I set mine up for the external phones to log into the internal profile > >> and it just worked (port 5060). But it might depend on the type of NAT > >> or something, I'm by no means an expert on the subject. > >> > >> Probably worth securing it better than the default config once you get > >> it going if it works on the internal profile. > >> > >> Ryan > >> > >> > >> On 18/01/2012 5:04 AM, Alexis de BRUYN [Mailinglists] wrote: > >>> Hi Everybody, > >>> > >>> I am trying to get a conversation between two natted clients and a > >>> natted freeswitch server. I am playing with a fresh install, the > >>> external profile and default directory (1000& 1001 users). > >>> > >>> I can see that my clients are registered on the server logs (on the > 5080 > >>> port). All others needed ports are opened too. > >>> > >>> I have followed the instructions from the wiki to get a double nat > >>> configuration, but my client phone are not ringing (and I cannot hear > >>> anything). > >>> > >>> All is working fine with defaults in a non nat configuration (with > >>> internal profile). > >>> > >>> Is there anything special to set up in the directory / user > configuration ? > >>> > >>> Can anyone give me some hints with his natted configuration ? I am > stuck. > >>> > >>> Thanks for your help. > >>> > >>> Regards, > >>> > >>> -------- Original Message -------- > >>> Subject: [Freeswitch-users] USER_NOT_REGISTERED with external profile > >>> Date: Mon, 16 Jan 2012 23:37:20 +0100 > >>> From: Alexis de BRUYN [Mailinglists] > >>> Reply-To: FreeSWITCH Users Help > >>> To: freeswitch-users at lists.freeswitch.org > >>> > >>> Hi Everybody, > >>> > >>> I am trying to use FreeSwitch in a (double) NAT Configuration from a > >>> fresh snapshot install (Debian Squeeze, server outside from my client > >>> LAN) with default directory for users. > >>> > >>> When I call 1001 from 1000 (or 1000 from 1001) , I cannot hear the > >>> callee, the phone doesn't ring and automatically hangup, I see in the > >>> console : > >>> > >>> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot > >>> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > >>> 2012-01-16 22:59:37.691663 [ERR] switch_ivr_originate.c:2447 Cannot > >>> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] > >>> > >>> However, 1000 and 1001 are registered (from the same LAN) : > >>> > >>> sofia status profile external > >>> > >>> > ================================================================================================= > >>> Name external > >>> Domain Name N/A > >>> Auto-NAT false > >>> DBName sofia_reg_external > >>> Pres Hosts > >>> Dialplan XML > >>> Context public > >>> Challenge Realm auto_to > >>> RTP-IP 192.168.1.6 > >>> SIP-IP 192.168.1.6 > >>> URL sip:mod_sofia at 192.168.1.6:5080 > >>> BIND-URL sip:mod_sofia at 192.168.1.6:5080 > >>> HOLD-MUSIC local_stream://moh > >>> OUTBOUND-PROXY N/A > >>> CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > >>> CODECS OUT PCMU,PCMA,GSM > >>> TEL-EVENT 101 > >>> DTMF-MODE rfc2833 > >>> CNG 13 > >>> SESSION-TO 0 > >>> MAX-DIALOG 0 > >>> NOMEDIA false > >>> LATE-NEG false > >>> PROXY-MEDIA false > >>> AGGRESSIVENAT false > >>> STUN-ENABLED true > >>> STUN-AUTO-DISABLE false > >>> CALLS-IN 3 > >>> FAILED-CALLS-IN 0 > >>> CALLS-OUT 0 > >>> FAILED-CALLS-OUT 0 > >>> > >>> Registrations: > >>> > ================================================================================================= > >>> Call-ID: aU.1ZlEZkW7nlHpwp9ig1lt7A3weNnwm > >>> User: 1001 at X.Y.Z.T > >>> Contact: "Freeswitch" > >>> Agent: Bria iOS 2.0.0 > >>> Status: Registered(UDP)(unknown) EXP(2012-01-16 23:16:00) > EXPSECS(934) > >>> Host: phone > >>> IP: A.B.C.D > >>> Port: 50193 > >>> Auth-User: 1001 > >>> Auth-Realm: X.Y.Z.T > >>> MWI-Account: 1001 at X.Y.Z.T > >>> > >>> Call-ID: 3c27b7424837-bsfi877ujb2n > >>> User: 1000 at X.Y.Z.T > >>> Contact: "freeswitch" > >>> Agent: snom300/8.4.32 > >>> Status: Registered(UDP)(unknown) EXP(2012-01-17 00:01:24) > EXPSECS(3658) > >>> Host: phone > >>> IP: A.B.C.D > >>> Port: 62061 > >>> Auth-User: 1000 > >>> Auth-Realm: X.Y.Z.T > >>> MWI-Account: 1000 at X.Y.Z.T > >>> > >>> Total items returned: 2 > >>> > ================================================================================================= > >>> > >>> All necessary ports are opened/forwarded on the server. > >>> > >>> I See on the 1000 configuration that this is the local ip address which > >>> is set as contact. Is there any other setups to do in the directory ? > >>> Or other parameters in the external profile ? > >>> > >>> Thanks for your help ! > >>> > >>> Regards, > >>> > >>> -- > >>> Alexis de BRUYN > >>> Mail : alexis.mailinglist at de-bruyn.fr > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > -- > > Alexis de BRUYN > > Mail : alexis.mailinglist at de-bruyn.fr > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/3061c120/attachment-0001.html From anton.jugatsu at gmail.com Thu Jan 19 07:53:54 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 19 Jan 2012 08:53:54 +0400 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: and what about udp, try ss -aup lsof -iudp 2012/1/19 Henrik Aagaard S?rensen > This is, by the way, my netstat -tunlp: > > tcp 0 0 0.0.0.0:22 0.0.0.0:* > LISTEN 906/sshd > tcp 0 0 127.0.0.1:25 0.0.0.0:* > LISTEN 1122/master > tcp 0 0 0.0.0.0:3306 0.0.0.0:* > LISTEN 1031/mysqld > tcp 0 0 :::80 :::* > LISTEN 1132/httpd > tcp 0 0 :::22 :::* > LISTEN 906/sshd > tcp 0 0 ::1:25 :::* > LISTEN 1122/master > > > 2012/1/18 Henrik Aagaard S?rensen > >> Could it be something with IPv6? >> >> >> On Wed, Jan 18, 2012 at 9:47 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Either its trying to bind to an ip that is not actually on the >>> interface, or the address is already in use. >>> >>> >>> 2012/1/18 Henrik Aagaard S?rensen >>> >>>> I've just completely reinstalled CentOS and FreeSwitch and still get >>>> the problem. >>>> >>>> I've tried what you've done and this is my log (freeswitch.log): >>>> 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA >>>> for profile: external >>>> >>>> >>>> On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: >>>> >>>>> 'sofia loglevel all 9' >>>>> >>>>> 'sofia profile xxx start' >>>>> >>>>> watch the error log. >>>>> >>>>> /b >>>>> >>>>> On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: >>>>> >>>>> I've had a server running only FreeSWITCH for about 6 months now. I >>>>> have >>>>> had it turned off for some while. >>>>> >>>>> Now, when trying to start FreeSWTICH, I get the errors: >>>>> Error Creating SIP UA for profile: X where X differs between internal, >>>>> external and others. >>>>> >>>>> When starting my CentOS 6 server I run: >>>>> service iptables stop >>>>> service ip6tables stop >>>>> just to make sure that they aren't blocking anything. >>>>> >>>>> I have nothing else running on CentOS. >>>>> >>>>> I've tried to log everything from console, warning etc. but there's >>>>> nothing >>>>> indicating what the problem is. >>>>> >>>>> Can someone help me out? >>>>> >>>>> >>>>> -- >>>>> Brian West >>>>> FreeSWITCH Solutions, LLC >>>>> Phone: +1 (918) 420-9266 >>>>> Fax: +1 (918) 420-9267 >>>>> brian at freeswitch.org >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/3eb4da3b/attachment.html From tonybecq at yahoo.fr Thu Jan 19 10:38:20 2012 From: tonybecq at yahoo.fr (obbyone) Date: Wed, 18 Jan 2012 23:38:20 -0800 (PST) Subject: [Freeswitch-users] Issues with Linksys ATA and DTMF In-Reply-To: <1326908671872-7200932.post@n2.nabble.com> References: <1326898273671-7200341.post@n2.nabble.com> <4F16FFEE.2060701@elder.hu> <1326908671872-7200932.post@n2.nabble.com> Message-ID: <1326958700591-7202978.post@n2.nabble.com> I've done what you told me and it works fine. Thanks a lot ! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issues-with-Linksys-ATA-and-DTMF-tp7200341p7202978.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mytemike72 at gmail.com Thu Jan 19 10:52:58 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 19 Jan 2012 08:52:58 +0100 Subject: [Freeswitch-users] bridging calls using mod_managed Message-ID: Hi, Is there someone who can explain me why after executing a bridge function in mod.managed, the hangup hook is executed immediately,?and my main code execution continues. And, even stranger, Context.session.bridged() returns false for both legs! This code snippet is the example, it skips the while loop (check for bridge) but does detect the answered(), even session.ready() returns false... FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); ManagedSession leg_b = new ManagedSession("{ignore_early_media=true,origination_caller_id_number=" + orig + ",originate_timeout=20}sofia/external/" + route); if (leg_b.Ready()) { if (leg_b.answered()) { leg_a.answer(); string apiResult = fsApi.ExecuteString(string.Format("uuid_bridge {1} {0}", leg_a.GetUuid(), leg_b.GetUuid())); while (leg_a.bridged() && leg_b.bridged()) { leg_a.sleep(500, 0); // Slow the loop down... } } } Thanks, Michael Lutz From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 19 12:24:05 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 19 Jan 2012 09:24:05 +0000 (GMT) Subject: [Freeswitch-users] feature list Message-ID: <1326965045.69511.YahooMailNeo@web29401.mail.ird.yahoo.com> How about someone puts a link to the changelog at the top of that page ? ?;-) >we should probably update that =p >there is also the changelog file in the build root which is pretty big but >sort of up to date. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/66fa623b/attachment.html From henrikaagaardsorensen at gmail.com Thu Jan 19 12:38:25 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Thu, 19 Jan 2012 10:38:25 +0100 Subject: [Freeswitch-users] Error Creating SIP UA for profile: X In-Reply-To: References: <70792043-1340-4287-8FBC-0EEA5A9DF611@freeswitch.org> Message-ID: <7972438173691683594@unknownmsgid> Adding my static IP fixed the problem. In FS conf. On 19/01/2012, at 08.30, Anton Kvashenkin wrote: and what about udp, try ss -aup lsof -iudp 2012/1/19 Henrik Aagaard S?rensen > This is, by the way, my netstat -tunlp: > > tcp 0 0 0.0.0.0:22 0.0.0.0:* > LISTEN 906/sshd > tcp 0 0 127.0.0.1:25 0.0.0.0:* > LISTEN 1122/master > tcp 0 0 0.0.0.0:3306 0.0.0.0:* > LISTEN 1031/mysqld > tcp 0 0 :::80 :::* > LISTEN 1132/httpd > tcp 0 0 :::22 :::* > LISTEN 906/sshd > tcp 0 0 ::1:25 :::* > LISTEN 1122/master > > > 2012/1/18 Henrik Aagaard S?rensen > >> Could it be something with IPv6? >> >> >> On Wed, Jan 18, 2012 at 9:47 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Either its trying to bind to an ip that is not actually on the >>> interface, or the address is already in use. >>> >>> >>> 2012/1/18 Henrik Aagaard S?rensen >>> >>>> I've just completely reinstalled CentOS and FreeSwitch and still get >>>> the problem. >>>> >>>> I've tried what you've done and this is my log (freeswitch.log): >>>> 2012-01-18 21:40:06.251164 [ERR] sofia.c:1930 Error Creating SIP UA >>>> for profile: external >>>> >>>> >>>> On Wed, Jan 18, 2012 at 8:58 PM, Brian West wrote: >>>> >>>>> 'sofia loglevel all 9' >>>>> >>>>> 'sofia profile xxx start' >>>>> >>>>> watch the error log. >>>>> >>>>> /b >>>>> >>>>> On Jan 18, 2012, at 10:15 AM, Henrik Aagaard S?rensen wrote: >>>>> >>>>> I've had a server running only FreeSWITCH for about 6 months now. I >>>>> have >>>>> had it turned off for some while. >>>>> >>>>> Now, when trying to start FreeSWTICH, I get the errors: >>>>> Error Creating SIP UA for profile: X where X differs between internal, >>>>> external and others. >>>>> >>>>> When starting my CentOS 6 server I run: >>>>> service iptables stop >>>>> service ip6tables stop >>>>> just to make sure that they aren't blocking anything. >>>>> >>>>> I have nothing else running on CentOS. >>>>> >>>>> I've tried to log everything from console, warning etc. but there's >>>>> nothing >>>>> indicating what the problem is. >>>>> >>>>> Can someone help me out? >>>>> >>>>> >>>>> -- >>>>> Brian West >>>>> FreeSWITCH Solutions, LLC >>>>> Phone: +1 (918) 420-9266 >>>>> Fax: +1 (918) 420-9267 >>>>> brian at freeswitch.org >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/cbe3df66/attachment-0001.html From ccesario at tecnomega.com.br Thu Jan 19 14:29:44 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Thu, 19 Jan 2012 09:29:44 -0200 Subject: [Freeswitch-users] Freetdm + OpenR2 Message-ID: <4F17FEA8.5080006@tecnomega.com.br> Hi guys! I'm try configure freeswith with E1 link R2 protocol, but I'm getting a lot errors when freeswitch services start. I have not found nothing about theses errors. Somebody have any idea about fix this ? All configsare made based inthe freeswitch wiki Errors Logs -> http://pastebin.freeswitch.org/18167 Freetdm.conf -> http://pastebin.freeswitch.org/18168 Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 /etc/dahdi/system.conf and dahi_status -> http://pastebin.freeswitch.org/18170 greats, Carlos From gerald.weber at besharp.at Thu Jan 19 15:49:38 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Thu, 19 Jan 2012 12:49:38 +0000 Subject: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice In-Reply-To: References: Message-ID: Ok thanks, that makes sense to me now ! :) Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Mittwoch, 18. J?nner 2012 19:28 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice You are confusing the envelope event of type socket data which contains the encapsulated custom event. As I said, we do not support blindly subscribing to every custom events, you must supply every subclass name. On Wed, Jan 18, 2012 at 3:58 AM, Gerald Weber > wrote: I get the same results with your script. (maybe I was not clear in my last reply and mixed things up, still new to fs) I was just curious why a plain telnet connect to fs shows "Event-Name CUSTOM" AND "Event-Name SOCKET_DATA" and fs_cli only shows Event-Name CUSTOM Seems there is a difference between fs_cli and a plain telnet connect to fs ? My real problem: When I want to receive all CUSTOM events (not being specific to a Event-Subclass) I don't receive them using $sock->sendRecv("event plain custom"); and $sock->recvEvent() in e.g. get.php when generating this events using ESLevent("CUSTOM","CONFIG::AGENT_LIST"); and sendEvent in send.php. I guess recvEvent is listening for the first Event-Name entry. If it doesn't match -> go ahead. It works using $sock->sendRecv("event plain custom CONFIG::AGENT_LIST"); But this means I have to know where the event is injected. Or am I on the wrong way, again ? Thx & regards, gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Dienstag, 17. J?nner 2012 18:11 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] PHP ESL Custom Events Event-Name sent twice you cannot do *all* custom events with /event plain custom that command is expecting also a subclass param /event plain custom CONFIG::AGENT_LIST you have made a mistake if you think this unlocks heartbeat etc, probably you did events plain all and forgot. When I do it from perl I do not see what you reported, perl test.pl [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [auth/request] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: auth/request [DEBUG] esl.c:1381 esl_send() SEND auth ClueCon [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK accepted] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK accepted [DEBUG] esl.c:502 esl_sendevent() SEND EVENT Event-Name: CUSTOM Event-Subclass: CONFIG::AGENT_LIST [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Content-Type] = [command/reply] [DEBUG] esl.c:1183 esl_recv_event() RECV HEADER [Reply-Text] = [+OK] [DEBUG] esl.c:1353 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: +OK +OK log level [7] freeswitch at internal> /event plain custom CONFIG::AGENT_LIST +OK event listener enabled plain RECV EVENT Event-Subclass: CONFIG::AGENT_LIST Core-UUID: 23c2fe9b-f686-45e6-b43b-a3431d32d3e2 FreeSWITCH-Hostname: deathstar.freeswitch.org FreeSWITCH-Switchname: DeathSTAR FreeSWITCH-IPv4: 8.19.97.170 FreeSWITCH-IPv6: ::1 Event-Date-Local: 2012-01-17 11:33:51 Event-Date-GMT: Tue, 17 Jan 2012 17:33:51 GMT Event-Date-Timestamp: 1326821631120687 Event-Calling-File: mod_event_socket.c Event-Calling-Function: read_packet Event-Calling-Line-Number: 1188 Command: sendevent CUSTOM Event-Name: CUSTOM freeswitch at internal> cat test.pl require ESL; ESL::eslSetLogLevel(7); my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $e = new ESL::ESLevent("CUSTOM","CONFIG::AGENT_LIST"); $con->sendEvent($e); On Tue, Jan 17, 2012 at 10:37 AM, Michael Collins > wrote: On Tue, Jan 17, 2012 at 12:17 AM, Gerald Weber > wrote: Hi, thanks for your answers @Anthony: I did a "make current" mins ago, fs starts with version FreeSWITCH Version 1.0.head (git-ef097a1 2012-01-16 17-26-35 -0600) @MC I connect using ./fs_cli -H 192.168.20.73 -P 8021 /log 0 /event plain custom (additionally I connect to fs using telnet 192.168.20.73 8021 on another terminal and issued "event plain all") Starting my php script, fs_cli shows nothing, telnet shows the event: Content-Length: 583 Content-Type: text/event-plain Event-Subclass: CONFIG%3A%3AAGENT_LIST Event-Name: SOCKET_DATA ... Event-Date-Timestamp: 1326785361276007 ... Event-Name: CUSTOM Agents: 2022 ZMQ-Msg-Cnt: 244 /event plain CUSTOM in fs_cli and restart php script -> no output. (first event subscription except all doesn't work ?) /event plain all in fs_cli and restart php script shows: Hmm, I guess it does not work. You can do a filter: /event plain all /filter Event-Name CUSTOM In any case I'll have to defer to Tony as to why you're seeing two different header's named "Event-Name" -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/b336cdf8/attachment-0001.html From dgarcia at anew.com.ve Thu Jan 19 16:12:16 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Thu, 19 Jan 2012 08:42:16 -0430 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F17FEA8.5080006@tecnomega.com.br> References: <4F17FEA8.5080006@tecnomega.com.br> Message-ID: <4F1816B0.5060800@anew.com.ve> Hi Carlos, Have you used freetdm before or it is your first time? Which hardware are you using? Digium? Sangoma? Could you capture a log of freeswitch from startup? One problem that a faced with freeswitch/freetdm and a digium compatible card was freeswitch and digium drivers (dahdi) were asociated with different users, so freeswitch/freetdm could not access dahdi files. To correct this, I adjusted "/etc/udev/rules.d/dahdi.rules". On 1/19/2012 6:59 AM, Carlos Cesario wrote: > Hi guys! > > I'm try configure freeswith with E1 link R2 protocol, but I'm getting a > lot errors when freeswitch services start. > I have not found nothing about theses errors. > > Somebody have any idea about fix this ? > > All configsare made based inthe freeswitch wiki > > > > Errors Logs -> http://pastebin.freeswitch.org/18167 > > Freetdm.conf -> http://pastebin.freeswitch.org/18168 > > Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 > > /etc/dahdi/system.conf and dahi_status -> > http://pastebin.freeswitch.org/18170 > > > greats, > > Carlos > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 > > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/71d77789/attachment.html From mytemike72 at gmail.com Thu Jan 19 16:56:15 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 19 Jan 2012 14:56:15 +0100 Subject: [Freeswitch-users] Loosing control when bridging calls using Lua Message-ID: Hi All, I am struggling with some code, I have (based on the wiki examples) the following piece of lua code: session:setHangupHook("HangUpDialAbort") legA = session legB = freeswitch.Session("sofia/external/{MYDESTINATION}@{MYEXTERNALGATEWAY}") dispoB = "None" while(legA:ready() and legB:ready() and dispoB ~= "ANSWER") do if ( not legA:ready() ) then freeswitch.consoleLog("NOTICE","It appears that leg_a disconnected...\n") else dispoB = legB:getVariable("endpoint_disposition") end end if ( legA:ready() and legB:ready() ) then freeswitch.bridge(legA,legB) while(legA:ready() and legB:ready()) do freeswitch.consoleLog("NOTICE","We?re in the loop!\n") end else freeswitch.consoleLog("NOTICE","It appears that " .. dialA .. " or " .. dialB .. " disconnected...\n") end Question 1: It seems freeswitch.bridge() is a blocking function, Is there a way to use it in a non-blocking way? (so it will reach my while loop checking for legA and B to be ready) I need control of my code while the two legs are bridged. Question 2: session:setHangupHook("HangUpDialAbort") How can I pass session object to my hangup hook? I need to abort the dial attempt in my hangup hook when legA disconnects. Or are there other (better) ways of doing this? Question3: Is it possible to set different hangup hooks for different session (legs)? (would need that when question 2 leads to a 'no-go' ...) Thanks for your help, Michael Lutz From ccesario at tecnomega.com.br Thu Jan 19 17:11:18 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Thu, 19 Jan 2012 12:11:18 -0200 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F1816B0.5060800@anew.com.ve> References: <4F17FEA8.5080006@tecnomega.com.br> <4F1816B0.5060800@anew.com.ve> Message-ID: <4F182486.5070701@tecnomega.com.br> Hi Dario, Is the my first time that I usage freetdm. I'm using a Digium TE122. Yes, the log captured is here -> http://pastebin.freeswitch.org/18178 Well,I already changed my /etc/udev/rules.d/dahdi.rules changing the owner to user=freeswitch and group=freeswitch. See it here -> http://pastebin.freeswitch.org/18177 I too fix perms in /usr/local/freeswitch - # chown -R freeswitch:freeswitch /usr/local/freeswitch And I startup freeswitch with -u freeswitch -g freeswitch params. Thanks Carlos Em 19-01-2012 11:12, Saugort Dario Garcia Tovar escreveu: > Hi Carlos, > > Have you used freetdm before or it is your first time? > > Which hardware are you using? Digium? Sangoma? > > Could you capture a log of freeswitch from startup? > One problem that a faced with freeswitch/freetdm and a digium > compatible card was freeswitch and digium drivers (dahdi) were > asociated with different users, so freeswitch/freetdm could not access > dahdi files. To correct this, I adjusted "/etc/udev/rules.d/dahdi.rules". > > > > On 1/19/2012 6:59 AM, Carlos Cesario wrote: >> Hi guys! >> >> I'm try configure freeswith with E1 link R2 protocol, but I'm getting a >> lot errors when freeswitch services start. >> I have not found nothing about theses errors. >> >> Somebody have any idea about fix this ? >> >> All configsare made based inthe freeswitch wiki >> >> >> >> Errors Logs -> http://pastebin.freeswitch.org/18167 >> >> Freetdm.conf -> http://pastebin.freeswitch.org/18168 >> >> Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 >> >> /etc/dahdi/system.conf and dahi_status -> >> http://pastebin.freeswitch.org/18170 >> >> >> greats, >> >> Carlos >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 >> >> > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/0c774368/attachment.html From brian at freeswitch.org Thu Jan 19 17:45:21 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Jan 2012 08:45:21 -0600 Subject: [Freeswitch-users] feature list In-Reply-To: <1326965045.69511.YahooMailNeo@web29401.mail.ird.yahoo.com> References: <1326965045.69511.YahooMailNeo@web29401.mail.ird.yahoo.com> Message-ID: <2023CC8D-07D2-4D6D-9522-DCE1AC52AB23@freeswitch.org> Sure go ahead and do that. You can create a login on the wiki and do that if you see that as a good addition /b On Jan 19, 2012, at 3:24 AM, Bob Smith wrote: > How about someone puts a link to the changelog at the top of that page ? ;-) -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/64285c96/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Thu Jan 19 18:27:56 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Thu, 19 Jan 2012 15:27:56 +0000 (GMT) Subject: [Freeswitch-users] feature list Message-ID: <1326986876.24955.YahooMailNeo@web29403.mail.ird.yahoo.com> I do have a wiki login actually... ;-) The reason for the post was that I'm not intimately familiar with the code tree, and hence there's a risk I'd link to the wrong thing .... I get 12 pages of results when I Search the git repository for the word "changelog".? Earlier Anthony said "the change log in the build root" ... but the one at?http://fisheye.freeswitch.org/browse/freeswitch.git/ChangeLog hasn't been updated since 2007 ? ? ?So I'm guessing?http://fisheye.freeswitch.org/browse/freeswitch.git/docs/ChangeLog .... but that's in docs.... not the root ... ;-) >Sure go ahead and do that. You can create a login on the wiki and do that if you see that as a good addition -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/58ce276e/attachment.html From dgarcia at anew.com.ve Thu Jan 19 18:39:22 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Thu, 19 Jan 2012 11:09:22 -0430 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F182486.5070701@tecnomega.com.br> References: <4F17FEA8.5080006@tecnomega.com.br> <4F1816B0.5060800@anew.com.ve> <4F182486.5070701@tecnomega.com.br> Message-ID: <4F18392A.8070801@anew.com.ve> Ok, First, Where are you deploying this? The remote point what is? a pbx? a pstn? Do you have documentation about the expected signaling by the remote end? The log tell you some clues: a. In freeswith log 1. 2012-01-19 12:06:45.798589 [DEBUG] ftmod_r2.c:250 [s1c1][1:1] Signalling link status changed to SUSPENDED 2. 2012-01-19 12:06:45.798602 [ERR] ftdm_io.c:3317 I/O backend does not support command 55! That means could be some discrepances about signaling. It could be causing the "channel_next_event method not implemented in module zt" message. b. The board receive a clear alarm signal but it go to "suspend" state again. Why? 1. 2012-01-19 12:06:45.800040 [DEBUG] mod_freetdm.c:2289 Got R2 channel sig [ALARM_CLEAR] in channel 31 2. 2012-01-19 12:06:45.800055 [NOTICE] mod_freetdm.c:1886 Alarm cleared on channel 1:30 ... 1. 2012-01-19 12:06:45.800297 [DEBUG] mod_freetdm.c:2289 Got R2 channel sig [SIGSTATUS_CHANGED] in channel 31 2. 2012-01-19 12:06:45.800301 [INFO] mod_freetdm.c:2390 1:30 signalling changed to: SUSPENDED Again, coudld be a differences in signaling On 1/19/2012 9:41 AM, Carlos Cesario wrote: > Hi Dario, > > Is the my first time that I usage freetdm. > > I'm using a Digium TE122. > > Yes, the log captured is here -> http://pastebin.freeswitch.org/18178 > > Well,I already changed my /etc/udev/rules.d/dahdi.rules changing the > owner to user=freeswitch and group=freeswitch. > See it here -> http://pastebin.freeswitch.org/18177 > > I too fix perms in /usr/local/freeswitch - > # chown -R freeswitch:freeswitch /usr/local/freeswitch > > And I startup freeswitch with -u freeswitch -g freeswitch params. > > Thanks > > Carlos > > Em 19-01-2012 11:12, Saugort Dario Garcia Tovar escreveu: >> Hi Carlos, >> >> Have you used freetdm before or it is your first time? >> >> Which hardware are you using? Digium? Sangoma? >> >> Could you capture a log of freeswitch from startup? >> One problem that a faced with freeswitch/freetdm and a digium >> compatible card was freeswitch and digium drivers (dahdi) were >> asociated with different users, so freeswitch/freetdm could not >> access dahdi files. To correct this, I adjusted >> "/etc/udev/rules.d/dahdi.rules". >> >> >> >> On 1/19/2012 6:59 AM, Carlos Cesario wrote: >>> Hi guys! >>> >>> I'm try configure freeswith with E1 link R2 protocol, but I'm getting a >>> lot errors when freeswitch services start. >>> I have not found nothing about theses errors. >>> >>> Somebody have any idea about fix this ? >>> >>> All configsare made based inthe freeswitch wiki >>> >>> >>> >>> Errors Logs -> http://pastebin.freeswitch.org/18167 >>> >>> Freetdm.conf -> http://pastebin.freeswitch.org/18168 >>> >>> Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 >>> >>> /etc/dahdi/system.conf and dahi_status -> >>> http://pastebin.freeswitch.org/18170 >>> >>> >>> greats, >>> >>> Carlos >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ----- >>> No virus found in this message. >>> Checked by AVG -www.avg.com >>> Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 >>> >>> >> >> >> -- >> Atentamente, >> *Dario Garc?a* >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/5b71aa71/attachment.html From jaasmailing at gmail.com Thu Jan 19 18:47:36 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Thu, 19 Jan 2012 16:47:36 +0100 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario Message-ID: <4F183B18.40203@gmail.com> Hi all, I would like to know if the track-calls feature could be used in a multi-primary - multi-backup scenario. What I think is an environment with Kamailio dispatcher to N freeswitch boxes with a (multi or) single freeswitch backup. In case of failure, one primary should be replaced with a backup (spare) box that will take its IP and active calls. If all active freeswitch track call state in the same table/database, how the backup box will know what calls should be recovered? Anyone could explain the call recovery algorithm? Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/b2a769ad/attachment-0001.html From avi at avimarcus.net Thu Jan 19 18:55:54 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 19 Jan 2012 17:55:54 +0200 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario In-Reply-To: <4F183B18.40203@gmail.com> References: <4F183B18.40203@gmail.com> Message-ID: BTW: A far simpler set up is simply dispatching to FS boxes. Then for maintenance, you pause calls (fsctl pause) or tell FS to shut down when calls are finished (fsctl asap, as per mod_commands#shutdown ). Are you looking into track-calls for FS crashes, or for maintenance to move calls around..? This satisfies the second just fine.. -Avi p.s. mod_ha_cluster in development should handle this beautifully... some info on mod_ha_cluster On Thu, Jan 19, 2012 at 5:47 PM, Carlo Dimaggio wrote: > Hi all, > > I would like to know if the track-calls feature could be used in a > multi-primary - multi-backup scenario. > What I think is an environment with Kamailio dispatcher to N freeswitch > boxes with a (multi or) single freeswitch backup. In case of failure, one > primary should be replaced with a backup (spare) box that will take its IP > and active calls. > > If all active freeswitch track call state in the same table/database, how > the backup box will know what calls should be recovered? > Anyone could explain the call recovery algorithm? > > > Best Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/fe1b9e6b/attachment.html From ccesario at tecnomega.com.br Thu Jan 19 18:56:50 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Thu, 19 Jan 2012 13:56:50 -0200 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F18392A.8070801@anew.com.ve> References: <4F17FEA8.5080006@tecnomega.com.br> <4F1816B0.5060800@anew.com.ve> <4F182486.5070701@tecnomega.com.br> <4F18392A.8070801@anew.com.ve> Message-ID: <4F183D42.5050801@tecnomega.com.br> Well, I'm deploying this in my test lab :) - in Brazil The remote point is a E1 link with R2 Protocol from Embratel Well, about documentation I only know is R2 Protocol. And the E1 link is, before test this Freeswitch deploy, I deployed a asterisk server, and all services and signal UP without problems. And the all configs set in freetdm.conf ... freetdms.xml.conf are based in my asteirsk configuration. The dahdi_tool show me the link is OK. See the picture attached. Thanks Carlos Em 19-01-2012 13:39, Saugort Dario Garcia Tovar escreveu: > Ok, > > First, > Where are you deploying this? > The remote point what is? a pbx? a pstn? > Do you have documentation about the expected signaling by the remote end? > > The log tell you some clues: > a. In freeswith log > > 1. > 2012-01-19 12:06:45.798589 [DEBUG] ftmod_r2.c:250 [s1c1][1:1] > Signalling link status changed to SUSPENDED > 2. > 2012-01-19 12:06:45.798602 [ERR] ftdm_io.c:3317 I/O backend does > not support command 55! > > That means could be some discrepances about signaling. It could be > causing the "channel_next_event method not implemented in module zt" > message. > > b. The board receive a clear alarm signal but it go to "suspend" state > again. Why? > > 1. > 2012-01-19 12:06:45.800040 [DEBUG] mod_freetdm.c:2289 Got R2 > channel sig [ALARM_CLEAR] in channel 31 > 2. > 2012-01-19 12:06:45.800055 [NOTICE] mod_freetdm.c:1886 Alarm > cleared on channel 1:30 > > ... > > 1. > 2012-01-19 12:06:45.800297 [DEBUG] mod_freetdm.c:2289 Got R2 > channel sig [SIGSTATUS_CHANGED] in channel 31 > 2. > 2012-01-19 12:06:45.800301 [INFO] mod_freetdm.c:2390 1:30 > signalling changed to: SUSPENDED > > > Again, coudld be a differences in signaling > > > > On 1/19/2012 9:41 AM, Carlos Cesario wrote: >> Hi Dario, >> >> Is the my first time that I usage freetdm. >> >> I'm using a Digium TE122. >> >> Yes, the log captured is here -> http://pastebin.freeswitch.org/18178 >> >> Well,I already changed my /etc/udev/rules.d/dahdi.rules changing the >> owner to user=freeswitch and group=freeswitch. >> See it here -> http://pastebin.freeswitch.org/18177 >> >> I too fix perms in /usr/local/freeswitch - >> # chown -R freeswitch:freeswitch /usr/local/freeswitch >> >> And I startup freeswitch with -u freeswitch -g freeswitch params. >> >> Thanks >> >> Carlos >> >> Em 19-01-2012 11:12, Saugort Dario Garcia Tovar escreveu: >>> Hi Carlos, >>> >>> Have you used freetdm before or it is your first time? >>> >>> Which hardware are you using? Digium? Sangoma? >>> >>> Could you capture a log of freeswitch from startup? >>> One problem that a faced with freeswitch/freetdm and a digium >>> compatible card was freeswitch and digium drivers (dahdi) were >>> asociated with different users, so freeswitch/freetdm could not >>> access dahdi files. To correct this, I adjusted >>> "/etc/udev/rules.d/dahdi.rules". >>> >>> >>> >>> On 1/19/2012 6:59 AM, Carlos Cesario wrote: >>>> Hi guys! >>>> >>>> I'm try configure freeswith with E1 link R2 protocol, but I'm getting a >>>> lot errors when freeswitch services start. >>>> I have not found nothing about theses errors. >>>> >>>> Somebody have any idea about fix this ? >>>> >>>> All configsare made based inthe freeswitch wiki >>>> >>>> >>>> >>>> Errors Logs -> http://pastebin.freeswitch.org/18167 >>>> >>>> Freetdm.conf -> http://pastebin.freeswitch.org/18168 >>>> >>>> Freetdm.conf.xml -> http://pastebin.freeswitch.org/18169 >>>> >>>> /etc/dahdi/system.conf and dahi_status -> >>>> http://pastebin.freeswitch.org/18170 >>>> >>>> >>>> greats, >>>> >>>> Carlos >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ----- >>>> No virus found in this message. >>>> Checked by AVG - www.avg.com >>>> Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 >>>> >>>> >>> >>> >>> -- >>> Atentamente, >>> *Dario Garc?a* >>> Consultor. >>> >>> CCCT, Nivel C2, Sector Yarey, Mz, >>> Ofc. MZ03a. >>> Caracas-Venezuela. >>> Tel?fono: +58 212 9081842 >>> Cel: +58 412 2221515 >>> dgarcia at anew.com.ve >>> http://www.anew.com.ve >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1901 / Virus Database: 2109/4753 - Release Date: 01/19/12 >> > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/6f96cb07/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: assinatura_cesario.png Type: image/png Size: 25162 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/6f96cb07/attachment-0003.png -------------- next part -------------- A non-text attachment was scrubbed... Name: dahdi2.png Type: image/png Size: 44843 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/6f96cb07/attachment-0004.png -------------- next part -------------- A non-text attachment was scrubbed... Name: dahdi1.png Type: image/png Size: 30446 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/6f96cb07/attachment-0005.png From jaasmailing at gmail.com Thu Jan 19 19:01:09 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Thu, 19 Jan 2012 17:01:09 +0100 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario In-Reply-To: References: <4F183B18.40203@gmail.com> Message-ID: <4F183E45.80604@gmail.com> Il 19/01/12 16.55, Avi Marcus ha scritto: > BTW: > A far simpler set up is simply dispatching to FS boxes. Then for > maintenance, you pause calls (fsctl pause) or tell FS to shut down > when calls are finished (fsctl asap, as per mod_commands#shutdown > ). > > Are you looking into track-calls for FS crashes, or for maintenance to > move calls around..? This satisfies the second just fine.. > -Avi Hi Avi, yes, it's not for maintenance but for real high availability (the freeswitch boxes will act as b2bua for sip and rtp). > p.s. mod_ha_cluster in development should handle this beautifully... > some info on mod_ha_cluster > I have seen, but i think it will not be production ready in a short period... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/a87a2959/attachment.html From msc at freeswitch.org Thu Jan 19 19:21:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jan 2012 08:21:33 -0800 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: <3b40b751b8da8b91b0894edf1ea98a98.squirrel@vm1.subformat.net> References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> <3b40b751b8da8b91b0894edf1ea98a98.squirrel@vm1.subformat.net> Message-ID: This is a strange one. I'd update to latest git and re-test. If you can consistently reproduce these symptoms after updating I would open a Jira ticket. -MC On Wed, Jan 18, 2012 at 3:35 AM, wrote: > The sound file does work, as i can just set the dial plan to go straight > to play the media and it plays fine. Its only when the play media is in > the middle of the dial plan that it doesnt seem to work. > > Should this work? Or should i be achieving this from a different > setup/dialplan? For instance, maybe specifying the voicemail file (wav) to > play for a certain extension number? > > Thanks > > Dan. > > > On Tue, Jan 17, 2012 at 2:03 PM, wrote: > > > >> Aaah sorry! That shows just how new I am to FS. > >> > >> Heres the rather large log: http://pastebin.freeswitch.org/18151 > >> > >> Thanks again. > >> > >> Dan. > >> > >> > > Well, no new info there. I would confirm that the sound files themselves > > are not broken. You might want to clean out the sounds directory and then > > re-run "make cd-sounds-install" > > > > -MC > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/4ed3abf4/attachment.html From msc at freeswitch.org Thu Jan 19 19:30:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jan 2012 08:30:41 -0800 Subject: [Freeswitch-users] feature list In-Reply-To: <1326986876.24955.YahooMailNeo@web29403.mail.ird.yahoo.com> References: <1326986876.24955.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: Well, I know this one. The ChangeLog file is in the docs/ subdirectory. That is the most up-to-date one. I'm about 2 months behind on updating it. *SIGH* Anyway, if you link to it on fisheye then you're golden. -MC On Thu, Jan 19, 2012 at 7:27 AM, Bob Smith wrote: > I do have a wiki login actually... ;-) > > The reason for the post was that I'm not intimately familiar with the code > tree, and hence there's a risk I'd link to the wrong thing .... I get 12 > pages of results when I Search the git repository for the word "changelog". > > Earlier Anthony said "the change log in the build root" ... but the one at > http://fisheye.freeswitch.org/browse/freeswitch.git/ChangeLog hasn't been > updated since 2007 ? So I'm guessing > http://fisheye.freeswitch.org/browse/freeswitch.git/docs/ChangeLog .... > but that's in docs.... not the root ... ;-) > > >Sure go ahead and do that. You can create a login on the wiki and do > that if you see that as a good addition > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/4cf27d28/attachment.html From djbinter at gmail.com Thu Jan 19 19:44:38 2012 From: djbinter at gmail.com (DJB International) Date: Thu, 19 Jan 2012 08:44:38 -0800 Subject: [Freeswitch-users] feature list In-Reply-To: References: <1326986876.24955.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: I've added it. Check again. -djbinter On Thu, Jan 19, 2012 at 8:30 AM, Michael Collins wrote: > Well, I know this one. The ChangeLog file is in the docs/ subdirectory. > That is the most up-to-date one. I'm about 2 months behind on updating it. > *SIGH* Anyway, if you link to it on fisheye then you're golden. > > -MC > > On Thu, Jan 19, 2012 at 7:27 AM, Bob Smith < > gb10hkzo-freeswitch at yahoo.co.uk> wrote: > >> I do have a wiki login actually... ;-) >> >> The reason for the post was that I'm not intimately familiar with the >> code tree, and hence there's a risk I'd link to the wrong thing .... I get >> 12 pages of results when I Search the git repository for the word >> "changelog". >> >> Earlier Anthony said "the change log in the build root" ... but the one >> at http://fisheye.freeswitch.org/browse/freeswitch.git/ChangeLog hasn't >> been updated since 2007 ? So I'm guessing >> http://fisheye.freeswitch.org/browse/freeswitch.git/docs/ChangeLog .... >> but that's in docs.... not the root ... ;-) >> >> >Sure go ahead and do that. You can create a login on the wiki and do >> that if you see that as a good addition >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/2cd466f4/attachment-0001.html From dan at subformat.net Thu Jan 19 20:08:46 2012 From: dan at subformat.net (dan at subformat.net) Date: Thu, 19 Jan 2012 17:08:46 -0000 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> <3b40b751b8da8b91b0894edf1ea98a98.squirrel@vm1.subformat.net> Message-ID: Thanks MC, will do. So I take it this sort of functionality should work then? Im not trying to do something out of the ordinary? Is there anyway to specify a different voicemail greeting per extension? Might be able to get round it that way. Thanks Dan. > This is a strange one. I'd update to latest git and re-test. If you can > consistently reproduce these symptoms after updating I would open a Jira > ticket. > > -MC > > On Wed, Jan 18, 2012 at 3:35 AM, wrote: > >> The sound file does work, as i can just set the dial plan to go straight >> to play the media and it plays fine. Its only when the play media is in >> the middle of the dial plan that it doesnt seem to work. >> >> Should this work? Or should i be achieving this from a different >> setup/dialplan? For instance, maybe specifying the voicemail file (wav) >> to >> play for a certain extension number? >> >> Thanks >> >> Dan. >> >> > On Tue, Jan 17, 2012 at 2:03 PM, wrote: >> > >> >> Aaah sorry! That shows just how new I am to FS. >> >> >> >> Heres the rather large log: http://pastebin.freeswitch.org/18151 >> >> >> >> Thanks again. >> >> >> >> Dan. >> >> >> >> >> > Well, no new info there. I would confirm that the sound files >> themselves >> > are not broken. You might want to clean out the sounds directory and >> then >> > re-run "make cd-sounds-install" >> > >> > -MC >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From moises.silva at gmail.com Thu Jan 19 20:43:21 2012 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 19 Jan 2012 12:43:21 -0500 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F183D42.5050801@tecnomega.com.br> References: <4F17FEA8.5080006@tecnomega.com.br> <4F1816B0.5060800@anew.com.ve> <4F182486.5070701@tecnomega.com.br> <4F18392A.8070801@anew.com.ve> <4F183D42.5050801@tecnomega.com.br> Message-ID: On Thu, Jan 19, 2012 at 10:56 AM, Carlos Cesario wrote: > Well, > > I'm deploying this in my test lab :) - in Brazil > The remote point is a E1 link with R2 Protocol from Embratel > Well, about documentation I only know is R2 Protocol. > > > And the E1 link is, before test this Freeswitch deploy, I deployed a > asterisk server, and all services and signal UP without problems. > > And the all configs set in freetdm.conf ... freetdms.xml.conf are based in > my asteirsk configuration. > > The dahdi_tool show me the link is OK. > > According to your error logs, this is a problem with the DAHDI module not implementing certain methods yet. I will add them in a few minutes. It would be better if you can give me ssh. I will contact you via IM. *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/dfd90387/attachment.html From jstewart at teksystems.com Thu Jan 19 20:07:02 2012 From: jstewart at teksystems.com (Stewart, Justin) Date: Thu, 19 Jan 2012 12:07:02 -0500 Subject: [Freeswitch-users] Urgent Need for a FreeSWITCH Developer in Sunny San Diego! (This is contract to hire opportunity and the client is looking to fill this role ASAP.) Message-ID: <1BC1CD68F962DD4E8353A050792B5ED4AB53628173@EXCH-MBX11.allegisgroup.com> HAPPY NEW YEAR! Thank you for taking the time to read this email... Our client is in need of a Sr. Level Software Engineer, who has strong embedded or networking software experience, and has built Windows code in C (an ideal candidate will be a developer who has worked on an Asterisk project, or has strong Asterisk experience). They will be producing simple GUI apps and simple install packages, so a windows GUI expert would not be the right person. If this person has telecom experience (Lucent, Nortel, Avaya, Cisco, other telecom etc and knows any one of the protocols SIP/MGCP/H323/etc) who has also built Windows apps is the best bet. Linux experience is VERY beneficial. Software Engineering skills C programming in Windows Visual Studio 2008 IDE Windows application packaging/installers (MSI) Network programming in Windows using winsock Application specific skills Needed Configuring FreeSWITCH or Asterisk on Windows XP/Server 2008 Installing FreeTDM drivers (for Windows) Implementing TCP/UDP IPv4 network applications in C Knowledge of VOIP Signalling protocols (either SIP, SCCP, MGCP, or H.323) Testing TCP/UDP network applications using packet impairment & network simulation This is contract to hire opportunity is located in Carlsbad, CA and the client is looking to fill this role ASAP. Compensation: Negotiable based on relevant experience. Please contact me at jstewart at teksystems.com or 858 320 2767 Regards, Justin Stewart, Account Manager 4180 La Jolla Village Drive, Suite #100 La Jolla, CA 92037 Toll Free: (888) 260-6282 Cell: (619) 318 8564 Office: (858) 320-2767 Fax: (858) 320-2790 http://www.teksystems.com/Locations/United-States/California/San-Diego.aspx [cid:image001.gif at 01CCD680.949C4910] ________________________________ This electronic mail (including any attachments) may contain information that is privileged, confidential, and/or otherwise protected from disclosure to anyone other than its intended recipient(s). Any dissemination or use of this electronic mail or its contents (including any attachments) by persons other than the intended recipient(s) is strictly prohibited. If you have received this message in error, please notify us immediately by reply e-mail so that we may correct our internal records. Please then delete the original message (including any attachments) in its entirety. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/5cfd74ed/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 3681 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/5cfd74ed/attachment-0001.gif From moises.silva at gmail.com Thu Jan 19 21:25:25 2012 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 19 Jan 2012 13:25:25 -0500 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: <4F17FEA8.5080006@tecnomega.com.br> References: <4F17FEA8.5080006@tecnomega.com.br> Message-ID: On Thu, Jan 19, 2012 at 6:29 AM, Carlos Cesario wrote: > Hi guys! > > I'm try configure freeswith with E1 link R2 protocol, but I'm getting a > lot errors when freeswitch services start. > I have not found nothing about theses errors. > > Somebody have any idea about fix this ? > > All configsare made based inthe freeswitch wiki > > > > Errors Logs -> http://pastebin.freeswitch.org/18167 > > These error logs were just fixed now. MFCR2 should work with DAHDI too now. commit e3cb0352b0411234934763784c93c017e7ac44ae Author: Moises Silva Date: Thu Jan 19 16:18:30 2012 -0200 freetdm: Fill in DAHDI function pointer to retrieve the next channel event *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/42c8dd6a/attachment.html From ccesario at tecnomega.com.br Thu Jan 19 21:39:34 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Thu, 19 Jan 2012 16:39:34 -0200 Subject: [Freeswitch-users] Freetdm + OpenR2 In-Reply-To: References: <4F17FEA8.5080006@tecnomega.com.br> <4F1816B0.5060800@anew.com.ve> <4F182486.5070701@tecnomega.com.br> <4F18392A.8070801@anew.com.ve> <4F183D42.5050801@tecnomega.com.br> Message-ID: <4F186366.6080100@tecnomega.com.br> Hi guys, Moises solve the problem. http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=e3cb0352b0411234934763784c93c017e7ac44ae thanks Carlos Em 19-01-2012 15:43, Moises Silva escreveu: > On Thu, Jan 19, 2012 at 10:56 AM, Carlos Cesario > > wrote: > > Well, > > I'm deploying this in my test lab :) - in Brazil > The remote point is a E1 link with R2 Protocol from Embratel > Well, about documentation I only know is R2 Protocol. > > > And the E1 link is, before test this Freeswitch deploy, I deployed > a asterisk server, and all services and signal UP without problems. > > And the all configs set in freetdm.conf ... freetdms.xml.conf are > based in my asteirsk configuration. > > The dahdi_tool show me the link is OK. > > > According to your error logs, this is a problem with the DAHDI module > not implementing certain methods yet. I will add them in a few minutes. > > It would be better if you can give me ssh. I will contact you via IM. > > *Moises Silva > **/Software Engineer, Development Manager/*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > > > ** > > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter > `| > | YouTube > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/8dc96203/attachment-0001.html From msc at freeswitch.org Thu Jan 19 22:01:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jan 2012 11:01:10 -0800 Subject: [Freeswitch-users] Play media dialplan In-Reply-To: References: <4F1478C1.2000909@cupis.co.uk> <0d107718525b81873c6eebca59d42270.squirrel@vm1.subformat.net> <7cbb1e1343f2aaff76e86f85e4ac5f41.squirrel@vm1.subformat.net> <3b40b751b8da8b91b0894edf1ea98a98.squirrel@vm1.subformat.net> Message-ID: Each extension can record and select his/her own greeting from the advanced options menu. -MC On Thu, Jan 19, 2012 at 9:08 AM, wrote: > Thanks MC, will do. > > So I take it this sort of functionality should work then? Im not trying to > do something out of the ordinary? > > Is there anyway to specify a different voicemail greeting per extension? > Might be able to get round it that way. > > Thanks > > Dan. > > > This is a strange one. I'd update to latest git and re-test. If you can > > consistently reproduce these symptoms after updating I would open a Jira > > ticket. > > > > -MC > > > > On Wed, Jan 18, 2012 at 3:35 AM, wrote: > > > >> The sound file does work, as i can just set the dial plan to go straight > >> to play the media and it plays fine. Its only when the play media is in > >> the middle of the dial plan that it doesnt seem to work. > >> > >> Should this work? Or should i be achieving this from a different > >> setup/dialplan? For instance, maybe specifying the voicemail file (wav) > >> to > >> play for a certain extension number? > >> > >> Thanks > >> > >> Dan. > >> > >> > On Tue, Jan 17, 2012 at 2:03 PM, wrote: > >> > > >> >> Aaah sorry! That shows just how new I am to FS. > >> >> > >> >> Heres the rather large log: http://pastebin.freeswitch.org/18151 > >> >> > >> >> Thanks again. > >> >> > >> >> Dan. > >> >> > >> >> > >> > Well, no new info there. I would confirm that the sound files > >> themselves > >> > are not broken. You might want to clean out the sounds directory and > >> then > >> > re-run "make cd-sounds-install" > >> > > >> > -MC > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/f8320e1d/attachment.html From notlikeme75 at yahoo.com Thu Jan 19 23:45:27 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 19 Jan 2012 12:45:27 -0800 (PST) Subject: [Freeswitch-users] announce count to user only entering conference - Parse Error Message-ID: <1327005927.62463.YahooMailNeo@web65311.mail.ac2.yahoo.com> Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means.? scenario: When I enter a room with people the system says "there are" ? say count ?"other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. this works fine when there is at least 1 other caller in the room.? but when there isn't another caller and the room is empty. it says "there are" say count is a parse error ?so no digit read ?(need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero"? ERR] mod_say.c:130 Parse Error! ?Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. ? ? ? ? ? ? ? ? ? ? ? ? ??? ? ? ? ? ? ? ?? ? ? ? ? ? ? ? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/ea0db9c0/attachment.html From notlikeme75 at yahoo.com Fri Jan 20 00:23:01 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 19 Jan 2012 13:23:01 -0800 (PST) Subject: [Freeswitch-users] anoymous calls Message-ID: <1327008181.34377.YahooMailNeo@web65313.mail.ac2.yahoo.com> I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/1deb6bc2/attachment.html From brian at freeswitch.org Fri Jan 20 00:42:07 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Jan 2012 15:42:07 -0600 Subject: [Freeswitch-users] anoymous calls In-Reply-To: <1327008181.34377.YahooMailNeo@web65313.mail.ac2.yahoo.com> References: <1327008181.34377.YahooMailNeo@web65313.mail.ac2.yahoo.com> Message-ID: <3C8D736E-235E-43C4-BC49-CE5EEEFA0157@freeswitch.org> What exactly are you logging? On Jan 19, 2012, at 3:23 PM, Rodney wrote: > I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/bfce3090/attachment-0001.html From notlikeme75 at yahoo.com Fri Jan 20 01:39:44 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 19 Jan 2012 14:39:44 -0800 (PST) Subject: [Freeswitch-users] anonymous calls In-Reply-To: References: Message-ID: <1327012784.28159.YahooMailNeo@web65308.mail.ac2.yahoo.com> I am using fusinpbx on windows which logs everything I suppose. I open the csv and it has multiple columns. one is caller id name and one is caller id number. both show "anonymous". some calls work some don't. this also appears when monitoring a conference with the fusion web panel and when i watch the console when the call comes in it also says?mod_dialplan_xml.c:481 Processing Anonymous ->DID (masked for troubleshooting) in context public the provider log shows? 2012-01-19 16:34:43-06 FULL ANI ?(masked for troubleshooting) DID? (masked for troubleshooting)?00:08 ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, January 19, 2012 4:42 PM Subject: FreeSWITCH-users Digest, Vol 67, Issue 183 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: Play media dialplan (Michael Collins) ? 2. announce count to user only entering conference??? - Parse Error ? ? ? (Rodney) ? 3. anoymous calls (Rodney) ? 4. Re: anoymous calls (Brian West) Each extension can record and select his/her own greeting from the advanced options menu. -MC On Thu, Jan 19, 2012 at 9:08 AM, wrote: Thanks MC, will do. > >So I take it this sort of functionality should work then? Im not trying to >do something out of the ordinary? > >Is there anyway to specify a different voicemail greeting per extension? >Might be able to get round it that way. > >Thanks > >Dan. > >> This is a strange one. I'd update to latest git and re-test. If you can >> consistently reproduce these symptoms after updating I would open a Jira >> ticket. >> >> -MC >> >> On Wed, Jan 18, 2012 at 3:35 AM, wrote: >> >>> The sound file does work, as i can just set the dial plan to go straight >>> to play the media and it plays fine. Its only when the play media is in >>> the middle of the dial plan that it doesnt seem to work. >>> >>> Should this work? Or should i be achieving this from a different >>> setup/dialplan? For instance, maybe specifying the voicemail file (wav) >>> to >>> play for a certain extension number? >>> >>> Thanks >>> >>> Dan. >>> >>> > On Tue, Jan 17, 2012 at 2:03 PM, wrote: >>> > >>> >> Aaah sorry! That shows just how new I am to FS. >>> >> >>> >> Heres the rather large log: http://pastebin.freeswitch.org/18151 >>> >> >>> >> Thanks again. >>> >> >>> >> Dan. >>> >> >>> >> >>> > Well, no new info there. I would confirm that the sound files >>> themselves >>> > are not broken. You might want to clean out the sounds directory and >>> then >>> > re-run "make cd-sounds-install" >>> > >>> > -MC >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means.? scenario: When I enter a room with people the system says "there are" ? say count ?"other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. this works fine when there is at least 1 other caller in the room.? but when there isn't another caller and the room is empty. it says "there are" say count is a parse error ?so no digit read ?(need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero"? ERR] mod_say.c:130 Parse Error! ?Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. ? ? ? ? ? ? ? ? ? ? ? ? ??? ? ? ? ? ? ? ?? ? ? ? ? ? ? ? ? I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix?? What exactly are you logging? On Jan 19, 2012, at 3:23 PM, Rodney wrote: I have anonymous calls showing up in my call logs. when i log into the provider side it shows full ANI/CID info. how can I trouble shoot the fix?? --? Brian West? FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266? Fax: ? +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/6f90967d/attachment-0001.html From krice at freeswitch.org Fri Jan 20 02:15:01 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Jan 2012 17:15:01 -0600 Subject: [Freeswitch-users] Urgent Need for a FreeSWITCH Developer in Sunny San Diego! (This is contract to hire opportunity and the client is looking to fill this role ASAP.) In-Reply-To: <1BC1CD68F962DD4E8353A050792B5ED4AB53628173@EXCH-MBX11.allegisgroup.com> References: <1BC1CD68F962DD4E8353A050792B5ED4AB53628173@EXCH-MBX11.allegisgroup.com> Message-ID: Contact consulting at freeswitch.org or use the freeswitch-biz list. that is the proper location for this sort of thing. thank you On Thu, Jan 19, 2012 at 11:07 AM, Stewart, Justin wrote: > ** ** > > HAPPY NEW YEAR!**** > > ** ** > > Thank you for taking the time to read this email? > > Our client is in need of a Sr. Level Software Engineer, who has strong > embedded or networking software experience, and has built Windows code in C > (an ideal candidate will be a developer who has worked on an Asterisk > project, or has strong Asterisk experience). They will be producing simple > GUI apps and simple install packages, so a windows GUI expert would not be > the right person. If this person has telecom experience (Lucent, Nortel, > Avaya, Cisco, other telecom etc and knows any one of the protocols > SIP/MGCP/H323/etc) who has also built Windows apps is the best bet. Linux > experience is VERY beneficial. > > *Software Engineering skills* > C programming in Windows Visual Studio 2008 IDE > Windows application packaging/installers (MSI) > Network programming in Windows using winsock > > *Application specific skills Needed* > Configuring FreeSWITCH or Asterisk on Windows XP/Server 2008 > Installing FreeTDM drivers (for Windows) > Implementing TCP/UDP IPv4 network applications in C > Knowledge of VOIP Signalling protocols (either SIP, SCCP, MGCP, or H.323) > Testing TCP/UDP network applications using packet impairment & network > simulation > > *This is contract to hire opportunity is located in Carlsbad, CA and the > client is looking to fill this role ASAP.* > > Compensation: Negotiable based on relevant experience. > > Please contact me at jstewart at teksystems.com or 858 320 2767**** > > ** ** > > ** ** > > ** ** > > ** ** > > ** ** > > Regards,**** > > ** ** > > ** ** > > ** ** > > *Justin Stewart, *Account Manager > 4180 La Jolla Village Drive, Suite #100**** > > La Jolla, CA 92037**** > > Toll Free: (888) 260-6282 **** > > Cell: (619) 318 8564**** > > Office: (858) 320-2767**** > > Fax: (858) 320-2790**** > > http://www.teksystems.com/Locations/United-States/California/San-Diego.aspx > > [image: http://www.teksystems.com] **** > > ** ** > > ** ** > > ------------------------------ > This electronic mail (including any attachments) may contain information > that is privileged, confidential, and/or otherwise protected from > disclosure to anyone other than its intended recipient(s). Any > dissemination or use of this electronic mail or its contents (including any > attachments) by persons other than the intended recipient(s) is strictly > prohibited. If you have received this message in error, please notify us > immediately by reply e-mail so that we may correct our internal records. > Please then delete the original message (including any attachments) in its > entirety. Thank you. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/4b26fcc3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 3681 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/4b26fcc3/attachment.gif From jock.mckechnie at gmail.com Fri Jan 20 01:08:29 2012 From: jock.mckechnie at gmail.com (Jock McKechnie) Date: Thu, 19 Jan 2012 16:08:29 -0600 Subject: [Freeswitch-users] mod_rtmp + Akamai stream push Message-ID: Greetings wise(r) FreeSWITCH users; We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP media conversion (audio only) and have been utterly delighted by how easy and (so far) reliable this has been to both set up and run on a daily basis. One of my superiors has gotten all excited and wonders if we can use the same mechanism to do a stream push to Akamai for content delivery amongst many thousands of RTMP clients (think radio broadcast). It appears that the FreeSWITCH mod_rtmp module is _not_ capable of this - it can send a fresh RTMP stream to a registered client, but it cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I correct in this assertion, or am I misunderstanding the documentation (and google hits) available? My thanks to all; - Jock From bdfoster at endigotech.com Fri Jan 20 05:50:00 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jan 2012 21:50:00 -0500 Subject: [Freeswitch-users] Running a BASH script within a dialplan Message-ID: Hola, I'm trying to run a BASH script which turns sends a message to Google's Transcription API. It's a modified version of the example posted on the wiki. Here we go: !. #!/bin/sh 2. sox $1 message.flac pad .1 0 rate 16k 3. wget -q -U "Mozilla/5.0" --post-file message.flac --header="Content-Type: audio/x-flac; rate=16000" -O - " http://www.google.com/speech-api/v1/recognize?lang=en-us&client=chromium" > message.ret 4. echo "$(cat message.ret | sed 's/.*utterance":"//' | sed 's/","confidence.*//')" 5. rm message.flac 6. rm message.ret 7. # rm $1 Before I stick that in a dialplan, I've been trying to run it within fs_cli to make sure it works. However, I get: -ERR no reply ...every time I run it. Is there some fundamental flaw that FreeSWITCH doesn't recognize as an output? I plan on putting together a few BASH scripts and a dialplan to basically do the same thing that Mundy has done with "Siriously". -BDF -- This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/b40b392c/attachment.html From bdfoster at endigotech.com Fri Jan 20 06:20:18 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jan 2012 22:20:18 -0500 Subject: [Freeswitch-users] Running a BASH script within a dialplan In-Reply-To: References: Message-ID: Solved. Thanks SwK! For anyone else who runs into this problem, you need to make sure that the pwd is /usr/local/freeswitch/scripts if you plan on putting your script in the scripts folder. to check this, open up fs_cli and do: system /bin/pwd >/tmp/fs_pwd ...then go and look at /tmp/fs_pwd. That should tell you what you need to know! You can always run the script from somewhere else by doing: system /path/to/script.sh arg1 -BDF On Thu, Jan 19, 2012 at 9:50 PM, Brian Foster wrote: > Hola, > > I'm trying to run a BASH script which turns sends a message to Google's > Transcription API. It's a modified version of the example posted on the > wiki. Here we go: > > > !. #!/bin/sh > 2. sox $1 message.flac pad .1 0 rate 16k > 3. wget -q -U "Mozilla/5.0" --post-file message.flac > --header="Content-Type: audio/x-flac; rate=16000" -O - " > http://www.google.com/speech-api/v1/recognize?lang=en-us&client=chromium" > > message.ret > 4. echo "$(cat message.ret | sed 's/.*utterance":"//' | sed > 's/","confidence.*//')" > 5. rm message.flac > 6. rm message.ret > 7. # rm $1 > > Before I stick that in a dialplan, I've been trying to run it within > fs_cli to make sure it works. However, I get: > > -ERR no reply > > ...every time I run it. Is there some fundamental flaw that FreeSWITCH > doesn't recognize as an output? I plan on putting together a few BASH > scripts and a dialplan to basically do the same thing that Mundy has done > with "Siriously". > > -BDF > -- > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120119/0e57e6d3/attachment-0001.html From fieldpeak at gmail.com Fri Jan 20 09:07:27 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Fri, 20 Jan 2012 14:07:27 +0800 Subject: [Freeswitch-users] Dynamic specify the outbound GW within source code In-Reply-To: References: Message-ID: Avi, Can you please give any advise, thanks ! Cheers, Charles 2012/1/9 fieldpeak > Hi Avi, > > Thanks so much for your kindly reply. > > Actually, now i'm using mod_nibble for billing, i write a function > "check_billing_before_routing" in nibble_state_handler, in this > func("check_billing_before_routing"), it will call an external command, > this command will query the backend database if the caller has enough money > to contiue the call, the mod_nibblebill will contiue the call or hangup the > call according to the result of the external command. i have realize all > above, it works well. > > switch_state_handler_table_t nibble_state_handler = { > /* on_init */ NULL, > /* on_routing */ check_billing_before_routing, /* Need to add a check > here for anything in their account before routing */ > /* on_execute */ sched_billing, /* Turn on heartbeat for this > session and do an initial account check */ > /* on_hangup */ process_hangup, /* On hangup - most important > place to go bill */ > /* on_exch_media */ NULL, > /* on_soft_exec */ NULL, > /* on_consume_med */ NULL, > /* on_hibernate */ NULL, > /* on_reset */ NULL, > /* on_park */ NULL, > /* on_reporting */ NULL, > /* on_destroy */ NULL > }; > > For PSTN call, i use dial plan below, "1.2.3.4" is the PSTN-GW > > > > > > > > > Now, as we add one more PSTN-GW for outbound call, and the FS have to > route call to the specific GW accoring to result of the external command > (the external command will return the IP address of GW as well), > > i can think out the FS own function like > "switch_channel_set_variable(channel, "caller_id_number")" can configure > the value of variable, however, what variable should i use for this case, > could you please advise, thank you very much! > > Regards, > Charels > > 2012/1/8 Avi Marcus > >> I'm not quite sure of the use case. Do any of these help? >> 1) specify a server, not an IP, and then let DNS determine where it goes. >> 2) use a small lua script to set the channel variable based on whatever >> you need - an sql query, some logic.. and then use that variable in the >> bridge string. >> >> Those help? If not, please explain more what problem you are trying to >> solve. >> >> -Avi >> >> >> On Sun, Jan 8, 2012 at 3:34 PM, fieldpeak wrote: >> >>> Dear friends, >>> >>> i have FS for PSTN outbound call using below dial plan, >>> >>> >>> >>> >>> >>> >>> >>> While, now i need dynamically specify the outbound GW?s IP address >>> according to the return result of the external command before routing in >>> the source code , e.g. if the external command return FS the IP address of >>> OB GW 6.7.8.9, then >>> >>> >>> >>> however, i don't know which function i should call within the source >>> code to realize it, could anybody help to advise, >>> >>> P.S. i know there is existing module ?mod_xml_curl? can realize similar >>> function, however, I could not use it for this case? >>> >>> >>> thanks a lot! >>> >>> Regards, >>> Charles >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Regards, > Charles > > -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/a5153a27/attachment.html From freeswitch at peely.com Fri Jan 20 11:11:19 2012 From: freeswitch at peely.com (peely) Date: Fri, 20 Jan 2012 00:11:19 -0800 (PST) Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> Message-ID: <1327047079032-7206746.post@n2.nabble.com> In some respects the Snom M9 is a step backwards from the M3 tho'. Lack of decent royalty free codec support for one (the M3 had iLBC). It also seems you need to edit the settings file (rather than use the GUI) to be able to clamp down the RTP port range. Still, I agree that for the currently available phones it tops the list, especially given its in-built PCAP logging, even settable from next from boot-up. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Cordless-IP-Phone-tp7191414p7206746.html Sent from the freeswitch-users mailing list archive at Nabble.com. From oseslija at gmail.com Fri Jan 20 17:25:33 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 20 Jan 2012 15:25:33 +0100 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F14EA33.8040505@communicatefreely.net> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> Message-ID: Since when Siemens A580IP can honour SRV? That's my biggest remark for it. On Tue, Jan 17, 2012 at 4:25 AM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Hello, > > I have been using the Gigaset S675IP and the A580IP with good success. > > They do G.722, can handle SIP SRV records, and the BLFs work (more than > I can say for the Aastra MBU400). > > They are also very inexpensive and sound great! > > They only caveats: > > SIP passwords have to be fairly short > No easy provisioning mechanism. > > Other than that, they work great. > > Not all models can transfer though - I believe the S675 can. I haven't > played with the C610 > > Good luck! > > -Tim > > ocset wrote: > > Hi > > > > I have had great success using Yealink phones with Freeswitch and a > > customer has asked me for a cordless phone for their office. I want to > > ensure that they can keep as much functionality with a cordless phone as > > they have with the Yealink T28. I am looking at the Siemens C610IP phone > > but don't know how well it plays with Freeswitch. > > > > Can someone please shed some light on the Siemens phone or any > > alternative that you have successfully implemented. > > > > Thanks in advance > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/b2febe80/attachment.html From bdfoster at endigotech.com Fri Jan 20 17:55:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jan 2012 09:55:11 -0500 Subject: [Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri Message-ID: In this example: send 1149 bytes to udp/[]:5060 at 14:47:46.972424: ------------------------------------------------------------------------ INVITE sip:@ SIP/2.0 Via: SIP/2.0/UDP 204.93.201.164;rport;branch=z9hG4bKDZKU0eQBtr3Be Max-Forwards: 64 From: "UNKNOWN" @>;tag=FFr3B9gSB0KZK To: @:5060> ... I need to change sip_invite_req_uri and sip_invite_to_uri so that the switch downstream can route the call based on the DID they have with me. I've tried setting this several different ways but no avail. I'd like it to be global if possible. Is there a good spot to set these variables? -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/3f9a157c/attachment-0001.html From brian at freeswitch.org Fri Jan 20 18:34:48 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 09:34:48 -0600 Subject: [Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri In-Reply-To: References: Message-ID: <62E72F9D-B9CA-4941-AFDA-07CCE2145D4A@freeswitch.org> OK how are you sending calls? I'll take dialplan for 300 Alex! /b On Jan 20, 2012, at 8:55 AM, Brian Foster wrote: > In this example: > > send 1149 bytes to udp/[]:5060 at 14:47:46.972424: > ------------------------------------------------------------------------ > INVITE sip:@ SIP/2.0 > Via: SIP/2.0/UDP 204.93.201.164;rport;branch=z9hG4bKDZKU0eQBtr3Be > Max-Forwards: 64 > From: "UNKNOWN" @>;tag=FFr3B9gSB0KZK > To: @:5060> > ... > > I need to change sip_invite_req_uri and sip_invite_to_uri so that the > switch downstream can route the call based on the DID they have with me. > I've tried setting this several different ways but no avail. I'd like it to > be global if possible. Is there a good spot to set these variables? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/16487fef/attachment.html From gustavo.espeche at upper-soft.com Fri Jan 20 18:41:35 2012 From: gustavo.espeche at upper-soft.com (Gustavo Espeche) Date: Fri, 20 Jan 2012 13:41:35 -0200 Subject: [Freeswitch-users] Build problem Message-ID: <20120120134135.18333mq475mpc9c0@www.easyipcall.com> hello all, i'm trying to build the las freeswitch version, in a ubutntu server 64 bit, but when i make it, i have the follow error Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo In file included from nua_subnotref.c:50:0: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:6:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:7:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:11:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:12:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:16:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:17:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:21:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:22:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:26:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:27:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:31:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:32:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:36:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:37:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:41:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:42:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:46:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:47:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:51:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:52:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:56:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:57:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:61:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:62:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:66:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:67:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:71:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:72:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:76:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:77:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:81:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:82:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:86:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:87:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:91:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:92:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:96:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:97:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:101:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:102:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:106:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:107:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:111:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:112:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:116:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:117:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:121:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:122:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:126:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:127:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:131:44: error: "/*" within comment [-Werror=comment] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:132:1: error: expected identifier or '(' before '}' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:47: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:53: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:53: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:54: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:54: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:393:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:397:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_refer_sub_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: error: 'sip_refer_sub_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: note: each undeclared identifier is reported only once for each function it appears in ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:28: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:425:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:429:1: error: unknown type name 'sip_refer_sub_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_refer_sub_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:11: error: 'sip_refer_sub_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:28: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:431:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:506:36: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:509:37: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:49: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:55: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:55: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:56: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:56: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:739:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:743:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_alert_info_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:11: error: 'sip_alert_info_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:29: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:771:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:775:1: error: unknown type name 'sip_alert_info_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_alert_info_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:11: error: 'sip_alert_info_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:29: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:777:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:852:38: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:855:39: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:45: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:51: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:51: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:52: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:52: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1085:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1089:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_reply_to_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:11: error: 'sip_reply_to_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:27: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1117:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1121:1: error: unknown type name 'sip_reply_to_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_reply_to_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:11: error: 'sip_reply_to_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:27: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1123:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1198:34: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1201:35: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:67: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:73: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:73: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:74: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:74: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1777:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1781:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_p_asserted_identity_make': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:11: error: 'sip_p_asserted_identity_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:38: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1809:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1813:1: error: unknown type name 'sip_p_asserted_identity_t' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function 'sip_p_asserted_identity_format': ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:11: error: 'sip_p_asserted_identity_t' undeclared (first use in this function) ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:38: error: expected expression before ')' token ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1815:17: error: variable 'h' set but not used [-Werror=unused-but-set-variable] ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: expected ')' before 'const' ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif without #if nua_subnotref.c: In function 'nua_refer_client_response': nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use in this function) nua_subnotref.c:915:23: error: expected ';' before 'const' nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function) cc1: all warnings being treated as errors make[9]: *** [nua_subnotref.lo] Error 1 make[8]: *** [all] Error 2 make[8]: *** No rule to make target `nua/libnua.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [/home/easyipcall/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Aprecite a lot any advice. Regards. Gustavo. From brian at freeswitch.org Fri Jan 20 18:47:43 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 09:47:43 -0600 Subject: [Freeswitch-users] Build problem In-Reply-To: <20120120134135.18333mq475mpc9c0@www.easyipcall.com> References: <20120120134135.18333mq475mpc9c0@www.easyipcall.com> Message-ID: <072BD719-F6D7-4BDF-8407-4539B77DD760@freeswitch.org> Please report bugs to jira.freeswitch.org /b On Jan 20, 2012, at 9:41 AM, Gustavo Espeche wrote: > hello all, i'm trying to build the las freeswitch version, in a > ubutntu server 64 bit, but when i make it, i have the follow error > Making all in nua > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > In file included from nua_subnotref.c:50:0: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:6:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:7:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:11:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:12:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:16:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:17:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:21:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:22:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:26:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:27:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:31:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:32:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:36:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:37:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:41:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:42:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:46:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:47:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:51:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:52:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:56:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:57:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:61:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:62:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:66:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:67:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:71:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:72:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:76:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:77:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:81:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:82:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:86:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:87:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:91:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:92:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:96:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:97:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:101:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:102:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:106:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:107:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:111:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:112:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:116:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:117:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:121:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:122:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:126:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:127:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:131:44: error: "/*" > within comment [-Werror=comment] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:132:1: error: expected > identifier or '(' before '}' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:252:47: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:318:53: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:323:53: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:362:54: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:367:54: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:393:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:397:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_refer_sub_make': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: error: > 'sip_refer_sub_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:11: note: each > undeclared identifier is reported only once for each function it > appears in > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:399:28: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:425:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:429:1: error: unknown > type name 'sip_refer_sub_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_refer_sub_format': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:11: error: > 'sip_refer_sub_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:438:28: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:431:17: error: > variable 'h' set but not used [-Werror=unused-but-set-variable] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:506:36: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:509:37: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:598:49: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:664:55: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:669:55: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:708:56: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:713:56: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:739:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:743:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_alert_info_make': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:11: error: > 'sip_alert_info_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:745:29: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:771:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:775:1: error: unknown > type name 'sip_alert_info_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_alert_info_format': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:11: error: > 'sip_alert_info_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:784:29: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:777:17: error: > variable 'h' set but not used [-Werror=unused-but-set-variable] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:852:38: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:855:39: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:944:45: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1010:51: error: > unknown type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1015:51: error: > unknown type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1054:52: error: > unknown type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1059:52: error: > unknown type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1085:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1089:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_reply_to_make': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:11: error: > 'sip_reply_to_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1091:27: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1117:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1121:1: error: unknown > type name 'sip_reply_to_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_reply_to_format': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:11: error: > 'sip_reply_to_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1130:27: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1123:17: error: > variable 'h' set but not used [-Werror=unused-but-set-variable] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1198:34: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1201:35: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1636:67: error: > unknown type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1702:73: error: > unknown type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1707:73: error: > unknown type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1746:74: error: > unknown type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1751:74: error: > unknown type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1777:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1781:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_p_asserted_identity_make': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:11: error: > 'sip_p_asserted_identity_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1783:38: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1809:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1813:1: error: unknown > type name 'sip_p_asserted_identity_t' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: In function > 'sip_p_asserted_identity_format': > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:11: error: > 'sip_p_asserted_identity_t' undeclared (first use in this function) > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1822:38: error: > expected expression before ')' token > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1815:17: error: > variable 'h' set but not used [-Werror=unused-but-set-variable] > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h: At top level: > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1890:56: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:1893:57: error: > expected ')' before 'const' > ../../libsofia-sip-ua/sip/sofia-sip/sip_extra.h:2945:2: error: #endif > without #if > nua_subnotref.c: In function 'nua_refer_client_response': > nua_subnotref.c:915:7: error: 'sip_refer_sub_t' undeclared (first use > in this function) > nua_subnotref.c:915:23: error: expected ';' before 'const' > nua_subnotref.c:917:11: error: 'rs' undeclared (first use in this function) > cc1: all warnings being treated as errors > > make[9]: *** [nua_subnotref.lo] Error 1 > make[8]: *** [all] Error 2 > make[8]: *** No rule to make target `nua/libnua.la', needed by > `libsofia-sip-ua.la'. Stop. > make[7]: *** [all-recursive] Error 1 > Making all in packages > make[6]: *** [all-recursive] Error 1 > make[5]: *** [all] Error 2 > make[4]: *** > [/home/easyipcall/freeswitch/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error > 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Aprecite a lot any advice. > Regards. > > Gustavo. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/74f81fd8/attachment-0001.html From jahnunya at gmail.com Fri Jan 20 11:37:55 2012 From: jahnunya at gmail.com (jahnunya at gmail.com) Date: Fri, 20 Jan 2012 21:37:55 +1300 Subject: [Freeswitch-users] RTMP from non-flash/flex code Message-ID: Hi. I'd like to ask if anybody succeeded to connect streams from RTMP session to a call, for example SIP, not using flex/Flash. I need a way to connect a SIP call to voice streams available from RTMP server (Red5). Which side initiates the call doesn't matter to me. I tried the flex client and it works fine: a call from Flash/flex could initiate a SIP call in FreeSWITCH. However, I need to connect streams from RTMP server (Red5). So far, every attempt to do the same failed: I can connect to FreeSWITCH via RTMP, of course, and make a call (by calling the "makeCall" method) but the streams cannot be connected. I tried the other way round: from SIP to RTMP, but it seems like the module was not meant to work this way, as it complaints about the session id/uuid. Unfortunately, I am not a C/C++ developer, so reading the code doesn't help me. Any solution or explanation of the RTMP module will be highly appreciated. Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/6f07e9ed/attachment.html From markus.lindenberg at gmail.com Fri Jan 20 12:45:39 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Fri, 20 Jan 2012 10:45:39 +0100 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F13AC5F.4030400@the800group.com> References: <4F13AC5F.4030400@the800group.com> Message-ID: Hi, On Mon, Jan 16, 2012 at 05:49, ocset wrote: > > Can someone please shed some light on the Siemens phone or any > alternative that you have successfully implemented. I'm using a quite old Gigaset C450IP with FreeSWITCH, works great, including callerid and call transfers. I still get firmware update notifications from time to time, so i'm happy with long term support. The Gigaset web ui doesn't have too many options and settings like Polycom and Snom stuff, but all the basics are there. We also use a bunch of Gigaset C610IPs with Asterisk and right now a N510IP Pro DECT base is sitting on my desk for evaluation. The pro DECT base can handle 6 handsets each with a different SIP account and 4 simultaneous calls. Voice quality and usability are superb - from a european perspective, that's mostly because the phones have a 'R'-key for consultation calls and can do transfer-on-hangup. Automated provisioining is possible as well, although you have to nag Gigset about the specs. Hopefully they'll open up more, some technical details at http://wiki.gigaset.com/ look like a good start. We had user acceptance problems using Snom m3 at a customer and anticipate better results rolling out Gigaset handsets, partly because almost every german with a landline is using a Gigaset phone at home (seriously!). The Gigaset division was split from Siemens and now the company is called Gigaset. If you need support for more handsets and multi cell setup with handover in call, have a look at Aastra's stuff. Setup is pretty straightforward as well, but also much more expensive. From henrikaagaardsorensen at gmail.com Fri Jan 20 13:11:31 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Fri, 20 Jan 2012 11:11:31 +0100 Subject: [Freeswitch-users] Log-rotation and uploading of .csv files. In-Reply-To: References: Message-ID: I'm in the beginning of development for billing on my FS instance. Currently I'm just in the log-rotation and uploading of the .csv-files stage. I'm having trouble understanding completely how this should work, to be "bullet-proof" in a production environment. I've read the "Example Perl Script for CDR into MySQL" in http://wiki.freeswitch.org/wiki/Mod_cdr_csv. What I would like to accomplish is uploading the Master.csv-file to another server, which should take care of all the billing, so it doesn't interfere with the VoIP-server. But how can I make sure, that when I'm uploading the Master.csv file, that FS isn't currently writing to the file as well. I'm thinking of moving the Master.csv to Master-YYYY-MM-DD-HH-MM.csv every minute and then uploading those files to another server. But then again, how can I make sure, that when moving the file to Master-YYYY-MM-DD-HH-MM.csv that FS isn't writing to the file? Can someone share some light on this matter? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/f5f4af1e/attachment.html From wstephen80 at gmail.com Fri Jan 20 19:14:28 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 17:14:28 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break Message-ID: When I have many traffic, sometimes it appear in the log: 2012-01-20 17:09:34.028067 [CRIT] switch_event.c:339 Event system overloading. Taking a 10 second break 2012-01-20 17:09:35.036984 [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. Any advise on avoid that? The cpu is running at 40%-50% of load. Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/734e3260/attachment.html From brian at freeswitch.org Fri Jan 20 19:16:25 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:16:25 -0600 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: Message-ID: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> What rev of FreeSWITCH are you running? On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: > Any advise on avoid that? The cpu is running at 40%-50% of load. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/3cbea3ea/attachment.html From brian at freeswitch.org Fri Jan 20 19:18:02 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:18:02 -0600 Subject: [Freeswitch-users] Log-rotation and uploading of .csv files. In-Reply-To: References: Message-ID: STEP 1. Don't move the files yourself. 'pkill -HUP freeswitch' or 'fsctl send_sighup' at the CLI or fs_cli -x 'fsctl send_sighup' will rotate them. STEP 2. Relax and don't try so hard. ;) /b On Jan 20, 2012, at 4:11 AM, Henrik Aagaard S?rensen wrote: > I'm in the beginning of development for billing on my FS instance. > Currently I'm just in the log-rotation and uploading of the .csv-files > stage. > > I'm having trouble understanding completely how this should work, to be > "bullet-proof" in a production environment. > > I've read the "Example Perl Script for CDR into MySQL" in > http://wiki.freeswitch.org/wiki/Mod_cdr_csv. > > What I would like to accomplish is uploading the Master.csv-file to another > server, which should take care of all the billing, so it doesn't interfere > with the VoIP-server. > > But how can I make sure, that when I'm uploading the Master.csv file, that > FS isn't currently writing to the file as well. > > I'm thinking of moving the Master.csv to Master-YYYY-MM-DD-HH-MM.csv every > minute and then uploading those files to another server. > But then again, how can I make sure, that when moving the file to > Master-YYYY-MM-DD-HH-MM.csv that FS isn't writing to the file? > > Can someone share some light on this matter? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/c43eba9d/attachment.html From wstephen80 at gmail.com Fri Jan 20 19:18:56 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 17:18:56 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> Message-ID: FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) On Fri, Jan 20, 2012 at 5:16 PM, Brian West wrote: > What rev of FreeSWITCH are you running? > > On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: > > Any advise on avoid that? The cpu is running at 40%-50% of load. > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/f8921cdc/attachment-0001.html From bdfoster at endigotech.com Fri Jan 20 19:21:43 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jan 2012 11:21:43 -0500 Subject: [Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri In-Reply-To: <62E72F9D-B9CA-4941-AFDA-07CCE2145D4A@freeswitch.org> References: <62E72F9D-B9CA-4941-AFDA-07CCE2145D4A@freeswitch.org> Message-ID: I'm sending it to a local extension. Nothing fancy. I'm asking where I should put it in a dialplan. -BDF On Fri, Jan 20, 2012 at 10:34 AM, Brian West wrote: > OK how are you sending calls? I'll take dialplan for 300 Alex! > > /b > > On Jan 20, 2012, at 8:55 AM, Brian Foster wrote: > > In this example: > > send 1149 bytes to udp/[]:5060 at 14:47:46.972424: > ------------------------------------------------------------------------ > INVITE sip:@ SIP/2.0 > Via: SIP/2.0/UDP 204.93.201.164;rport;branch=z9hG4bKDZKU0eQBtr3Be > Max-Forwards: 64 > From: "UNKNOWN" @>;tag=FFr3B9gSB0KZK > To: @:5060> > ... > > I need to change sip_invite_req_uri and sip_invite_to_uri so that the > switch downstream can route the call based on the DID they have with me. > I've tried setting this several different ways but no avail. I'd like it to > be global if possible. Is there a good spot to set these variables? > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/b24662f0/attachment.html From roland at haenel.me Fri Jan 20 19:28:09 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Fri, 20 Jan 2012 17:28:09 +0100 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox Message-ID: Hi, It seems that FreeSwitch cannot negotiate G.722 with an AVM Fritzbox device (*the* most popular VoIP box/router in Germany) because of what seems to be a concatenation of several bugs. First of all, the FritzBox sends this SDP in the INVITE: INVITE sip:+4921173062190950 at 192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. [...] v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. *a=rtpmap:9 G722/16000.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. which is wrong since RFC 3551, Section 4.5.2 clearly states that the RTP clock is 8kHz. We could all blame it on AVM here, but when taking a look at the "200 OK" from FreeSwitch, that's really not what should happen: SIP/2.0 200 OK. [...] v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio *0* RTP/AVP 96. *a=rtpmap:96 G722/8000.* So FreeSwitch announces G.722/8000 (correct), but as RTP/AVP ID 96 (9 was requested) and with *UDP Port 0*. The FritzBox BYE's the call when it receives that. What's even more funny is that in the FreeSwitch debugging, it says about the local SDP: [...] 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007 at qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007 at qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 *m=audio 23418 RTP/AVP 9* *a=rtpmap:9 G722/8000* a=silenceSupp:off - - - - a=ptime:20 a=sendrecv So there, everything is correct. Does anyone have an idea about this one? Actually, I have taken a look at the source but I was not able to find the point where the UDP port gets lost... The full debugging trace and packet capture ist attached. Greetings, Roland # U 2012/01/20 17:01:04.323112 213.148.136.222:5060 -> 192.168.232.164:6060 INVITE sip:+4921173062190950 at 192.168.232.164:6060;transport=udp;gw=duro SIP/2.0. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. To: . From: "+4921653258007" ;tag=SD24vi901-88lt6slg-CC-35. CSeq: 1 INVITE. Max-Forwards: 65. P-Asserted-Identity: . Contact: . Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER. User-Agent: Huawei SoftX3000 V300R010. Supported: 100rel. Content-Length: 352. Content-Type: application/sdp. P-Called-Party-ID: sip:+4921173062190950 at qsc.de. X-ORIGINAL-DDI-URI: sip:+4921173062190950 at qsc.de. X-CID: mrg5ml6g71ggh5tty7yh7hc5osy5os8y at SoftX3000. . v=0. o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222. s=Sip Call. c=IN IP4 213.148.136.222. t=0 0. m=audio 28800 RTP/AVP 9 8 0 102 100 99 97. a=rtpmap:9 G722/16000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:102 G726-32/8000. a=rtpmap:100 G726-40/8000. a=rtpmap:99 G726-24/8000. a=rtpmap:97 iLBC/8000. a=ptime:30. a=fmtp:97 mode=30. # U 2012/01/20 17:01:04.323965 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" ;tag=SD24vi901-88lt6slg-CC-35. To: . Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Content-Length: 0. . # U 2012/01/20 17:01:04.354937 192.168.232.164:6060 -> 213.148.136.222:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 213.148.136.222:5060;branch=z9hG4bK7u3e5920b061csoit5d0.1. From: "+4921653258007" ;tag=SD24vi901-88lt6slg-CC-35. To: ;tag=ejS9yr5v3a6Qj. Call-ID: SD24vi901-73c7d5e35e297c6dd1cbccbf5ab175ee-l65h8l3. CSeq: 1 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-cfa926d 2012-01-10 17-33-40 -0600. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 156. P-Asserted-Identity: "+4921173062190950" . . v=0. o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164. s=FreeSWITCH. c=IN IP4 192.168.232.164. t=0 0. m=audio 0 RTP/AVP 96. a=rtpmap:96 G722/8000. ------------------------ 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007 at qsc.de entering state [received][100] 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5505 Remote SDP: v=0 o=HuaweiSoftX3000 8451448 8451448 IN IP4 213.148.136.222 s=Sip Call c=IN IP4 213.148.136.222 t=0 0 m=audio 28800 RTP/AVP 9 8 0 102 100 99 97 a=rtpmap:9 G722/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:102 G726-32/8000 a=rtpmap:100 G726-40/8000 a=rtpmap:99 G726-24/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=ptime:30 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:380 (sofia/external/+4921653258007 at qsc.de) State NEW 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G722:9:16000:30:64000]/[G722:9:8000:20:64000] 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4814 Bah HUMBUG! Sticking with G722 at 8000h@20i 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/+4921653258007 at qsc.de G722/8000 20 ms 160 samples 64000 bits 2012-01-20 17:01:04.324734 [DEBUG] sofia_glue.c:4930 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-01-20 17:01:04.324734 [DEBUG] sofia.c:5717 (sofia/external/+ 4921653258007 at qsc.de) State Change CS_NEW -> CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_INIT 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007 at qsc.de) State INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:85 sofia/external/+ 4921653258007 at qsc.de SOFIA INIT 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:125 (sofia/external/+ 4921653258007 at qsc.de) State Change CS_INIT -> CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:401 (sofia/external/+4921653258007 at qsc.de) State INIT going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_channel.c:1890 (sofia/external/+ 4921653258007 at qsc.de) Callstate Change DOWN -> RINGING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007 at qsc.de) State ROUTING 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:148 sofia/external/+ 4921653258007 at qsc.de SOFIA ROUTING 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:104 sofia/external/+4921653258007 at qsc.de Standard ROUTING 2012-01-20 17:01:04.324734 [INFO] mod_dialplan_xml.c:481 Processing +4921653258007 <+4921653258007>->+4921173062190950 in context default Dialplan: sofia/external/+4921653258007 at qsc.de parsing [default->default] continue=false Dialplan: sofia/external/+4921653258007 at qsc.de Absolute Condition [default] Dialplan: sofia/external/+4921653258007 at qsc.de Action park() 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:154 (sofia/external/+4921653258007 at qsc.de) State Change CS_ROUTING -> CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:410 (sofia/external/+4921653258007 at qsc.de) State ROUTING going to sleep 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007 at qsc.de) State EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] mod_sofia.c:241 sofia/external/+ 4921653258007 at qsc.de SOFIA EXECUTE 2012-01-20 17:01:04.324734 [DEBUG] switch_core_state_machine.c:192 sofia/external/+4921653258007 at qsc.de Standard EXECUTE EXECUTE sofia/external/+4921653258007 at qsc.de park() 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007 at qsc.de Command Execute answer() EXECUTE sofia/external/+4921653258007 at qsc.de answer() 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [INFO] switch_nat.c:590 NAT port mapping disabled 2012-01-20 17:01:04.344716 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/+4921653258007 at qsc.de] 192.168.232.164 port 23418 -> 213.148.136.222 port 28800 codec: 9 ms: 20 2012-01-20 17:01:04.344716 [DEBUG] switch_rtp.c:1659 Starting timer [soft] 160 bytes per 20ms 2012-01-20 17:01:04.344716 [DEBUG] mod_sofia.c:750 Local SDP sofia/external/+4921653258007 at qsc.de: v=0 o=FreeSWITCH 1327051846 1327051847 IN IP4 192.168.232.164 s=FreeSWITCH c=IN IP4 192.168.232.164 t=0 0 m=audio 23418 RTP/AVP 9 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:729 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] switch_channel.c:3194 (sofia/external/+ 4921653258007 at qsc.de) Callstate Change RINGING -> ACTIVE 2012-01-20 17:01:04.344716 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/+4921653258007 at qsc.de] has been answered 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.344716 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007 at qsc.de entering state [completed][200] 2012-01-20 17:01:04.344716 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007 at qsc.de Command Execute playback(silence_stream://100) EXECUTE sofia/external/+4921653258007 at qsc.de playback(silence_stream://100) 2012-01-20 17:01:04.364714 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 16000hz 1 channels 20ms 2012-01-20 17:01:04.464734 [DEBUG] switch_ivr_play_say.c:1678 done playing file silence_stream://100 2012-01-20 17:01:04.464734 [DEBUG] switch_core_session.c:1012 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.484715 [DEBUG] switch_ivr.c:591 sofia/external/+ 4921653258007 at qsc.de Command Execute sleep(900) EXECUTE sofia/external/+4921653258007 at qsc.de sleep(900) 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.624743 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.644728 [DEBUG] sofia.c:5494 Channel sofia/external/+ 4921653258007 at qsc.de entering state [ready][200] 2012-01-20 17:01:04.664735 [DEBUG] switch_core_session.c:875 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2852 (sofia/external/+ 4921653258007 at qsc.de) Callstate Change ACTIVE -> HANGUP 2012-01-20 17:01:04.684730 [NOTICE] sofia.c:623 Hangup sofia/external/+ 4921653258007 at qsc.de [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-20 17:01:04.684730 [DEBUG] switch_channel.c:2875 Send signal sofia/external/+4921653258007 at qsc.de [KILL] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007 at qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:2285 sofia/external/+4921653258007 at qsc.de skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:417 (sofia/external/+4921653258007 at qsc.de) State EXECUTE going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_HANGUP 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007 at qsc.de) State HANGUP 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:463 sofia/external/+ 4921653258007 at qsc.de Overriding SIP cause 480 with 200 from the other leg 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:469 Channel sofia/external/+ 4921653258007 at qsc.de hanging up, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:47 sofia/external/+4921653258007 at qsc.de Standard HANGUP, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:602 (sofia/external/+4921653258007 at qsc.de) State HANGUP going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:393 (sofia/external/+4921653258007 at qsc.de) State Change CS_HANGUP -> CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:362 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007 at qsc.de) State REPORTING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:79 sofia/external/+4921653258007 at qsc.de Standard REPORTING, cause: NORMAL_CLEARING 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:662 (sofia/external/+4921653258007 at qsc.de) State REPORTING going to sleep 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:387 (sofia/external/+4921653258007 at qsc.de) State Change CS_REPORTING -> CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/+4921653258007 at qsc.de [BREAK] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_session.c:1380 Session 3 (sofia/external/+4921653258007 at qsc.de) Locked, Waiting on external entities 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/external/+4921653258007 at qsc.de) Ended 2012-01-20 17:01:04.684730 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/+4921653258007 at qsc.de [CS_DESTROY] 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:491 (sofia/external/+4921653258007 at qsc.de) Callstate Change HANGUP -> DOWN 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:494 (sofia/external/+4921653258007 at qsc.de) Running State Change CS_DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007 at qsc.de) State DESTROY 2012-01-20 17:01:04.684730 [DEBUG] mod_sofia.c:374 sofia/external/+ 4921653258007 at qsc.de SOFIA DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:86 sofia/external/+4921653258007 at qsc.de Standard DESTROY 2012-01-20 17:01:04.684730 [DEBUG] switch_core_state_machine.c:504 (sofia/external/+4921653258007 at qsc.de) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/28e1e3ad/attachment-0001.html From brian at freeswitch.org Fri Jan 20 19:29:00 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:29:00 -0600 Subject: [Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri In-Reply-To: References: <62E72F9D-B9CA-4941-AFDA-07CCE2145D4A@freeswitch.org> Message-ID: Show me what you are doing in your dialplan because thats going to influence what I tell you to do. /b On Jan 20, 2012, at 10:21 AM, Brian Foster wrote: > I'm sending it to a local extension. Nothing fancy. I'm asking where I > should put it in a dialplan. > > -BDF -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/f8abb82d/attachment.html From brian at freeswitch.org Fri Jan 20 19:29:22 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:29:22 -0600 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> Message-ID: <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Are you load testing? I would update btw. /b On Jan 20, 2012, at 10:18 AM, Stephen Wilde wrote: > FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) > > > On Fri, Jan 20, 2012 at 5:16 PM, Brian West wrote: > >> What rev of FreeSWITCH are you running? >> >> On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: >> >> Any advise on avoid that? The cpu is running at 40%-50% of load. >> > -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/3a79ef3a/attachment.html From bdfoster at endigotech.com Fri Jan 20 19:31:15 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jan 2012 11:31:15 -0500 Subject: [Freeswitch-users] Modify sip_invite_req_uri, sip_invite_to_uri In-Reply-To: References: <62E72F9D-B9CA-4941-AFDA-07CCE2145D4A@freeswitch.org> Message-ID: On Fri, Jan 20, 2012 at 11:29 AM, Brian West wrote: > Show me what you are doing in your dialplan because thats going to > influence what I tell you to do. > > /b > > On Jan 20, 2012, at 10:21 AM, Brian Foster wrote: > > I'm sending it to a local extension. Nothing fancy. I'm asking where I > should put it in a dialplan. > > -BDF > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/1ff029e2/attachment.html From brian at freeswitch.org Fri Jan 20 19:31:20 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 10:31:20 -0600 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: Message-ID: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> This would be because the AVM Fritzbox is broken. Its not G722/16000 its G722/8000 (even tho its a 16k codec). On Jan 20, 2012, at 10:28 AM, Roland H?nel wrote: > It seems that FreeSwitch cannot negotiate G.722 with an AVM Fritzbox device > (*the* most popular VoIP box/router in Germany) because of what seems to be > a concatenation of several bugs. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/b1368b09/attachment.html From wstephen80 at gmail.com Fri Jan 20 19:33:09 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 17:33:09 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: No, it's real traffic on a production server. On Fri, Jan 20, 2012 at 5:29 PM, Brian West wrote: > Are you load testing? I would update btw. > > /b > > On Jan 20, 2012, at 10:18 AM, Stephen Wilde wrote: > > FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) > > > On Fri, Jan 20, 2012 at 5:16 PM, Brian West wrote: > > What rev of FreeSWITCH are you running? > > > On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: > > > Any advise on avoid that? The cpu is running at 40%-50% of load. > > > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/d00d5c26/attachment-0001.html From mario_fs at mgtech.com Fri Jan 20 20:02:20 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 20 Jan 2012 09:02:20 -0800 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: References: <4F13AC5F.4030400@the800group.com> Message-ID: <0CB347D7-9D6B-45F6-ACE0-4B3CF4FBB482@mgtech.com> Not what you are looking for but we were looking for a good one as well and wound up.... using our iPad with SipPhone, works perfectly with FS. We stopped looking for SIP phones and use the iPhones/iPads with SipPhone, (there are other clients). On Jan 20, 2012, at 1:45 AM, Markus Lindenberg wrote: > Hi, > > On Mon, Jan 16, 2012 at 05:49, ocset wrote: >> >> Can someone please shed some light on the Siemens phone or any >> alternative that you have successfully implemented. > > I'm using a quite old Gigaset C450IP with FreeSWITCH, works great, > including callerid and call transfers. I still get firmware update > notifications from time to time, so i'm happy with long term support. > The Gigaset web ui doesn't have too many options and settings like > Polycom and Snom stuff, but all the basics are there. > > We also use a bunch of Gigaset C610IPs with Asterisk and right now a > N510IP Pro DECT base is sitting on my desk for evaluation. The pro > DECT base can handle 6 handsets each with a different SIP account and > 4 simultaneous calls. Voice quality and usability are superb - from a > european perspective, that's mostly because the phones have a 'R'-key > for consultation calls and can do transfer-on-hangup. Automated > provisioining is possible as well, although you have to nag Gigset > about the specs. Hopefully they'll open up more, some technical > details at http://wiki.gigaset.com/ look like a good start. > > We had user acceptance problems using Snom m3 at a customer and > anticipate better results rolling out Gigaset handsets, partly because > almost every german with a landline is using a Gigaset phone at home > (seriously!). The Gigaset division was split from Siemens and now the > company is called Gigaset. > > If you need support for more handsets and multi cell setup with > handover in call, have a look at Aastra's stuff. Setup is pretty > straightforward as well, but also much more expensive. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vetali100 at gmail.com Fri Jan 20 20:35:10 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Fri, 20 Jan 2012 09:35:10 -0800 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: How many simultaneous calls are you reaching when you are getting this message? Thanks Vitalie 2012/1/20 Stephen Wilde > No, it's real traffic on a production server. > > On Fri, Jan 20, 2012 at 5:29 PM, Brian West wrote: > >> Are you load testing? I would update btw. >> >> /b >> >> On Jan 20, 2012, at 10:18 AM, Stephen Wilde wrote: >> >> FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) >> >> >> On Fri, Jan 20, 2012 at 5:16 PM, Brian West wrote: >> >> What rev of FreeSWITCH are you running? >> >> >> On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: >> >> >> Any advise on avoid that? The cpu is running at 40%-50% of load. >> >> >> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/2fcdeb56/attachment.html From wstephen80 at gmail.com Fri Jan 20 20:40:29 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 18:40:29 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: Near 1500 calls, but the CPU is 60% idle. On Fri, Jan 20, 2012 at 6:35 PM, Vitalie Colosov wrote: > How many simultaneous calls are you reaching when you are getting this > message? > > Thanks > Vitalie > > > 2012/1/20 Stephen Wilde > >> No, it's real traffic on a production server. >> >> On Fri, Jan 20, 2012 at 5:29 PM, Brian West wrote: >> >>> Are you load testing? I would update btw. >>> >>> /b >>> >>> On Jan 20, 2012, at 10:18 AM, Stephen Wilde wrote: >>> >>> FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) >>> >>> >>> On Fri, Jan 20, 2012 at 5:16 PM, Brian West >>> wrote: >>> >>> What rev of FreeSWITCH are you running? >>> >>> >>> On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: >>> >>> >>> Any advise on avoid that? The cpu is running at 40%-50% of load. >>> >>> >>> >>> >>> -- >>> Brian West >>> FreeSWITCH Solutions, LLC >>> Phone: +1 (918) 420-9266 >>> Fax: +1 (918) 420-9267 >>> brian at freeswitch.org >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/8e41a32c/attachment.html From wstephen80 at gmail.com Fri Jan 20 21:10:21 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 19:10:21 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: The problem seems to be related to the number of session per second because if I lower the maximum sps value (for example with ./fs_cli -x "fsctl sps 30") then in the log appear only "Throttle Error" but not the "Event system overloading". There is a way to increment the "event system capacity" so I can handle more sessions per second? On Fri, Jan 20, 2012 at 6:40 PM, Stephen Wilde wrote: > Near 1500 calls, but the CPU is 60% idle. > > > On Fri, Jan 20, 2012 at 6:35 PM, Vitalie Colosov wrote: > >> How many simultaneous calls are you reaching when you are getting this >> message? >> >> Thanks >> Vitalie >> >> >> 2012/1/20 Stephen Wilde >> >>> No, it's real traffic on a production server. >>> >>> On Fri, Jan 20, 2012 at 5:29 PM, Brian West wrote: >>> >>>> Are you load testing? I would update btw. >>>> >>>> /b >>>> >>>> On Jan 20, 2012, at 10:18 AM, Stephen Wilde wrote: >>>> >>>> FreeSWITCH Version 1.0.head (git-187abe0 2012-01-13 15-55-54 -0600) >>>> >>>> >>>> On Fri, Jan 20, 2012 at 5:16 PM, Brian West >>>> wrote: >>>> >>>> What rev of FreeSWITCH are you running? >>>> >>>> >>>> On Jan 20, 2012, at 10:14 AM, Stephen Wilde wrote: >>>> >>>> >>>> Any advise on avoid that? The cpu is running at 40%-50% of load. >>>> >>>> >>>> >>>> >>>> -- >>>> Brian West >>>> FreeSWITCH Solutions, LLC >>>> Phone: +1 (918) 420-9266 >>>> Fax: +1 (918) 420-9267 >>>> brian at freeswitch.org >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/26ce7552/attachment-0001.html From msc at freeswitch.org Fri Jan 20 21:18:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jan 2012 10:18:05 -0800 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: On Fri, Jan 20, 2012 at 10:10 AM, Stephen Wilde wrote: > The problem seems to be related to the number of session per second > because if I lower the maximum sps value (for example with ./fs_cli -x > "fsctl sps 30") then in the log appear only "Throttle Error" but not the > "Event system overloading". > There is a way to increment the "event system capacity" so I can handle > more sessions per second? > "Event system capacity" may very well be a function of the combination of operating system and hardware. Can you tell us the specs on the server? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/77f68dd8/attachment.html From wstephen80 at gmail.com Fri Jan 20 21:33:55 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 19:33:55 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: The server is a Fujitsu rack server with 2 x Xeon 5660 (a total of 12 core @2.8GHz). This is dstat: ----total-cpu-usage---- --net/eth0- usr sys idl wai hiq siq| recv send 28 6 62 0 0 4|5255k 5136k 27 6 63 0 0 3|5139k 5027k The OS is CentOS 5.7 64bit On Fri, Jan 20, 2012 at 7:18 PM, Michael Collins wrote: > > > On Fri, Jan 20, 2012 at 10:10 AM, Stephen Wilde wrote: > >> The problem seems to be related to the number of session per second >> because if I lower the maximum sps value (for example with ./fs_cli -x >> "fsctl sps 30") then in the log appear only "Throttle Error" but not the >> "Event system overloading". >> There is a way to increment the "event system capacity" so I can handle >> more sessions per second? >> > > "Event system capacity" may very well be a function of the combination of > operating system and hardware. Can you tell us the specs on the server? > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/4b525f85/attachment.html From wstephen80 at gmail.com Fri Jan 20 21:54:31 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 19:54:31 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: I saw "increase the event system capacity" because in the log there was a row: [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. Where it seems that all event dispatch threads are "busy" but I see that the cpu has many idle cycles so why not increase the number of dispatch threads? Or I'm wrong? On Fri, Jan 20, 2012 at 7:18 PM, Michael Collins wrote: > > > On Fri, Jan 20, 2012 at 10:10 AM, Stephen Wilde wrote: > >> The problem seems to be related to the number of session per second >> because if I lower the maximum sps value (for example with ./fs_cli -x >> "fsctl sps 30") then in the log appear only "Throttle Error" but not the >> "Event system overloading". >> There is a way to increment the "event system capacity" so I can handle >> more sessions per second? >> > > "Event system capacity" may very well be a function of the combination of > operating system and hardware. Can you tell us the specs on the server? > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/5a0d9195/attachment.html From roland at haenel.me Fri Jan 20 21:55:55 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Fri, 20 Jan 2012 19:55:55 +0100 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: Hello Brian, As I wrote, I was already aware of that. However I think it would make sense to improve the handling on the FS side because - the response of FS is clearly faulty, if FS doesn't like the FB offer it could reject the call, but sending 200 OK with port 0 is incorrect. Especially when the SDP data in debugging is not the same as the data we see in the SIP packet - though incorrect, the G722/16000 thing is not uncommon. According to some reports I googled Grandstream and others also misbehave in that way. Because FS is friendly, I'd suggest to answer the request for G722/16000 with G722/8000 and everything will probably just work out fine. By the way, the debugging output suggests that exactly this was the intent of the author - but there seems to be a bug that prevents this SDP from making it into the 200 OK packet. Roland Am 20.01.2012 17:37 schrieb "Brian West" : > This would be because the AVM Fritzbox is broken. Its not G722/16000 its > G722/8000 (even tho its a 16k codec). > > On Jan 20, 2012, at 10:28 AM, Roland H?nel wrote: > > It seems that FreeSwitch cannot negotiate G.722 with an AVM Fritzbox device > (*the* most popular VoIP box/router in Germany) because of what seems to be > a concatenation of several bugs. > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/5992b3e1/attachment.html From greg at brilliantecho.com Fri Jan 20 21:58:52 2012 From: greg at brilliantecho.com (Greg Millam) Date: Fri, 20 Jan 2012 10:58:52 -0800 Subject: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? Message-ID: Hi - The video refresh request does the job. recorded videos are very good quality, now, and start only when record_fsv is called. Thank you very much! - Greg Millam > ---------- Forwarded message ---------- > From:?Anthony Minessale > To:?FreeSWITCH Users Help > Cc: > Date:?Wed, 18 Jan 2012 13:41:49 -0600 > Subject:?Re: [Freeswitch-users] record_fsv buffers - Any way to flush incoming video? > try this: > > commit 67f559685f97b273dc5f5d23b69cc85cf90e32b0 > Author: Anthony Minessale > Date: ? Wed Jan 18 14:08:55 2012 -0600 > > ? ? req vid refresh in fsv > > > On Wed, Jan 18, 2012 at 11:33 AM, Greg Millam wrote: >> >> > commit c358f67fe4348b8b5209328660aef02d8ffaf15f >> > Author: Anthony Minessale >> > Date: ? Tue Jan 17 12:19:23 2012 -0600 >> > >> > ? ? eat inbound vid while playing fsv files >> > >> > >> > I didn't have a way to test it setup so i'll need feedback. >> >> It works, but now since it's missing the first frame, the resulting >> video is only partial, showing new bits. Is there a way to send a >> message to the client to "clear and send a full frame of video" ? >> >> > On Mon, Jan 16, 2012 at 11:38 PM, Greg Millam wrote: >> >> >> >> Hi folks - >> >> >> >> I have a freeswitch dialplan + script that first calls play_fsv to >> >> play a greeting, then record_fsv to record incoming video to fsv. >> >> >> >> It works fine, but there's one issue: Apparently, freeswitch is >> >> buffering incoming video during play_fsv. When record_fsv is called, >> >> that buffer is dumped into the .fsv file, resulting in several seconds >> >> of unneeded video. >> >> >> >> Is there a way to empty that incoming video buffer before record_fsv >> >> begins recording? >> >> >> >> Thank you! >> >> >> >> - Greg Millam From cmrienzo at gmail.com Fri Jan 20 22:22:18 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Fri, 20 Jan 2012 14:22:18 -0500 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: Did you bind any callbacks to events that might be taking a long time to process? On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: > I saw "increase the event system capacity" because in the log there was a > row: > > [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things > down. > > Where it seems that all event dispatch threads are "busy" but I see that the > cpu has many idle cycles so why not increase the number of dispatch threads? > Or I'm wrong? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/1247b9b7/attachment.html From brian at freeswitch.org Fri Jan 20 22:41:59 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 20 Jan 2012 13:41:59 -0600 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: Port 0 is absolutely correct its part of SOA and its rejected the media so it sends port 0 to reject it. Now the payload changing is odd but port zero is def. correct if its rejected. /b On Jan 20, 2012, at 12:55 PM, Roland H?nel wrote: > Hello Brian, > > As I wrote, I was already aware of that. However I think it would make > sense to improve the handling on the FS side because > > - the response of FS is clearly faulty, if FS doesn't like the FB offer > it could reject the call, but sending 200 OK with port 0 is incorrect. > Especially when the SDP data in debugging is not the same as the data we > see in the SIP packet > > - though incorrect, the G722/16000 thing is not uncommon. According to > some reports I googled Grandstream and others also misbehave in that way. > Because FS is friendly, I'd suggest to answer the request for G722/16000 > with G722/8000 and everything will probably just work out fine. By the way, > the debugging output suggests that exactly this was the intent of the > author - but there seems to be a bug that prevents this SDP from making it > into the 200 OK packet. > > Roland -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/a1d0b68d/attachment.html From edpimentl at gmail.com Fri Jan 20 23:04:41 2012 From: edpimentl at gmail.com (EdPimentl) Date: Fri, 20 Jan 2012 15:04:41 -0500 Subject: [Freeswitch-users] WebRTC ... Google Real Time Communication Message-ID: http://blog.chromium.org/2012/01/real-time-communications-in-chrome.html Just think of the possibilities with FS From anthony.minessale at gmail.com Fri Jan 20 23:45:38 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Jan 2012 14:45:38 -0600 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: the box can't handle the load, the disk io from the sql stmts is backing up the events. get a nicer box with the money saved from free softswitch =p On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: > Did you bind any callbacks to events that might be taking a long time to > process? > > > > On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: > >> I saw "increase the event system capacity" because in the log there was a >> row: >> >> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things >> down. >> >> Where it seems that all event dispatch threads are "busy" but I see that the >> cpu has many idle cycles so why not increase the number of dispatch threads? >> Or I'm wrong? >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/82fddd58/attachment.html From msc at freeswitch.org Fri Jan 20 23:47:17 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jan 2012 12:47:17 -0800 Subject: [Freeswitch-users] WebRTC ... Google Real Time Communication In-Reply-To: References: Message-ID: Is anyone out there building anything using WebRTC? Just curious. -MC On Fri, Jan 20, 2012 at 12:04 PM, EdPimentl wrote: > http://blog.chromium.org/2012/01/real-time-communications-in-chrome.html > > Just think of the possibilities with FS > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/ea9b06a0/attachment.html From msc at freeswitch.org Sat Jan 21 00:09:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jan 2012 13:09:33 -0800 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: On Fri, Jan 20, 2012 at 12:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > the box can't handle the load, the disk io from the sql stmts is backing > up the events. > get a nicer box with the money saved from free softswitch =p > Perhaps putting the db into a ramdrive might help: http://wiki.freeswitch.org/wiki/FreeSWITCH_DB_In_RAMdrive -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/18b7f01a/attachment.html From wstephen80 at gmail.com Sat Jan 21 00:11:35 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jan 2012 22:11:35 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: I'm using ram disk for the FS database "freeswitch/db". Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect the performance? On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > the box can't handle the load, the disk io from the sql stmts is backing > up the events. > get a nicer box with the money saved from free softswitch =p > > > On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: > >> Did you bind any callbacks to events that might be taking a long time to >> process? >> >> >> >> On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: >> >>> I saw "increase the event system capacity" because in the log there was >>> a row: >>> >>> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things >>> down. >>> >>> Where it seems that all event dispatch threads are "busy" but I see that the >>> cpu has many idle cycles so why not increase the number of dispatch threads? >>> Or I'm wrong? >>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/7b4647bf/attachment-0001.html From roland at haenel.me Sat Jan 21 01:09:40 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Fri, 20 Jan 2012 23:09:40 +0100 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: OK you're right, I read the RfC and in fact a zero port means that the stream is rejected. However the whole "negotiation" is still broken. If FS rejects the incorrect G.722, OK, but what about all the other (perfectly valid) codecs in the request (PCMA, PCMU)? Because FS only rejects G.722, the call is doomed to fail, even though it could work nicely if FS just selected one of the acceptable offers. Roland Am 20.01.2012 20:52 schrieb "Brian West" : > Port 0 is absolutely correct its part of SOA and its rejected the media so > it sends port 0 to reject it. Now the payload changing is odd but port > zero is def. correct if its rejected. > > /b > > On Jan 20, 2012, at 12:55 PM, Roland H?nel wrote: > > Hello Brian, > > As I wrote, I was already aware of that. However I think it would make > sense to improve the handling on the FS side because > > - the response of FS is clearly faulty, if FS doesn't like the FB offer > it could reject the call, but sending 200 OK with port 0 is incorrect. > Especially when the SDP data in debugging is not the same as the data we > see in the SIP packet > > - though incorrect, the G722/16000 thing is not uncommon. According to > some reports I googled Grandstream and others also misbehave in that way. > Because FS is friendly, I'd suggest to answer the request for G722/16000 > with G722/8000 and everything will probably just work out fine. By the way, > the debugging output suggests that exactly this was the intent of the > author - but there seems to be a bug that prevents this SDP from making it > into the 200 OK packet. > > Roland > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/16363d66/attachment.html From krice at freeswitch.org Sat Jan 21 01:51:42 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 20 Jan 2012 16:51:42 -0600 Subject: [Freeswitch-users] RTMP from non-flash/flex code In-Reply-To: References: Message-ID: you need to keep in mind that RTMP isn't just a streaming media protocol, its kinda like JSON or XML in the fact that you can encode a variety of things into RTMP such as arrays etc... you need to review the RTMP client implementation in freeswitch and make a matching plugin for Red5 or look at some things like red5phone and see how to make red5 talk sip... Keeping in mind that while red5 does work and work well in many instances I have found it to leak memory something fierce ... K On Fri, Jan 20, 2012 at 2:37 AM, wrote: > Hi. > > I'd like to ask if anybody succeeded to connect streams from RTMP session > to a call, for example SIP, not using flex/Flash. > > I need a way to connect a SIP call to voice streams available from RTMP > server (Red5). Which side initiates the call doesn't matter to me. > > I tried the flex client and it works fine: a call from Flash/flex could > initiate a SIP call in FreeSWITCH. > > However, I need to connect streams from RTMP server (Red5). So far, every > attempt to do the same failed: > I can connect to FreeSWITCH via RTMP, of course, and make a call (by > calling the "makeCall" method) but the streams cannot be connected. > > I tried the other way round: from SIP to RTMP, but it seems like the > module was not meant to work this way, as it complaints about the session > id/uuid. > > Unfortunately, I am not a C/C++ developer, so reading the code doesn't > help me. > > Any solution or explanation of the RTMP module will be highly appreciated. > > Thanks in advance > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/0eac1c9f/attachment.html From anthony.minessale at gmail.com Sat Jan 21 02:30:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Jan 2012 17:30:07 -0600 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: you can tell that by taking system vitals its hard to tell from the small amount of data. I do know that to get those errors, you have to push the core so hard that the sql stmts queuing up for transactions are getting too large for the rate at which they are written to the DB. Try a ramdisk like Michael suggested. On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: > I'm using ram disk for the FS database "freeswitch/db". > Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect > the performance? > > > On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> the box can't handle the load, the disk io from the sql stmts is backing >> up the events. >> get a nicer box with the money saved from free softswitch =p >> >> >> On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: >> >>> Did you bind any callbacks to events that might be taking a long time to >>> process? >>> >>> >>> >>> On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: >>> >>>> I saw "increase the event system capacity" because in the log there was >>>> a row: >>>> >>>> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things >>>> down. >>>> >>>> Where it seems that all event dispatch threads are "busy" but I see >>>> that the cpu has many idle cycles so why not increase the number of >>>> dispatch threads? >>>> Or I'm wrong? >>>> >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/38886d09/attachment-0001.html From anthony.minessale at gmail.com Sat Jan 21 02:39:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Jan 2012 17:39:03 -0600 Subject: [Freeswitch-users] mod_rtmp + Akamai stream push In-Reply-To: References: Message-ID: I don't recall any initiative being done to facilitate what you describe but its open source code so nothing's impossible. On Thu, Jan 19, 2012 at 4:08 PM, Jock McKechnie wrote: > Greetings wise(r) FreeSWITCH users; > > We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP > media conversion (audio only) and have been utterly delighted by how > easy and (so far) reliable this has been to both set up and run on a > daily basis. One of my superiors has gotten all excited and wonders if > we can use the same mechanism to do a stream push to Akamai for > content delivery amongst many thousands of RTMP clients (think radio > broadcast). > > It appears that the FreeSWITCH mod_rtmp module is _not_ capable of > this - it can send a fresh RTMP stream to a registered client, but it > cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I > correct in this assertion, or am I misunderstanding the documentation > (and google hits) available? > > My thanks to all; > > - Jock > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/5bfb1454/attachment.html From msc at freeswitch.org Sat Jan 21 03:28:01 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jan 2012 16:28:01 -0800 Subject: [Freeswitch-users] announce count to user only entering conference - Parse Error In-Reply-To: <1327005927.62463.YahooMailNeo@web65311.mail.ac2.yahoo.com> References: <1327005927.62463.YahooMailNeo@web65311.mail.ac2.yahoo.com> Message-ID: I would use a phrase macro for this. In fact, this seems like such an obvious thing to do that I added a phrase macro to conf/lang/en/ivr/sounds.xml. Update to the latest latest like two seconds ago git and you will have it available. (You'll need to copy it over from the source directory to your install directory.) The new macro is called other_callers_in_conf. In your example the extension would be this: The phrase macro does all the work. NOTE: I need to record a few new phrase files and then tweak the macro. In the meantime I encourage you to look at the macro that I added and tailor it to your needs. -MC On Thu, Jan 19, 2012 at 12:45 PM, Rodney wrote: > Okay, I finally figured out how to do this hopefully since I know what > conference they are entering and all seems to work accept when the room is > empty. I am putting phrases before and after the say to make it more > understandable to "some people" what the number means. > > scenario: > > When I enter a room with people the system says "there are" say count > "other callers" because it is a count without them in the room yet, unlike > the normal caller control to call the announce extension when they are > already in the room. > > this works fine when there is at least 1 other caller in the room. > > but when there isn't another caller and the room is empty. it says "there > are" say count is a parse error so no digit read (need it to say zero) > then it say "other callers" then says my normal you are in the conference > alone message which for now i have changed to say "zero" so it says sort of > backwards "there are" "other callers' "zero" > > ERR] mod_say.c:130 Parse Error! > > Could someone please help me understand why this Parse error could be > happening on zero count. I would like to fix it. thank you. > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120120/963999e2/attachment.html From brian at freeswitch.org Sat Jan 21 17:32:03 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Jan 2012 08:32:03 -0600 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland H?nel wrote: > OK you're right, I read the RfC and in fact a zero port means that the > stream is rejected. However the whole "negotiation" is still broken. If FS > rejects the incorrect G.722, OK, but what about all the other (perfectly > valid) codecs in the request (PCMA, PCMU)? > > Because FS only rejects G.722, the call is doomed to fail, even though it > could work nicely if FS just selected one of the acceptable offers. > > Roland -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/56849f12/attachment-0001.html From brian at freeswitch.org Sat Jan 21 17:37:02 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Jan 2012 08:37:02 -0600 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <4F14EA33.8040505@communicatefreely.net> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> Message-ID: <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> I hear these are awesome: http://www.amazon.com/Panasonic-Warranty-CORDLESS-TELEPHONE-Networking/dp/B004K3601G/ref=sr_1_4?s=electronics&ie=UTF8&qid=1327156543&sr=1-4 And I kinda like this: http://www.amazon.com/Siemens-Bluetooth-Connectivity-Answering-SL785/dp/B001V5J67Q/ref=wl_it_dp_o_npd?ie=UTF8&coliid=I29FGHCGGPF6GQ&colid=1BWDJUX5LYQE0 I added both to my wish list, a boy can hope! /b On Jan 16, 2012, at 9:25 PM, Tim St. Pierre wrote: > Hello, > > I have been using the Gigaset S675IP and the A580IP with good success. > > They do G.722, can handle SIP SRV records, and the BLFs work (more than > I can say for the Aastra MBU400). > > They are also very inexpensive and sound great! > > They only caveats: > > SIP passwords have to be fairly short > No easy provisioning mechanism. > > Other than that, they work great. > > Not all models can transfer though - I believe the S675 can. I haven't > played with the C610 > > Good luck! > > -Tim > > ocset wrote: >> Hi >> >> I have had great success using Yealink phones with Freeswitch and a >> customer has asked me for a cordless phone for their office. I want to >> ensure that they can keep as much functionality with a cordless phone as >> they have with the Yealink T28. I am looking at the Siemens C610IP phone >> but don't know how well it plays with Freeswitch. >> >> Can someone please shed some light on the Siemens phone or any >> alternative that you have successfully implemented. >> >> Thanks in advance >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org From henrikaagaardsorensen at gmail.com Sat Jan 21 16:41:45 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sat, 21 Jan 2012 14:41:45 +0100 Subject: [Freeswitch-users] MySQL type of read_codec and write_codec in CDR's. Message-ID: I've seen the SQL script, cdr.sql, in the src-directory which should match the CDR's. But it is missing the read_codec and write_codec. I guess it should be VARCHAR's in MySQL, but can anyone verify that and what is the longest possible string (VARCHAR(80)?)? From david.villasmil.work at gmail.com Sat Jan 21 18:24:29 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 21 Jan 2012 16:24:29 +0100 Subject: [Freeswitch-users] Profiles and gateways Message-ID: Hello All, I'm receiving calls on, say IP 1.2.3.4 and i want to send them out on IP 1.2.3.4 or IP 5.6.7.8 depending on the gateway I want to use. For this, i created TWO profiles, one binding to IP1, the other to IP2. Then I created one gateway on each profile. BUT, FS is loading both GWs on BOTH PROFILES! Is this normal? Can anyone give me a hand with this? Thanks! David my fs-cli output: freeswitch at 127.0.0.1@internal> sofia xmlstatus IP_REAL profile sip:mod_sofia at 1.2.3.4:5060 RUNNING (0) example.com gateway sip:@example.com NOREG test gateway sip:@Y.Y.Y.Y NOREG GREAT gateway sip:@X.X.X.X NOREG IP_SPAIN_1 profile sip:mod_sofia at 5.6.7.8:5060 RUNNING (0) example.com gateway sip:@example.com NOREG test gateway sip:@Y.Y.Y.Y NOREG GREAT gateway sip:@X.X.X.X NOREG My sofia: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/ce35d79c/attachment-0001.html From anthony.minessale at gmail.com Sat Jan 21 18:50:17 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Jan 2012 09:50:17 -0600 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: There is both a bug in the RFC itself and in the device in question. We might be able to wedge in tolerance but it may be a disservice because if you take the other option and fix the device then it will work correctly with every other compliant device. It may take a lot of code to make several exceptions in many places to tolerate this. It was painful enough wedging in the change to say 8000 everywhere. If the device is popular I would expect they would be eager to fix this. Germans are known for their attention to detail and quality after all. =] Plus we don't even have one to test with. On Jan 21, 2012 8:33 AM, "Brian West" wrote: > What is your codec selection set to in FreeSWITCH and what rev of > FreeSWITCH are you running? > > > On Jan 20, 2012, at 4:09 PM, Roland H?nel wrote: > > OK you're right, I read the RfC and in fact a zero port means that the > stream is rejected. However the whole "negotiation" is still broken. If FS > rejects the incorrect G.722, OK, but what about all the other (perfectly > valid) codecs in the request (PCMA, PCMU)? > > Because FS only rejects G.722, the call is doomed to fail, even though it > could work nicely if FS just selected one of the acceptable offers. > > Roland > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/ab761e21/attachment.html From anthony.minessale at gmail.com Sat Jan 21 18:58:00 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 21 Jan 2012 09:58:00 -0600 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> Message-ID: >From my experience the Panasonic cordless can completely replace a traditional analog cordless and the user will never know the difference. The base unit is OpenBSD and apart from some challenging docs they are fire and forget and look and feel like normal phones which oddly is a challenge to find. I have both aastra and snom and they have fallen into disuse because they are flimsy and unnatural. On Jan 21, 2012 8:37 AM, "Brian West" wrote: > I hear these are awesome: > > > http://www.amazon.com/Panasonic-Warranty-CORDLESS-TELEPHONE-Networking/dp/B004K3601G/ref=sr_1_4?s=electronics&ie=UTF8&qid=1327156543&sr=1-4 > > And I kinda like this: > > > http://www.amazon.com/Siemens-Bluetooth-Connectivity-Answering-SL785/dp/B001V5J67Q/ref=wl_it_dp_o_npd?ie=UTF8&coliid=I29FGHCGGPF6GQ&colid=1BWDJUX5LYQE0 > > I added both to my wish list, a boy can hope! > > /b > > On Jan 16, 2012, at 9:25 PM, Tim St. Pierre wrote: > > > Hello, > > > > I have been using the Gigaset S675IP and the A580IP with good success. > > > > They do G.722, can handle SIP SRV records, and the BLFs work (more than > > I can say for the Aastra MBU400). > > > > They are also very inexpensive and sound great! > > > > They only caveats: > > > > SIP passwords have to be fairly short > > No easy provisioning mechanism. > > > > Other than that, they work great. > > > > Not all models can transfer though - I believe the S675 can. I haven't > > played with the C610 > > > > Good luck! > > > > -Tim > > > > ocset wrote: > >> Hi > >> > >> I have had great success using Yealink phones with Freeswitch and a > >> customer has asked me for a cordless phone for their office. I want to > >> ensure that they can keep as much functionality with a cordless phone as > >> they have with the Yealink T28. I am looking at the Siemens C610IP phone > >> but don't know how well it plays with Freeswitch. > >> > >> Can someone please shed some light on the Siemens phone or any > >> alternative that you have successfully implemented. > >> > >> Thanks in advance > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/a07a9ce4/attachment.html From wstephen80 at gmail.com Sun Jan 22 00:04:20 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Sat, 21 Jan 2012 22:04:20 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: I'm already using the ramdisk. The problem happens when I have a provider that give congestion. In this case Freeswitch receives many tries but few connected calls and the number of session per second is high. To avoid the "event system overloading" (avoiding to lower the global session per second 'sps' parameter) I have insert in dialplan: In this way I have limited the session rate for the congestioned destination where I have so many tries. My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage of disk is near zero (dstat) why I cannot handle this session rate? On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can tell that by taking system vitals its hard to tell from the small > amount of data. > I do know that to get those errors, you have to push the core so hard that > the sql stmts queuing up for transactions are getting too large for the > rate at which they are written to the DB. Try a ramdisk like Michael > suggested. > > > On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: > >> I'm using ram disk for the FS database "freeswitch/db". >> Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect >> the performance? >> >> >> On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> the box can't handle the load, the disk io from the sql stmts is backing >>> up the events. >>> get a nicer box with the money saved from free softswitch =p >>> >>> >>> On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: >>> >>>> Did you bind any callbacks to events that might be taking a long time >>>> to process? >>>> >>>> >>>> >>>> On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: >>>> >>>>> I saw "increase the event system capacity" because in the log there >>>>> was a row: >>>>> >>>>> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing >>>>> things down. >>>>> >>>>> Where it seems that all event dispatch threads are "busy" but I see >>>>> that the cpu has many idle cycles so why not increase the number of >>>>> dispatch threads? >>>>> Or I'm wrong? >>>>> >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/4c8d1430/attachment-0001.html From darcy at primrose.ws Sun Jan 22 00:15:19 2012 From: darcy at primrose.ws (Darcy) Date: Sat, 21 Jan 2012 16:15:19 -0500 (Eastern Standard Time) Subject: [Freeswitch-users] Privacy on shared lines Message-ID: <4F1B2AE7.000064.17136@DWP> Is there a setting to set lines to private on shared lines in the freeswitch I am using snom phones and have the shared lines working except that I would like to automatically default to private lines versus allowing barge in. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/be1f358d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/gif Size: 45518 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/be1f358d/attachment-0001.gif From errotan at elder.hu Sun Jan 22 00:24:53 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sat, 21 Jan 2012 22:24:53 +0100 Subject: [Freeswitch-users] Profiles and gateways In-Reply-To: References: Message-ID: <4F1B2D25.4010109@elder.hu> Have you checked the log/freeswitch.xml.fsxml file to be sure only one gateway is present in each profile ? 2012-01-21 16:24 keltez?ssel, David Villasmil ?rta: > Hello All, > > > I'm receiving calls on, say IP 1.2.3.4 and i want to send them out on > IP 1.2.3.4 or IP 5.6.7.8 depending on the gateway I want to use. > For this, i created TWO profiles, one binding to IP1, the other to IP2. > Then I created one gateway on each profile. > > BUT, FS is loading both GWs on BOTH PROFILES! > > Is this normal? Can anyone give me a hand with this? > > Thanks! > > David > > my fs-cli output: > > freeswitch at 127.0.0.1@internal> sofia xmlstatus > > > > > IP_REAL > profile > sip:mod_sofia at 1.2.3.4:5060 > > RUNNING (0) > > > example.com > gateway > sip:@example.com > NOREG > > > > test > gateway > sip:@Y.Y.Y.Y > NOREG > > > > GREAT > gateway > sip:@X.X.X.X > NOREG > > > > IP_SPAIN_1 > profile > sip:mod_sofia at 5.6.7.8:5060 > > RUNNING (0) > > > example.com > gateway > sip:@example.com > NOREG > > > > test > gateway > sip:@Y.Y.Y.Y > NOREG > > > > GREAT > gateway > sip:@X.X.X.X > NOREG > > > > > My sofia: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/1c437023/attachment-0001.html From ochere at gmail.com Sat Jan 21 18:49:36 2012 From: ochere at gmail.com (Frank Ochere) Date: Sat, 21 Jan 2012 18:49:36 +0300 Subject: [Freeswitch-users] MySQL type of read_codec and write_codec in CDR's. In-Reply-To: References: Message-ID: Henrik, I use VARCHAR(30) for both, it has worked well so far. 2012/1/21 Henrik Aagaard S?rensen > I've seen the SQL script, cdr.sql, in the src-directory which should > match the CDR's. But it is missing the read_codec and write_codec. > > I guess it should be VARCHAR's in MySQL, but can anyone verify that > and what is the longest possible string (VARCHAR(80)?)? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/d98d80b4/attachment.html From markus.lindenberg at gmail.com Sat Jan 21 23:32:43 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Sat, 21 Jan 2012 21:32:43 +0100 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> Message-ID: Beware, that's "just" an analog POTS phone. On Sat, Jan 21, 2012 at 15:37, Brian West wrote: > > And I kinda like this: > > http://www.amazon.com/Siemens-Bluetooth-Connectivity-Answering-SL785/dp/B001V5J67Q/ref=wl_it_dp_o_npd?ie=UTF8&coliid=I29FGHCGGPF6GQ&colid=1BWDJUX5LYQE0 > From brian at freeswitch.org Sun Jan 22 03:33:05 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 21 Jan 2012 18:33:05 -0600 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> Message-ID: <10B5B749-69B6-44F0-A9ED-B9E5031CB9AD@freeswitch.org> Yes but its SEXSAY! /b On Jan 21, 2012, at 2:32 PM, Markus Lindenberg wrote: > Beware, that's "just" an analog POTS phone. > > On Sat, Jan 21, 2012 at 15:37, Brian West wrote: >> >> And I kinda like this: >> >> http://www.amazon.com/Siemens-Bluetooth-Connectivity-Answering-SL785/dp/B001V5J67Q/ref=wl_it_dp_o_npd?ie=UTF8&coliid=I29FGHCGGPF6GQ&colid=1BWDJUX5LYQE0 > -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/99023a00/attachment.html From notlikeme75 at yahoo.com Sun Jan 22 03:33:13 2012 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 21 Jan 2012 16:33:13 -0800 (PST) Subject: [Freeswitch-users] announce count to user only entering conference - Parse Error (Michael Collins) In-Reply-To: References: Message-ID: <1327192393.29454.YahooMailNeo@web65301.mail.ac2.yahoo.com> MC, thank you for adding this macro. it seems to work fine. for my current purposes i just created silent wavs until i need the complete voice instructions. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Saturday, January 21, 2012 9:32 AM Subject: FreeSWITCH-users Digest, Vol 67, Issue 195 ----- Forwarded Message ----- Send FreeSWITCH-users mailing list submissions to ??? freeswitch-users at lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit ??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to ??? freeswitch-users-request at lists.freeswitch.org You can reach the person managing the list at ??? freeswitch-users-owner at lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: ? 1. Re: mod_rtmp + Akamai stream push (Anthony Minessale) ? 2. Re: announce count to user only entering conference - Parse ? ? ? Error (Michael Collins) ? 3. Re: G.722 negotiation issue with AVM Fritzbox (Brian West) I don't recall any initiative being done to?facilitate what you describe but its open source code so nothing's impossible. ? On Thu, Jan 19, 2012 at 4:08 PM, Jock McKechnie wrote: Greetings wise(r) FreeSWITCH users; > >We've recently started using FreeSWITCH and mod_rtmp to do RTMP -> SIP >media conversion (audio only) and have been utterly delighted by how >easy and (so far) reliable this has been to both set up and run on a >daily basis. One of my superiors has gotten all excited and wonders if >we can use the same mechanism to do a stream push to Akamai for >content delivery amongst many thousands of RTMP clients (think radio >broadcast). > >It appears that the FreeSWITCH mod_rtmp module is _not_ capable of >this - it can send a fresh RTMP stream to a registered client, but it >cannot arbitrarily start pushing RTMP to an Akamai EntryPoint. Am I >correct in this assertion, or am I misunderstanding the documentation >(and google hits) available? > >My thanks to all; > >?- Jock > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 I would use a phrase macro for this. In fact, this seems like such an obvious thing to do that I added a phrase macro to conf/lang/en/ivr/sounds.xml. Update to the latest latest like two seconds ago git and you will have it available. (You'll need to copy it over from the source directory to your install directory.) The new macro is called other_callers_in_conf. In your example the extension would be this: ? ? ? ??????? ? ? ? ? ? ? ? ? ? ??????? ??????? ? ? ? ?? ? The phrase macro does all the work. NOTE: I need to record a few new phrase files and then tweak the macro. In the meantime I encourage you to look at the macro that I added and tailor it to your needs. -MC On Thu, Jan 19, 2012 at 12:45 PM, Rodney wrote: Okay, I finally figured out how to do this hopefully since I know what conference they are entering and all seems to work accept when the room is empty. I am putting phrases before and after the say to make it more understandable to "some people" what the number means.? > > >scenario: > > >When I enter a room with people the system says "there are" ? say count ?"other callers" because it is a count without them in the room yet, unlike the normal caller control to call the announce extension when they are already in the room. > > >this works fine when there is at least 1 other caller in the room.? > > >but when there isn't another caller and the room is empty. it says "there are" say count is a parse error ?so no digit read ?(need it to say zero) then it say "other callers" then says my normal you are in the conference alone message which for now i have changed to say "zero" so it says sort of backwards "there are" "other callers' "zero"? > > >ERR] mod_say.c:130 Parse Error! > > > >?Could someone please help me understand why this Parse error could be happening on zero count. I would like to fix it. thank you. > > > > > >? ? ? > >? >? ? ? ? >? ? ? ? >??? ? ? >? ? ? ? ?? > > >? ? ? ? >? ? ? ? > > > > > > > > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland H?nel wrote: OK you're right, I read the RfC and in fact a zero port means that the >stream is rejected. However the whole "negotiation" is still broken. If FS >rejects the incorrect G.722, OK, but what about all the other (perfectly >valid) codecs in the request (PCMA, PCMU)? > >Because FS only rejects G.722, the call is doomed to fail, even though it >could work nicely if FS just selected one of the acceptable offers. > >Roland --? Brian West? FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266? Fax: ? +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/caacca8c/attachment-0001.html From sherifomran2000 at yahoo.com Sun Jan 22 04:34:54 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 21 Jan 2012 17:34:54 -0800 (PST) Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: Message-ID: <1327196094.93390.YahooMailClassic@web110810.mail.gq1.yahoo.com> I have fritzbox and it is working well with freeswitch, however you should use i30 codec instead. For old devices this would be true because they dont support other codecs, however new firmwares exist and one can update his device. If you need some testing, please let me know. In my setting i changed the codec into greedy kind regards, Sherif Omran --- On Sat, 1/21/12, Anthony Minessale wrote: From: Anthony Minessale Subject: Re: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox To: "FreeSWITCH Users Help" Date: Saturday, January 21, 2012, 5:50 PM There is both a bug in the RFC itself and in the device in question.?? We might be able to wedge in tolerance but it may be a disservice because if you take the other option and fix the device then it will work correctly with every other compliant device. It may take a lot of code to make several exceptions in many places to tolerate this.? It was painful enough wedging in the change to say 8000 everywhere.?? If the device is popular I would expect they would be eager to fix this.? Germans are known for their attention to detail and quality after all. =] Plus we don't even have one to test with. On Jan 21, 2012 8:33 AM, "Brian West" wrote: What is your codec selection set to in FreeSWITCH and what rev of FreeSWITCH are you running? On Jan 20, 2012, at 4:09 PM, Roland H?nel wrote: OK you're right, I read the RfC and in fact a zero port means that the stream is rejected. However the whole "negotiation" is still broken. If FS rejects the incorrect G.722, OK, but what about all the other (perfectly valid) codecs in the request (PCMA, PCMU)? Because FS only rejects G.722, the call is doomed to fail, even though it could work nicely if FS just selected one of the acceptable offers. Roland --?Brian West?FreeSWITCH Solutions, LLCPhone: +1 (918) 420-9266?Fax: ? +1 (918) 420-9267 brian at freeswitch.orghttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120121/29a5825b/attachment.html From steveu at coppice.org Sun Jan 22 11:16:49 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 22 Jan 2012 16:16:49 +0800 Subject: [Freeswitch-users] Cordless IP Phone In-Reply-To: <10B5B749-69B6-44F0-A9ED-B9E5031CB9AD@freeswitch.org> References: <4F13AC5F.4030400@the800group.com> <4F14EA33.8040505@communicatefreely.net> <54D6EEA4-D7ED-4800-A865-D15C860DB73F@freeswitch.org> <10B5B749-69B6-44F0-A9ED-B9E5031CB9AD@freeswitch.org> Message-ID: <4F1BC5F1.3080602@coppice.org> On 01/22/2012 08:33 AM, Brian West wrote: > Yes but its SEXSAY! There's nothing sexy about a narrow band voice :-) > > /b > > On Jan 21, 2012, at 2:32 PM, Markus Lindenberg wrote: > >> Beware, that's "just" an analog POTS phone. >> >> On Sat, Jan 21, 2012 at 15:37, Brian West > > wrote: >>> >>> And I kinda like this: >>> >>> http://www.amazon.com/Siemens-Bluetooth-Connectivity-Answering-SL785/dp/B001V5J67Q/ref=wl_it_dp_o_npd?ie=UTF8&coliid=I29FGHCGGPF6GQ&colid=1BWDJUX5LYQE0 >>> >> > Steve From amit.nakum2009 at gmail.com Mon Jan 23 09:23:02 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Mon, 23 Jan 2012 11:53:02 +0530 Subject: [Freeswitch-users] How to Set Call Priority in mod_callcenter Message-ID: Dear All, I have set queue parameter to ,so now how to raise some caller's score,allows them to receive priority over other normal calls that might have been in the queue longer. I have test above scenario using which is not fulfill my requirement,if there is any parameter to set this scenario. Pl help me to set this scenario. Thanks and Regards Amit Nakum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/f49f84f7/attachment.html From bdfoster at endigotech.com Mon Jan 23 09:55:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 23 Jan 2012 01:55:58 -0500 Subject: [Freeswitch-users] How to Set Call Priority in mod_callcenter In-Reply-To: References: Message-ID: Hola, Try this: -BDF On Mon, Jan 23, 2012 at 1:23 AM, amit nakum wrote: > Dear All, > > I have set queue parameter to value="system"/>,so now how to raise some caller's score,allows them to > receive priority over other normal calls that might have been in the queue > longer. > > I have test above scenario using value="system"/> which is not fulfill my requirement,if there is any > parameter to set this scenario. > > Pl help me to set this scenario. > > > Thanks and Regards > Amit Nakum > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/9e9775f9/attachment.html From ruslan at symsol.com.au Mon Jan 23 02:04:44 2012 From: ruslan at symsol.com.au (Ruslan) Date: Mon, 23 Jan 2012 10:04:44 +1100 Subject: [Freeswitch-users] Invalid authorization header Message-ID: Hi please somebody help me!! I have this message "Invalid authorization header" on every inbound call I make. I couldn't find any reference (apart from chunk of c code, which I can't comprehend). My sip config looks like that: Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/565e82ec/attachment-0001.html From red.rain.seven at gmail.com Mon Jan 23 10:49:46 2012 From: red.rain.seven at gmail.com (Henry Huang) Date: Mon, 23 Jan 2012 15:49:46 +0800 Subject: [Freeswitch-users] upnp issue I think Message-ID: I am running Asterisk on my internal network. And I had port forwarding of 5060 from public IP to my Asterisk's internal IP address. I have recently installed a few FreeSWITCH from git on remote locations. And the weird thing is whenever I am trying to register to those FreeSWITCH servers with a softphone on my laptop, the returning SIP message end up routed to the Asterisk instead of my laptop since my softphone default SIP port is 5060. But the same wouldn't happen if I am registering to remote Asterisk servers. The return SIP message will routed correctly back to my softphone. So in summary, I can register to remote Asterisk server even when I've already forwarded port 5060 to my internal Asterisk server, but I can't register to remote FreeSWITCH servers. Does anyone know the reason behind this? Henry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/d84f8cad/attachment.html From roland at haenel.me Mon Jan 23 11:00:11 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Mon, 23 Jan 2012 09:00:11 +0100 Subject: [Freeswitch-users] G.722 negotiation issue with AVM Fritzbox In-Reply-To: References: <858081B1-9EE5-4C59-A6EC-2974E7491E8E@freeswitch.org> Message-ID: My codec selection list is: and I'm running a git version bebf3028d87cfcfbf094373e83ef92a391049d1f (4th Jan 2012). G722 is first because I want high quality to be selected if offered by the calling device (if I put PCMA first, then this gets selected without any problem). All in all, I think I can agree to what Anthony said, the first mistake is clearly a Fritzbox problem, and there is some risk that other things might break when we try to fix a thing that is already broken. Maybe I'll just implement a workaround to dynamically modify the codec list if I see a Fritzbox in the user-agent string... Greetings, Roland ----- 2012/1/21 Brian West > What is your codec selection set to in FreeSWITCH and what rev of > FreeSWITCH are you running? > > > On Jan 20, 2012, at 4:09 PM, Roland H?nel wrote: > > OK you're right, I read the RfC and in fact a zero port means that the > stream is rejected. However the whole "negotiation" is still broken. If FS > rejects the incorrect G.722, OK, but what about all the other (perfectly > valid) codecs in the request (PCMA, PCMU)? > > Because FS only rejects G.722, the call is doomed to fail, even though it > could work nicely if FS just selected one of the acceptable offers. > > Roland > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/02e747ec/attachment.html From roland at haenel.me Mon Jan 23 11:14:09 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Mon, 23 Jan 2012 09:14:09 +0100 Subject: [Freeswitch-users] Trigger re-INVITE with different SDP during a call Message-ID: Hi, Is there a way to trigger a SIP re-INVITE for a channel that has already been answered, and to make FreeSwitch offer a certain list of codecs to the endpoint? Let's assume the call is answered with PCMA/PCMU, then the user enters an IVR, now we know that he's going to join a conference. For the conference, I might want to enable G.722 and H.264 video (which should not generally be enabled for all incoming calls). I found stuff about codec-related channel variables, but no 'trigger' to tell FS to make a codec re-negotiation. Greetings, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/3c1db98b/attachment.html From thomasirvin85 at gmail.com Mon Jan 23 10:46:17 2012 From: thomasirvin85 at gmail.com (thomasirvin) Date: Sun, 22 Jan 2012 23:46:17 -0800 (PST) Subject: [Freeswitch-users] VoIP SIP SDK with Cisco Message-ID: <1327304777884-7215291.post@n2.nabble.com> Hi everyone, I have a small company and would like to use VoIP service with my Cisco System. Currently I am testing the SDK of Ozeki and it works fine with Cisco Unified CM (http://www.voip-sip-sdk.com/p_42-how-to-setup-ozeki-voip-sip-sdk-with-cisco-unified-communications-manager-voip.html) so far. Has anyone experienced anything with this solution after a longer test period? Thanks in advance. Thomas -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/VoIP-SIP-SDK-with-Cisco-tp7215291p7215291.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at earthspike.net Mon Jan 23 11:29:12 2012 From: freeswitch at earthspike.net (John) Date: Mon, 23 Jan 2012 08:29:12 +0000 Subject: [Freeswitch-users] VoIP SIP SDK with Cisco In-Reply-To: <1327304777884-7215291.post@n2.nabble.com> References: <1327304777884-7215291.post@n2.nabble.com> Message-ID: <4F1D1A58.5010302@earthspike.net> Thomas, I am struggling to see the relevance of this question on a FreeSWITCH users' list. Perhaps you could make clearer the FreeSWITCH aspect of your query? John On 23/01/12 07:46, thomasirvin wrote: > Hi everyone, > > I have a small company and would like to use VoIP service with my Cisco > System. Currently I am testing the SDK of Ozeki and it works fine with Cisco > Unified CM > (http://www.voip-sip-sdk.com/p_42-how-to-setup-ozeki-voip-sip-sdk-with-cisco-unified-communications-manager-voip.html) > so far. Has anyone experienced anything with this solution after a longer > test period? > > Thanks in advance. > > Thomas > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/VoIP-SIP-SDK-with-Cisco-tp7215291p7215291.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From shaheryarkh at googlemail.com Mon Jan 23 12:59:19 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 23 Jan 2012 14:59:19 +0500 Subject: [Freeswitch-users] Trigger re-INVITE with different SDP during a call In-Reply-To: References: Message-ID: Not sure but i think FS can't initiate re-invite, however if caller (or callee) initiate a re-invite then FS responds accordingly, including codec renegotiation. Thank you. 2012/1/23 Roland H?nel > Hi, > > Is there a way to trigger a SIP re-INVITE for a channel that has already > been answered, and to make FreeSwitch offer a certain list of codecs to the > endpoint? > > Let's assume the call is answered with PCMA/PCMU, then the user enters an > IVR, now we know that he's going to join a conference. For the conference, > I might want to enable G.722 and H.264 video (which should not generally be > enabled for all incoming calls). I found stuff about codec-related channel > variables, but no 'trigger' to tell FS to make a codec re-negotiation. > > Greetings, > Roland > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/8b652a1e/attachment.html From engineerzuhairraza at gmail.com Mon Jan 23 13:18:42 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Mon, 23 Jan 2012 14:18:42 +0400 Subject: [Freeswitch-users] Trigger re-INVITE with different SDP during a call In-Reply-To: References: Message-ID: Hi, Try setting bypass_media_after_bridge=true with in sofia profile and then pass absolute_codec_string before bridge on the new leg like bypass_media_after_bridge=true is a bit similar to canreinvite of asterisk Regards, Zohair Raza On Mon, Jan 23, 2012 at 1:59 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Not sure but i think FS can't initiate re-invite, however if caller (or > callee) initiate a re-invite then FS responds accordingly, including > codec renegotiation. > > Thank you. > > > 2012/1/23 Roland H?nel > >> Hi, >> >> Is there a way to trigger a SIP re-INVITE for a channel that has already >> been answered, and to make FreeSwitch offer a certain list of codecs to the >> endpoint? >> >> Let's assume the call is answered with PCMA/PCMU, then the user enters an >> IVR, now we know that he's going to join a conference. For the conference, >> I might want to enable G.722 and H.264 video (which should not generally be >> enabled for all incoming calls). I found stuff about codec-related channel >> variables, but no 'trigger' to tell FS to make a codec re-negotiation. >> >> Greetings, >> Roland >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/94a5112c/attachment-0001.html From devel at omninet.eu Mon Jan 23 13:40:30 2012 From: devel at omninet.eu (Anestis Mavro) Date: Mon, 23 Jan 2012 12:40:30 +0200 Subject: [Freeswitch-users] NDLB-connectile-dysfunction does not work Message-ID: <90CD446A38F84C33BAAD4FE6592259D4@omni1.local> Hello, I have tried to add to the user according to the Wiki page about NAT traversal, but it is not working as expected. I see the answer to the registration request going to the correct (Ethernet) IP address, but the port is wrong. It still uses the port from within the SIP packet, instead of the Ethernet port. I have tried also to add at the same time without success I am using XML curl for the directory and I confirmed that both settings are in the xml. Is there any other additional setting missing? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/3e136a06/attachment.html From david.villasmil.work at gmail.com Mon Jan 23 14:52:44 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 23 Jan 2012 12:52:44 +0100 Subject: [Freeswitch-users] Profiles and gateways In-Reply-To: <4F1B2D25.4010109@elder.hu> References: <4F1B2D25.4010109@elder.hu> Message-ID: Hello, Inthe log it shows the correct XML, one different GW on each profile... David On Sat, Jan 21, 2012 at 10:24 PM, Pusk?s Zsolt wrote: > > Have you checked the log/freeswitch.xml.fsxml file to be sure only one > gateway is present in each profile ? > > > 2012-01-21 16:24 keltez?ssel, David Villasmil ?rta: > > Hello All, > > > I'm receiving calls on, say IP 1.2.3.4 and i want to send them out on IP > 1.2.3.4 or IP 5.6.7.8 depending on the gateway I want to use. > For this, i created TWO profiles, one binding to IP1, the other to IP2. > Then I created one gateway on each profile. > > BUT, FS is loading both GWs on BOTH PROFILES! > > Is this normal? Can anyone give me a hand with this? > > Thanks! > > David > > my fs-cli output: > > freeswitch at 127.0.0.1@internal> sofia xmlstatus > > > > > IP_REAL > profile > sip:mod_sofia at 1.2.3.4:5060 > RUNNING (0) > > > example.com > gateway > sip:@example.com > NOREG > > > > test > gateway > sip:@Y.Y.Y.Y > NOREG > > > > GREAT > gateway > sip:@X.X.X.X > NOREG > > > > IP_SPAIN_1 > profile > sip:mod_sofia at 5.6.7.8:5060 > RUNNING (0) > > > example.com > gateway > sip:@example.com > NOREG > > > > test > gateway > sip:@Y.Y.Y.Y > NOREG > > > > GREAT > gateway > sip:@X.X.X.X > NOREG > > > > > My sofia: > > > > > > > > > > > > > > > > > > > > > > > > > value="true"/> > > > > > > > > > > > > > > > > > > > > > > > > value="$${global_codec_prefs}"/> > value="$${outbound_codec_prefs}"/> > > > > > value="localnet.auto"/> > > > > > > > > > > > > > value="true"/> > value="generous"/> > > > > > > > > > > > > > > > value="transport=tls"/> > > value="$${external_tls_port}"/> > > value="$${external_ssl_dir}"/> > > value="$${sip_tls_version}"/> > > > > > > > > > > > > > > > > > value="true"/> > > > > > > > > > > > > > > > > > > > > > > > > value="$${global_codec_prefs}"/> > value="$${outbound_codec_prefs}"/> > > > > > value="localnet.auto"/> > > > > > > > > > > > > > value="true"/> > value="generous"/> > > > > > > > > > > > > > > > value="transport=tls"/> > > value="$${external_tls_port}"/> > > value="$${external_ssl_dir}"/> > > value="$${sip_tls_version}"/> > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/516ad244/attachment-0001.html From gopalakrishnan.an at gmail.com Mon Jan 23 15:12:53 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 23 Jan 2012 17:42:53 +0530 Subject: [Freeswitch-users] Internal profile not showing Message-ID: Hi, I have installed Freeswitch from git in my Ubuntu 11.10. There are some default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. But when I try to register one extension to IP Phone or Softphone the account is not registered and responding as Authentication failuere. 403 Forbidden error, even though I have changed the password in 1000.xml file. Also in FS CLI I am not able to see the output for "sofia status profile internal". Please advice. Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/5f6630f2/attachment.html From roland at haenel.me Mon Jan 23 15:36:30 2012 From: roland at haenel.me (=?ISO-8859-1?Q?Roland_H=E4nel?=) Date: Mon, 23 Jan 2012 13:36:30 +0100 Subject: [Freeswitch-users] Trigger re-INVITE with different SDP during a call In-Reply-To: References: Message-ID: Hi, All these solutions only work for the specific case that the re-INVITE is done to match an outgoing call (bridge / transfer). What I'm looking for is a way to initiate a re-negotiation, not necessarily because the call is going to be bridged. For example to change the codec prior to entering a conference. Greetings, Roland 2012/1/23 Zohair Raza > Hi, > > Try setting bypass_media_after_bridge=true > > with in sofia > profile and then pass absolute_codec_string > before bridge on the new leg like > > data="nolocal:absolute_codec_string=PCMA,PCMU"/> > > bypass_media_after_bridge=true is a bit similar to canreinvite of asterisk > > Regards, > Zohair Raza > > > On Mon, Jan 23, 2012 at 1:59 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> Not sure but i think FS can't initiate re-invite, however if caller (or >> callee) initiate a re-invite then FS responds accordingly, including >> codec renegotiation. >> >> Thank you. >> >> >> 2012/1/23 Roland H?nel >> >>> Hi, >>> >>> Is there a way to trigger a SIP re-INVITE for a channel that has already >>> been answered, and to make FreeSwitch offer a certain list of codecs to the >>> endpoint? >>> >>> Let's assume the call is answered with PCMA/PCMU, then the user enters >>> an IVR, now we know that he's going to join a conference. For the >>> conference, I might want to enable G.722 and H.264 video (which should not >>> generally be enabled for all incoming calls). I found stuff about >>> codec-related channel variables, but no 'trigger' to tell FS to make a >>> codec re-negotiation. >>> >>> Greetings, >>> Roland >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Gru?, Roland -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/617b120b/attachment.html From govoiper at gmail.com Mon Jan 23 16:23:57 2012 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 23 Jan 2012 18:23:57 +0500 Subject: [Freeswitch-users] Trigger re-INVITE with different SDP during a call In-Reply-To: References: Message-ID: Hey, I'm not very sure about this. I pretty much guess that the kind of re-negotiation trigger must always come from UAC from either end for FS to relay it to other end, otherwise FS is a server and I doubt this may not get "triggered" from server itself. Anyways, that was just my humble opinion some other SIP+FreeSwitch guru could guide you better. Regards, Sammy 2012/1/23 Roland H?nel > Hi, > > All these solutions only work for the specific case that the re-INVITE is > done to match an outgoing call (bridge / transfer). What I'm looking for is > a way to initiate a re-negotiation, not necessarily because the call is > going to be bridged. For example to change the codec prior to entering a > conference. > > Greetings, > Roland > > > 2012/1/23 Zohair Raza > >> Hi, >> >> Try setting bypass_media_after_bridge=true >> >> with in sofia >> profile and then pass absolute_codec_string >> before bridge on the new leg like >> >> > data="nolocal:absolute_codec_string=PCMA,PCMU"/> >> >> bypass_media_after_bridge=true is a bit similar to canreinvite of asterisk >> >> Regards, >> Zohair Raza >> >> >> On Mon, Jan 23, 2012 at 1:59 PM, Muhammad Shahzad < >> shaheryarkh at googlemail.com> wrote: >> >>> Not sure but i think FS can't initiate re-invite, however if caller (or >>> callee) initiate a re-invite then FS responds accordingly, including >>> codec renegotiation. >>> >>> Thank you. >>> >>> >>> 2012/1/23 Roland H?nel >>> >>>> Hi, >>>> >>>> Is there a way to trigger a SIP re-INVITE for a channel that has >>>> already been answered, and to make FreeSwitch offer a certain list of >>>> codecs to the endpoint? >>>> >>>> Let's assume the call is answered with PCMA/PCMU, then the user enters >>>> an IVR, now we know that he's going to join a conference. For the >>>> conference, I might want to enable G.722 and H.264 video (which should not >>>> generally be enabled for all incoming calls). I found stuff about >>>> codec-related channel variables, but no 'trigger' to tell FS to make a >>>> codec re-negotiation. >>>> >>>> Greetings, >>>> Roland >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +92 334 422 40 88 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Gru?, > Roland > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/ecc1de83/attachment-0001.html From michal.zubac at comgate.cz Mon Jan 23 16:40:20 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Mon, 23 Jan 2012 14:40:20 +0100 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header Message-ID: <4F1D6344.6000501@comgate.cz> Hello. I'd like to remove UPDATE value from SIP Accept header when creating SIP calls. We're sending it in INVITE message and our provider uses that for in-call keep-alive checks every 10 minutes. FreeSwitch doesn't respond to that, so our provider disconnects RTP and call is dropped. According to RFC3311 we can indicate that we don't support this by not sending UPDATE in Accept header. Is this gonna help? Is there any way to drop that from SIP headers from dialplan? Or do I have to change source code? Or better, is there any other (cleaner) way to resolve this? Regards Michal Zubac ComGate Interactive s.r.o. From kris at kriskinc.com Mon Jan 23 17:51:33 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 23 Jan 2012 09:51:33 -0500 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: <4F1D6344.6000501@comgate.cz> References: <4F1D6344.6000501@comgate.cz> Message-ID: What are your Sofia profile session timer values set to? Can you post a complete SIP trace? Can your provider send a re-INVITE instead of UPDATE? On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? wrote: > Hello. > > I'd like to remove UPDATE value from SIP Accept header when creating SIP > calls. We're sending it in INVITE message and our provider uses that for > in-call keep-alive checks every 10 minutes. > FreeSwitch doesn't respond to that, so our provider disconnects RTP and > call is dropped. According to RFC3311 we can indicate that we don't > support this by not sending UPDATE in Accept header. Is this gonna help? > > Is there any way to drop that from SIP headers from dialplan? Or do I > have to change source code? > Or better, is there any other (cleaner) way to resolve this? > > Regards > > Michal Zubac > ComGate Interactive s.r.o. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From jaasmailing at gmail.com Mon Jan 23 17:58:59 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 23 Jan 2012 15:58:59 +0100 Subject: [Freeswitch-users] Response Custom header in CDR Message-ID: <4F1D75B3.9080601@gmail.com> Hi all, I need to store a custom response header in CDR. Let me explain the scenario. I have a Kamailio (SIP Proxy) that sends invite toFreeswitch (B2BUA)for call processing (prepaid/CDR); the call is then relayed back to kamailio for LCR (it choose the right destination/carrier): user -> kamailio -> freeswitch -> kamailio -> provider I can set a custom header in the reply route of kamailio (X-Provider: 1.2.3.4) but I don't know how take this information in freeswitch... Could ${sip_ph_X-Provider}/${sip_rh_X-Provider}/${sip_bye_X-Provider} CDR? user <- kamailio <- freeswitch <- kamailio (X-Provider) <- provider (100 or 200 response code) I want to store this information in order to do statistics and profits tracking. Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/cf05c4c7/attachment.html From emss.mail at gmail.com Mon Jan 23 16:32:06 2012 From: emss.mail at gmail.com (Eric Masson) Date: Mon, 23 Jan 2012 14:32:06 +0100 Subject: [Freeswitch-users] HEAD & french sounds Message-ID: Hello, I'd like to test HEAD on my home setup to replace my oldish 1.0.6. While installing, gmake cd-sounds-fr-install barfs at me with the following message, Unknown target cd-sounds-fr-install and inspecting the Makefile confirms this. So, what's the status of JIRA FS-2213, please ? Any hope to get it committed anytime soon ? Kind Regards ?ric Masson From dineshkapoor27 at gmail.com Mon Jan 23 17:30:20 2012 From: dineshkapoor27 at gmail.com (Dinesh Kapoor) Date: Mon, 23 Jan 2012 20:00:20 +0530 Subject: [Freeswitch-users] Problems with setinputCallback Message-ID: Hello, I have a functionality that I am trying to implement via Freeswitch and mod_python, in which a user calls in, Freeswitch hangs up the call and then call the user back. All should happen from inside Freeswitch. Here is the code that I had written for it: def handler(session, args): if is_callback_set(app_instance): schedule_callback(app_instance, session.getCallerID()) is_callback_set(app_instance) : checks if we need to callback on this application instance. schedule_callback(app_instance, session.getCallerID()) : function to call the caller back inside schedule_callback, I create a new thread called CallerbackThread, and start the callback in another thread. Inside CallBackThread, following happens: def run(self): new_session = Session(url) if new_session.answered(): new_session.ai = self.app_instance new_session.getCallerID = lambda: new_session.getVariable("destination_number") new_session.setInputCallback(call_back) new_session.isHungup = lambda: new_session.getState() == 'CS_HANGUP' while not new_session.isHungup(): new_session.streamFile("abcd.mp3") new_session.hangup() def call_back(session,what,obj): logger.debug("hi callback") The originate occurs and I am able to hear the file getting streamed correctly. But when I press any dtmf input, then on first input nothing happens, and in the next input I start getting "TypeError : call_back expects 3 args, 0 given" error. I dont know what can be the solution. Any help regarding this is highly appreciated! Regards, Dinesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/a94b02e2/attachment.html From ccesario at tecnomega.com.br Mon Jan 23 18:19:05 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Mon, 23 Jan 2012 13:19:05 -0200 Subject: [Freeswitch-users] Freeswitch + Freetdm errors Message-ID: <4F1D7A69.1070806@tecnomega.com.br> Hello guys, In randon moments when I start the freeswitch service, I get the following errors (patebin link)... This is as infinit loop. And I need restart the service 5 times to it UP. In my try to discovery the problem, I believe that the problem happen when my E1 link cable is disconnected. http://pastebin.freeswitch.org/18203 Somebody have idea about this ? Greats Carlos From anthony.minessale at gmail.com Mon Jan 23 19:06:47 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Jan 2012 10:06:47 -0600 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: <4F1D6344.6000501@comgate.cz> References: <4F1D6344.6000501@comgate.cz> Message-ID: try setting ignore_display_updates=true globally or on the calls in question On Mon, Jan 23, 2012 at 7:40 AM, Michal Zub?? wrote: > Hello. > > I'd like to remove UPDATE value from SIP Accept header when creating SIP > calls. We're sending it in INVITE message and our provider uses that for > in-call keep-alive checks every 10 minutes. > FreeSwitch doesn't respond to that, so our provider disconnects RTP and > call is dropped. According to RFC3311 we can indicate that we don't > support this by not sending UPDATE in Accept header. Is this gonna help? > > Is there any way to drop that from SIP headers from dialplan? Or do I > have to change source code? > Or better, is there any other (cleaner) way to resolve this? > > Regards > > Michal Zubac > ComGate Interactive s.r.o. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/35a33e47/attachment-0001.html From michal.zubac at comgate.cz Mon Jan 23 19:24:59 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Mon, 23 Jan 2012 17:24:59 +0100 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: References: <4F1D6344.6000501@comgate.cz> Message-ID: <4F1D89DB.7030900@comgate.cz> I don't have "enable-timer" or "session-timeout" variables set in Sofia profile, so maybe defaults? I think these are what you are referring to. SIP trace is at http://pastebin.com/DTQU3nMr Yes, provider offered us re-INVITE method as an alternative for UPDATE. Would this one work? What do I have to set up? Michal Zubac ComGate Interactive s.r.o. On 23.1.2012 15:51, Kristian Kielhofner wrote: > What are your Sofia profile session timer values set to? > > Can you post a complete SIP trace? > > Can your provider send a re-INVITE instead of UPDATE? > > On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? wrote: >> Hello. >> >> I'd like to remove UPDATE value from SIP Accept header when creating SIP >> calls. We're sending it in INVITE message and our provider uses that for >> in-call keep-alive checks every 10 minutes. >> FreeSwitch doesn't respond to that, so our provider disconnects RTP and >> call is dropped. According to RFC3311 we can indicate that we don't >> support this by not sending UPDATE in Accept header. Is this gonna help? >> >> Is there any way to drop that from SIP headers from dialplan? Or do I >> have to change source code? >> Or better, is there any other (cleaner) way to resolve this? >> >> Regards >> >> Michal Zubac >> ComGate Interactive s.r.o. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From michal.zubac at comgate.cz Mon Jan 23 19:27:19 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Mon, 23 Jan 2012 17:27:19 +0100 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: References: <4F1D6344.6000501@comgate.cz> Message-ID: <4F1D8A67.4060904@comgate.cz> I already tried that (as channel variable) with no help. FS is still sending UPDATE in Accept header. Michal Zubac ComGate Interactive s.r.o. On 23.1.2012 17:06, Anthony Minessale wrote: > try setting ignore_display_updates=true globally or on the calls in > question > > On Mon, Jan 23, 2012 at 7:40 AM, Michal Zub?? > wrote: > > Hello. > > I'd like to remove UPDATE value from SIP Accept header when > creating SIP > calls. We're sending it in INVITE message and our provider uses > that for > in-call keep-alive checks every 10 minutes. > FreeSwitch doesn't respond to that, so our provider disconnects > RTP and > call is dropped. According to RFC3311 we can indicate that we don't > support this by not sending UPDATE in Accept header. Is this gonna > help? > > Is there any way to drop that from SIP headers from dialplan? Or do I > have to change source code? > Or better, is there any other (cleaner) way to resolve this? > > Regards > > Michal Zubac > ComGate Interactive s.r.o. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From hynek.cihlar at gmail.com Mon Jan 23 21:03:40 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Mon, 23 Jan 2012 19:03:40 +0100 Subject: [Freeswitch-users] sip_gateway dialplan variable Message-ID: Dear, is there a dialplan variable that would hold the gateway name an incoming call came through? Searching through the source code and wiki didn't yield any positive result. Thanks! Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/ac85f478/attachment.html From potxoka at gmail.com Mon Jan 23 21:55:17 2012 From: potxoka at gmail.com (Anto) Date: Mon, 23 Jan 2012 19:55:17 +0100 Subject: [Freeswitch-users] Mod_SPANDSP In-Reply-To: References: Message-ID: Hi, Thanks ;-) Regards ! 2012/1/3 Michael Collins : > You may want to check out the "execute_on_fax*" channel variables: > > http://wiki.freeswitch.org/wiki/Mod_spandsp#Execute_based_on_fax_session_outcome > > These allow for some elegant solutions. > > -MC > > On Mon, Jan 2, 2012 at 5:46 AM, Anto wrote: >> >> hello >> >> I'm adapting the files are in >> http://wiki.freeswitch.org/wiki/Mod_spandsp to make faxing my >> business. I wanted to know whether to send the fax, an error occurs >> (destination busy, etc.), this is still retrying successive times or >> is there a way to handle this. I've been looking fax_result_code and >> fax_result_text, to make a correct treatment but have not found >> information on them. Does anyone have any examples or can guide me? >> Thank you very much >> >> Best Regards >> Anto >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kris at kriskinc.com Mon Jan 23 22:57:52 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 23 Jan 2012 14:57:52 -0500 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: <4F1D89DB.7030900@comgate.cz> References: <4F1D6344.6000501@comgate.cz> <4F1D89DB.7030900@comgate.cz> Message-ID: Michal, I can barely read these parsed out SIP traces (yuck) but from the skimming I did it looks like you'd benefit from a few changes: - Explicitly disable session timers in your Sofia profile - Set ignore_display_updates=true (so FreeSWITCH doesn't send an UPDATE - won't help in this case but a good thing to do) - Request your provider uses re-INVITEs instead of UPDATE On Mon, Jan 23, 2012 at 11:24 AM, Michal Zub?? wrote: > I don't have "enable-timer" or "session-timeout" variables set in Sofia > profile, so maybe defaults? I think these are what you are referring to. > > SIP trace is at > ? http://pastebin.com/DTQU3nMr > > Yes, provider offered us re-INVITE method as an alternative for UPDATE. > Would this one work? What do I have to set up? > > Michal Zubac > ComGate Interactive s.r.o. > > > On 23.1.2012 15:51, Kristian Kielhofner wrote: >> What are your Sofia profile session timer values set to? >> >> Can you post a complete SIP trace? >> >> Can your provider send a re-INVITE instead of UPDATE? >> >> On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? ?wrote: >>> Hello. >>> >>> I'd like to remove UPDATE value from SIP Accept header when creating SIP >>> calls. We're sending it in INVITE message and our provider uses that for >>> in-call keep-alive checks every 10 minutes. >>> FreeSwitch doesn't respond to that, so our provider disconnects RTP and >>> call is dropped. According to RFC3311 we can indicate that we don't >>> support this by not sending UPDATE in Accept header. Is this gonna help? >>> >>> Is there any way to drop that from SIP headers from dialplan? Or do I >>> have to change source code? >>> Or better, is there any other (cleaner) way to resolve this? >>> >>> Regards >>> >>> Michal Zubac >>> ComGate Interactive s.r.o. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From brad at tech21.com Mon Jan 23 23:26:30 2012 From: brad at tech21.com (Brad Mina) Date: Mon, 23 Jan 2012 12:26:30 -0800 Subject: [Freeswitch-users] HEAD & french sounds In-Reply-To: References: Message-ID: Although I'm not sure if the installation automates any other languages other than English, you can find language specific sounds here: http://files.freeswitch.org http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-8000-1.0.14.tar.gz http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-16000-1.0.14.tar.gz http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-32000-1.0.14.tar.gz http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-48000-1.0.14.tar.gz On Mon, Jan 23, 2012 at 5:32 AM, Eric Masson wrote: > Hello, > > I'd like to test HEAD on my home setup to replace my oldish 1.0.6. > > While installing, gmake cd-sounds-fr-install barfs at me with the > following message, Unknown target cd-sounds-fr-install and inspecting > the Makefile confirms this. > > So, what's the status of JIRA FS-2213, please ? > > Any hope to get it committed anytime soon ? > > Kind Regards > > ?ric Masson > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/6417641f/attachment.html From errotan at elder.hu Tue Jan 24 01:15:32 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Mon, 23 Jan 2012 23:15:32 +0100 Subject: [Freeswitch-users] Invalid authorization header In-Reply-To: References: Message-ID: <4F1DDC04.70401@elder.hu> Hi. Please put your console log of a failed inbound call to pastebin so we can help. Turn siptrace on for your external profile for additional information. 2012-01-23 00:04 keltez?ssel, Ruslan ?rta: > Hi please somebody help me!! I have this message "Invalid > authorization header" on every inbound call I make. I couldn't find > any reference (apart from chunk of c code, which I can't comprehend). > > My sip config looks like that: > > > > > > > > > > > > > > > > > > Thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/099835a3/attachment-0001.html From wstephen80 at gmail.com Tue Jan 24 01:20:10 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 23 Jan 2012 23:20:10 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: Can be useful to run Freeswitch with "-nosql" option? What I lose in this case? On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde wrote: > I'm already using the ramdisk. > The problem happens when I have a provider that give congestion. > In this case Freeswitch receives many tries but few connected calls and > the number of session per second is high. > To avoid the "event system overloading" (avoiding to lower the global > session per second 'sps' parameter) I have insert in dialplan: > > break="never"> > > > > In this way I have limited the session rate for the congestioned > destination where I have so many tries. > > My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage > of disk is near zero (dstat) why I cannot handle this session rate? > > > > On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> you can tell that by taking system vitals its hard to tell from the small >> amount of data. >> I do know that to get those errors, you have to push the core so hard >> that the sql stmts queuing up for transactions are getting too large for >> the rate at which they are written to the DB. Try a ramdisk like Michael >> suggested. >> >> >> On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: >> >>> I'm using ram disk for the FS database "freeswitch/db". >>> Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid >>> 1) affect the performance? >>> >>> >>> On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> the box can't handle the load, the disk io from the sql stmts is >>>> backing up the events. >>>> get a nicer box with the money saved from free softswitch =p >>>> >>>> >>>> On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo >>> > wrote: >>>> >>>>> Did you bind any callbacks to events that might be taking a long time >>>>> to process? >>>>> >>>>> >>>>> >>>>> On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: >>>>> >>>>>> I saw "increase the event system capacity" because in the log there >>>>>> was a row: >>>>>> >>>>>> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing >>>>>> things down. >>>>>> >>>>>> Where it seems that all event dispatch threads are "busy" but I see >>>>>> that the cpu has many idle cycles so why not increase the number of >>>>>> dispatch threads? >>>>>> Or I'm wrong? >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/1380d687/attachment.html From errotan at elder.hu Tue Jan 24 01:22:24 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Mon, 23 Jan 2012 23:22:24 +0100 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: Message-ID: <4F1DDDA0.1080805@elder.hu> Hi, Does your softphone runs on the same machine as freeswitch ? Maybe your softphone already binds to the 5060 port and freeswitch can't start the internal profile which also listens on 5060. What it is the output of "sofia status" command ? 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: > Hi, > > I have installed Freeswitch from git in my Ubuntu 11.10. There are > some default sip users in prefix/conf/directory/default/1000.xml to > 1019.xml. But when I try to register one extension to IP Phone or > Softphone the account is not registered and responding as > Authentication failuere. 403 Forbidden error, even though I have > changed the password in 1000.xml file. > > Also in FS CLI I am not able to see the output for "sofia status > profile internal". > > Please advice. > > Regards. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/321993c5/attachment.html From mgende at gendesign.com Tue Jan 24 01:36:43 2012 From: mgende at gendesign.com (Michael Gende) Date: Mon, 23 Jan 2012 16:36:43 -0600 Subject: [Freeswitch-users] CAMA Stack Message-ID: I'm just starting a search on this subject, but does a stack exist for FS implementing CAMA, or at least a subset of that protocol? Thanks in Advance for any info, an email address of someone working on this now, or anyone who has considered it. Regards, Mike G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/750d2e08/attachment-0001.html From krice at freeswitch.org Tue Jan 24 01:44:12 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 23 Jan 2012 16:44:12 -0600 Subject: [Freeswitch-users] SouthAmerican Users Message-ID: Hey guys, Maybe one of you can help me out... Looking for someone in South America (Brazil preferably) that has good Colocation Space K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/bb85a232/attachment.html From itamar at ispbrasil.com.br Tue Jan 24 01:51:19 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Mon, 23 Jan 2012 20:51:19 -0200 Subject: [Freeswitch-users] SouthAmerican Users In-Reply-To: References: Message-ID: On Mon, Jan 23, 2012 at 8:44 PM, Ken Rice wrote: > Hey guys, > > Maybe one of you can help me out... Looking for someone in South America > (Brazil preferably) that has good Colocation Space > > K I think I can help you. please contact me ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From itamar at ispbrasil.com.br Tue Jan 24 01:58:37 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Mon, 23 Jan 2012 20:58:37 -0200 Subject: [Freeswitch-users] SouthAmerican Users In-Reply-To: References: Message-ID: On Mon, Jan 23, 2012 at 8:44 PM, Ken Rice wrote: > Hey guys, > > Maybe one of you can help me out... Looking for someone in South America > (Brazil preferably) that has good Colocation Space > give a call and I will try to help you +1-941-870-1670 -- ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From ruslan at symsol.com.au Tue Jan 24 02:12:04 2012 From: ruslan at symsol.com.au (Ruslan) Date: Tue, 24 Jan 2012 10:12:04 +1100 Subject: [Freeswitch-users] Invalid authorization header In-Reply-To: <4F1DDC04.70401@elder.hu> References: <4F1DDC04.70401@elder.hu> Message-ID: yeah, thanks for that. I got that with ngrep http://pastebin.freeswitch.org/18211 How do I set up the proxy credentials? I do have username, auth-user and password in my config. Thank you so much On Tue, Jan 24, 2012 at 9:15 AM, Pusk?s Zsolt wrote: > Hi. > > Please put your console log of a failed inbound call to pastebin so we can > help. Turn siptrace on for your external profile for additional information. > > 2012-01-23 00:04 keltez?ssel, Ruslan ?rta: > > Hi please somebody help me!! I have this message "Invalid authorization > header" on every inbound call I make. I couldn't find any reference (apart > from chunk of c code, which I can't comprehend). > > My sip config looks like that: > > > > > > > > > > > > > > > > > > Thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/281feb53/attachment.html From mario.mania at gmail.com Tue Jan 24 03:06:09 2012 From: mario.mania at gmail.com (Mario Augusto Mania) Date: Tue, 24 Jan 2012 01:06:09 +0100 Subject: [Freeswitch-users] SouthAmerican Users In-Reply-To: References: Message-ID: Hi What is a "good colocation" for you? What is your needs? I'm from brazil. Mario 2012/1/23 Ken Rice > Hey guys, > > Maybe one of you can help me out... Looking for someone in South America > (Brazil preferably) that has good Colocation Space > > K > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/81ed67de/attachment.html From vetali100 at gmail.com Tue Jan 24 07:17:37 2012 From: vetali100 at gmail.com (Vitaly Colosov) Date: Mon, 23 Jan 2012 20:17:37 -0800 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: <4F1DDDA0.1080805@elder.hu> References: <4F1DDDA0.1080805@elder.hu> Message-ID: <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> After recent change you need to type "sofia status profile internal reg" Sent from my iPad On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: > Hi, > > Does your softphone runs on the same machine as freeswitch ? Maybe your softphone already binds to the 5060 port and freeswitch can't start the internal profile which also listens on 5060. > > What it is the output of "sofia status" command ? > > > 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >> >> Hi, >> >> I have installed Freeswitch from git in my Ubuntu 11.10. There are some default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. But when I try to register one extension to IP Phone or Softphone the account is not registered and responding as Authentication failuere. 403 Forbidden error, even though I have changed the password in 1000.xml file. >> >> Also in FS CLI I am not able to see the output for "sofia status profile internal". >> >> Please advice. >> >> Regards. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120123/503a2667/attachment-0001.html From amit.nakum2009 at gmail.com Tue Jan 24 10:13:24 2012 From: amit.nakum2009 at gmail.com (amit nakum) Date: Tue, 24 Jan 2012 12:43:24 +0530 Subject: [Freeswitch-users] How to Set Call Priority in mod_callcenter In-Reply-To: References: Message-ID: I also try this scenario with this parameter but result is same. On Mon, Jan 23, 2012 at 12:25 PM, Brian Foster wrote: > Hola, > > Try this: > > > > -BDF > > On Mon, Jan 23, 2012 at 1:23 AM, amit nakum wrote: > >> Dear All, >> >> I have set queue parameter to > value="system"/>,so now how to raise some caller's score,allows them to >> receive priority over other normal calls that might have been in the queue >> longer. >> >> I have test above scenario using > value="system"/> which is not fulfill my requirement,if there is any >> parameter to set this scenario. >> >> Pl help me to set this scenario. >> >> >> Thanks and Regards >> Amit Nakum >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-429-1069 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/9df64a4e/attachment.html From hynek.cihlar at gmail.com Tue Jan 24 10:27:38 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 24 Jan 2012 08:27:38 +0100 Subject: [Freeswitch-users] sip_gateway dialplan variable In-Reply-To: References: Message-ID: In one of the examples on wiki I found ${sip_gateway}, but that doesn't seem to work. Also, there is ${sip_gateway_name}, but that works on outbound calls only. Hynek On Mon, Jan 23, 2012 at 7:03 PM, Hynek Cihlar wrote: > Dear, is there a dialplan variable that would hold the gateway name an > incoming call came through? Searching through the source code and wiki > didn't yield any positive result. > > Thanks! > > Hynek > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/4db374fe/attachment.html From mytemike72 at gmail.com Tue Jan 24 10:42:40 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Tue, 24 Jan 2012 08:42:40 +0100 Subject: [Freeswitch-users] sip_gateway dialplan variable In-Reply-To: References: Message-ID: Hi Hynek, {sip_req_host} should give your own gateway (local), and {network_addr} should give the remote gateway. Otherwise writing a cdr and look will help you with this. Regards, Mike Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar het volgende geschreven: > In one of the examples on wiki I found ${sip_gateway}, but that doesn't seem to work. Also, there is ${sip_gateway_name}, but that works on outbound calls only. > > Hynek > > > > On Mon, Jan 23, 2012 at 7:03 PM, Hynek Cihlar wrote: > Dear, is there a dialplan variable that would hold the gateway name an incoming call came through? Searching through the source code and wiki didn't yield any positive result. > > Thanks! > > Hynek > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/921c426c/attachment.html From gopalakrishnan.an at gmail.com Tue Jan 24 12:35:46 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Tue, 24 Jan 2012 15:05:46 +0530 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: I am not able to run "sofia status profile internal reg", In my fs_cli I am getting till sofia status profile external not internal, I think my internal profile is not activated. On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: > After recent change you need to type "sofia status profile internal reg" > > Sent from my iPad > > On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: > > Hi, > > Does your softphone runs on the same machine as freeswitch ? Maybe your > softphone already binds to the 5060 port and freeswitch can't start the > internal profile which also listens on 5060. > > What it is the output of "sofia status" command ? > > > 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: > > Hi, > > I have installed Freeswitch from git in my Ubuntu 11.10. There are some > default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. > But when I try to register one extension to IP Phone or Softphone the > account is not registered and responding as Authentication failuere. 403 > Forbidden error, even though I have changed the password in 1000.xml file. > > Also in FS CLI I am not able to see the output for "sofia status profile > internal". > > Please advice. > > Regards. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/0d484460/attachment-0001.html From gopalakrishnan.an at gmail.com Tue Jan 24 12:47:57 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Tue, 24 Jan 2012 15:17:57 +0530 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: My "sofia status" output is, freeswitch at ubuntu> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG ================================================================================================= 1 profile 0 aliases freeswitch at ubuntu> On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > I am not able to run "sofia status profile internal reg", In my fs_cli I > am getting till sofia status profile external not internal, I think my > internal profile is not activated. > > > On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: > >> After recent change you need to type "sofia status profile internal reg" >> >> Sent from my iPad >> >> On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: >> >> Hi, >> >> Does your softphone runs on the same machine as freeswitch ? Maybe your >> softphone already binds to the 5060 port and freeswitch can't start the >> internal profile which also listens on 5060. >> >> What it is the output of "sofia status" command ? >> >> >> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >> >> Hi, >> >> I have installed Freeswitch from git in my Ubuntu 11.10. There are some >> default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. >> But when I try to register one extension to IP Phone or Softphone the >> account is not registered and responding as Authentication failuere. 403 >> Forbidden error, even though I have changed the password in 1000.xml file. >> >> Also in FS CLI I am not able to see the output for "sofia status >> profile internal". >> >> Please advice. >> >> Regards. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/5ba761ee/attachment.html From jaasmailing at gmail.com Tue Jan 24 17:18:29 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Tue, 24 Jan 2012 15:18:29 +0100 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario In-Reply-To: <4F183B18.40203@gmail.com> References: <4F183B18.40203@gmail.com> Message-ID: <4F1EBDB5.7010805@gmail.com> Does anybody use fs in a full high availability environment and/or call recovery feature? Regards, Il 19/01/12 16.47, Carlo Dimaggio ha scritto: > Hi all, > > I would like to know if the track-calls feature could be used in a > multi-primary - multi-backup scenario. > What I think is an environment with Kamailio dispatcher to N > freeswitch boxes with a (multi or) single freeswitch backup. In case > of failure, one primary should be replaced with a backup (spare) box > that will take its IP and active calls. > > If all active freeswitch track call state in the same table/database, > how the backup box will know what calls should be recovered? > Anyone could explain the call recovery algorithm? > > > Best Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/ae58a252/attachment.html From bdfoster at endigotech.com Tue Jan 24 18:08:04 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 24 Jan 2012 10:08:04 -0500 Subject: [Freeswitch-users] How to Set Call Priority in mod_callcenter In-Reply-To: References: Message-ID: You also need to play with this variable, link: http://wiki.freeswitch.org/wiki/Mod_callcenter#cc_base_score On Jan 24, 2012 2:14 AM, "amit nakum" wrote: > I also try this scenario with this parameter value="queue"/> but result is same. > > On Mon, Jan 23, 2012 at 12:25 PM, Brian Foster wrote: > >> Hola, >> >> Try this: >> >> >> >> -BDF >> >> On Mon, Jan 23, 2012 at 1:23 AM, amit nakum wrote: >> >>> Dear All, >>> >>> I have set queue parameter to >> value="system"/>,so now how to raise some caller's score,allows them to >>> receive priority over other normal calls that might have been in the queue >>> longer. >>> >>> I have test above scenario using >> value="system"/> which is not fulfill my requirement,if there is any >>> parameter to set this scenario. >>> >>> Pl help me to set this scenario. >>> >>> >>> Thanks and Regards >>> Amit Nakum >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-429-1069 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/f2f3a9f4/attachment-0001.html From moises.silva at gmail.com Tue Jan 24 19:31:56 2012 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 24 Jan 2012 11:31:56 -0500 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: Message-ID: On Mon, Jan 23, 2012 at 5:36 PM, Michael Gende wrote: > I'm just starting a search on this subject, but does a stack exist for FS > implementing CAMA, or at least a subset of that protocol? > > Someone asked me about this a while ago, can't remember nor find his name or email. Just to confirm, are you talking about this?: http://en.wikipedia.org/wiki/Automatic_message_accounting I don't think there is a stack or code to handle that in the open source community (though, you never know what is out there). From what I see requires MF tones, which at least spandsp is able to generate and detect. Some logic is required on top to interpret and respond to the tones accordingly. *Moises Silva **Software Engineer, Development Manager*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/aaeacfd1/attachment.html From henrikaagaardsorensen at gmail.com Tue Jan 24 12:33:11 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Tue, 24 Jan 2012 10:33:11 +0100 Subject: [Freeswitch-users] Rotate cdr-csv on hub without rotating log-files. Message-ID: It does not seem possible to rotate cdr-csv files on kill -HUB ... without also having to rotate the log-files. Is that correct? When setting rotate-on-hub to false in the log-conf, but true in cdr-csv nothing happings when HUB'ing. From henrikaagaardsorensen at gmail.com Tue Jan 24 13:17:32 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Tue, 24 Jan 2012 11:17:32 +0100 Subject: [Freeswitch-users] fs_cli -x 'cdr_csv rotate' does not always rotate cdr_csv-files. Message-ID: I've setup a cron script every 5 minute running the command: fs_cli -x 'cdr_csv rotate' which should rotate the cdr_csv-files. But sometimes this does not happen. Have anyone had experience with such and can anyone comment on it? From starach at gmail.com Tue Jan 24 13:15:14 2012 From: starach at gmail.com (Sebastian Tarach) Date: Tue, 24 Jan 2012 11:15:14 +0100 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem Message-ID: Hello, I'm pretty new to the whole idea of using Linux as telephone central . I tried to setup Asterisk at first but failed. Although I have managed to install it I was unable to "call my computer". That's why I was hoping to deal with FreeSWITCH better. Please tell a noob like myself how to make this finally work. Keep in mind that something that is clear to you to me is probably a dark magic. For instance in installation guide there are mentioned few different ways to connect to a modem. Which method should I use if it comes to my modem : $ lspci -d 8086:24c6 -vvv 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) Subsystem: Toshiba America Info Systems Device 0001 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- SERR- Hi I would like to use freeswitch as a Voip Gate way like Analog,ISDN,GSM. Is this possible? if yes provide some hint. Thanks From gabler at abx.de Tue Jan 24 18:54:33 2012 From: gabler at abx.de (Andreas Gabler, ABX) Date: Tue, 24 Jan 2012 16:54:33 +0100 Subject: [Freeswitch-users] missing digit from destination number, overlap dialing Message-ID: Dear all, May I ask you for help? We use Freeswitch with Sangoma B700 FlexBRI on Windows XP professional SP3. We have 2 BRI lines. Some external calls lack the last digit of the destination number. Many destination numbers are ok. It depends from the source of the call. Calls from German mobile network have complete number. Many calls from German landline have complete number. A few calls from German landline lack the last digit. This callers hear "Called number is not available". Freeswitch received the complete number from 68% of the calling numbers at all times. This means 7 digits after the country code. Incomplete number from 14% at all times. Only 6 digits. 18% varied. Sometimes we received 6 digits. Sometimes 7. I tried in the file freetdm.conf.xml. This makes the external caller do not receive an answer. He hears nothing. ---------------------------------------------------------------------- This is an extraction of the freeswitch.log without overlap=yes: Line numbers 02, 04 and 19 lack the last digit. Line numbers 01, 03 and 10 from mobile network. Line number 13 anonymous caller. Other from landline. 01 incoming call: called no:[4166065] calling no:[1722051138] 02 incoming call: called no:[416604] calling no:[3517969878] 03 incoming call: called no:[4166030] calling no:[1738920856] 04 incoming call: called no:[416603] calling no:[6423969575] 05 incoming call: called no:[4166033] calling no:[6423969575] 06 incoming call: called no:[4166041] calling no:[35284193228] 07 incoming call: called no:[4166038] calling no:[970891000] 08 incoming call: called no:[4166012] calling no:[3514129670] 09 incoming call: called no:[4166012] calling no:[35284193212] 10 incoming call: called no:[4166054] calling no:[1733583343] 11 incoming call: called no:[4166052] calling no:[35284153322] 12 incoming call: called no:[4166021] calling no:[3571608532] 13 incoming call: called no:[4166020] calling no:[] 14 incoming call: called no:[4166030] calling no:[2518347351] 15 incoming call: called no:[4166033] calling no:[352371791] 16 incoming call: called no:[4166054] calling no:[35284595951] 17 incoming call: called no:[4166030] calling no:[2518347351] 18 incoming call: called no:[4166019] calling no:[35183142120] 19 incoming call: called no:[416602] calling no:[351828240] 20 incoming call: called no:[4166010] calling no:[7022609250] ---------------------------------------------------------------------- This is a part of freeswitch.log without overlap=yes: Calling from 35284041831 to 4166069 is ok. 07:59:04.835904 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s2c1][2:1] Incoming call: Called No:[4166069] Calling No:[35284041831] 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:199 [s2c1][2:1] Changed state from DOWN to RING 07:59:04.835904 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for RING 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to RING 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:647 [s2c1][2:1] Sending incoming call from 35284041831 to 4166069 to FTDM core 07:59:04.835904 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from DOWN to RING in 1ms ... ---------------------------------------------------------------------- This is a part of freeswitch.log with overlap=yes: Calling from 35284041728 to 4166065 failed. 10:17:43.696875 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s2c1][2:1] Incoming call: Called No:[4166065] Calling No:[35284041728] 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:197 [s2c1][2:1] Changed state from DOWN to COLLECT 10:17:43.696875 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for COLLECT 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to COLLECT 10:17:43.696875 [INFO] ftmod_sangoma_isdn_stack_out.c:102 [s2c1][2:1] Sending SETUP ACK (suId:1 suInstId:5 spInstId:5 dchan:2 ces:0) 10:17:43.696875 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from DOWN to COLLECT in 0ms 10:17:55.775000 [INFO] ftmod_sangoma_isdn_stack_rcv.c:172 [s2c1][2:1] Received INFO (suId:1 suInstId:5 spInstId:5 ces:0) 10:17:55.775000 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:368 [s2c1][2:1] Processing INFO (suId:1 suInstId:5 spInstId:5 ces:0) 10:17:55.775000 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:464 [s2c1][2:1] Processing INFO (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [INFO] ftmod_sangoma_isdn_stack_rcv.c:210 [s2c1][2:1] Received DISCONNECT (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:523 [s2c1][2:1] Processing DISCONNECT (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:547 [s2c1][2:1] Changed state from COLLECT to CANCEL 10:18:08.337500 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for CANCEL 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to CANCEL 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:753 [s2c1][2:1] Hanging up call before informing user! 10:18:08.337500 [INFO] ftmod_sangoma_isdn_stack_out.c:420 [s2c1][2:1] Sending RELEASE/RELEASE COMPLETE (suId:1 suInstId:5 spInstId:5) 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:759 [s2c1][2:1] Completed state change from COLLECT to CANCEL in 0ms 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:759 [s2c1][2:1] Changed state from CANCEL to HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftdm_state.c:511 [s2c1][2:1] Executing state processor for HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:623 [s2c1][2:1] processing state change to HANGUP_COMPLETE 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:831 [s2c1][2:1] Waiting for release from stack 10:18:08.337500 [DEBUG] ftmod_sangoma_isdn.c:884 [s2c1][2:1] Completed state change from CANCEL to HANGUP_COMPLETE in 1ms ---------------------------------------------------------------------- You can download an attachment (3 MBytes): http://www.abx.de/temp/freeswitch/2012-01-24/attachment.zip The attachment contains dialplan: default.xml default-anrufverteilung-nach-isdn.xml public.xml (Public directory is empty.) config: freeswitch.xml freetdm.conf freetdm.conf.xml Wanpipe.pdf log: full freeswitch.log ---------------------------------------------------------------------- Thank you for help. Best regards Andreas Gabler gabler at abx.de ------------------------------------------- ABX advanced biochemical compounds Heinrich-Glaeser-Str. 10-14 D-01454 Radeberg Germany telephone: +49 3528 4041 740 facsimile: +49 3528 4041 8877 email: info at abx.de web: www.abx.de ------------------------------------------- general manager: Dr. Peter Moll company headquarter: D-01454 Radeberg VAT number: DE 812 136 673 Commercial Registry: HRB 14041 Amtsgericht Dresden tax authority: Finanzamt 3213 Hoyerswerda tax number: 213/105/00531 ------------------------------------------- Please, use plain text format instead of html in your emails. Thank you. ------------------------------------------- From krice at freeswitch.org Tue Jan 24 20:01:59 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 24 Jan 2012 11:01:59 -0600 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: Message-ID: Unfortunately most softmodems are not supported as a Telephony interface... The problem is lack of drivers. The simple answer is it wont work... You're best bet is to get an ATA with both FXO and FXS ports and connect via SIP... On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: > Hello, > > I'm pretty new to the whole idea of using Linux as telephone central . > I tried to setup Asterisk at first but failed. Although I have managed > to install it I was unable to "call my computer". That's why I was > hoping to deal with FreeSWITCH better. Please tell a noob like myself > how to make this finally work. Keep in mind that something that is > clear to you to me is probably a dark magic. For instance in > installation guide there are mentioned few different ways to connect > to a modem. Which method should I use if it comes to my modem : > > > $ lspci -d 8086:24c6 -vvv > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) > Subsystem: Toshiba America Info Systems Device 0001 > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> TAbort- SERR- Latency: 0 > Interrupt: pin B routed to IRQ 4 > Region 0: I/O ports at 2400 [size=256] > Region 1: I/O ports at 2000 [size=128] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA > PME(D0+,D1-,D2-,D3hot+,D3cold+) > Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Intel ICH Modem > > Thanks in advance > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Tue Jan 24 20:02:42 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 24 Jan 2012 11:02:42 -0600 Subject: [Freeswitch-users] Freeswitch as gateway In-Reply-To: <4F1EA660.8080408@cem-solutions.net> Message-ID: Easy Answer Yes... See FreeTDM on the FreeSWITCH Wiki On 1/24/12 6:38 AM, "Nikhil" wrote: > > Hi > > I would like to use freeswitch as a Voip Gate way like > Analog,ISDN,GSM. Is this possible? if yes provide some hint. > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Jan 24 20:20:53 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Jan 2012 09:20:53 -0800 Subject: [Freeswitch-users] fs_cli -x 'cdr_csv rotate' does not always rotate cdr_csv-files. In-Reply-To: References: Message-ID: is there any information in the freeswitch.log file? -MC 2012/1/24 Henrik Aagaard S?rensen > I've setup a cron script every 5 minute running the command: > > fs_cli -x 'cdr_csv rotate' > > which should rotate the cdr_csv-files. But sometimes this does not happen. > > Have anyone had experience with such and can anyone comment on it? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/4c6a2c27/attachment.html From msc at freeswitch.org Tue Jan 24 20:21:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Jan 2012 09:21:59 -0800 Subject: [Freeswitch-users] Rotate cdr-csv on hub without rotating log-files. In-Reply-To: References: Message-ID: This might be a bug. Test on latest git and if you can reproduce it then open a tick on jira.freeswitch.org. -MC 2012/1/24 Henrik Aagaard S?rensen > It does not seem possible to rotate cdr-csv files on kill -HUB ... > without also having to rotate the log-files. Is that correct? > When setting rotate-on-hub to false in the log-conf, but true in > cdr-csv nothing happings when HUB'ing. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/c817aca1/attachment.html From msc at freeswitch.org Tue Jan 24 20:35:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Jan 2012 09:35:06 -0800 Subject: [Freeswitch-users] Invalid authorization header In-Reply-To: References: <4F1DDC04.70401@elder.hu> Message-ID: You are better off enabling sip trace in the freeswitch console and getting the sip trace and console logs together. From fs_cli: sofia global siptrace on Make the call, capturing the console output and then drop it into a pastebin. Be sure to use "FreeSWITCH Log" as the syntax highlighting so that we have pretty colors. :) -MC On Mon, Jan 23, 2012 at 3:12 PM, Ruslan wrote: > yeah, thanks for that. I got that with ngrep > > http://pastebin.freeswitch.org/18211 > > How do I set up the proxy credentials? I do have username, auth-user and > password in my config. > > Thank you so much > > > > > On Tue, Jan 24, 2012 at 9:15 AM, Pusk?s Zsolt wrote: > >> Hi. >> >> Please put your console log of a failed inbound call to pastebin so we >> can help. Turn siptrace on for your external profile for additional >> information. >> >> 2012-01-23 00:04 keltez?ssel, Ruslan ?rta: >> >> Hi please somebody help me!! I have this message "Invalid authorization >> header" on every inbound call I make. I couldn't find any reference (apart >> from chunk of c code, which I can't comprehend). >> >> My sip config looks like that: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Thanks >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/d2fe7d2f/attachment-0001.html From msc at freeswitch.org Tue Jan 24 20:37:34 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Jan 2012 09:37:34 -0800 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: Try: sofia profile internal reload If you get any errors on output then put them in pastebin and let us know. -MC On Tue, Jan 24, 2012 at 1:47 AM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > My "sofia status" output is, > freeswitch at ubuntu> sofia status > > Name Type Data > State > > ================================================================================================= > external profile > sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > > ================================================================================================= > 1 profile 0 aliases > > freeswitch at ubuntu> > > > On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> I am not able to run "sofia status profile internal reg", In my fs_cli I >> am getting till sofia status profile external not internal, I think my >> internal profile is not activated. >> >> >> On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: >> >>> After recent change you need to type "sofia status profile internal reg" >>> >>> Sent from my iPad >>> >>> On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: >>> >>> Hi, >>> >>> Does your softphone runs on the same machine as freeswitch ? Maybe your >>> softphone already binds to the 5060 port and freeswitch can't start the >>> internal profile which also listens on 5060. >>> >>> What it is the output of "sofia status" command ? >>> >>> >>> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >>> >>> Hi, >>> >>> I have installed Freeswitch from git in my Ubuntu 11.10. There are >>> some default sip users in prefix/conf/directory/default/1000.xml to >>> 1019.xml. But when I try to register one extension to IP Phone or Softphone >>> the account is not registered and responding as Authentication failuere. >>> 403 Forbidden error, even though I have changed the password in 1000.xml >>> file. >>> >>> Also in FS CLI I am not able to see the output for "sofia status >>> profile internal". >>> >>> Please advice. >>> >>> Regards. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/6b8b8dbe/attachment.html From curriegrad2004 at gmail.com Tue Jan 24 20:43:24 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 24 Jan 2012 09:43:24 -0800 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: Don't even bother getting a x100p or it's clones either. It's been reported to work very poorly as an fxo. If you dug deeper you'll find that it's really a softmodem by Intel. On 2012-01-24 9:02 AM, "Ken Rice" wrote: > Unfortunately most softmodems are not supported as a Telephony interface... > The problem is lack of drivers. The simple answer is it wont work... > > You're best bet is to get an ATA with both FXO and FXS ports and connect > via > SIP... > > > On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: > > > Hello, > > > > I'm pretty new to the whole idea of using Linux as telephone central . > > I tried to setup Asterisk at first but failed. Although I have managed > > to install it I was unable to "call my computer". That's why I was > > hoping to deal with FreeSWITCH better. Please tell a noob like myself > > how to make this finally work. Keep in mind that something that is > > clear to you to me is probably a dark magic. For instance in > > installation guide there are mentioned few different ways to connect > > to a modem. Which method should I use if it comes to my modem : > > > > > > $ lspci -d 8086:24c6 -vvv > > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) > > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) > > Subsystem: Toshiba America Info Systems Device 0001 > > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > > ParErr- Stepping- SERR- FastB2B- DisINTx- > > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > >> TAbort- SERR- > Latency: 0 > > Interrupt: pin B routed to IRQ 4 > > Region 0: I/O ports at 2400 [size=256] > > Region 1: I/O ports at 2000 [size=128] > > Capabilities: [50] Power Management version 2 > > Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA > > PME(D0+,D1-,D2-,D3hot+,D3cold+) > > Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- > > Kernel driver in use: Intel ICH Modem > > > > Thanks in advance > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/92a62e70/attachment.html From curriegrad2004 at gmail.com Tue Jan 24 20:48:10 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 24 Jan 2012 09:48:10 -0800 Subject: [Freeswitch-users] WebRTC ... Google Real Time Communication In-Reply-To: References: Message-ID: For the windows users out there the vcproj files for iSAC is on the mainline git tree. If you're interested you're more than welcome to try it out and see how well it works for your application. On 2012-01-20 12:53 PM, "Michael Collins" wrote: > Is anyone out there building anything using WebRTC? Just curious. > -MC > > On Fri, Jan 20, 2012 at 12:04 PM, EdPimentl wrote: > >> http://blog.chromium.org/2012/01/real-time-communications-in-chrome.html >> >> Just think of the possibilities with FS >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/ce0d6554/attachment-0001.html From krice at freeswitch.org Tue Jan 24 20:48:16 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 24 Jan 2012 11:48:16 -0600 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: Message-ID: Not to mention intel discontinued the chipset used on that card ages ago... You?ll be lucky to find them today On 1/24/12 11:43 AM, "curriegrad2004" wrote: > Don't even bother getting a x100p or it's clones either. It's been reported to > work very poorly as an fxo. If you dug deeper you'll find that it's really a > softmodem by Intel. > > On 2012-01-24 9:02 AM, "Ken Rice" wrote: >> Unfortunately most softmodems are not supported as a Telephony interface... >> The problem is lack of drivers. The simple answer is it wont work... >> >> You're best bet is to get an ATA with both FXO and FXS ports and connect via >> SIP... >> >> >> On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: >> >>> > Hello, >>> > >>> > I'm pretty new to the whole idea of using Linux as telephone central . >>> > I tried to setup Asterisk at first but failed. Although I have managed >>> > to install it I was unable to "call my computer". That's why I was >>> > hoping to deal with FreeSWITCH better. Please tell a noob like myself >>> > how to make this finally work. Keep in mind that something that is >>> > clear to you to me is probably a dark magic. For instance in >>> > installation guide there are mentioned few different ways to connect >>> > to a modem. Which method should I use if it comes to my modem : >>> > >>> > >>> > $ lspci -d 8086:24c6 -vvv >>> > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >>> > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >>> > ? ? ? ? Subsystem: Toshiba America Info Systems Device 0001 >>> > ? ? ? ? Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >>> > ParErr- Stepping- SERR- FastB2B- DisINTx- >>> > ? ? ? ? Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >>>> >> TAbort- SERR- >> > ? ? ? ? Latency: 0 >>> > ? ? ? ? Interrupt: pin B routed to IRQ 4 >>> > ? ? ? ? Region 0: I/O ports at 2400 [size=256] >>> > ? ? ? ? Region 1: I/O ports at 2000 [size=128] >>> > ? ? ? ? Capabilities: [50] Power Management version 2 >>> > ? ? ? ? ? ? ? ? Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >>> > PME(D0+,D1-,D2-,D3hot+,D3cold+) >>> > ? ? ? ? ? ? ? ? Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >>> > ? ? ? ? Kernel driver in use: Intel ICH Modem >>> > >>> > Thanks in advance >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/148c7acb/attachment.html From gcd at i.ph Tue Jan 24 20:51:26 2012 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 25 Jan 2012 01:51:26 +0800 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: FreeSWITCH doesn't need softmodem unlike Asterisk where an X100P (or a Digium card) is needed as a timing device. On Tue, Jan 24, 2012 at 6:15 PM, Sebastian Tarach wrote: > Hello, > > I'm pretty new to the whole idea of using Linux as telephone central . > I tried to setup Asterisk at first but failed. Although I have managed > to install it I was unable to "call my computer". That's why I was > hoping to deal with FreeSWITCH better. Please tell a noob like myself > how to make this finally work. Keep in mind that something that is > clear to you to me is probably a dark magic. For instance in > installation guide there are mentioned few different ways to connect > to a modem. Which method should I use if it comes to my modem : > > > $ lspci -d 8086:24c6 -vvv > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) > Subsystem: Toshiba America Info Systems Device 0001 > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > >TAbort- SERR- Latency: 0 > Interrupt: pin B routed to IRQ 4 > Region 0: I/O ports at 2400 [size=256] > Region 1: I/O ports at 2000 [size=128] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA > PME(D0+,D1-,D2-,D3hot+,D3cold+) > Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Intel ICH Modem > > Thanks in advance > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/d37ae9d9/attachment.html From curriegrad2004 at gmail.com Tue Jan 24 20:52:08 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 24 Jan 2012 09:52:08 -0800 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: Yeah, seen a lot of fake x100ps on eBay too so steer clear of that place too. On 2012-01-24 9:49 AM, "Ken Rice" wrote: > Not to mention intel discontinued the chipset used on that card ages > ago... You?ll be lucky to find them today > > > On 1/24/12 11:43 AM, "curriegrad2004" wrote: > > Don't even bother getting a x100p or it's clones either. It's been > reported to work very poorly as an fxo. If you dug deeper you'll find that > it's really a softmodem by Intel. > > On 2012-01-24 9:02 AM, "Ken Rice" wrote: > > Unfortunately most softmodems are not supported as a Telephony interface... > The problem is lack of drivers. The simple answer is it wont work... > > You're best bet is to get an ATA with both FXO and FXS ports and connect > via > SIP... > > > On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: > > > Hello, > > > > I'm pretty new to the whole idea of using Linux as telephone central . > > I tried to setup Asterisk at first but failed. Although I have managed > > to install it I was unable to "call my computer". That's why I was > > hoping to deal with FreeSWITCH better. Please tell a noob like myself > > how to make this finally work. Keep in mind that something that is > > clear to you to me is probably a dark magic. For instance in > > installation guide there are mentioned few different ways to connect > > to a modem. Which method should I use if it comes to my modem : > > > > > > $ lspci -d 8086:24c6 -vvv > > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) > > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) > > Subsystem: Toshiba America Info Systems Device 0001 > > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > > ParErr- Stepping- SERR- FastB2B- DisINTx- > > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium > >> TAbort- SERR- > Latency: 0 > > Interrupt: pin B routed to IRQ 4 > > Region 0: I/O ports at 2400 [size=256] > > Region 1: I/O ports at 2000 [size=128] > > Capabilities: [50] Power Management version 2 > > Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA > > PME(D0+,D1-,D2-,D3hot+,D3cold+) > > Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- > > Kernel driver in use: Intel ICH Modem > > > > Thanks in advance > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/87eebd76/attachment-0001.html From steveu at coppice.org Tue Jan 24 21:13:11 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 25 Jan 2012 02:13:11 +0800 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: <4F1EF4B7.3090004@coppice.org> As others have said, the problem with softmodems is nobody has been motivated to produce drivers, even though it wouldn't be that hard in most cases. I think it would be especially hard to get people motivated to produce drivers for your ICH4 modem, as that's a pretty old chipset. Intel have moved from AC97 to the high definition audio scheme since then. I think that means any drivers for a modern chipset won't work for yours, and the modern chipsets are seldom plumbed to a modem port. Its all a bit of a dead end, really. Steve On 01/24/2012 06:15 PM, Sebastian Tarach wrote: > Hello, > > I'm pretty new to the whole idea of using Linux as telephone central . > I tried to setup Asterisk at first but failed. Although I have managed > to install it I was unable to "call my computer". That's why I was > hoping to deal with FreeSWITCH better. Please tell a noob like myself > how to make this finally work. Keep in mind that something that is > clear to you to me is probably a dark magic. For instance in > installation guide there are mentioned few different ways to connect > to a modem. Which method should I use if it comes to my modem : > > > $ lspci -d 8086:24c6 -vvv > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) > Subsystem: Toshiba America Info Systems Device 0001 > Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> TAbort-SERR- Latency: 0 > Interrupt: pin B routed to IRQ 4 > Region 0: I/O ports at 2400 [size=256] > Region 1: I/O ports at 2000 [size=128] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA > PME(D0+,D1-,D2-,D3hot+,D3cold+) > Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Intel ICH Modem > > Thanks in advance > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From freeswitch-list at puzzled.xs4all.nl Tue Jan 24 21:23:07 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 24 Jan 2012 19:23:07 +0100 Subject: [Freeswitch-users] Rotate cdr-csv on hub without rotating log-files. In-Reply-To: References: Message-ID: <4F1EF70B.3070500@puzzled.xs4all.nl> On 24-01-12 18:21, Michael Collins wrote: [snip] > It does not seem possible to rotate cdr-csv files on kill -HUB ... > without also having to rotate the log-files. Is that correct? > When setting rotate-on-hub to false in the log-conf, but true in > cdr-csv nothing happings when HUB'ing. On my Fedora 16 box the signal "HUB" does not exist. Was that a typo? Perhaps you meant kill -HUP. You can see which signals your kill command supports with: $ kill -l Regards, Patrick From msc at freeswitch.org Tue Jan 24 22:09:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 24 Jan 2012 11:09:30 -0800 Subject: [Freeswitch-users] WebRTC ... Google Real Time Communication In-Reply-To: References: Message-ID: Thanks for the info! On Tue, Jan 24, 2012 at 9:48 AM, curriegrad2004 wrote: > For the windows users out there the vcproj files for iSAC is on the > mainline git tree. If you're interested you're more than welcome to try it > out and see how well it works for your application. > On 2012-01-20 12:53 PM, "Michael Collins" wrote: > >> Is anyone out there building anything using WebRTC? Just curious. >> -MC >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/f8d1d888/attachment.html From emss.mail at gmail.com Tue Jan 24 20:52:57 2012 From: emss.mail at gmail.com (Eric Masson) Date: Tue, 24 Jan 2012 18:52:57 +0100 Subject: [Freeswitch-users] HEAD & french sounds In-Reply-To: References: Message-ID: Le 23/01/2012 21:26, Brad Mina a ?crit : Hi Brad, > Although I'm not sure if the installation automates any other languages > other than English, It seems that only russian is supported except english. > you can find language specific sounds here: > http://files.freeswitch.org > > http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-8000-1.0.14.tar.gz > http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-16000-1.0.14.tar.gz > http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-32000-1.0.14.tar.gz > http://files.freeswitch.org/freeswitch-sounds-fr-ca-june-48000-1.0.14.tar.gz Ok, FS-2213 states that other files exist for fr-fr at www.archive.org I was just wondering whether mods to say_fr were about to be committed. So I'll stick with manual installation. Regards ?ric Masson From jack at livecall.com Tue Jan 24 22:01:39 2012 From: jack at livecall.com (Jack) Date: Tue, 24 Jan 2012 11:01:39 -0800 Subject: [Freeswitch-users] WebRTC ... Google Real Time Communication In-Reply-To: References: Message-ID: <4F1F0013.9030100@livecall.com> Do you have a link available? jack On 1/24/2012 9:48 AM, curriegrad2004 wrote: > > For the windows users out there the vcproj files for iSAC is on the > mainline git tree. If you're interested you're more than welcome to > try it out and see how well it works for your application. > > On 2012-01-20 12:53 PM, "Michael Collins" > wrote: > > Is anyone out there building anything using WebRTC? Just curious. > -MC > > On Fri, Jan 20, 2012 at 12:04 PM, EdPimentl > wrote: > > http://blog.chromium.org/2012/01/real-time-communications-in-chrome.html > > Just think of the possibilities with FS > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/dac28e7a/attachment.html From mgende at gendesign.com Tue Jan 24 22:19:49 2012 From: mgende at gendesign.com (Michael Gende) Date: Tue, 24 Jan 2012 13:19:49 -0600 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: Message-ID: Yes, that's the one. Not the entire protocol, just the subset used in E9-1-1 (while it still is). On Tue, Jan 24, 2012 at 10:31 AM, Moises Silva wrote: > On Mon, Jan 23, 2012 at 5:36 PM, Michael Gende wrote: > >> I'm just starting a search on this subject, but does a stack exist for FS >> implementing CAMA, or at least a subset of that protocol? >> >> > Someone asked me about this a while ago, can't remember nor find his name > or email. > > Just to confirm, are you talking about this?: > http://en.wikipedia.org/wiki/Automatic_message_accounting > > I don't think there is a stack or code to handle that in the open source > community (though, you never know what is out there). From what I see > requires MF tones, which at least spandsp is able to generate and detect. > Some logic is required on top to interpret and respond to the tones > accordingly. > > *Moises Silva > **Software Engineer, Development Manager*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/7e31ef30/attachment-0001.html From krice at freeswitch.org Tue Jan 24 22:59:41 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 24 Jan 2012 13:59:41 -0600 Subject: [Freeswitch-users] =?iso-8859-1?q?Wondering_what_modules_you_guys?= =?iso-8859-1?q?_are_using_that_don=B9t_get_all_the_love?= Message-ID: Hey Guys, Help out with a little feedback for the FreeSWITCH project it?ll take about 1 minute of your time. I?ve selected a handful of modules that add some good functionality but don?t seem to get the attention that some of the others get. Let us know which modules you are using from the list. http://www.easypolls.net/poll.html?p=4f1f0147c2e1b0e4f77cc294 K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120124/4e34a51d/attachment.html From starach at gmail.com Tue Jan 24 23:07:49 2012 From: starach at gmail.com (Sebastian Tarach) Date: Tue, 24 Jan 2012 21:07:49 +0100 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: ~I'm hoping I'm not doing something wrong by replying to this message. Sorry I'm using message boards and irc most of the time and not mailing lists. Well after installing sl-modem-daemon and modem-cmd I was able to call my mobile phone bo using command modem-cmd /dev/modem ATD9 - 9 tells my Panasonic KX-T61610B to use first outside line available. After a while I heard my mobile phone ringing. That's why I'm not really convinced that it won't work because it will lack the drivers. Well maybe FreeSWITCH needs some special drivers I don't know of. I probably should have said it before but my telephone line isn't plugged in directly into my modem. Its connected via Panasonic KX-T61610B. I'm planning on using some old junk as its well I wouldn't call it replacement. More like supplement because I want them to work side by side. Since abilities of that KX-... are really poor I want to put some old crappy computer with telephone modem inside next to it which will handle things like IVR. But before I will get to this I want to check if I'm not stupid enough to even set it up on my Toshiba Satellite M30 laptop which happens has a modem. I would like to explore all the possibilities before I will start investing money in something unnecessary or in something I can't handle. So despite the fact I was able to use it to call myself on cellphone are you really sure I won't be able to set it up this way? > ---------- Forwarded message ---------- > From:?Ken Rice > To:?FreeSWITCH Users Help > Cc: > Date:?Tue, 24 Jan 2012 11:48:16 -0600 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > Not to mention intel discontinued the chipset used on that card ages ago... You?ll be lucky to find them today > > > On 1/24/12 11:43 AM, "curriegrad2004" wrote: > > Don't even bother getting a x100p or it's clones either. It's been reported to work very poorly as an fxo. If you dug deeper you'll find that it's really a softmodem by Intel. > > On 2012-01-24 9:02 AM, "Ken Rice" wrote: > > Unfortunately most softmodems are not supported as a Telephony interface... > The problem is lack of drivers. The simple answer is it wont work... > > You're best bet is to get an ATA with both FXO and FXS ports and connect via > SIP... > > > On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: > >> Hello, >> >> I'm pretty new to the whole idea of using Linux as telephone central . >> I tried to setup Asterisk at first but failed. Although I have managed >> to install it I was unable to "call my computer". That's why I was >> hoping to deal with FreeSWITCH better. Please tell a noob like myself >> how to make this finally work. Keep in mind that something that is >> clear to you to me is probably a dark magic. For instance in >> installation guide there are mentioned few different ways to connect >> to a modem. Which method should I use if it comes to my modem : >> >> >> $ lspci -d 8086:24c6 -vvv >> 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> ? ? ? ? Subsystem: Toshiba America Info Systems Device 0001 >> ? ? ? ? Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> ParErr- Stepping- SERR- FastB2B- DisINTx- >> ? ? ? ? Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >>> TAbort- SERR- > ? ? ? ? Latency: 0 >> ? ? ? ? Interrupt: pin B routed to IRQ 4 >> ? ? ? ? Region 0: I/O ports at 2400 [size=256] >> ? ? ? ? Region 1: I/O ports at 2000 [size=128] >> ? ? ? ? Capabilities: [50] Power Management version 2 >> ? ? ? ? ? ? ? ? Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> PME(D0+,D1-,D2-,D3hot+,D3cold+) >> ? ? ? ? ? ? ? ? Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> ? ? ? ? Kernel driver in use: Intel ICH Modem >> >> Thanks in advance >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ---------- Forwarded message ---------- > From:?Nandy Dagondon > To:?FreeSWITCH Users Help > Cc: > Date:?Wed, 25 Jan 2012 01:51:26 +0800 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > FreeSWITCH doesn't need softmodem unlike Asterisk where an X100P (or a Digium card) is needed as a timing device. > > > On Tue, Jan 24, 2012 at 6:15 PM, Sebastian Tarach wrote: >> >> Hello, >> >> I'm pretty new to the whole idea of using Linux as telephone central . >> I tried to setup Asterisk at first but failed. Although I have managed >> to install it I was unable to "call my computer". That's why I was >> hoping to deal with FreeSWITCH better. Please tell a noob like myself >> how to make this finally work. Keep in mind that something that is >> clear to you to me is probably a dark magic. For instance in >> installation guide there are mentioned few different ways to connect >> to a modem. Which method should I use if it comes to my modem : >> >> >> $ lspci -d 8086:24c6 -vvv >> 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> ? ? ? ?Subsystem: Toshiba America Info Systems Device 0001 >> ? ? ? ?Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> ParErr- Stepping- SERR- FastB2B- DisINTx- >> ? ? ? ?Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> >TAbort- SERR- > ? ? ? ?Latency: 0 >> ? ? ? ?Interrupt: pin B routed to IRQ 4 >> ? ? ? ?Region 0: I/O ports at 2400 [size=256] >> ? ? ? ?Region 1: I/O ports at 2000 [size=128] >> ? ? ? ?Capabilities: [50] Power Management version 2 >> ? ? ? ? ? ? ? ?Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> PME(D0+,D1-,D2-,D3hot+,D3cold+) >> ? ? ? ? ? ? ? ?Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> ? ? ? ?Kernel driver in use: Intel ICH Modem >> >> Thanks in advance >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From:?curriegrad2004 > To:?FreeSWITCH Users Help > Cc: > Date:?Tue, 24 Jan 2012 09:52:08 -0800 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > > Yeah, seen a lot of fake x100ps on eBay too so steer clear of that place too. > > On 2012-01-24 9:49 AM, "Ken Rice" wrote: >> >> Not to mention intel discontinued the chipset used on that card ages ago... You?ll be lucky to find them today >> >> >> On 1/24/12 11:43 AM, "curriegrad2004" wrote: >> >> Don't even bother getting a x100p or it's clones either. It's been reported to work very poorly as an fxo. If you dug deeper you'll find that it's really a softmodem by Intel. >> >> On 2012-01-24 9:02 AM, "Ken Rice" wrote: >> >> Unfortunately most softmodems are not supported as a Telephony interface... >> The problem is lack of drivers. The simple answer is it wont work... >> >> You're best bet is to get an ATA with both FXO and FXS ports and connect via >> SIP... >> >> >> On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: >> >> > Hello, >> > >> > I'm pretty new to the whole idea of using Linux as telephone central . >> > I tried to setup Asterisk at first but failed. Although I have managed >> > to install it I was unable to "call my computer". That's why I was >> > hoping to deal with FreeSWITCH better. Please tell a noob like myself >> > how to make this finally work. Keep in mind that something that is >> > clear to you to me is probably a dark magic. For instance in >> > installation guide there are mentioned few different ways to connect >> > to a modem. Which method should I use if it comes to my modem : >> > >> > >> > $ lspci -d 8086:24c6 -vvv >> > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> > ? ? ? ? Subsystem: Toshiba America Info Systems Device 0001 >> > ? ? ? ? Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> > ParErr- Stepping- SERR- FastB2B- DisINTx- >> > ? ? ? ? Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> >> TAbort- SERR- > > ? ? ? ? Latency: 0 >> > ? ? ? ? Interrupt: pin B routed to IRQ 4 >> > ? ? ? ? Region 0: I/O ports at 2400 [size=256] >> > ? ? ? ? Region 1: I/O ports at 2000 [size=128] >> > ? ? ? ? Capabilities: [50] Power Management version 2 >> > ? ? ? ? ? ? ? ? Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> > PME(D0+,D1-,D2-,D3hot+,D3cold+) >> > ? ? ? ? ? ? ? ? Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> > ? ? ? ? Kernel driver in use: Intel ICH Modem >> > >> > Thanks in advance >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peder at networkoblivion.com Tue Jan 24 23:28:20 2012 From: peder at networkoblivion.com (peder at networkoblivion.com) Date: Tue, 24 Jan 2012 14:28:20 -0600 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: <007801ccdad6$b76444c0$262cce40$@networkoblivion.com> There is a big difference between voice modem and data modem. Doing an ATD just dials via the data side. It doesn't mean that it will work for voice. You can't just get any old modem and run voice over it, it needs to be a modem with a DSP and then the appropriate drivers that can convert voice. You are better off buying an ATA and doing it that way. You can find used ATAs on ebay for $30 US or less. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sebastian Tarach Sent: Tuesday, January 24, 2012 2:08 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Running FreeSWITCH on softmodem ~I'm hoping I'm not doing something wrong by replying to this message. Sorry I'm using message boards and irc most of the time and not mailing lists. Well after installing sl-modem-daemon and modem-cmd I was able to call my mobile phone bo using command modem-cmd /dev/modem ATD9 - 9 tells my Panasonic KX-T61610B to use first outside line available. After a while I heard my mobile phone ringing. That's why I'm not really convinced that it won't work because it will lack the drivers. Well maybe FreeSWITCH needs some special drivers I don't know of. I probably should have said it before but my telephone line isn't plugged in directly into my modem. Its connected via Panasonic KX-T61610B. I'm planning on using some old junk as its well I wouldn't call it replacement. More like supplement because I want them to work side by side. Since abilities of that KX-... are really poor I want to put some old crappy computer with telephone modem inside next to it which will handle things like IVR. But before I will get to this I want to check if I'm not stupid enough to even set it up on my Toshiba Satellite M30 laptop which happens has a modem. I would like to explore all the possibilities before I will start investing money in something unnecessary or in something I can't handle. So despite the fact I was able to use it to call myself on cellphone are you really sure I won't be able to set it up this way? > ---------- Forwarded message ---------- > From:?Ken Rice > To:?FreeSWITCH Users Help > Cc: > Date:?Tue, 24 Jan 2012 11:48:16 -0600 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > Not to mention intel discontinued the chipset used on that card ages ago... You?ll be lucky to find them today > > > On 1/24/12 11:43 AM, "curriegrad2004" wrote: > > Don't even bother getting a x100p or it's clones either. It's been reported to work very poorly as an fxo. If you dug deeper you'll find that it's really a softmodem by Intel. > > On 2012-01-24 9:02 AM, "Ken Rice" wrote: > > Unfortunately most softmodems are not supported as a Telephony interface... > The problem is lack of drivers. The simple answer is it wont work... > > You're best bet is to get an ATA with both FXO and FXS ports and connect via > SIP... > > > On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: > >> Hello, >> >> I'm pretty new to the whole idea of using Linux as telephone central . >> I tried to setup Asterisk at first but failed. Although I have managed >> to install it I was unable to "call my computer". That's why I was >> hoping to deal with FreeSWITCH better. Please tell a noob like myself >> how to make this finally work. Keep in mind that something that is >> clear to you to me is probably a dark magic. For instance in >> installation guide there are mentioned few different ways to connect >> to a modem. Which method should I use if it comes to my modem : >> >> >> $ lspci -d 8086:24c6 -vvv >> 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> ? ? ? ? Subsystem: Toshiba America Info Systems Device 0001 >> ? ? ? ? Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> ParErr- Stepping- SERR- FastB2B- DisINTx- >> ? ? ? ? Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >>> TAbort- SERR- > ? ? ? ? Latency: 0 >> ? ? ? ? Interrupt: pin B routed to IRQ 4 >> ? ? ? ? Region 0: I/O ports at 2400 [size=256] >> ? ? ? ? Region 1: I/O ports at 2000 [size=128] >> ? ? ? ? Capabilities: [50] Power Management version 2 >> ? ? ? ? ? ? ? ? Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> PME(D0+,D1-,D2-,D3hot+,D3cold+) >> ? ? ? ? ? ? ? ? Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> ? ? ? ? Kernel driver in use: Intel ICH Modem >> >> Thanks in advance >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ---------- Forwarded message ---------- > From:?Nandy Dagondon > To:?FreeSWITCH Users Help > Cc: > Date:?Wed, 25 Jan 2012 01:51:26 +0800 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > FreeSWITCH doesn't need softmodem unlike Asterisk where an X100P (or a Digium card) is needed as a timing device. > > > On Tue, Jan 24, 2012 at 6:15 PM, Sebastian Tarach wrote: >> >> Hello, >> >> I'm pretty new to the whole idea of using Linux as telephone central . >> I tried to setup Asterisk at first but failed. Although I have managed >> to install it I was unable to "call my computer". That's why I was >> hoping to deal with FreeSWITCH better. Please tell a noob like myself >> how to make this finally work. Keep in mind that something that is >> clear to you to me is probably a dark magic. For instance in >> installation guide there are mentioned few different ways to connect >> to a modem. Which method should I use if it comes to my modem : >> >> >> $ lspci -d 8086:24c6 -vvv >> 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> ? ? ? ?Subsystem: Toshiba America Info Systems Device 0001 >> ? ? ? ?Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> ParErr- Stepping- SERR- FastB2B- DisINTx- >> ? ? ? ?Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> >TAbort- SERR- > ? ? ? ?Latency: 0 >> ? ? ? ?Interrupt: pin B routed to IRQ 4 >> ? ? ? ?Region 0: I/O ports at 2400 [size=256] >> ? ? ? ?Region 1: I/O ports at 2000 [size=128] >> ? ? ? ?Capabilities: [50] Power Management version 2 >> ? ? ? ? ? ? ? ?Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> PME(D0+,D1-,D2-,D3hot+,D3cold+) >> ? ? ? ? ? ? ? ?Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> ? ? ? ?Kernel driver in use: Intel ICH Modem >> >> Thanks in advance >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From:?curriegrad2004 > To:?FreeSWITCH Users Help > Cc: > Date:?Tue, 24 Jan 2012 09:52:08 -0800 > Subject:?Re: [Freeswitch-users] Running FreeSWITCH on softmodem > > Yeah, seen a lot of fake x100ps on eBay too so steer clear of that place too. > > On 2012-01-24 9:49 AM, "Ken Rice" wrote: >> >> Not to mention intel discontinued the chipset used on that card ages ago... You?ll be lucky to find them today >> >> >> On 1/24/12 11:43 AM, "curriegrad2004" wrote: >> >> Don't even bother getting a x100p or it's clones either. It's been reported to work very poorly as an fxo. If you dug deeper you'll find that it's really a softmodem by Intel. >> >> On 2012-01-24 9:02 AM, "Ken Rice" wrote: >> >> Unfortunately most softmodems are not supported as a Telephony interface... >> The problem is lack of drivers. The simple answer is it wont work... >> >> You're best bet is to get an ATA with both FXO and FXS ports and connect via >> SIP... >> >> >> On 1/24/12 4:15 AM, "Sebastian Tarach" wrote: >> >> > Hello, >> > >> > I'm pretty new to the whole idea of using Linux as telephone central . >> > I tried to setup Asterisk at first but failed. Although I have managed >> > to install it I was unable to "call my computer". That's why I was >> > hoping to deal with FreeSWITCH better. Please tell a noob like myself >> > how to make this finally work. Keep in mind that something that is >> > clear to you to me is probably a dark magic. For instance in >> > installation guide there are mentioned few different ways to connect >> > to a modem. Which method should I use if it comes to my modem : >> > >> > >> > $ lspci -d 8086:24c6 -vvv >> > 00:1f.6 Modem: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) >> > AC'97 Modem Controller (rev 03) (prog-if 00 [Generic]) >> > ? ? ? ? Subsystem: Toshiba America Info Systems Device 0001 >> > ? ? ? ? Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- >> > ParErr- Stepping- SERR- FastB2B- DisINTx- >> > ? ? ? ? Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >> >> TAbort- SERR- > > ? ? ? ? Latency: 0 >> > ? ? ? ? Interrupt: pin B routed to IRQ 4 >> > ? ? ? ? Region 0: I/O ports at 2400 [size=256] >> > ? ? ? ? Region 1: I/O ports at 2000 [size=128] >> > ? ? ? ? Capabilities: [50] Power Management version 2 >> > ? ? ? ? ? ? ? ? Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA >> > PME(D0+,D1-,D2-,D3hot+,D3cold+) >> > ? ? ? ? ? ? ? ? Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- >> > ? ? ? ? Kernel driver in use: Intel ICH Modem >> > >> > Thanks in advance >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From georg at riseup.net Wed Jan 25 00:50:16 2012 From: georg at riseup.net (georg at riseup.net) Date: Tue, 24 Jan 2012 22:50:16 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: References: Message-ID: > I don't believe you can reliable put these tones in synch. I've never > heard > of any telephony system that does so, although I admit my experience is > limited to legacy PBXes and FreeSWITCH. Allright. But now it's like this: When someone is calling, the internal phone is ringing, and after ~ two seconds, the calling party is getting the ringback tone, which is quite long, isn't it? And it might be confusing aswell... Thanks, Georg From frankjr at mcpeekdodge.com Wed Jan 25 01:27:47 2012 From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr) Date: Tue, 24 Jan 2012 22:27:47 +0000 Subject: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? Message-ID: Probably not the right way to do this, but currently I implement "RingGroups" by bridging to a group call (as opposed to FIFO). Is there any way to determine after the bridge is successful which extension in the group call actually answered the call? Example: I ask as a followup to a question long ago about how to determine the actual call path each call takes. I want to be able to determine that a call came in, was transferred to ext 1200 (The Call Group), and then know which actual extension picked it up. I am doing that in my "Local_Transfer" extension by appending/exporting to a channel variable I have created called ${call_path}. I then have my cdr_csv include ${call_path} in its output. Works nicely except with the "group call". I get 7145551212:1200 Inbound Call_id : transferred to 1200 What I would like to somehow accomplish is: 7145551212:1200:1250 Inbound Call_ID : transferred to 1200 (RingGroup) : Transferred to 1250 (Actual extension in RingGroup that answered). Hope this doesn't sound like gibberish... Thanks for any ideas -Frank From db3l.net at gmail.com Wed Jan 25 01:49:00 2012 From: db3l.net at gmail.com (David Bolen) Date: Tue, 24 Jan 2012 17:49:00 -0500 Subject: [Freeswitch-users] announce count mod conference References: <1326851672.23930.YahooMailNeo@web65315.mail.ac2.yahoo.com> Message-ID: Rodney writes: > yes i would love to implement the announce conference count to user > before entering but cant figure out how to "calculate" that. I use a small script to control entry into the conference (it also handles some messaging around moderators), and get a conference count through the "conference xxx list count" api command, then play it to the current caller with a custom phrase before adding them. I ended up announcing which person they are ("you are the nth listener in the conference"), which obviously also lets them know the current count. I think I just chose that due to the existing recorded messages I was reusing sounding better that way. E.g., something like (in the lua script): conf_count = tonumber(api:executeString("conference " .. conf_name_api .. " list count")); session:sayPhrase("gp_conf_size", tostring(conf_count+1), "en"); Using the phrase definition: There's a race condition if two people are entering simultaneously, since the count won't increment for the new person until the phrase finishes playing and they are technically put into conference, but I don't have any high entry rate conferences and was willing to accept that. -- David From georg at riseup.net Wed Jan 25 02:18:19 2012 From: georg at riseup.net (georg at riseup.net) Date: Wed, 25 Jan 2012 00:18:19 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: References: Message-ID: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> > But now it's like this: When someone is calling, the internal phone is > ringing, and after ~ two seconds, the calling party is getting the > ringback tone, which is quite long, isn't it? And it might be confusing > aswell... It's working quite well now with ring_ready. Are there any disadvantages using this? Thanks, Georg From georg at riseup.net Wed Jan 25 03:33:39 2012 From: georg at riseup.net (georg at riseup.net) Date: Wed, 25 Jan 2012 01:33:39 +0100 Subject: [Freeswitch-users] ACLs / changing to which IPs FS binds to Message-ID: <8d97bc5b095dee0208f316ff08702637.squirrel@fulvetta.riseup.net> Hi all, I've got a server running FS with five nets associated. There are just two, from where I receive calls and my phones are registering. I would like to exclude all the nets by default from being allowed to contact / register at FS, and only allow - one net 172.251.X.XXX - one net 192.168.X.XXX I tried achieving this trough acl.conf, however, had no success. I disabled NAT at startup trough -nonat. 'sofia status profile internal' is showing me a public ip of my server next to "Pres Hosts" (but also one ip out of the mentioned 192.168.X.XXX net, which is fine). In internal.xml, I set rtp-ip and sip-ip to this (correct) ip. I think my main mistake is that I don't understand how things are handled in acl.conf. So far it looks like this: Thanks in advance, Georg From gopalakrishnan.an at gmail.com Wed Jan 25 08:04:46 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 25 Jan 2012 10:34:46 +0530 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: I have pasted the output here http://pastebin.com/54mxJCsT On Tue, Jan 24, 2012 at 11:07 PM, Michael Collins wrote: > Try: > > sofia profile internal reload > > If you get any errors on output then put them in pastebin and let us know. > -MC > > > On Tue, Jan 24, 2012 at 1:47 AM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> My "sofia status" output is, >> freeswitch at ubuntu> sofia status >> >> Name Type >> Data State >> >> ================================================================================================= >> external profile >> sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) >> external::example.com gateway >> sip:joeuser at example.com NOREG >> >> ================================================================================================= >> 1 profile 0 aliases >> >> freeswitch at ubuntu> >> >> >> On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N < >> gopalakrishnan.an at gmail.com> wrote: >> >>> I am not able to run "sofia status profile internal reg", In my fs_cli I >>> am getting till sofia status profile external not internal, I think my >>> internal profile is not activated. >>> >>> >>> On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: >>> >>>> After recent change you need to type "sofia status profile internal reg" >>>> >>>> Sent from my iPad >>>> >>>> On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: >>>> >>>> Hi, >>>> >>>> Does your softphone runs on the same machine as freeswitch ? Maybe your >>>> softphone already binds to the 5060 port and freeswitch can't start the >>>> internal profile which also listens on 5060. >>>> >>>> What it is the output of "sofia status" command ? >>>> >>>> >>>> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >>>> >>>> Hi, >>>> >>>> I have installed Freeswitch from git in my Ubuntu 11.10. There are >>>> some default sip users in prefix/conf/directory/default/1000.xml to >>>> 1019.xml. But when I try to register one extension to IP Phone or Softphone >>>> the account is not registered and responding as Authentication failuere. >>>> 403 Forbidden error, even though I have changed the password in 1000.xml >>>> file. >>>> >>>> Also in FS CLI I am not able to see the output for "sofia status >>>> profile internal". >>>> >>>> Please advice. >>>> >>>> Regards. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/ffda1922/attachment-0001.html From gopalakrishnan.an at gmail.com Wed Jan 25 08:06:29 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 25 Jan 2012 10:36:29 +0530 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: Also in my fs cli i got some error message while starting freeswitch which i pasted here http://pastebin.com/BLegHUGG On Wed, Jan 25, 2012 at 10:34 AM, Gopalakrishnan N < gopalakrishnan.an at gmail.com> wrote: > I have pasted the output here http://pastebin.com/54mxJCsT > > > On Tue, Jan 24, 2012 at 11:07 PM, Michael Collins wrote: > >> Try: >> >> sofia profile internal reload >> >> If you get any errors on output then put them in pastebin and let us know. >> -MC >> >> >> On Tue, Jan 24, 2012 at 1:47 AM, Gopalakrishnan N < >> gopalakrishnan.an at gmail.com> wrote: >> >>> My "sofia status" output is, >>> freeswitch at ubuntu> sofia status >>> >>> Name Type >>> Data State >>> >>> ================================================================================================= >>> external profile >>> sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) >>> external::example.com gateway >>> sip:joeuser at example.com NOREG >>> >>> ================================================================================================= >>> 1 profile 0 aliases >>> >>> freeswitch at ubuntu> >>> >>> >>> On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N < >>> gopalakrishnan.an at gmail.com> wrote: >>> >>>> I am not able to run "sofia status profile internal reg", In my fs_cli >>>> I am getting till sofia status profile external not internal, I think my >>>> internal profile is not activated. >>>> >>>> >>>> On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: >>>> >>>>> After recent change you need to type "sofia status profile internal >>>>> reg" >>>>> >>>>> Sent from my iPad >>>>> >>>>> On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: >>>>> >>>>> Hi, >>>>> >>>>> Does your softphone runs on the same machine as freeswitch ? Maybe >>>>> your softphone already binds to the 5060 port and freeswitch can't start >>>>> the internal profile which also listens on 5060. >>>>> >>>>> What it is the output of "sofia status" command ? >>>>> >>>>> >>>>> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >>>>> >>>>> Hi, >>>>> >>>>> I have installed Freeswitch from git in my Ubuntu 11.10. There are >>>>> some default sip users in prefix/conf/directory/default/1000.xml to >>>>> 1019.xml. But when I try to register one extension to IP Phone or Softphone >>>>> the account is not registered and responding as Authentication failuere. >>>>> 403 Forbidden error, even though I have changed the password in 1000.xml >>>>> file. >>>>> >>>>> Also in FS CLI I am not able to see the output for "sofia status >>>>> profile internal". >>>>> >>>>> Please advice. >>>>> >>>>> Regards. >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/7b8d3bc8/attachment.html From steveu at coppice.org Wed Jan 25 08:09:12 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 25 Jan 2012 13:09:12 +0800 Subject: [Freeswitch-users] Running FreeSWITCH on softmodem In-Reply-To: References: Message-ID: <4F1F8E78.8080104@coppice.org> On 01/25/2012 04:07 AM, Sebastian Tarach wrote: > ~I'm hoping I'm not doing something wrong by replying to this message. > Sorry I'm using message boards and irc most of the time and not > mailing lists. > > Well after installing sl-modem-daemon and modem-cmd I was able to call > my mobile phone bo using command > modem-cmd /dev/modem ATD9 - 9 tells my Panasonic > KX-T61610B to use first outside line available. > > After a while I heard my mobile phone ringing. That's why I'm not > really convinced that it won't work because it will lack the drivers. > Well maybe FreeSWITCH needs some special drivers I don't know of. Did you get good voice results during your test call? No? Maybe there are good reasons. The sl-modem-daemon runs the modem interface at 9600 samples/second. Most modem interfaces, when used for modem work, run at that sampling rate. It happens to simplify the modem's DSP a bit. So, you have a driver which can control your modem port, but you have two problems. One is the sampling rate is not the 8000 samples/second rate all the telephony systems works at. The second is the driver APIs are proprietary, and not like the APIs used to control mainstream telephony cards, like those made by Sangoma and Digium. The source code for the driver module of sl-modem is available. For some hardware you can find the information that will allow you to set the hardware to sample at 8000 samples/second. All the elements are available for building a driver that would be compatible with FreeSwitch (or Yate, or CallWeaver, or Asterisk, etc.). None exists today. If you have the time, skills and enthusiasm quite a few people might love to see such a driver. It used to be quite an interesting thing to have, as a lot of notebooks had similar modem ports. Now they are disappearing, its becoming less interesting every time the makers refresh their models. Steve From steveu at coppice.org Wed Jan 25 08:19:27 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 25 Jan 2012 13:19:27 +0800 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: Message-ID: <4F1F90DF.30405@coppice.org> Hi, Freeswitch and Asterisk (probably Yate and others too) have been used for E911 services for some time, yet I haven't heard anyone ask for CAMA before. Its usually things like TDD for the deaf support they ask for. However, it appears CAMA is a popular choice for E911, as it overcomes limitations in the ISDN protocols for a service eager to know the origins of a dropped call. http://wholesale.att.com/reference_library/.../TR-73576.pdfseems to be the relevant spec. Its basically a conventional Bell MF type T1, with a few modifications (or analogue pairs for smaller setups). It doesn't look like a big job to implement it, although finding suitable test beds might be a pain. Regards, Steve On 01/25/2012 03:19 AM, Michael Gende wrote: > Yes, that's the one. Not the entire protocol, just the subset used in > E9-1-1 (while it still is). > > On Tue, Jan 24, 2012 at 10:31 AM, Moises Silva > wrote: > > On Mon, Jan 23, 2012 at 5:36 PM, Michael Gende > > wrote: > > I'm just starting a search on this subject, but does a stack > exist for FS implementing CAMA, or at least a subset of that > protocol? > > > Someone asked me about this a while ago, can't remember nor find > his name or email. > > Just to confirm, are you talking about this?: > http://en.wikipedia.org/wiki/Automatic_message_accounting > > I don't think there is a stack or code to handle that in the open > source community (though, you never know what is out there). From > what I see requires MF tones, which at least spandsp is able to > generate and detect. Some logic is required on top to interpret > and respond to the tones accordingly. > > *Moises Silva > **/Software Engineer, Development Manager/*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > > Products > | > Solutions > | > Events > | > Contact > | > Wiki > | > Facebook > | > Twitter > `| > | YouTube > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gerald.weber at besharp.at Wed Jan 25 14:17:51 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 25 Jan 2012 11:17:51 +0000 Subject: [Freeswitch-users] exit lua script in hangup hook Message-ID: Hello, i'm trying to exit my lua script (called from dialplan) to exit in the the hanguphook: function on_hangup(s,status) freeswitch.consoleLog("NOTICE","---- on_hangup: "..status.."\n"); error(); end freeswitch.consoleLog("NOTICE","---- ANSWER:\n"); session:answer(); freeswitch.consoleLog("NOTICE","---- SETHOOK: \n"); session:setHangupHook("on_hangup"); while (session:ready() == true) do freeswitch.consoleLog("NOTICE","---- START OF LOOP \n"); freeswitch.consoleLog("NOTICE","---- STREAMFILE \n"); session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 1 -> HANGUP\n"); end freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 2 -> HANGUP\n"); session:hangup() When I hangup during the streamFile Call, I geht the following output: 2012-01-25 12:10:19.936097 [NOTICE] switch_channel.c:930 New Channel sofia/internal/2001 at 192.168.20.73 [2b1cd9f6-4745-11e1-8c71-0d2938abf0b5] 2012-01-25 12:10:19.936097 [INFO] mod_dialplan_xml.c:481 Processing B#-FS-2001 <2001>->3001 in context default 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- ANSWER: 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:599 Channel [sofia/internal/2001 at 192.168.20.73] has been answered 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- SETHOOK: 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- START OF LOOP 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- STREAMFILE 2012-01-25 12:10:21.116050 [NOTICE] sofia.c:624 Hangup sofia/internal/2001 at 192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- on_hangup: hangup 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF SCRIPT 1 -> HANGUP 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF SCRIPT 2 -> HANGUP 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1398 Session 2 (sofia/internal/2001 at 192.168.20.73) Ended 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/2001 at 192.168.20.73 [CS_DESTROY] Why does the hangup handler return and the rest of the script is executed ? I event tried "do return end;" , "exit;" ," exit();" instead of "error(); - none of them works. If i put the same logic into javascript, the hook works and the script is terminated: function on_hangup(s,status) { console_log("NOTICE","---- hangup "+status+"\n"); return "exit"; } console_log("NOTICE","---- ANSWER\n"); session.answer(); console_log("NOTICE","---- ANSWER\n"); session.setHangupHook(on_hangup); while(session.ready()) { console_log("NOTICE","---- START OF LOOP\n"); session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); console_log("NOTICE","---- END OF LOOP\n"); } console_log("NOTICE","---- END OF SCRIPT\n"); session.hangup(); 2012-01-25 12:13:08.396041 [NOTICE] switch_channel.c:930 New Channel sofia/internal/2001 at 192.168.20.73 [8f85a15c-4745-11e1-8c76-0d2938abf0b5] 2012-01-25 12:13:08.396041 [INFO] mod_dialplan_xml.c:481 Processing B#-FS-2001 <2001>->3002 in context default 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER 2012-01-25 12:13:08.396041 [NOTICE] mod_spidermonkey.c:2068 Channel [sofia/internal/2001 at 192.168.20.73] has been answered 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- START OF LOOP 2012-01-25 12:13:09.875996 [NOTICE] sofia.c:624 Hangup sofia/internal/2001 at 192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING] 2012-01-25 12:13:09.875996 [NOTICE] hangup.js:2 ---- hangup hangup 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1398 Session 3 (sofia/internal/2001 at 192.168.20.73) Ended 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/2001 at 192.168.20.73 [CS_DESTROY] Any suggestions ? Or do i miss something here ? The dialplan part: (replace hangup.lua with hangup.js for the javascript version) Freeswitch Version: FreeSWITCH Version 1.0.head (git-9be51d5 2012-01-21 13-45-21 -0500) thx®ards, gw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/0b8c81de/attachment.html From michal.zubac at comgate.cz Wed Jan 25 14:40:40 2012 From: michal.zubac at comgate.cz (=?ISO-8859-2?Q?Michal_Zub=E1=E8?=) Date: Wed, 25 Jan 2012 12:40:40 +0100 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: References: <4F1D6344.6000501@comgate.cz> <4F1D89DB.7030900@comgate.cz> Message-ID: <4F1FEA38.2020803@comgate.cz> Hi. I tried all the things you advised me, but with no effect. When provider send us re-INVITE, FreeSwitch doesn't react to it at all. Traces for test call are here: PCAP: http://et5.comgate.cz/zubacm/voice/dt_10m_4_sip.zip TEXT SUMMARY: http://pastebin.com/BgPUfmuc I looked into FS source code and I didn't see anything what looks like response to SIP re-INVITE message. It looks like FS can't process any of incoming RFC 4028/3311 mechanisms. I solved this issue by instructing FS to send re-INVITEs from our side before provider does it, effectively avoiding processing provider's re-INVITEs. Michal Zubac ComGate Interactive s.r.o. On 23.1.2012 20:57, Kristian Kielhofner wrote: > Michal, > > I can barely read these parsed out SIP traces (yuck) but from the > skimming I did it looks like you'd benefit from a few changes: > > - Explicitly disable session timers in your Sofia profile > - Set ignore_display_updates=true (so FreeSWITCH doesn't send an > UPDATE - won't help in this case but a good thing to do) > - Request your provider uses re-INVITEs instead of UPDATE > > On Mon, Jan 23, 2012 at 11:24 AM, Michal Zub?? wrote: >> I don't have "enable-timer" or "session-timeout" variables set in Sofia >> profile, so maybe defaults? I think these are what you are referring to. >> >> SIP trace is at >> http://pastebin.com/DTQU3nMr >> >> Yes, provider offered us re-INVITE method as an alternative for UPDATE. >> Would this one work? What do I have to set up? >> >> Michal Zubac >> ComGate Interactive s.r.o. >> >> >> On 23.1.2012 15:51, Kristian Kielhofner wrote: >>> What are your Sofia profile session timer values set to? >>> >>> Can you post a complete SIP trace? >>> >>> Can your provider send a re-INVITE instead of UPDATE? >>> >>> On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? wrote: >>>> Hello. >>>> >>>> I'd like to remove UPDATE value from SIP Accept header when creating SIP >>>> calls. We're sending it in INVITE message and our provider uses that for >>>> in-call keep-alive checks every 10 minutes. >>>> FreeSwitch doesn't respond to that, so our provider disconnects RTP and >>>> call is dropped. According to RFC3311 we can indicate that we don't >>>> support this by not sending UPDATE in Accept header. Is this gonna help? >>>> >>>> Is there any way to drop that from SIP headers from dialplan? Or do I >>>> have to change source code? >>>> Or better, is there any other (cleaner) way to resolve this? >>>> >>>> Regards >>>> >>>> Michal Zubac >>>> ComGate Interactive s.r.o. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From henrikaagaardsorensen at gmail.com Wed Jan 25 16:25:49 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 25 Jan 2012 14:25:49 +0100 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? Message-ID: I'm using the script on: http://convergence.pk/freeswitch-load-cdr-csv-to-mysql.html which also can be found on the http://wiki.freeswitch.org/wiki/Mod_cdr_csv. I'm having in running every 5 minute. And everything works. But I do have some concern. What if the job suddenly takes more than 5 minutes to complete and another job start, doing the same thing. Couldn't that be a problem and can it be fixed? From mgende at gendesign.com Wed Jan 25 17:00:29 2012 From: mgende at gendesign.com (Michael Gende) Date: Wed, 25 Jan 2012 08:00:29 -0600 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: <4F1F90DF.30405@coppice.org> References: <4F1F90DF.30405@coppice.org> Message-ID: Hey Steve, I know that Asterisk was used, we were once using it ourselves and had Digium produce a CAMA stack (as you noted, no great shakes, I think one guy did it in a weekend. It sure wasn't priced that way!). Do you know of anyone else using FS for E911? I'd be interested to speak to them. I happen to have test equipment that we used when testing our Digium stuff, so that's no problem. As a matter of fact, I'm also looking to see if TDD has been adapted for FS. As I'm looking at solution possibilities, I don't want to re-invent the wheel if someone else has already been down this road. Thanks for responding, Mike G. On Tue, Jan 24, 2012 at 11:19 PM, Steve Underwood wrote: > Hi, > > Freeswitch and Asterisk (probably Yate and others too) have been used > for E911 services for some time, yet I haven't heard anyone ask for CAMA > before. Its usually things like TDD for the deaf support they ask for. > However, it appears CAMA is a popular choice for E911, as it overcomes > limitations in the ISDN protocols for a service eager to know the > origins of a dropped call. > > http://wholesale.att.com/reference_library/.../TR-73576.pdfseems to be > the relevant spec. Its basically a conventional Bell MF type T1, with a > few modifications (or analogue pairs for smaller setups). It doesn't > look like a big job to implement it, although finding suitable test beds > might be a pain. > > Regards, > Steve > > On 01/25/2012 03:19 AM, Michael Gende wrote: > > Yes, that's the one. Not the entire protocol, just the subset used in > > E9-1-1 (while it still is). > > > > On Tue, Jan 24, 2012 at 10:31 AM, Moises Silva > > wrote: > > > > On Mon, Jan 23, 2012 at 5:36 PM, Michael Gende > > > wrote: > > > > I'm just starting a search on this subject, but does a stack > > exist for FS implementing CAMA, or at least a subset of that > > protocol? > > > > > > Someone asked me about this a while ago, can't remember nor find > > his name or email. > > > > Just to confirm, are you talking about this?: > > http://en.wikipedia.org/wiki/Automatic_message_accounting > > > > I don't think there is a stack or code to handle that in the open > > source community (though, you never know what is out there). From > > what I see requires MF tones, which at least spandsp is able to > > generate and detect. Some logic is required on top to interpret > > and respond to the tones accordingly. > > > > *Moises Silva > > **/Software Engineer, Development Manager/*** > > > > msilva at sangoma.com > > > > Sangoma Technologies > > > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > > > > > > t. +1 800 388 2475 (N. America) > > > > t. +1 905 474 1990 x128 > > > > f. +1 905 474 9223 > > > > > > > > ** > > < > http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures > > > > > > Products > > < > http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Solutions > > < > http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Events > > < > http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Contact > > < > http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Wiki > > < > http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Facebook > > < > http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> > | > > Twitter > > < > http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures > >`| > > | YouTube > > < > http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/3335b85c/attachment.html From mike at jerris.com Wed Jan 25 17:23:57 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 09:23:57 -0500 Subject: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? In-Reply-To: References: Message-ID: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> It depends where you are trying to use this information. The xml cdr will already have much of this information. Can you explain what your trying to do with this information and when? Mike On Jan 24, 2012, at 5:27 PM, Frank Busalacchi Jr wrote: > Probably not the right way to do this, but currently I implement "RingGroups" by bridging to a group call (as opposed to FIFO). > > Is there any way to determine after the bridge is successful which extension in the group call actually answered the call? > > Example: > > > > > > > > > > > > > > > > > > > I ask as a followup to a question long ago about how to determine the actual call path each call takes. I want to be able to determine that a call came in, was transferred to ext 1200 (The Call Group), and then know which actual extension picked it up. I am doing that in my "Local_Transfer" extension by appending/exporting to a channel variable I have created called ${call_path}. I then have my cdr_csv include ${call_path} in its output. Works nicely except with the "group call". > > I get 7145551212:1200 > Inbound Call_id : transferred to 1200 > > What I would like to somehow accomplish is: > > 7145551212:1200:1250 > Inbound Call_ID : transferred to 1200 (RingGroup) : Transferred to 1250 (Actual extension in RingGroup that answered). From mike at jerris.com Wed Jan 25 17:27:10 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 09:27:10 -0500 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> Message-ID: <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> if you are explicitly doing ring_ready, its telling it to ringback before we actually know the phone is ringing. The approach we take passes the ringing message all the way back to the original device. Take a look at the debug to see what is actually taking that long. Mike On Jan 24, 2012, at 6:18 PM, georg at riseup.net wrote: >> But now it's like this: When someone is calling, the internal phone is >> ringing, and after ~ two seconds, the calling party is getting the >> ringback tone, which is quite long, isn't it? And it might be confusing >> aswell... > > It's working quite well now with ring_ready. Are there any disadvantages > using this? > > Thanks, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mike at jerris.com Wed Jan 25 17:29:07 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 09:29:07 -0500 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> Message-ID: <2CFA639E-4530-4B8C-BF67-E2DA60BC2319@jerris.com> you probably need to turn on nua or nta debug to see the real error here, but it is almost always an error binding to the IP or port due to IP not existing, or port being in use. On Jan 25, 2012, at 12:06 AM, Gopalakrishnan N wrote: > Also in my fs cli i got some error message while starting freeswitch which i pasted here http://pastebin.com/BLegHUGG > > On Wed, Jan 25, 2012 at 10:34 AM, Gopalakrishnan N wrote: > I have pasted the output here http://pastebin.com/54mxJCsT > > > On Tue, Jan 24, 2012 at 11:07 PM, Michael Collins wrote: > Try: > > sofia profile internal reload > > If you get any errors on output then put them in pastebin and let us know. > -MC > > > On Tue, Jan 24, 2012 at 1:47 AM, Gopalakrishnan N wrote: > My "sofia status" output is, > freeswitch at ubuntu> sofia status > > Name Type Data State > ================================================================================================= > external profile sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) > external::example.com gateway sip:joeuser at example.com NOREG > ================================================================================================= > 1 profile 0 aliases > > freeswitch at ubuntu> > > > On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N wrote: > I am not able to run "sofia status profile internal reg", In my fs_cli I am getting till sofia status profile external not internal, I think my internal profile is not activated. > > > On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: > After recent change you need to type "sofia status profile internal reg" > > Sent from my iPad > > On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: > >> Hi, >> >> Does your softphone runs on the same machine as freeswitch ? Maybe your softphone already binds to the 5060 port and freeswitch can't start the internal profile which also listens on 5060. >> >> What it is the output of "sofia status" command ? >> >> >> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >>> >>> Hi, >>> >>> I have installed Freeswitch from git in my Ubuntu 11.10. There are some default sip users in prefix/conf/directory/default/1000.xml to 1019.xml. But when I try to register one extension to IP Phone or Softphone the account is not registered and responding as Authentication failuere. 403 Forbidden error, even though I have changed the password in 1000.xml file. >>> >>> Also in FS CLI I am not able to see the output for "sofia status profile internal". -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/887853cc/attachment-0001.html From barnyritchley at hotmail.com Wed Jan 25 15:22:08 2012 From: barnyritchley at hotmail.com (bennygeorge) Date: Wed, 25 Jan 2012 04:22:08 -0800 (PST) Subject: [Freeswitch-users] Outbound Caller ID on anoymous calls Message-ID: <1327494128468-7223625.post@n2.nabble.com> Hello all we have a situation where we receive the caller id in from as per below: INVITE sip:123456789 at 1.1.1.1 SIP/2.0. From: "anonymous" ; when this happens, freeswitch sends the call out and inserts the caller id number of 0000000000 I have changed the vars.xml to the following: however this doesn't work. What I'm trying to achieve is that if FS doesn't receive a good caller id, then it will format it with anonymous at anonymous.invalid This particular carrier is not using PAI. It would be fine if FS left the from number empty - i did try changing the vars.xml to the following: but that didn't work either any help greatly appreciated as always brgds -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoymous-calls-tp7223625p7223625.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Jan 25 17:33:10 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 09:33:10 -0500 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: <4F1FEA38.2020803@comgate.cz> References: <4F1D6344.6000501@comgate.cz> <4F1D89DB.7030900@comgate.cz> <4F1FEA38.2020803@comgate.cz> Message-ID: <247B15DD-2240-4D6A-86C4-A5E9A242A530@jerris.com> I know sofia will respond properly to session timers like this.. Its kind of a bother to pull a pcap up to look at but if you post the text of the sip (like that from freeswitch logs with siptrace enabled) I would take a look. On Jan 25, 2012, at 6:40 AM, Michal Zub?? wrote: > Hi. > > I tried all the things you advised me, but with no effect. When provider > send us re-INVITE, FreeSwitch doesn't react to it at all. > > Traces for test call are here: > PCAP: http://et5.comgate.cz/zubacm/voice/dt_10m_4_sip.zip > TEXT SUMMARY: http://pastebin.com/BgPUfmuc > > I looked into FS source code and I didn't see anything what looks like > response to SIP re-INVITE message. It looks like FS can't process any of > incoming RFC 4028/3311 mechanisms. > > > I solved this issue by instructing FS to send re-INVITEs from our side > before provider does it, effectively avoiding processing provider's > re-INVITEs. > > Michal Zubac > ComGate Interactive s.r.o. > > > On 23.1.2012 20:57, Kristian Kielhofner wrote: >> Michal, >> >> I can barely read these parsed out SIP traces (yuck) but from the >> skimming I did it looks like you'd benefit from a few changes: >> >> - Explicitly disable session timers in your Sofia profile >> - Set ignore_display_updates=true (so FreeSWITCH doesn't send an >> UPDATE - won't help in this case but a good thing to do) >> - Request your provider uses re-INVITEs instead of UPDATE >> >> On Mon, Jan 23, 2012 at 11:24 AM, Michal Zub?? wrote: >>> I don't have "enable-timer" or "session-timeout" variables set in Sofia >>> profile, so maybe defaults? I think these are what you are referring to. >>> >>> SIP trace is at >>> http://pastebin.com/DTQU3nMr >>> >>> Yes, provider offered us re-INVITE method as an alternative for UPDATE. >>> Would this one work? What do I have to set up? >>> >>> Michal Zubac >>> ComGate Interactive s.r.o. >>> >>> >>> On 23.1.2012 15:51, Kristian Kielhofner wrote: >>>> What are your Sofia profile session timer values set to? >>>> >>>> Can you post a complete SIP trace? >>>> >>>> Can your provider send a re-INVITE instead of UPDATE? >>>> >>>> On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? wrote: >>>>> Hello. >>>>> >>>>> I'd like to remove UPDATE value from SIP Accept header when creating SIP >>>>> calls. We're sending it in INVITE message and our provider uses that for >>>>> in-call keep-alive checks every 10 minutes. >>>>> FreeSwitch doesn't respond to that, so our provider disconnects RTP and >>>>> call is dropped. According to RFC3311 we can indicate that we don't >>>>> support this by not sending UPDATE in Accept header. Is this gonna help? >>>>> >>>>> Is there any way to drop that from SIP headers from dialplan? Or do I >>>>> have to change source code? >>>>> Or better, is there any other (cleaner) way to resolve this? >>>>> >>>>> Regards >>>>> >>>>> Michal Zubac >>>>> ComGate Interactive s.r.o. >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kris at kriskinc.com Wed Jan 25 17:36:31 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 25 Jan 2012 09:36:31 -0500 Subject: [Freeswitch-users] sofia: disable support for UPDATE message in SIP Accept header In-Reply-To: <247B15DD-2240-4D6A-86C4-A5E9A242A530@jerris.com> References: <4F1D6344.6000501@comgate.cz> <4F1D89DB.7030900@comgate.cz> <4F1FEA38.2020803@comgate.cz> <247B15DD-2240-4D6A-86C4-A5E9A242A530@jerris.com> Message-ID: +1 +1 2012/1/25 Michael Jerris : > I know sofia will respond properly to session timers like this.. Its kind of a bother to pull a pcap up to look at but if you post the text of the sip (like that from freeswitch logs with siptrace enabled) I would take a look. > > On Jan 25, 2012, at 6:40 AM, Michal Zub?? wrote: > >> Hi. >> >> I tried all the things you advised me, but with no effect. When provider >> send us re-INVITE, FreeSwitch doesn't react to it at all. >> >> Traces for test call are here: >> ? PCAP: http://et5.comgate.cz/zubacm/voice/dt_10m_4_sip.zip >> ? TEXT SUMMARY: http://pastebin.com/BgPUfmuc >> >> I looked into FS source code and I didn't see anything what looks like >> response to SIP re-INVITE message. It looks like FS can't process any of >> incoming RFC 4028/3311 mechanisms. >> >> >> I solved this issue by instructing FS to send re-INVITEs from our side >> before provider does it, effectively avoiding processing provider's >> re-INVITEs. >> >> Michal Zubac >> ComGate Interactive s.r.o. >> >> >> On 23.1.2012 20:57, Kristian Kielhofner wrote: >>> Michal, >>> >>> ? I can barely read these parsed out SIP traces (yuck) but from the >>> skimming I did it looks like you'd benefit from a few changes: >>> >>> - Explicitly disable session timers in your Sofia profile >>> - Set ?ignore_display_updates=true (so FreeSWITCH doesn't send an >>> UPDATE - won't help in this case but a good thing to do) >>> - Request your provider uses re-INVITEs instead of UPDATE >>> >>> On Mon, Jan 23, 2012 at 11:24 AM, Michal Zub?? ?wrote: >>>> I don't have "enable-timer" or "session-timeout" variables set in Sofia >>>> profile, so maybe defaults? I think these are what you are referring to. >>>> >>>> SIP trace is at >>>> ? http://pastebin.com/DTQU3nMr >>>> >>>> Yes, provider offered us re-INVITE method as an alternative for UPDATE. >>>> Would this one work? What do I have to set up? >>>> >>>> Michal Zubac >>>> ComGate Interactive s.r.o. >>>> >>>> >>>> On 23.1.2012 15:51, Kristian Kielhofner wrote: >>>>> What are your Sofia profile session timer values set to? >>>>> >>>>> Can you post a complete SIP trace? >>>>> >>>>> Can your provider send a re-INVITE instead of UPDATE? >>>>> >>>>> On Mon, Jan 23, 2012 at 8:40 AM, Michal Zub?? ? ?wrote: >>>>>> Hello. >>>>>> >>>>>> I'd like to remove UPDATE value from SIP Accept header when creating SIP >>>>>> calls. We're sending it in INVITE message and our provider uses that for >>>>>> in-call keep-alive checks every 10 minutes. >>>>>> FreeSwitch doesn't respond to that, so our provider disconnects RTP and >>>>>> call is dropped. According to RFC3311 we can indicate that we don't >>>>>> support this by not sending UPDATE in Accept header. Is this gonna help? >>>>>> >>>>>> Is there any way to drop that from SIP headers from dialplan? Or do I >>>>>> have to change source code? >>>>>> Or better, is there any other (cleaner) way to resolve this? >>>>>> >>>>>> Regards >>>>>> >>>>>> Michal Zubac >>>>>> ComGate Interactive s.r.o. >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From miha at softnet.si Wed Jan 25 17:48:55 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 25 Jan 2012 15:48:55 +0100 Subject: [Freeswitch-users] Call FW on phone Message-ID: <4F201657.9040005@softnet.si> Hi, I have phone registered on FS and phone is set to make call FW. I noticed that FS do not make a call FW the external number but looks on FS if the phone is registered on FS (because the calling number is not on FS the call is being rejected). How to tell FS that it should make a call fw to a number that is not on FS. Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From curriegrad2004 at gmail.com Wed Jan 25 18:08:32 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 25 Jan 2012 07:08:32 -0800 Subject: [Freeswitch-users] Outbound Caller ID on anoymous calls In-Reply-To: <1327494128468-7223625.post@n2.nabble.com> References: <1327494128468-7223625.post@n2.nabble.com> Message-ID: Try the effective_caller_name and effective_caller_id variables instead. On Wed, Jan 25, 2012 at 4:22 AM, bennygeorge wrote: > Hello all > > we have a situation where we receive the caller id in from as per below: > > > INVITE sip:123456789 at 1.1.1.1 SIP/2.0. > From: "anonymous" ; > > when this happens, freeswitch sends the call out and inserts the caller id > number of 0000000000 > > I have changed the vars.xml to the following: > > ? > ? data="outbound_caller_id=anonymous at anonymous.invalid"/> > > however this doesn't work. > > What I'm trying to achieve is that if FS doesn't receive a good caller id, > then it will format it with anonymous at anonymous.invalid > > This particular carrier is not using PAI. > > It would be fine if FS left the from number empty - i did try changing the > vars.xml to the following: > > ? > > but that didn't work either > > any help greatly appreciated as always > > brgds > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoymous-calls-tp7223625p7223625.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kris at kriskinc.com Wed Jan 25 18:09:08 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 25 Jan 2012 10:09:08 -0500 Subject: [Freeswitch-users] FreeSWITCH ICE Support Message-ID: Hello, Has anyone ever investigated adding ICE functionality to FreeSWITCH? While FreeSWITCH happily ignores ICE candidates in the SDP now it would be "really neat" if it could participate in ICE negotiation to find optimal media paths with other ICE implementations. -- Kristian Kielhofner From barnyritchley at hotmail.com Wed Jan 25 18:22:12 2012 From: barnyritchley at hotmail.com (bennygeorge) Date: Wed, 25 Jan 2012 07:22:12 -0800 (PST) Subject: [Freeswitch-users] Outbound Caller ID on anoymous calls In-Reply-To: References: <1327494128468-7223625.post@n2.nabble.com> Message-ID: <1327504932620-7224077.post@n2.nabble.com> hi there the problem is that I'm trying to set it up so that it uses the variables set in vars.xml the dial plan is quite large and is curl based so don't really want to increase the load on every call checking the cli on each call attempt are these variables accepted in the main vars.xml section, or is there a global way to specify the caller id to send when freeswitch would normally set it to 0000000000? ideally, it would be great to be able to say to use the anonymous at anonymous.invalid for cli in the from field rather than the 0000000000 many thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoymous-calls-tp7223625p7224077.html Sent from the freeswitch-users mailing list archive at Nabble.com. From moises.silva at gmail.com Wed Jan 25 18:50:15 2012 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 25 Jan 2012 10:50:15 -0500 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: <4F1F90DF.30405@coppice.org> Message-ID: On Wed, Jan 25, 2012 at 9:00 AM, Michael Gende wrote: > Hey Steve, > > I know that Asterisk was used, we were once using it ourselves and had > Digium produce a CAMA stack (as you noted, no great shakes, I think one guy > did it in a weekend. It sure wasn't priced that way!). LOL ... I'm sure you've heard some variation of this one: "Nikola Tesla visited Henry Ford at his factory, which was having some kind of difficulty. Ford asked Tesla if he could help identify the problem area. Tesla walked up to a wall of boilerplate and made a small X in chalk on one of the plates. Ford was thrilled, and told him to send an invoice. The bill arrived, for $10,000. Ford asked for a breakdown. Tesla sent another invoice, indicating a $1 charge for marking the wall with an X, and $9,999 for knowing where to put it." Taken from: http://www.snopes.com/business/genius/where.asp - Moy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/6070c678/attachment-0001.html From frankjr at mcpeekdodge.com Wed Jan 25 19:32:44 2012 From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr) Date: Wed, 25 Jan 2012 16:32:44 +0000 Subject: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? In-Reply-To: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> References: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> Message-ID: Ultimately what I am trying to do is this: I would like to write a Webpage that provides managers the ability to enter a date range and a phone extension and get a list of each phone call in which that extension participated (originated, transferred etc etc..Any participation in a call..)...Then, if the manager clicks on that call, the webpage would dig up the call recording for that call, and play it back. In order to accomplish that, I somehow need to be able to log a record of the path that a call went through...ex. A call comes in...answered by the operator (ext 1250), transferred to a group call (ext 1300), answered by a sales rep (ext 1310), who then transferred the call to the manager (ext 1320). What I am successfully doing now, is getting the "call path" with the exception of when a call is bridged to a group ( I append/export a variable in my "transfer" extension)...My example above ends up in the csv cdr as this, "7145551212->1250:1300:1320" .. I am unable to snag that ext 1310. What I would really like to do is to have the csv cdr have: "71455551212->1250:1300:1310:1320" ^^^ If I can get that, the rest of what I am trying to do should be trivial (Famous last words). Frank Busalacchi Jr???????? -----Original Message----- From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 6:24 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? It depends where you are trying to use this information. The xml cdr will already have much of this information. Can you explain what your trying to do with this information and when? Mike On Jan 24, 2012, at 5:27 PM, Frank Busalacchi Jr wrote: > Probably not the right way to do this, but currently I implement "RingGroups" by bridging to a group call (as opposed to FIFO). > > Is there any way to determine after the bridge is successful which extension in the group call actually answered the call? > > Example: > > > > > > > > > > > > > > > > > > > I ask as a followup to a question long ago about how to determine the actual call path each call takes. I want to be able to determine that a call came in, was transferred to ext 1200 (The Call Group), and then know which actual extension picked it up. I am doing that in my "Local_Transfer" extension by appending/exporting to a channel variable I have created called ${call_path}. I then have my cdr_csv include ${call_path} in its output. Works nicely except with the "group call". > > I get 7145551212:1200 > Inbound Call_id : transferred to 1200 > > What I would like to somehow accomplish is: > > 7145551212:1200:1250 > Inbound Call_ID : transferred to 1200 (RingGroup) : Transferred to 1250 (Actual extension in RingGroup that answered). From frankjr at mcpeekdodge.com Wed Jan 25 19:34:02 2012 From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr) Date: Wed, 25 Jan 2012 16:34:02 +0000 Subject: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? In-Reply-To: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> References: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> Message-ID: Oh, my bad...I didn't fully answer your question. The when portion is sometime after the call ends....Might be same day, might be next week, might be next month. Frank Busalacchi Jr???????? Direct (714) 254-2612 -----Original Message----- From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 6:24 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? It depends where you are trying to use this information. The xml cdr will already have much of this information. Can you explain what your trying to do with this information and when? Mike On Jan 24, 2012, at 5:27 PM, Frank Busalacchi Jr wrote: > Probably not the right way to do this, but currently I implement "RingGroups" by bridging to a group call (as opposed to FIFO). > > Is there any way to determine after the bridge is successful which extension in the group call actually answered the call? > > Example: > > > > > > > > > > > > > > > > > > > I ask as a followup to a question long ago about how to determine the actual call path each call takes. I want to be able to determine that a call came in, was transferred to ext 1200 (The Call Group), and then know which actual extension picked it up. I am doing that in my "Local_Transfer" extension by appending/exporting to a channel variable I have created called ${call_path}. I then have my cdr_csv include ${call_path} in its output. Works nicely except with the "group call". > > I get 7145551212:1200 > Inbound Call_id : transferred to 1200 > > What I would like to somehow accomplish is: > > 7145551212:1200:1250 > Inbound Call_ID : transferred to 1200 (RingGroup) : Transferred to 1250 (Actual extension in RingGroup that answered). From asilva at wirelessmundi.com Wed Jan 25 19:37:35 2012 From: asilva at wirelessmundi.com (Antonio) Date: Wed, 25 Jan 2012 17:37:35 +0100 Subject: [Freeswitch-users] Registration Failed with status DNS Error Message-ID: <1327509455.10812.221.camel@marces.madrid.commsmundi.com> Hi I have a problem registering a gateway, the return error is DNS error [503]. My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. how does freeswitch resolves the destination domain? if you need more information, please let me know. Btw i just compile the last git head. Thanks, Ant?nio ******** Log register. 2012-01-25 17:20:42.637192 [NOTICE] sofia_reg.c:414 Registering sol 2012-01-25 17:20:42.737189 [ERR] sofia_reg.c:2004 sol Registration Failed with status DNS Error [503]. failure #1 2012-01-25 17:20:43.637195 [WARNING] sofia_reg.c:473 sol Failed Registration [0], setting retry to 30 seconds. ******** sofia status gateway sol: freeswitch at internal> sofia status gateway sol ================================================================================================= Name sol Profile 192.168.10.25 Scheme Digest Realm mobile.commsmundi.com:5060 Username 2203 Password yes From Contact Exten 2203 To sip:2203 at mobile.commsmundi.com:5060 Proxy sip:mobile.commsmundi.com:5060 Context incoming Expires 1800 Freq 1800 Ping 1327508466 PingFreq 25 PingState -1/0/1 State FAIL_WAIT Status DOWN CallsIN 0 CallsOUT 0 FailedCallsIN 0 FailedCallsOUT 0 ================================================================================================= ******** sofia_dig: freeswitch at internal> sofia_dig mobile.commsmundi.com Preference Weight Transport Port Address ================================================================================ 1 0.200 udp 5060 192.168.10.1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/bfc873e0/attachment.html From steveu at coppice.org Wed Jan 25 19:39:27 2012 From: steveu at coppice.org (Steve Underwood) Date: Thu, 26 Jan 2012 00:39:27 +0800 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: <4F1F90DF.30405@coppice.org> Message-ID: <4F20303F.2040909@coppice.org> Hi Michael, I know about people using both Asterisk and FS for E911 because they used the TDD facility in spandsp, and I worked to improve aspects of the TDD receiver with a couple of them. There is supposed to be TDD code in Asterisk, but its completely useless. I have no idea what other elements have been used in these E911 systems. Some of them might have implemented CAMA themselves. Regards, Steve On 01/25/2012 10:00 PM, Michael Gende wrote: > Hey Steve, > > I know that Asterisk was used, we were once using it ourselves and had > Digium produce a CAMA stack (as you noted, no great shakes, I think > one guy did it in a weekend. It sure wasn't priced that way!). Do you > know of anyone else using FS for E911? I'd be interested to speak to > them. > > I happen to have test equipment that we used when testing our Digium > stuff, so that's no problem. As a matter of fact, I'm also looking to > see if TDD has been adapted for FS. As I'm looking at solution > possibilities, I don't want to re-invent the wheel if someone else has > already been down this road. > > Thanks for responding, > > Mike G. > > On Tue, Jan 24, 2012 at 11:19 PM, Steve Underwood > wrote: > > Hi, > > Freeswitch and Asterisk (probably Yate and others too) have been used > for E911 services for some time, yet I haven't heard anyone ask > for CAMA > before. Its usually things like TDD for the deaf support they ask for. > However, it appears CAMA is a popular choice for E911, as it overcomes > limitations in the ISDN protocols for a service eager to know the > origins of a dropped call. > > http://wholesale.att.com/reference_library/.../TR-73576.pdfseems to be > the relevant spec. Its basically a conventional Bell MF type T1, > with a > few modifications (or analogue pairs for smaller setups). It doesn't > look like a big job to implement it, although finding suitable > test beds > might be a pain. > > Regards, > Steve > > On 01/25/2012 03:19 AM, Michael Gende wrote: > > Yes, that's the one. Not the entire protocol, just the subset > used in > > E9-1-1 (while it still is). > > > > On Tue, Jan 24, 2012 at 10:31 AM, Moises Silva > > > >> > wrote: > > > > On Mon, Jan 23, 2012 at 5:36 PM, Michael Gende > > > >> wrote: > > > > I'm just starting a search on this subject, but does a stack > > exist for FS implementing CAMA, or at least a subset of that > > protocol? > > > > > > Someone asked me about this a while ago, can't remember nor find > > his name or email. > > > > Just to confirm, are you talking about this?: > > http://en.wikipedia.org/wiki/Automatic_message_accounting > > > > I don't think there is a stack or code to handle that in the > open > > source community (though, you never know what is out there). > From > > what I see requires MF tones, which at least spandsp is able to > > generate and detect. Some logic is required on top to interpret > > and respond to the tones accordingly. > > > > *Moises Silva > > **/Software Engineer, Development Manager/*** > > > > msilva at sangoma.com > > > > > > Sangoma Technologies > > > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > > > > > > > > t. +1 800 388 2475 > (N. America) > > > > t. +1 905 474 1990 x128 > > > > > f. +1 905 474 9223 > > > > > > > > > ** > > > > > > > > Products > > > > > | > > Solutions > > > > > | > > Events > > > > > | > > Contact > > > > > | > > Wiki > > > > > | > > Facebook > > > > > | > > Twitter > > > >`| > > | YouTube > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Wed Jan 25 19:41:28 2012 From: steveu at coppice.org (Steve Underwood) Date: Thu, 26 Jan 2012 00:41:28 +0800 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: <4F1F90DF.30405@coppice.org> Message-ID: <4F2030B8.201@coppice.org> Hi Moy, On 01/25/2012 11:50 PM, Moises Silva wrote: > On Wed, Jan 25, 2012 at 9:00 AM, Michael Gende > wrote: > > Hey Steve, > > I know that Asterisk was used, we were once using it ourselves and > had Digium produce a CAMA stack (as you noted, no great shakes, I > think one guy did it in a weekend. It sure wasn't priced that way!). > > > LOL ... I'm sure you've heard some variation of this one: > > "Nikola Tesla visited Henry Ford at his factory, which was having some > kind of difficulty. Ford asked Tesla if he could help identify the > problem area. > Tesla walked up to a wall of boilerplate and made a small X in chalk > on one of the plates. Ford was thrilled, and told him to send an invoice. > > The bill arrived, for $10,000. Ford asked for a breakdown. Tesla > sent another invoice, indicating a $1 charge for marking the wall with > an X, and $9,999 for knowing where to put it." > > Taken from: http://www.snopes.com/business/genius/where.asp > > - > Moy > You omitted to offer your own reasonably priced services. Does Sangoma already have a CAMA protocol? Steve From frank at rosengart.de Wed Jan 25 21:06:19 2012 From: frank at rosengart.de (Frank Rosengart) Date: Wed, 25 Jan 2012 19:06:19 +0100 Subject: [Freeswitch-users] Opus codec passthru (or terminate) Message-ID: <4F20449B.8040108@rosengart.de> Hi, how do I have to configure my global codec prefs to pass thru opus codec calls? I guess with mod_opus it's already possible to terminate these calls, right? But anyway, pass thru would be ok. I'm now using baresip as useragent on both ends. Thanks! Frank From krice at freeswitch.org Wed Jan 25 21:09:32 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 25 Jan 2012 12:09:32 -0600 Subject: [Freeswitch-users] Opus codec passthru (or terminate) In-Reply-To: <4F20449B.8040108@rosengart.de> Message-ID: Opus should passthru just fine if both ends are opus and no transcoding is required. Just make sure you have open config'd in your global codec pref's and mod_opus loaded K On 1/25/12 12:06 PM, "Frank Rosengart" wrote: > Hi, > > how do I have to configure my global codec prefs to pass thru opus codec > calls? I guess with mod_opus it's already possible to terminate these > calls, right? But anyway, pass thru would be ok. > > I'm now using baresip as useragent on both ends. > > > Thanks! > Frank > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From brian at freeswitch.org Wed Jan 25 21:12:34 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 25 Jan 2012 12:12:34 -0600 Subject: [Freeswitch-users] Opus codec passthru (or terminate) In-Reply-To: <4F20449B.8040108@rosengart.de> References: <4F20449B.8040108@rosengart.de> Message-ID: <1C668906-6A8C-4BB8-913E-A9601743AFF2@freeswitch.org> What are you trying to overcome? /b On Jan 25, 2012, at 12:06 PM, Frank Rosengart wrote: > Hi, > > how do I have to configure my global codec prefs to pass thru opus codec > calls? I guess with mod_opus it's already possible to terminate these > calls, right? But anyway, pass thru would be ok. > > I'm now using baresip as useragent on both ends. > > > Thanks! > Frank -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/3af7d1b7/attachment.html From rmorin at blie-ent.com Wed Jan 25 21:31:10 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Wed, 25 Jan 2012 13:31:10 -0500 Subject: [Freeswitch-users] Sendmail segfaulting before second message is sent Message-ID: <11f001ccdb8f$83124280$8936c780$@blie-ent.com> For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). Thank you, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/c8675861/attachment.html From mike at jerris.com Wed Jan 25 21:50:55 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 13:50:55 -0500 Subject: [Freeswitch-users] FreeSWITCH ICE Support In-Reply-To: References: Message-ID: <64BBC2E4-CBAC-4401-B475-FCA4144449A6@jerris.com> We have some code that is very similar to sip ice support already because mod_dingaling/jingle requires it, but as implemented it does not exactly match the sip spec. We would need to adapt that code to support the sip ICE requirements and add code to mod_sofia to handle the negotiation. Mike On Jan 25, 2012, at 10:09 AM, Kristian Kielhofner wrote: > Has anyone ever investigated adding ICE functionality to FreeSWITCH? > > While FreeSWITCH happily ignores ICE candidates in the SDP now it > would be "really neat" if it could participate in ICE negotiation to > find optimal media paths with other ICE implementations. From mike at jerris.com Wed Jan 25 21:52:38 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 13:52:38 -0500 Subject: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? In-Reply-To: References: <5CD1556D-BE05-48EA-B314-F309A9D8E5E1@jerris.com> Message-ID: <82523141-EF25-4F26-991D-0DB76869F011@jerris.com> Please take a look at the xml cdr for the call, I think all that information will be found there. Mike On Jan 25, 2012, at 11:34 AM, Frank Busalacchi Jr wrote: > Oh, my bad...I didn't fully answer your question. > > The when portion is sometime after the call ends....Might be same day, might be next week, might be next month. > > Frank Busalacchi Jr > Direct (714) 254-2612 > > > > -----Original Message----- > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Wednesday, January 25, 2012 6:24 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Determine what extension picked up after Bridging to a Group Call? > > It depends where you are trying to use this information. The xml cdr will already have much of this information. Can you explain what your trying to do with this information and when? > > Mike From mike at jerris.com Wed Jan 25 21:53:47 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 13:53:47 -0500 Subject: [Freeswitch-users] Registration Failed with status DNS Error In-Reply-To: <1327509455.10812.221.camel@marces.madrid.commsmundi.com> References: <1327509455.10812.221.camel@marces.madrid.commsmundi.com> Message-ID: <29F00015-5507-4795-8BD6-5981463324A6@jerris.com> Is it even sending the request? try turning on nua and sresolv debug in sofia to see more about what is going on? Mike On Jan 25, 2012, at 11:37 AM, Antonio wrote: > > Hi > > I have a problem registering a gateway, the return error is DNS error [503]. > > My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. > > > how does freeswitch resolves the destination domain? From mike at jerris.com Wed Jan 25 21:56:58 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 25 Jan 2012 13:56:58 -0500 Subject: [Freeswitch-users] Sendmail segfaulting before second message is sent In-Reply-To: <11f001ccdb8f$83124280$8936c780$@blie-ent.com> References: <11f001ccdb8f$83124280$8936c780$@blie-ent.com> Message-ID: <81CBC66D-D9D9-418D-B400-9B38FAAE9FD9@jerris.com> http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > For some extensions, I?ve got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. > > Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. > > If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/f845e10b/attachment-0001.html From paul at cupis.co.uk Wed Jan 25 22:06:33 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Wed, 25 Jan 2012 19:06:33 +0000 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? In-Reply-To: References: Message-ID: <4F2052B9.2070706@cupis.co.uk> On 25/01/12 13:25, Henrik Aagaard S?rensen wrote: > What if the job suddenly takes more than 5 minutes to complete and > another job start, doing the same thing. > > Couldn't that be a problem and can it be fixed? Use Perls Fcntl module to ensure the program can only be running once - if a second instance starts it will quit immediately and the next scheduled run will pick up the remaining files. Of course, if this happens to often you will take a long time to catch up with the importing. At the beginning of the script (after 'use File::Copy;') add: use Fcntl qw(:flock); unless (flock(DATA, LOCK_EX|LOCK_NB)) { exit; print "$0 is already running. Exiting.\n"; exit(1); } and at the very end add: __DATA__ This exists so flock() code above works. DO NOT REMOVE THIS DATA SECTION. Regards, From paul at cupis.co.uk Wed Jan 25 22:07:35 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Wed, 25 Jan 2012 19:07:35 +0000 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? In-Reply-To: <4F2052B9.2070706@cupis.co.uk> References: <4F2052B9.2070706@cupis.co.uk> Message-ID: <4F2052F7.70904@cupis.co.uk> On 25/01/12 19:06, Paul Cupis wrote: > unless (flock(DATA, LOCK_EX|LOCK_NB)) { > exit; > print "$0 is already running. Exiting.\n"; > exit(1); > } Er. editing the commands inside the braces above to suit. :) Regards, From Claudio.Cavalera at italtel.it Wed Jan 25 22:22:25 2012 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Wed, 25 Jan 2012 20:22:25 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 secondbreak In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org><409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: Hello, what about some limit in the network? Do you have ethernet bonding? Are you also "processing" media with your 1500 calls (3000 channels?) If you lower the Session per Seconds can you beyond that number of calls (provided that the calls are alive longer)? I think that once ago I got network bound (maybe too many small packets per second) but I did not get those messages. Regards, Claudio From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stephen Wilde Sent: Monday, January 23, 2012 11:20 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Event system overloading. Taking a 10 secondbreak Can be useful to run Freeswitch with "-nosql" option? What I lose in this case? On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde wrote: I'm already using the ramdisk. The problem happens when I have a provider that give congestion. In this case Freeswitch receives many tries but few connected calls and the number of session per second is high. To avoid the "event system overloading" (avoiding to lower the global session per second 'sps' parameter) I have insert in dialplan: In this way I have limited the session rate for the congestioned destination where I have so many tries. My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage of disk is near zero (dstat) why I cannot handle this session rate? On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale wrote: you can tell that by taking system vitals its hard to tell from the small amount of data. I do know that to get those errors, you have to push the core so hard that the sql stmts queuing up for transactions are getting too large for the rate at which they are written to the DB. Try a ramdisk like Michael suggested. On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: I'm using ram disk for the FS database "freeswitch/db". Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect the performance? On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale wrote: the box can't handle the load, the disk io from the sql stmts is backing up the events. get a nicer box with the money saved from free softswitch =p On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: Did you bind any callbacks to events that might be taking a long time to process? On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: I saw "increase the event system capacity" because in the log there was a row: [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. Where it seems that all event dispatch threads are "busy" but I see that the cpu has many idle cycles so why not increase the number of dispatch threads? Or I'm wrong? ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/d342c185/attachment-0001.html From henrikaagaardsorensen at gmail.com Wed Jan 25 22:28:36 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 25 Jan 2012 20:28:36 +0100 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? In-Reply-To: <4F2052F7.70904@cupis.co.uk> References: <4F2052B9.2070706@cupis.co.uk> <4F2052F7.70904@cupis.co.uk> Message-ID: Should __DATA__ be at the very very end, after exit 0; So: exit 0; __DATA__ On Wed, Jan 25, 2012 at 8:07 PM, Paul Cupis wrote: > On 25/01/12 19:06, Paul Cupis wrote: >> unless (flock(DATA, LOCK_EX|LOCK_NB)) { >> ? exit; >> ? print "$0 is already running. Exiting.\n"; >> ? exit(1); >> } > > Er. editing the commands inside the braces above to suit. :) > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anthony.minessale at gmail.com Wed Jan 25 22:36:20 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Jan 2012 13:36:20 -0600 Subject: [Freeswitch-users] Event system overloading. Taking a 10 secondbreak In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: also, do you use presence? We did find an issue recently that caused excess events for presence endpoints. That's the only other thing I can think of, try HEAD On Wed, Jan 25, 2012 at 1:22 PM, Cavalera Claudio Luigi < Claudio.Cavalera at italtel.it> wrote: > Hello,**** > > what about some limit in the network? Do you have ethernet bonding? Are > you also "processing" media with your 1500 calls (3000 channels?) If you > lower the Session per Seconds can you beyond that number of calls (provided > that the calls are alive longer)?**** > > I think that once ago I got network bound (maybe too many small packets > per second) but I did not get those messages.**** > > Regards,**** > > Claudio**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stephen > Wilde > *Sent:* Monday, January 23, 2012 11:20 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Event system overloading. Taking a 10 > secondbreak**** > > ** ** > > Can be useful to run Freeswitch with "-nosql" option?**** > > What I lose in this case?**** > > ** ** > > On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde > wrote:**** > > I'm already using the ramdisk.**** > > The problem happens when I have a provider that give congestion.**** > > In this case Freeswitch receives many tries but few connected calls and > the number of session per second is high.**** > > To avoid the "event system overloading" (avoiding to lower the global > session per second 'sps' parameter) I have insert in dialplan:**** > > ** ** > > break="never">**** > > **** > > **** > > ** ** > > In this way I have limited the session rate for the congestioned > destination where I have so many tries.**** > > ** ** > > My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage > of disk is near zero (dstat) why I cannot handle this session rate?**** > > ** ** > > ** ** > > ** ** > > On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote:**** > > you can tell that by taking system vitals its hard to tell from the small > amount of data.**** > > I do know that to get those errors, you have to push the core so hard that > the sql stmts queuing up for transactions are getting too large for the > rate at which they are written to the DB. Try a ramdisk like Michael > suggested.**** > > ** ** > > On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde > wrote:**** > > I'm using ram disk for the FS database "freeswitch/db".**** > > Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect > the performance?**** > > ** ** > > ** ** > > On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote:**** > > the box can't handle the load, the disk io from the sql stmts is backing > up the events.**** > > get a nicer box with the money saved from free softswitch =p**** > > ** ** > > On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo > wrote:**** > > Did you bind any callbacks to events that might be taking a long time to > process?**** > > > > **** > > On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde > wrote:**** > > I saw "increase the event system capacity" because in the log there was a > row:**** > > ** ** > > [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things > down.**** > > ** ** > > Where it seems that all event dispatch threads are "busy" but I see > that the cpu has many idle cycles so why not increase the number of > dispatch threads?**** > > Or I'm wrong?**** > > **** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > **** > > ** ** > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > ** ** > > > > 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. > > ******************************************************************************************************************************************** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/38a973ac/attachment-0001.html From wstephen80 at gmail.com Wed Jan 25 23:37:02 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Wed, 25 Jan 2012 21:37:02 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 secondbreak In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: Thank you for your reply. I use only external profile with: I use limit, for example: The server handle media for some calls (1/3 of the total, the others are with bypass_media=true) but the problem is not the number of total calls because it can handle 5000 channels with media without issue. The problem is present only when there are many session per second due to a provider with low ASR (in this case there are many inbound calls due to tries, more then 10.000.000 sessions per day). About ethernet I don't know what can be the issue: the error seems to be internal of the box, related to core freeswitch db. Any suggestion is very appreciated. I'll try with HEAD. Stephen On Wed, Jan 25, 2012 at 8:36 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > also, > > do you use presence? We did find an issue recently that caused excess > events for presence endpoints. > That's the only other thing I can think of, try HEAD > > > On Wed, Jan 25, 2012 at 1:22 PM, Cavalera Claudio Luigi < > Claudio.Cavalera at italtel.it> wrote: > >> Hello,**** >> >> what about some limit in the network? Do you have ethernet bonding? Are >> you also "processing" media with your 1500 calls (3000 channels?) If you >> lower the Session per Seconds can you beyond that number of calls (provided >> that the calls are alive longer)?**** >> >> I think that once ago I got network bound (maybe too many small packets >> per second) but I did not get those messages.**** >> >> Regards,**** >> >> Claudio**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stephen >> Wilde >> *Sent:* Monday, January 23, 2012 11:20 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Event system overloading. Taking a 10 >> secondbreak**** >> >> ** ** >> >> Can be useful to run Freeswitch with "-nosql" option?**** >> >> What I lose in this case?**** >> >> ** ** >> >> On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde >> wrote:**** >> >> I'm already using the ramdisk.**** >> >> The problem happens when I have a provider that give congestion.**** >> >> In this case Freeswitch receives many tries but few connected calls and >> the number of session per second is high.**** >> >> To avoid the "event system overloading" (avoiding to lower the global >> session per second 'sps' parameter) I have insert in dialplan:**** >> >> ** ** >> >> > break="never">**** >> >> **** >> >> **** >> >> ** ** >> >> In this way I have limited the session rate for the congestioned >> destination where I have so many tries.**** >> >> ** ** >> >> My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage >> of disk is near zero (dstat) why I cannot handle this session rate?**** >> >> ** ** >> >> ** ** >> >> ** ** >> >> On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote:**** >> >> you can tell that by taking system vitals its hard to tell from the small >> amount of data.**** >> >> I do know that to get those errors, you have to push the core so hard >> that the sql stmts queuing up for transactions are getting too large for >> the rate at which they are written to the DB. Try a ramdisk like Michael >> suggested.**** >> >> ** ** >> >> On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde >> wrote:**** >> >> I'm using ram disk for the FS database "freeswitch/db".**** >> >> Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect >> the performance?**** >> >> ** ** >> >> ** ** >> >> On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote:**** >> >> the box can't handle the load, the disk io from the sql stmts is backing >> up the events.**** >> >> get a nicer box with the money saved from free softswitch =p**** >> >> ** ** >> >> On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo >> wrote:**** >> >> Did you bind any callbacks to events that might be taking a long time to >> process?**** >> >> >> >> **** >> >> On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde >> wrote:**** >> >> I saw "increase the event system capacity" because in the log there was a >> row:**** >> >> ** ** >> >> [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things >> down.**** >> >> ** ** >> >> Where it seems that all event dispatch threads are "busy" but I see >> that the cpu has many idle cycles so why not increase the number of >> dispatch threads?**** >> >> Or I'm wrong?**** >> >> **** >> >> ** ** >> >> ** ** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> >> **** >> >> ** ** >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> ** ** >> >> >> >> 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. >> >> ******************************************************************************************************************************************** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/3e4821ab/attachment-0001.html From kris at kriskinc.com Wed Jan 25 23:44:59 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Wed, 25 Jan 2012 15:44:59 -0500 Subject: [Freeswitch-users] FreeSWITCH ICE Support In-Reply-To: <64BBC2E4-CBAC-4401-B475-FCA4144449A6@jerris.com> References: <64BBC2E4-CBAC-4401-B475-FCA4144449A6@jerris.com> Message-ID: Mike, Interesting... You may here more from me on this in the future :). On Wed, Jan 25, 2012 at 1:50 PM, Michael Jerris wrote: > We have some code that is very similar to sip ice support already because mod_dingaling/jingle requires it, but as implemented it does not exactly match the sip spec. ?We would need to adapt that code to support the sip ICE requirements and add code to mod_sofia to handle the negotiation. > > Mike > > On Jan 25, 2012, at 10:09 AM, Kristian Kielhofner wrote: >> ?Has anyone ever investigated adding ICE functionality to FreeSWITCH? >> >> ?While FreeSWITCH happily ignores ICE candidates in the SDP now it >> would be "really neat" if it could participate in ICE negotiation to >> find optimal media paths with other ICE implementations. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From frank at rosengart.de Wed Jan 25 23:55:55 2012 From: frank at rosengart.de (Frank Rosengart) Date: Wed, 25 Jan 2012 21:55:55 +0100 Subject: [Freeswitch-users] Opus codec passthru (or terminate) In-Reply-To: References: Message-ID: <4F206C5B.5010808@rosengart.de> On 25.01.2012 19:09, Ken Rice wrote: > Just make sure you have open config'd in your global codec pref's this should read "make sure you have opus config'd..." ? Yes, this was my question. What codec name do I put in the list? G722,PCMA, ... ? Frank From craigesmith at gmail.com Thu Jan 26 00:32:30 2012 From: craigesmith at gmail.com (Craig Smith) Date: Wed, 25 Jan 2012 16:32:30 -0500 Subject: [Freeswitch-users] Noobie question Message-ID: How do I do a "dialplan show" in FreeSWITCH? I'm trying to troubleshoot why calls aren't going out through a gateway. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/fbfa4993/attachment.html From elliott at zoogmedia.com Thu Jan 26 00:52:21 2012 From: elliott at zoogmedia.com (Elliott Vogel) Date: Wed, 25 Jan 2012 21:52:21 +0000 Subject: [Freeswitch-users] Dialplan help Message-ID: <461821C20115054F9056ED0E52EAC4DAE4921D@BY2PRD0710MB378.namprd07.prod.outlook.com> Hello, I'm trying to create a dial plan where a user is not registered (online) or an error sip message (480,486,500) is received; I want the call be sent to another sip address. I want to use within this context: > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/df9dbac7/attachment.html From fcintrono at gmail.com Thu Jan 26 01:12:17 2012 From: fcintrono at gmail.com (=?ISO-8859-1?Q?Francisco_Javier_Cintr=F3n_Olgu=EDn?=) Date: Wed, 25 Jan 2012 16:12:17 -0600 Subject: [Freeswitch-users] Noobie question In-Reply-To: References: Message-ID: If you are inside fs_cli a complete log debug output is going to be written. Have a good day. On Wed, Jan 25, 2012 at 3:32 PM, Craig Smith wrote: > How do I do a "dialplan show" in FreeSWITCH? I'm trying to troubleshoot > why calls aren't going out through a gateway. Thanks in advance. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/20de5a0c/attachment-0001.html From johnrose at google.hm Thu Jan 26 01:13:59 2012 From: johnrose at google.hm (John Rose) Date: Wed, 25 Jan 2012 17:13:59 -0500 Subject: [Freeswitch-users] MESSAGE Body Encoding Message-ID: <008801ccdbae$a43fd160$ecbf7420$@google.hm> Does anyone know if the MESSAGE payload can have two encodings in one string when multi-byte text is present? Example: UTF-8 and CP-1252 ? Or is this dependent on the UA or API that populates the body? John From paul at cupis.co.uk Thu Jan 26 01:19:55 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Wed, 25 Jan 2012 22:19:55 +0000 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? In-Reply-To: References: <4F2052B9.2070706@cupis.co.uk> <4F2052F7.70904@cupis.co.uk> Message-ID: <4F20800B.3010700@cupis.co.uk> On 25/01/12 19:28, Henrik Aagaard S?rensen wrote: > Should __DATA__ be at the very very end, after exit 0; > > So: > exit 0; > __DATA__ Yes, that's right, it should be right at the end. Regards, From barnyritchley at hotmail.com Thu Jan 26 01:53:07 2012 From: barnyritchley at hotmail.com (bennygeorge) Date: Wed, 25 Jan 2012 14:53:07 -0800 (PST) Subject: [Freeswitch-users] Outbound Caller ID on anoymous calls In-Reply-To: <1327504932620-7224077.post@n2.nabble.com> References: <1327494128468-7223625.post@n2.nabble.com> <1327504932620-7224077.post@n2.nabble.com> Message-ID: <1327531987122-7225416.post@n2.nabble.com> Does anyone have any ideas on this one? Are there any non documented ways to set a callerid as anonymous at anonymous.invalid instead of 0000000? Brgds -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/RE-Outbound-Caller-ID-on-anoymous-calls-tp7223625p7225416.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mgende at gendesign.com Thu Jan 26 03:36:56 2012 From: mgende at gendesign.com (Michael Gende) Date: Wed, 25 Jan 2012 18:36:56 -0600 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: <4F2030B8.201@coppice.org> References: <4F1F90DF.30405@coppice.org> <4F2030B8.201@coppice.org> Message-ID: Hey Moy, Along the lines of Steve's comment, I spoke to Anthony (probably only his parents call him that) and Brian yesterday about this. They suggested you and I speak. I note your phone number on the posts so I'll probably give you a call, assuming that is acceptable to you. Regards, Mike G. On Wed, Jan 25, 2012 at 10:41 AM, Steve Underwood wrote: > Hi Moy, > > On 01/25/2012 11:50 PM, Moises Silva wrote: > > On Wed, Jan 25, 2012 at 9:00 AM, Michael Gende > > wrote: > > > > Hey Steve, > > > > I know that Asterisk was used, we were once using it ourselves and > > had Digium produce a CAMA stack (as you noted, no great shakes, I > > think one guy did it in a weekend. It sure wasn't priced that way!). > > > > > > LOL ... I'm sure you've heard some variation of this one: > > > > "Nikola Tesla visited Henry Ford at his factory, which was having some > > kind of difficulty. Ford asked Tesla if he could help identify the > > problem area. > > Tesla walked up to a wall of boilerplate and made a small X in chalk > > on one of the plates. Ford was thrilled, and told him to send an > invoice. > > > > The bill arrived, for $10,000. Ford asked for a breakdown. Tesla > > sent another invoice, indicating a $1 charge for marking the wall with > > an X, and $9,999 for knowing where to put it." > > > > Taken from: http://www.snopes.com/business/genius/where.asp > > > > - > > Moy > > > You omitted to offer your own reasonably priced services. Does Sangoma > already have a CAMA protocol? > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/cc039710/attachment.html From bobc at devassert.com Thu Jan 26 03:37:45 2012 From: bobc at devassert.com (Bob Coleman) Date: Thu, 26 Jan 2012 13:37:45 +1300 Subject: [Freeswitch-users] Remote Extension Message-ID: Hi, I have a remote extension( Cisco SPA504G ) that connects to our FreeSWITCH server across the internet. I am using a typical double nat scenario as per wiki and the remote extension registers properly against the right profile. The remote extension via a dialplan can dial any of the extensions that are local to the FS boxes network successfully with no problems However when we dial out to the remote extension we get a "Cannot locate any authentication on credentials to complete an authentication request for realm '' I have tried using both the default force register options and also setting up another directory realm as well. When looking at the log you see the originate attempting to connect to the appropriate ip/port of the external phone, but after it "entering state [calling]" it fails with the above error. Just to test I have installed the 3cx softphone client on the remote network, and it can be dialled with no problems. The difference being that the 3cx seems to register a port in the dynamic range (eg 52789) where as the cisco phone uses the 5090 port Has anyone struck that issue before Thanks Bob From georg at riseup.net Thu Jan 26 04:07:50 2012 From: georg at riseup.net (georg at riseup.net) Date: Thu, 26 Jan 2012 02:07:50 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> Message-ID: <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> Hi Mike, > if you are explicitly doing ring_ready, its telling it to ringback before > we actually know the phone is ringing. The approach we take passes the > ringing message all the way back to the original device. Take a look at > the debug to see what is actually taking that long. Allright. I thought of something similar, so I step back of using ring_ready. I tried several calls, and it seems just random: Like 50%/50%, the ringback tone the caller hears is nearly in sync with the internal one, in the other 50%, there's a delay of ~ three seconds. However, the debug output is looking the same, I posted one to [1]. My provider isn't supporting early media. Could this be a problem? Thanks, Georg [1] http://dpaste.de/no6bO/ From rmorin at blie-ent.com Thu Jan 26 06:43:53 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Wed, 25 Jan 2012 22:43:53 -0500 Subject: [Freeswitch-users] Sendmail segfaulting Message-ID: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120125/6079c292/attachment-0001.html From vishal.kakkar at gmail.com Thu Jan 26 11:34:40 2012 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Thu, 26 Jan 2012 14:04:40 +0530 Subject: [Freeswitch-users] Gateway Configuration with No Authentication Message-ID: Hi all, We are trying to use FS to handle incoming call from SIP provider. Provider doesnt have any authentication because its all inbound traffic. They have configured their SIP trunk to point to our FreeSwitch Box. We have also configured one gateway in eternal/provider.xml with the IP of provider and disable the registration as provider is not expecting the same(they dont have any user configured for us, as they are using their box in trunk mode). Issue is that when an incoming call reaches to provider, we are not getting any INVITE from them and its being dropped at the IP that we have added as gateway. Reason shared is that they are expecting us to have heartbeat with their IP. When we turn ping option, it sends OPTION message but username is getting appended to TO and FROM uri's so that cause 404 No User found response. Please help to configure FS to send option something like this- *OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0* Via: SIP/2.0/UDP 10.129.43.23:5060; rport;branch=z9hG4bK- 6df4-1156281802-19999-423 Call-ID: 5aa9-1e61-7222006212322-chiloe-0 at 10.129.43.23 CSeq: 1 OPTIONS Max-Forwards: 70 *To: * *From: * ;tag=95ffcd055e0f78f7d5d397020e8 9288db5f2 User-Agent: Dialogic-SIP/10.3.2.57 chiloe 0 Contact: Accept: application/ sdp Content-Length: 0 i.e. They want OPTION with only IP address not user at IP. So that on recieving any call they can transfer it to our FS. Please Help. Thanks -Manav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/81ba123e/attachment.html From georg at riseup.net Thu Jan 26 12:08:12 2012 From: georg at riseup.net (georg at riseup.net) Date: Thu, 26 Jan 2012 10:08:12 +0100 Subject: [Freeswitch-users] Gateway Configuration with No Authentication In-Reply-To: References: Message-ID: <89ffbc8eca512e18dd483b1a2f072324.squirrel@fulvetta.riseup.net> He Vishal, > We are trying to use FS to handle incoming call from SIP provider. > Provider > doesnt have any authentication because its all inbound traffic. They have > configured their SIP trunk to point to our FreeSwitch Box. I don't think you have to use a username. I've got a similar setup, my external.xml config is looking like: HTH, Georg From avi at avimarcus.net Thu Jan 26 12:58:56 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 26 Jan 2012 11:58:56 +0200 Subject: [Freeswitch-users] Gateway Configuration with No Authentication In-Reply-To: <89ffbc8eca512e18dd483b1a2f072324.squirrel@fulvetta.riseup.net> References: <89ffbc8eca512e18dd483b1a2f072324.squirrel@fulvetta.riseup.net> Message-ID: I usually have my origination providers send to $number@$myserver:5080 so it goes to external and hits the public context. Or, if they won't, I create a user with their ACL and set the context on the user to public. Hmm, perhaps external gateway that doesn't register would work and be cleaner.. -Avi On Thu, Jan 26, 2012 at 11:08 AM, wrote: > He Vishal, > > > We are trying to use FS to handle incoming call from SIP provider. > > Provider > > doesnt have any authentication because its all inbound traffic. They have > > configured their SIP trunk to point to our FreeSwitch Box. > > I don't think you have to use a username. I've got a similar setup, my > external.xml config is looking like: > > > > > > > > > > > > HTH, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/b4cf1bce/attachment.html From asilva at wirelessmundi.com Thu Jan 26 13:05:06 2012 From: asilva at wirelessmundi.com (Antonio) Date: Thu, 26 Jan 2012 11:05:06 +0100 Subject: [Freeswitch-users] Registration Failed with status DNS Error In-Reply-To: <29F00015-5507-4795-8BD6-5981463324A6@jerris.com> References: <1327509455.10812.221.camel@marces.madrid.commsmundi.com> <29F00015-5507-4795-8BD6-5981463324A6@jerris.com> Message-ID: <1327572306.10812.240.camel@marces.madrid.commsmundi.com> Mike, thanks for the answer. Actually the problem is the DNS server doesn't have an A entry for the destination domain. When i add this entry to the domain, the register works nice. Shouldn't freeswitch try to resolve first to the SRV entry and then if none is found, try to resolve the A entry? you can find my logs in http://pastebin.freeswitch.org/18231 I read something in the jira for some options to disable SRV in the profile, i look at the code and it seams to be enabled by default. Just to be sure i also try to put in the profile all the possibilities to disable-srv and disable-srv503, without luck. Thanks, Ant?nio On Wed, 2012-01-25 at 13:53 -0500, Michael Jerris wrote: > Is it even sending the request? try turning on nua and sresolv debug in sofia to see more about what is going on? > > Mike > > On Jan 25, 2012, at 11:37 AM, Antonio wrote: > > > > > Hi > > > > I have a problem registering a gateway, the return error is DNS error [503]. > > > > My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. > > > > > > how does freeswitch resolves the destination domain? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/426100fd/attachment.html From Stefan.Weigel at allianz-warranty.com Thu Jan 26 14:03:16 2012 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Thu, 26 Jan 2012 12:03:16 +0100 Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing Message-ID: <5003D7D3E06F514E8C682F18D223265C05121CD349@AZWSMS03.azwarranty.int> Hi all, we're using successfully mod_callcenter. If a caller enters the queue, he/she is getting default moh sound. The sound is also playing when the call is passed to an agent and the phone is ringing. Is it possible to play a normal ringtone when a call is passed to an agent ? Thanks and best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/5c462a47/attachment-0001.html From daggelinckxmichel at gmail.com Thu Jan 26 15:17:18 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Thu, 26 Jan 2012 13:17:18 +0100 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario In-Reply-To: <4F1EBDB5.7010805@gmail.com> References: <4F183B18.40203@gmail.com> <4F1EBDB5.7010805@gmail.com> Message-ID: 2600hz.org uses freeswitch in its whistle/winkstart project for high availability and call recovery. On Tue, Jan 24, 2012 at 3:18 PM, Carlo Dimaggio wrote: > Does anybody use fs in a full high availability environment and/or call > recovery feature? > > Regards, > > > Il 19/01/12 16.47, Carlo Dimaggio ha scritto: > > Hi all, > > I would like to know if the track-calls feature could be used in a > multi-primary - multi-backup scenario. > What I think is an environment with Kamailio dispatcher to N freeswitch > boxes with a (multi or) single freeswitch backup. In case of failure, one > primary should be replaced with a backup (spare) box that will take its IP > and active calls. > > If all active freeswitch track call state in the same table/database, how > the backup box will know what calls should be recovered? > Anyone could explain the call recovery algorithm? > > > Best Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/02a9596a/attachment.html From mike at jerris.com Thu Jan 26 16:04:14 2012 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Jan 2012 08:04:14 -0500 Subject: [Freeswitch-users] FreeSWITCH load CDR CSV to Mysql, theoretical problem? In-Reply-To: <4F2052B9.2070706@cupis.co.uk> References: <4F2052B9.2070706@cupis.co.uk> Message-ID: http://search.cpan.org/dist/Proc-Pidfile/Pidfile.pm On Jan 25, 2012, at 2:06 PM, Paul Cupis wrote: > On 25/01/12 13:25, Henrik Aagaard S?rensen wrote: >> What if the job suddenly takes more than 5 minutes to complete and >> another job start, doing the same thing. >> >> Couldn't that be a problem and can it be fixed? > > Use Perls Fcntl module to ensure the program can only be running once - > if a second instance starts it will quit immediately and the next > scheduled run will pick up the remaining files. > > Of course, if this happens to often you will take a long time to catch > up with the importing. > > At the beginning of the script (after 'use File::Copy;') add: > > > use Fcntl qw(:flock); > unless (flock(DATA, LOCK_EX|LOCK_NB)) { > exit; > print "$0 is already running. Exiting.\n"; > exit(1); > } > > > and at the very end add: > > __DATA__ > This exists so flock() code above works. > DO NOT REMOVE THIS DATA SECTION. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/a436de81/attachment.html From mike at jerris.com Thu Jan 26 16:07:11 2012 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Jan 2012 08:07:11 -0500 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> Message-ID: <11171047-CD59-4868-BAD7-844F068FD3C2@jerris.com> seems you have truncated anything useful in this paste. On Jan 25, 2012, at 8:07 PM, georg at riseup.net wrote: > Hi Mike, > >> if you are explicitly doing ring_ready, its telling it to ringback before >> we actually know the phone is ringing. The approach we take passes the >> ringing message all the way back to the original device. Take a look at >> the debug to see what is actually taking that long. > > Allright. I thought of something similar, so I step back of using ring_ready. > > I tried several calls, and it seems just random: Like 50%/50%, the > ringback tone the caller hears is nearly in sync with the internal one, in > the other 50%, there's a delay of ~ three seconds. > > However, the debug output is looking the same, I posted one to [1]. > > My provider isn't supporting early media. Could this be a problem? > > Thanks, > Georg > > [1] http://dpaste.de/no6bO/ > From mike at jerris.com Thu Jan 26 16:08:38 2012 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Jan 2012 08:08:38 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> Message-ID: <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: > I saw that, and tried everything I could. Still no luck. > > My stack limit is 8192, so that shouldn?t be the issue. I?m running CentOS x_64. > > I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn?t work either (I?m still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it?s completely written and released the email it?s sending. Any idea how to make that work? > > Thank you, > Rob > > From: Michael Jerris [mailto:mike at jerris.com] > Sent: Wednesday, January 25, 2012 1:57 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > > > For some extensions, I?ve got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. > > Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. > > If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/64c643fc/attachment.html From vishal.kakkar at gmail.com Thu Jan 26 16:19:22 2012 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Thu, 26 Jan 2012 18:49:22 +0530 Subject: [Freeswitch-users] Gateway Configuration with No Authentication Message-ID: Hi, Not using username parameter in gateway is causing following OPTION message being published- OPTIONS sip:192.168.1.30;transport=udp SIP/2.0 Via: SIP/2.0/UDP 59.90.193.54:5080;rport;branch=z9hG4bKZ5D672rZgta9N Max-Forwards: 70 From: *;tag=ZS9p1ZN164S4Q To: * Call-ID: a78e7d46-c2c1-122f-d9a3-0013a9866b3c CSeq: 23455584 OPTIONS Contact: User-Agent: FreeSwitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 On Thu, Jan 26, 2012 at 4:33 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users at lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request at lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner at lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Gateway Configuration with No Authentication (Vishal Kakkar) > 2. Re: Gateway Configuration with No Authentication > (georg at riseup.net) > 3. Re: Gateway Configuration with No Authentication (Avi Marcus) > 4. Re: Registration Failed with status DNS Error (Antonio) > 5. mod_callcenter - ring tones when phone is ringing (Weigel, Stefan) > > > ---------- Forwarded message ---------- > From: Vishal Kakkar > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Thu, 26 Jan 2012 14:04:40 +0530 > Subject: [Freeswitch-users] Gateway Configuration with No Authentication > Hi all, > > We are trying to use FS to handle incoming call from SIP provider. > Provider doesnt have any authentication because its all inbound traffic. > They have configured their SIP trunk to point to our FreeSwitch Box. > > We have also configured one gateway in eternal/provider.xml with the IP of > provider and disable the registration as provider is not expecting the > same(they dont have any user configured for us, as they are using their box > in trunk mode). > > Issue is that when an incoming call reaches to provider, we are not > getting any INVITE from them and its being dropped at the IP that we have > added as gateway. Reason shared is that they are expecting us to have > heartbeat with their IP. > > When we turn ping option, it sends OPTION message but username is getting > appended to TO and FROM uri's so that cause 404 No User found response. > > Please help to configure FS to send option something like this- > > > *OPTIONS sip:10.129.43.154:5060;ttl=0 SIP/2.0* > > Via: SIP/2.0/UDP 10.129.43.23:5060; > rport;branch=z9hG4bK- > > 6df4-1156281802-19999-423 > > Call-ID: > 5aa9-1e61-7222006212322-chiloe-0 at 10.129.43.23 > > CSeq: 1 OPTIONS > > Max-Forwards: 70 > > *To: * > > *From: * > ;tag=95ffcd055e0f78f7d5d397020e8 > > 9288db5f2 > > User-Agent: Dialogic-SIP/10.3.2.57 chiloe 0 > > Contact: > > Accept: application/ sdp > > Content-Length: 0 > > i.e. They want OPTION with only IP address not user at IP. So that on > recieving any call they can transfer it to our FS. > > Please Help. > Thanks > -Manav > > > ---------- Forwarded message ---------- > From: georg at riseup.net > To: "FreeSWITCH Users Help" > Cc: > Date: Thu, 26 Jan 2012 10:08:12 +0100 > Subject: Re: [Freeswitch-users] Gateway Configuration with No > Authentication > He Vishal, > > > We are trying to use FS to handle incoming call from SIP provider. > > Provider > > doesnt have any authentication because its all inbound traffic. They have > > configured their SIP trunk to point to our FreeSwitch Box. > > I don't think you have to use a username. I've got a similar setup, my > external.xml config is looking like: > > > > > > > > > > > > HTH, > Georg > > > > > > ---------- Forwarded message ---------- > From: Avi Marcus > To: FreeSWITCH Users Help > Cc: > Date: Thu, 26 Jan 2012 11:58:56 +0200 > Subject: Re: [Freeswitch-users] Gateway Configuration with No > Authentication > I usually have my origination providers send to $number@$myserver:5080 so > it goes to external and hits the public context. > Or, if they won't, I create a user with their ACL and set the context on > the user to public. > > Hmm, perhaps external gateway that doesn't register would work and be > cleaner.. > -Avi > > > On Thu, Jan 26, 2012 at 11:08 AM, wrote: > >> He Vishal, >> >> > We are trying to use FS to handle incoming call from SIP provider. >> > Provider >> > doesnt have any authentication because its all inbound traffic. They >> have >> > configured their SIP trunk to point to our FreeSwitch Box. >> >> I don't think you have to use a username. I've got a similar setup, my >> external.xml config is looking like: >> >> >> >> >> >> >> >> >> >> >> >> HTH, >> Georg >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > ---------- Forwarded message ---------- > From: Antonio > To: FreeSWITCH Users Help > Cc: > Date: Thu, 26 Jan 2012 11:05:06 +0100 > Subject: Re: [Freeswitch-users] Registration Failed with status DNS Error > ** > Mike, thanks for the answer. > > Actually the problem is the DNS server doesn't have an A entry for the > destination domain. When i add this entry to the domain, the register works > nice. > > Shouldn't freeswitch try to resolve first to the SRV entry and then if > none is found, try to resolve the A entry? > > you can find my logs in http://pastebin.freeswitch.org/18231 > > > I read something in the jira for some options to disable SRV in the > profile, i look at the code and it seams to be enabled by default. > Just to be sure i also try to put in the profile all the possibilities to > disable-srv and disable-srv503, without luck. > > > > Thanks, > Ant?nio > > > > On Wed, 2012-01-25 at 13:53 -0500, Michael Jerris wrote: > > Is it even sending the request? try turning on nua and sresolv debug in sofia to see more about what is going on? > > Mike > > On Jan 25, 2012, at 11:37 AM, Antonio wrote: > > > > > Hi > > > > I have a problem registering a gateway, the return error is DNS error [503]. > > > > My DNS is OK, i can dig to the destination domain in question, and even when o do sofia_dig in fresswith it resolves the domain without problems. > > > > > > how does freeswitch resolves the destination domain? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: "Weigel, Stefan" > To: "'freeswitch-users at lists.freeswitch.org'" < > freeswitch-users at lists.freeswitch.org> > Cc: > Date: Thu, 26 Jan 2012 12:03:16 +0100 > Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is > ringing > > Hi all,**** > > ** ** > > we?re using successfully mod_callcenter. If a caller enters the queue, > he/she is getting default moh sound. The sound is also playing when the > call is passed to an agent and the phone is ringing. Is it possible to play > a normal ringtone when a call is passed to an agent ?**** > > ** ** > > ** ** > > Thanks and best regards**** > > ** ** > > Stefan**** > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/be3d71a3/attachment-0001.html From jaasmailing at gmail.com Thu Jan 26 16:35:38 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Thu, 26 Jan 2012 14:35:38 +0100 Subject: [Freeswitch-users] Call recovery - Multi-primary / multi-backup scenario In-Reply-To: References: <4F183B18.40203@gmail.com> <4F1EBDB5.7010805@gmail.com> Message-ID: <4F2156AA.4020203@gmail.com> Hi Michel, thanks, but I would likt to know the internal algorithm implemented in freeswitch and an ha scenario with kamailio. Regards, Il 26/01/12 13.17, Michel Daggelinckx ha scritto: > 2600hz.org uses freeswitch in its > whistle/winkstart project for high availability and call recovery. > > > On Tue, Jan 24, 2012 at 3:18 PM, Carlo Dimaggio > wrote: > > Does anybody use fs in a full high availability environment and/or > call recovery feature? > > Regards, > > > Il 19/01/12 16.47, Carlo Dimaggio ha scritto: >> Hi all, >> >> I would like to know if the track-calls feature could be used in >> a multi-primary - multi-backup scenario. >> What I think is an environment with Kamailio dispatcher to N >> freeswitch boxes with a (multi or) single freeswitch backup. In >> case of failure, one primary should be replaced with a backup >> (spare) box that will take its IP and active calls. >> >> If all active freeswitch track call state in the same >> table/database, how the backup box will know what calls should be >> recovered? >> Anyone could explain the call recovery algorithm? >> >> >> Best Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/ab13f118/attachment.html From all.eforums at gmail.com Thu Jan 26 17:15:09 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Thu, 26 Jan 2012 09:15:09 -0500 Subject: [Freeswitch-users] Skypopen on debian squeeze - All sorts of fonts and alsa/oss driver and binary issues Message-ID: Hello All, anyone tried to get Skypopen to work in Debian? I am not really sure what the real issue is as I get multiple errors when trying to start but the biggest one I guess is the one claiming that the skype binary cannot be executed. Will provide further info as requested as I really don't know what the issue really is. Does the Skype client need a real sound card/driver present? I doubt it. Does it really need a real screen attached to it? I doubt it since I see some magic being done with the Xserver to create fake displays and what not. Anyway, here's what I get when I try to start the start_skype_clients.sh *ERROR: Module snd_pcm_oss does not exist in /proc/modules* *ERROR: Module snd_mixer_oss does not exist in /proc/modules* *ERROR: Module snd_seq_oss does not exist in /proc/modules* *mknod: `/dev/dsp': File exists* *insmod: error inserting '/opt/freeswitch/skypopen/sound-drv/skypopen.ko': -1 File exists* *SELinux: Disabled on system, not enabling in X server* *[dix] Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!* *bash: /opt/freeswitch/skypopen/clientsymlnk/skype101: cannot execute binary file* Any ideas? Giovanni? :) Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/6eecaa43/attachment.html From mike at jerris.com Thu Jan 26 17:23:39 2012 From: mike at jerris.com (Michael Jerris) Date: Thu, 26 Jan 2012 09:23:39 -0500 Subject: [Freeswitch-users] Gateway Configuration with No Authentication In-Reply-To: References: Message-ID: <354FE91E-4C74-4E52-A30E-36D40157C8A5@jerris.com> please open a bug on this http://wiki.freeswitch.org/wiki/Reporting_Bugs#Reporting_A_Bug_With_JIRA On Jan 26, 2012, at 8:19 AM, Vishal Kakkar wrote: > Hi, > > Not using username parameter in gateway is causing following OPTION message being published- > > OPTIONS sip:192.168.1.30;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 59.90.193.54:5080;rport;branch=z9hG4bKZ5D672rZgta9N > Max-Forwards: 70 > From: ;tag=ZS9p1ZN164S4Q > To: > Call-ID: a78e7d46-c2c1-122f-d9a3-0013a9866b3c > CSeq: 23455584 OPTIONS > Contact: > User-Agent: FreeSwitch > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Length: 0 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/7b97900f/attachment.html From rmorin at blie-ent.com Thu Jan 26 18:13:50 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Thu, 26 Jan 2012 10:13:50 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> Message-ID: <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> I tried to modify the eximcompat.sh script to work with sendmail. I'm not sure why it didn't work. It might be that, for starters, I don't have exim installed, I have sendmail. I haven't tried the python script. Is that the one you use? Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, January 26, 2012 8:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/bb8616d0/attachment-0001.html From georg at riseup.net Thu Jan 26 19:52:03 2012 From: georg at riseup.net (georg at riseup.net) Date: Thu, 26 Jan 2012 17:52:03 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <11171047-CD59-4868-BAD7-844F068FD3C2@jerris.com> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> <11171047-CD59-4868-BAD7-844F068FD3C2@jerris.com> Message-ID: <7d4c72305700590f9b3a813729aec63b.squirrel@fulvetta.riseup.net> > seems you have truncated anything useful in this paste. You mean the "XXX"? Or is the paste "to short" or the loglevel not "deep" enough? Thanks, Georg From bote_radio at botecomm.com Thu Jan 26 19:55:06 2012 From: bote_radio at botecomm.com (Bote Man) Date: Thu, 26 Jan 2012 11:55:06 -0500 Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing In-Reply-To: <5003D7D3E06F514E8C682F18D223265C05121CD349@AZWSMS03.azwarranty.int> References: <5003D7D3E06F514E8C682F18D223265C05121CD349@AZWSMS03.azwarranty.int> Message-ID: <02fe01ccdc4b$42985840$c7c908c0$@com> If you do not need the music on hold, create the desired ringback tone as a sound file and install that in place of the moh sound. Because the moh sound repeats forever you only need to create a file long enough for the ringback tone and silence period. I do not know a method to change from moh to generated ringback because I am new to FreeSWITCH and still learning. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Weigel, Stefan Sent: Thursday, 26 January, 2012 06:03 To: 'freeswitch-users at lists.freeswitch.org' Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing Hi all, we?re using successfully mod_callcenter. If a caller enters the queue, he/she is getting default moh sound. The sound is also playing when the call is passed to an agent and the phone is ringing. Is it possible to play a normal ringtone when a call is passed to an agent ? Thanks and best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/33d4a393/attachment.html From freeswitch at earthspike.net Thu Jan 26 21:48:41 2012 From: freeswitch at earthspike.net (John) Date: Thu, 26 Jan 2012 18:48:41 +0000 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> Message-ID: <4F21A009.8080400@earthspike.net> Slightly unrelated, but I have been using the python script sendemail.py until recently when a user noticed that they were not being emailed some voicemails. I have since changed to using ssmtp (rather than nullmailer as my mail server requires a login, TLS on tcp/587). One thing I noticed, and it may be me that caused it, but the sendemail.py script had some indentation using (4) spaces and some using tabs. But the main problem is that the script keeps warning about the MimeWriter class being deprecated and has no error reporting or recovery, it seems. For me, it was easier to install ssmtp than rewrite the python script. If you don't need a full mail server, I would recommend using a null mailer like ssmtp or nullmailer. John On 26/01/12 15:13, Rob Morin wrote: > > I tried to modify the eximcompat.sh script to work with sendmail. I'm > not sure why it didn't work. It might be that, for starters, I don't > have exim installed, I have sendmail. > > I haven't tried the python script. Is that the one you use? > > Rob > > *From:*Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, January 26, 2012 8:09 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > delay won't help you. Why didn't the script work? I know we do this > and it works fine. Pretty sure we use a modified version of the > sample in tree. > > Mike > > On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: > > > > I saw that, and tried everything I could. Still no luck. > > My stack limit is 8192, so that shouldn't be the issue. I'm running > CentOS x_64. > > I tried creating a script as the wiki suggests, so that I could make > changes without having to restart FS. But that didn't work either (I'm > still a noob). My goal was to add a sleep delay, in the case that FS > is calling the sendmail command before it's completely written and > released the email it's sending. Any idea how to make that work? > > Thank you, > > Rob > > *From:*Michael Jerris [mailto:mike at jerris.com] > > *Sent:*Wednesday, January 25, 2012 1:57 PM > *To:*FreeSWITCH Users Help > *Subject:*Re: [Freeswitch-users] Sendmail segfaulting before second > message is sent > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > > > > > For some extensions, I've got FreeSWITCH configured to send both the > voicemail to an email address and a notification to a cell phone that > there is a message. > > Sendmail is segfaulting (error 6) before the second email message is > sent. Sometimes the notification message is going out, sometimes the > email message goes. > > If I am just sending an email, it is **generally** successful. > Although sometimes it fails, so I really need to fix this. > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/b05bd055/attachment-0001.html From mgende at gendesign.com Thu Jan 26 22:19:01 2012 From: mgende at gendesign.com (Michael Gende) Date: Thu, 26 Jan 2012 13:19:01 -0600 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: <4F1F90DF.30405@coppice.org> <4F2030B8.201@coppice.org> Message-ID: Hello Moy, Responding to your Tesla post - tongue in cheek yet with truth - while Mr Tesla was a legitimate genius (do many understand how much we owe to his accomplishments?) his money handling was not the best. I shot you a phone call but please let me leave my number here: 815.953.5620. Call anytime, I'm usually near that phone (when not tinkering with my home-made, over-sized Tesla coils). Regards, Mike G. On Wed, Jan 25, 2012 at 6:36 PM, Michael Gende wrote: > Hey Moy, > > Along the lines of Steve's comment, I spoke to Anthony (probably only his > parents call him that) and Brian yesterday about this. They suggested you > and I speak. I note your phone number on the posts so I'll probably give > you a call, assuming that is acceptable to you. > > Regards, > > Mike G. > > > On Wed, Jan 25, 2012 at 10:41 AM, Steve Underwood wrote: > >> Hi Moy, >> >> On 01/25/2012 11:50 PM, Moises Silva wrote: >> > On Wed, Jan 25, 2012 at 9:00 AM, Michael Gende > > > wrote: >> > >> > Hey Steve, >> > >> > I know that Asterisk was used, we were once using it ourselves and >> > had Digium produce a CAMA stack (as you noted, no great shakes, I >> > think one guy did it in a weekend. It sure wasn't priced that way!). >> > >> > >> > LOL ... I'm sure you've heard some variation of this one: >> > >> > "Nikola Tesla visited Henry Ford at his factory, which was having some >> > kind of difficulty. Ford asked Tesla if he could help identify the >> > problem area. >> > Tesla walked up to a wall of boilerplate and made a small X in chalk >> > on one of the plates. Ford was thrilled, and told him to send an >> invoice. >> > >> > The bill arrived, for $10,000. Ford asked for a breakdown. Tesla >> > sent another invoice, indicating a $1 charge for marking the wall with >> > an X, and $9,999 for knowing where to put it." >> > >> > Taken from: http://www.snopes.com/business/genius/where.asp >> > >> > - >> > Moy >> > >> You omitted to offer your own reasonably priced services. Does Sangoma >> already have a CAMA protocol? >> >> Steve >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/489cf74e/attachment.html From rmorin at blie-ent.com Fri Jan 27 01:04:44 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Thu, 26 Jan 2012 17:04:44 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <4F21A009.8080400@earthspike.net> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> Message-ID: <193001ccdc76$830b28a0$892179e0$@blie-ent.com> John, That's helpful, but ssmtp doesn't support attachments so it won't support sending the voicemail files. Additionally, I'm somewhat stumped on the default 'sendmail' option. The references in the Wiki are basically all 'configuration' issues - stack size, etc. The problem I have is that it isn't a configuration issue - sometimes it goes through, sometimes it doesn't. If it were a configuration issue, it would never go through (unless it was a resource that was near its margin). So that's why I wanted to insert a delay. It appears to me that the segfault is being caused by the file getting moved, deleted, or still being open when sendmail attempts to access it. I wish I had more to go on though. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 1:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Slightly unrelated, but I have been using the python script sendemail.py until recently when a user noticed that they were not being emailed some voicemails. I have since changed to using ssmtp (rather than nullmailer as my mail server requires a login, TLS on tcp/587). One thing I noticed, and it may be me that caused it, but the sendemail.py script had some indentation using (4) spaces and some using tabs. But the main problem is that the script keeps warning about the MimeWriter class being deprecated and has no error reporting or recovery, it seems. For me, it was easier to install ssmtp than rewrite the python script. If you don't need a full mail server, I would recommend using a null mailer like ssmtp or nullmailer. John On 26/01/12 15:13, Rob Morin wrote: I tried to modify the eximcompat.sh script to work with sendmail. I'm not sure why it didn't work. It might be that, for starters, I don't have exim installed, I have sendmail. I haven't tried the python script. Is that the one you use? Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, January 26, 2012 8:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/a4fe4b7d/attachment-0001.html From brian at neotiq.com Fri Jan 27 01:15:00 2012 From: brian at neotiq.com (Brian) Date: Thu, 26 Jan 2012 23:15:00 +0100 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk Message-ID: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> Hi List, I'm quite new to FreeSwitch and I'm trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven't managed to figure it out how to do it with FreeSwitch. That is about the "s" extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the "s" extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/4924c59e/attachment.html From freeswitch at earthspike.net Fri Jan 27 02:23:04 2012 From: freeswitch at earthspike.net (John) Date: Thu, 26 Jan 2012 23:23:04 +0000 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <193001ccdc76$830b28a0$892179e0$@blie-ent.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> Message-ID: <4F21E058.9000004@earthspike.net> Rob, ssmtp sends whatever it's given: FreeSWITCH puts the message together (in src/switch_utils.c:switch_simple_email()) to make a fully MIME-compliant message with the attachments read out of their files and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, sendemail.py or whatever) program for it to send/relay it. FreeSWITCH waits for the 'cat | sendmail' program to return before continuing execution and deleting files, so file deletion is unlikely to be the issue. Like you say, it sounds like the problem is not with configuration but it's also very unlikely to be with file deletion. Can you reproduce the fault by running sendmail from the command line with a suitably formatted mail message[*] as input? This is pretty much what FreeSWITCH does: cat message.txt | -f noreply at mydomain.com recipient at mydomain.com ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? [* 'suitably formatted message' could be one of the successful ones, view source, then edit to trim off the transport headers] If that is successful, try to replicate your 2-message-scenario by running two of the above commands in parallel with different input files and see if that generates a segfault. [If you are a linux noob as well as a FS noob I can spell this out for you.] John On 26/01/12 22:04, Rob Morin wrote: > > John, > > > > That's helpful, but ssmtp doesn't support attachments so it won't > support sending the voicemail files. > > > > Additionally, I'm somewhat stumped on the default 'sendmail' option. > The references in the Wiki are basically all 'configuration' issues -- > stack size, etc. The problem I have is that it isn't a configuration > issue -- sometimes it goes through, sometimes it doesn't. If it were a > configuration issue, it would never go through (unless it was a > resource that was near its margin). So that's why I wanted to insert > a delay. It appears to me that the segfault is being caused by the > file getting moved, deleted, or still being open when sendmail > attempts to access it. I wish I had more to go on though. > > > > Thank you, > > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 1:49 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Slightly unrelated, but I have been using the python script > sendemail.py until recently when a user noticed that they were not > being emailed some voicemails. I have since changed to using ssmtp > (rather than nullmailer as my mail server requires a login, TLS on > tcp/587). One thing I noticed, and it may be me that caused it, but > the sendemail.py script had some indentation using (4) spaces and some > using tabs. But the main problem is that the script keeps warning > about the MimeWriter class being deprecated and has no error reporting > or recovery, it seems. For me, it was easier to install ssmtp than > rewrite the python script. If you don't need a full mail server, I > would recommend using a null mailer like ssmtp or nullmailer. > > John > > On 26/01/12 15:13, Rob Morin wrote: > > I tried to modify the eximcompat.sh script to work with sendmail. I'm > not sure why it didn't work. It might be that, for starters, I don't > have exim installed, I have sendmail. > > > > I haven't tried the python script. Is that the one you use? > > > > Rob > > > > *From:*Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, January 26, 2012 8:09 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > delay won't help you. Why didn't the script work? I know we do this > and it works fine. Pretty sure we use a modified version of the > sample in tree. > > > > Mike > > > > On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: > > > > > I saw that, and tried everything I could. Still no luck. > > > > My stack limit is 8192, so that shouldn't be the issue. I'm running > CentOS x_64. > > > > I tried creating a script as the wiki suggests, so that I could make > changes without having to restart FS. But that didn't work either (I'm > still a noob). My goal was to add a sleep delay, in the case that FS > is calling the sendmail command before it's completely written and > released the email it's sending. Any idea how to make that work? > > Thank you, > > Rob > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > > *Sent:* Wednesday, January 25, 2012 1:57 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting before second > message is sent > > > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > > > On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > > > > > > For some extensions, I've got FreeSWITCH configured to send both the > voicemail to an email address and a notification to a cell phone that > there is a message. > > > > Sendmail is segfaulting (error 6) before the second email message is > sent. Sometimes the notification message is going out, sometimes the > email message goes. > > > > If I am just sending an email, it is **generally** successful. > Although sometimes it fails, so I really need to fix this. > > > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/f502a6a3/attachment-0001.html From rmorin at blie-ent.com Fri Jan 27 03:46:56 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Thu, 26 Jan 2012 19:46:56 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <4F21E058.9000004@earthspike.net> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> <4F21E058.9000004@earthspike.net> Message-ID: <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> John, Thank you for your help. I ran the command line as you described, and everything worked fine. I should also mention, that some of my extensions do not have the notifications turned on. And even for them, getting the message through is hit or miss - sometimes it segfaults, sometimes it doesn't. So I don't necessarily think it's a "two message" issue. Also, in looking at the logs, sometimes I see the segfault appear in the middle of the maillog logging for the other message. So it doesn't look like FS is waiting for the sendmail command to return before executing the next command. Here's what I mean. Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: Authentication-Warning: blie-fs.blie-ent.com: freeswitch set sender to voicemail at blie-ent.com using -f Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: from=voicemail at blie-ent.com, size=18662, class=0, nrcpts=1, msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, relay=freeswitch at localhost Jan 26 14:28:51 blie-fs sendmail[25201]: q0QESpVj025201: from=, size=19026, class=0, nrcpts=1, msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, proto=ESMTP, daemon=MTA, relay=localhost [127.0.0.1] Jan 26 14:28:51 blie-fs sendmail[25200]: q0QESoi9025200: to=rmorin at blie-ent.com, ctladdr=voicemail at blie-ent.com (501/501), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=48662, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (q0QESpVj025201 Message accepted for delivery) Jan 26 14:28:51 blie-fs kernel: [41082.268638] sendmail[25207]: segfault at 7fff96dffed8 rip 7f158ee0b7fb rsp 7fff96dffea0 error 6 Jan 26 14:28:51 blie-fs sendmail[25203]: STARTTLS=client, relay=smtp.sendgrid.net., version=TLSv1/SSLv3, verify=FAIL, cipher=DHE-RSA-AES256-SHA, bits=256/256 Jan 26 14:28:51 blie-fs sendmail[25203]: q0QESpVj025201: to=, delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=139026, relay=smtp.sendgrid.net. [174.36.32.204], dsn=2.0.0, stat=Sent (Delivery in progress) On this instance, the notification didn't get through, but the message did. All of that said, I'm not sure how to run the 'parallel' test, though. (I did run the single test many times in quick succession with no failures.) My ulimit -s is set at 8192. And I don't have any other arguments, other than '-t'. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 6:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting Rob, ssmtp sends whatever it's given: FreeSWITCH puts the message together (in src/switch_utils.c:switch_simple_email()) to make a fully MIME-compliant message with the attachments read out of their files and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, sendemail.py or whatever) program for it to send/relay it. FreeSWITCH waits for the 'cat | sendmail' program to return before continuing execution and deleting files, so file deletion is unlikely to be the issue. Like you say, it sounds like the problem is not with configuration but it's also very unlikely to be with file deletion. Can you reproduce the fault by running sendmail from the command line with a suitably formatted mail message[*] as input? This is pretty much what FreeSWITCH does: cat message.txt | -f noreply at mydomain.com recipient at mydomain.com ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? [* 'suitably formatted message' could be one of the successful ones, view source, then edit to trim off the transport headers] If that is successful, try to replicate your 2-message-scenario by running two of the above commands in parallel with different input files and see if that generates a segfault. [If you are a linux noob as well as a FS noob I can spell this out for you.] John On 26/01/12 22:04, Rob Morin wrote: John, That's helpful, but ssmtp doesn't support attachments so it won't support sending the voicemail files. Additionally, I'm somewhat stumped on the default 'sendmail' option. The references in the Wiki are basically all 'configuration' issues - stack size, etc. The problem I have is that it isn't a configuration issue - sometimes it goes through, sometimes it doesn't. If it were a configuration issue, it would never go through (unless it was a resource that was near its margin). So that's why I wanted to insert a delay. It appears to me that the segfault is being caused by the file getting moved, deleted, or still being open when sendmail attempts to access it. I wish I had more to go on though. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 1:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Slightly unrelated, but I have been using the python script sendemail.py until recently when a user noticed that they were not being emailed some voicemails. I have since changed to using ssmtp (rather than nullmailer as my mail server requires a login, TLS on tcp/587). One thing I noticed, and it may be me that caused it, but the sendemail.py script had some indentation using (4) spaces and some using tabs. But the main problem is that the script keeps warning about the MimeWriter class being deprecated and has no error reporting or recovery, it seems. For me, it was easier to install ssmtp than rewrite the python script. If you don't need a full mail server, I would recommend using a null mailer like ssmtp or nullmailer. John On 26/01/12 15:13, Rob Morin wrote: I tried to modify the eximcompat.sh script to work with sendmail. I'm not sure why it didn't work. It might be that, for starters, I don't have exim installed, I have sendmail. I haven't tried the python script. Is that the one you use? Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, January 26, 2012 8:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/44dde311/attachment-0001.html From freeswitch at earthspike.net Fri Jan 27 04:40:52 2012 From: freeswitch at earthspike.net (John) Date: Fri, 27 Jan 2012 01:40:52 +0000 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> <4F21E058.9000004@earthspike.net> <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> Message-ID: <4F2200A4.3010300@earthspike.net> Rob, sendmail is running as a service, so the segfault could be being thrown by the sendmail service, not necessarily by the command-line-app-version that adds mails to the queue (and then terminates). Most mailers work this way: separate processes to add the mail to the queue, process the queue, accept connections on various sockets. FreeSWITCH is written to wait until the queue-insertion invocation of sendmail is completed. Thankfully, it doesn't wait until the message is actually sent on to its next location as, in some cases, that could take hours if the remote host is down. In your log below, it looks like sendmail [25200] has happily accepted the message from FreeSWITCH and queued it for delivery. sendmail [25207] is the process that has then barfed, after the message has been queued, which suggests that it is nothing to do with FreeSWITCH itself. Perhaps there might be a copy of the message from the crashing process in the message queues in /var/spool/mail (or similar) which you could look at to see if there is anything weird in it? But the bottom line is that it looks like you have pure sendmail problem. Unfortunately, that's where my knowledge stops as I run Postfix as my main mailer and ssmtp as a dumb relay. Maybe someone on a CentOS or sendmail list can help you with the sendmail issue? Or, as I mentioned earlier, if you don't need a full mailer, you could remove sendmail and install a simpler mailer such as ssmtp (as I see you have a relay that needs TLS and for which authentication appears to be failing). John PS. The parallel test is to wrap the original command in parentheses and duplicate it several times on a single command line, each time followed by an ampersand to put it into the background, ie: rmorin at blie-fs$ ( cat .... | sendmail ...... ... ) & ( cat ..... | sendmail .... ) & .... and so on [It maybe that your sendmail queues faster than your shell can invoke sendmail, so it isn't really in parallel, but then that should be similar for FreeSWITCH too, so it would be a 'rapid-fire' test instead of a parallel test.] On 27/01/12 00:46, Rob Morin wrote: > > John, > > > > Thank you for your help. > > > > I ran the command line as you described, and everything worked fine. > > > > I should also mention, that some of my extensions do not have the > notifications turned on. And even for them, getting the message > through is hit or miss -- sometimes it segfaults, sometimes it > doesn't. So I don't necessarily think it's a "two message" issue. > Also, in looking at the logs, sometimes I see the segfault appear in > the middle of the maillog logging for the other message. So it doesn't > look like FS is waiting for the sendmail command to return before > executing the next command. Here's what I mean... > > > > Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: > Authentication-Warning: blie-fs.blie-ent.com: freeswitch set sender to > voicemail at blie-ent.com using -f > > Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: > from=voicemail at blie-ent.com, size=18662, class=0, nrcpts=1, > msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, > relay=freeswitch at localhost > > Jan 26 14:28:51 blie-fs sendmail[25201]: q0QESpVj025201: > from=, size=19026, class=0, nrcpts=1, > msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, proto=ESMTP, > daemon=MTA, relay=localhost [127.0.0.1] > > Jan 26 14:28:51 blie-fs sendmail[25200]: q0QESoi9025200: > to=rmorin at blie-ent.com, ctladdr=voicemail at blie-ent.com (501/501), > delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=48662, > relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (q0QESpVj025201 > Message accepted for delivery) > > Jan 26 14:28:51 blie-fs kernel: [41082.268638] sendmail[25207]: > segfault at 7fff96dffed8 rip 7f158ee0b7fb rsp 7fff96dffea0 error 6 > > Jan 26 14:28:51 blie-fs sendmail[25203]: STARTTLS=client, > relay=smtp.sendgrid.net., version=TLSv1/SSLv3, verify=FAIL, > cipher=DHE-RSA-AES256-SHA, bits=256/256 > > Jan 26 14:28:51 blie-fs sendmail[25203]: q0QESpVj025201: > to=, delay=00:00:00, xdelay=00:00:00, > mailer=relay, pri=139026, relay=smtp.sendgrid.net. [174.36.32.204], > dsn=2.0.0, stat=Sent (Delivery in progress) > > > > On this instance, the notification didn't get through, but the message > did. > > > > All of that said, I'm not sure how to run the 'parallel' test, though. > (I did run the single test many times in quick succession with no > failures.) > > > > My ulimit --s is set at 8192. And I don't have any other arguments, > other than '-t'. > > > > Thank you, > > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 6:23 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Rob, > > ssmtp sends whatever it's given: FreeSWITCH puts the message together > (in src/switch_utils.c:switch_simple_email()) to make a fully > MIME-compliant message with the attachments read out of their files > and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, > sendemail.py or whatever) program for it to send/relay it. FreeSWITCH > waits for the 'cat | sendmail' program to return before continuing > execution and deleting files, so file deletion is unlikely to be the > issue. Like you say, it sounds like the problem is not with > configuration but it's also very unlikely to be with file deletion. > > Can you reproduce the fault by running sendmail from the command line > with a suitably formatted mail message[*] as input? This is pretty > much what FreeSWITCH does: > > cat message.txt | -f noreply at mydomain.com > recipient at mydomain.com > > > ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? > > [* 'suitably formatted message' could be one of the successful ones, > view source, then edit to trim off the transport headers] > > If that is successful, try to replicate your 2-message-scenario by > running two of the above commands in parallel with different input > files and see if that generates a segfault. [If you are a linux noob > as well as a FS noob I can spell this out for you.] > > John > > On 26/01/12 22:04, Rob Morin wrote: > > John, > > > > That's helpful, but ssmtp doesn't support attachments so it won't > support sending the voicemail files. > > > > Additionally, I'm somewhat stumped on the default 'sendmail' option. > The references in the Wiki are basically all 'configuration' issues -- > stack size, etc. The problem I have is that it isn't a configuration > issue -- sometimes it goes through, sometimes it doesn't. If it were a > configuration issue, it would never go through (unless it was a > resource that was near its margin). So that's why I wanted to insert > a delay. It appears to me that the segfault is being caused by the > file getting moved, deleted, or still being open when sendmail > attempts to access it. I wish I had more to go on though. > > > > Thank you, > > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 1:49 PM > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Slightly unrelated, but I have been using the python script > sendemail.py until recently when a user noticed that they were not > being emailed some voicemails. I have since changed to using ssmtp > (rather than nullmailer as my mail server requires a login, TLS on > tcp/587). One thing I noticed, and it may be me that caused it, but > the sendemail.py script had some indentation using (4) spaces and some > using tabs. But the main problem is that the script keeps warning > about the MimeWriter class being deprecated and has no error reporting > or recovery, it seems. For me, it was easier to install ssmtp than > rewrite the python script. If you don't need a full mail server, I > would recommend using a null mailer like ssmtp or nullmailer. > > John > > On 26/01/12 15:13, Rob Morin wrote: > > I tried to modify the eximcompat.sh script to work with sendmail. I'm > not sure why it didn't work. It might be that, for starters, I don't > have exim installed, I have sendmail. > > > > I haven't tried the python script. Is that the one you use? > > > > Rob > > > > *From:*Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, January 26, 2012 8:09 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > delay won't help you. Why didn't the script work? I know we do this > and it works fine. Pretty sure we use a modified version of the > sample in tree. > > > > Mike > > > > On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: > > > > > > I saw that, and tried everything I could. Still no luck. > > > > My stack limit is 8192, so that shouldn't be the issue. I'm running > CentOS x_64. > > > > I tried creating a script as the wiki suggests, so that I could make > changes without having to restart FS. But that didn't work either (I'm > still a noob). My goal was to add a sleep delay, in the case that FS > is calling the sendmail command before it's completely written and > released the email it's sending. Any idea how to make that work? > > Thank you, > > Rob > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > > *Sent:* Wednesday, January 25, 2012 1:57 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting before second > message is sent > > > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > > > On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > > > > > > > For some extensions, I've got FreeSWITCH configured to send both the > voicemail to an email address and a notification to a cell phone that > there is a message. > > > > Sendmail is segfaulting (error 6) before the second email message is > sent. Sometimes the notification message is going out, sometimes the > email message goes. > > > > If I am just sending an email, it is **generally** successful. > Although sometimes it fails, so I really need to fix this. > > > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/84397df5/attachment-0001.html From elliott at zoogmedia.com Fri Jan 27 06:00:36 2012 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 27 Jan 2012 03:00:36 +0000 Subject: [Freeswitch-users] Options ping Message-ID: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com> Hello, how can I make FS send Option pings to users that are registered? I would like to send a ping every 120 seconds because we are having problems with remote users that are behind firewalls using polycom phones, their firewalls are closing ports if there is no activity and calls fail. Also if anyone has better way to keep alive a firewall port I would love to hear it. Elliott -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/6e67075b/attachment.html From georg at riseup.net Fri Jan 27 06:08:30 2012 From: georg at riseup.net (georg at riseup.net) Date: Fri, 27 Jan 2012 04:08:30 +0100 Subject: [Freeswitch-users] Options ping In-Reply-To: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com> References: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com> Message-ID: <89a8f0cef8dc002798a684ad051ff204.squirrel@fulvetta.riseup.net> Hi Elliot, You could use to exactly address your issue as described in [1]. Greetings, Georg [1] http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files From rmorin at blie-ent.com Fri Jan 27 06:14:12 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Thu, 26 Jan 2012 22:14:12 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <4F2200A4.3010300@earthspike.net> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> <4F21E058.9000004@earthspike.net> <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> <4F2200A4.3010300@earthspike.net> Message-ID: <19c501ccdca1$beb1a3e0$3c14eba0$@blie-ent.com> Okay, this is not expected. I switched over to ssmtp, since from what I can tell, that attaching of the voicemail file occurs within Freeswitch. It isn't sending anything, although, again, I can send it from the command line. When I run freeswitch from the command line and leave a voicemail, it gives the following errors: ssmtp: recipients with -t option not supported ssmtp: recipients with -t option not supported The only problem is, my switch.conf file looks like this, So, where is it getting the "-t " ??? Just to be safe, I removed the line and rebooted. But it is still there? Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 8:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Rob, sendmail is running as a service, so the segfault could be being thrown by the sendmail service, not necessarily by the command-line-app-version that adds mails to the queue (and then terminates). Most mailers work this way: separate processes to add the mail to the queue, process the queue, accept connections on various sockets. FreeSWITCH is written to wait until the queue-insertion invocation of sendmail is completed. Thankfully, it doesn't wait until the message is actually sent on to its next location as, in some cases, that could take hours if the remote host is down. In your log below, it looks like sendmail [25200] has happily accepted the message from FreeSWITCH and queued it for delivery. sendmail [25207] is the process that has then barfed, after the message has been queued, which suggests that it is nothing to do with FreeSWITCH itself. Perhaps there might be a copy of the message from the crashing process in the message queues in /var/spool/mail (or similar) which you could look at to see if there is anything weird in it? But the bottom line is that it looks like you have pure sendmail problem. Unfortunately, that's where my knowledge stops as I run Postfix as my main mailer and ssmtp as a dumb relay. Maybe someone on a CentOS or sendmail list can help you with the sendmail issue? Or, as I mentioned earlier, if you don't need a full mailer, you could remove sendmail and install a simpler mailer such as ssmtp (as I see you have a relay that needs TLS and for which authentication appears to be failing). John PS. The parallel test is to wrap the original command in parentheses and duplicate it several times on a single command line, each time followed by an ampersand to put it into the background, ie: rmorin at blie-fs$ ( cat .... | sendmail ...... ... ) & ( cat ..... | sendmail .... ) & .... and so on [It maybe that your sendmail queues faster than your shell can invoke sendmail, so it isn't really in parallel, but then that should be similar for FreeSWITCH too, so it would be a 'rapid-fire' test instead of a parallel test.] On 27/01/12 00:46, Rob Morin wrote: John, Thank you for your help. I ran the command line as you described, and everything worked fine. I should also mention, that some of my extensions do not have the notifications turned on. And even for them, getting the message through is hit or miss - sometimes it segfaults, sometimes it doesn't. So I don't necessarily think it's a "two message" issue. Also, in looking at the logs, sometimes I see the segfault appear in the middle of the maillog logging for the other message. So it doesn't look like FS is waiting for the sendmail command to return before executing the next command. Here's what I mean. Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: Authentication-Warning: blie-fs.blie-ent.com: freeswitch set sender to voicemail at blie-ent.com using -f Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: from=voicemail at blie-ent.com, size=18662, class=0, nrcpts=1, msgid= <201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, relay=freeswitch at localhost Jan 26 14:28:51 blie-fs sendmail[25201]: q0QESpVj025201: from= , size=19026, class=0, nrcpts=1, msgid= <201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, proto=ESMTP, daemon=MTA, relay=localhost [127.0.0.1] Jan 26 14:28:51 blie-fs sendmail[25200]: q0QESoi9025200: to=rmorin at blie-ent.com, ctladdr=voicemail at blie-ent.com (501/501), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=48662, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (q0QESpVj025201 Message accepted for delivery) Jan 26 14:28:51 blie-fs kernel: [41082.268638] sendmail[25207]: segfault at 7fff96dffed8 rip 7f158ee0b7fb rsp 7fff96dffea0 error 6 Jan 26 14:28:51 blie-fs sendmail[25203]: STARTTLS=client, relay=smtp.sendgrid.net., version=TLSv1/SSLv3, verify=FAIL, cipher=DHE-RSA-AES256-SHA, bits=256/256 Jan 26 14:28:51 blie-fs sendmail[25203]: q0QESpVj025201: to= , delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=139026, relay=smtp.sendgrid.net. [174.36.32.204], dsn=2.0.0, stat=Sent (Delivery in progress) On this instance, the notification didn't get through, but the message did. All of that said, I'm not sure how to run the 'parallel' test, though. (I did run the single test many times in quick succession with no failures.) My ulimit -s is set at 8192. And I don't have any other arguments, other than '-t'. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 6:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting Rob, ssmtp sends whatever it's given: FreeSWITCH puts the message together (in src/switch_utils.c:switch_simple_email()) to make a fully MIME-compliant message with the attachments read out of their files and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, sendemail.py or whatever) program for it to send/relay it. FreeSWITCH waits for the 'cat | sendmail' program to return before continuing execution and deleting files, so file deletion is unlikely to be the issue. Like you say, it sounds like the problem is not with configuration but it's also very unlikely to be with file deletion. Can you reproduce the fault by running sendmail from the command line with a suitably formatted mail message[*] as input? This is pretty much what FreeSWITCH does: cat message.txt | -f noreply at mydomain.com recipient at mydomain.com ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? [* 'suitably formatted message' could be one of the successful ones, view source, then edit to trim off the transport headers] If that is successful, try to replicate your 2-message-scenario by running two of the above commands in parallel with different input files and see if that generates a segfault. [If you are a linux noob as well as a FS noob I can spell this out for you.] John On 26/01/12 22:04, Rob Morin wrote: John, That's helpful, but ssmtp doesn't support attachments so it won't support sending the voicemail files. Additionally, I'm somewhat stumped on the default 'sendmail' option. The references in the Wiki are basically all 'configuration' issues - stack size, etc. The problem I have is that it isn't a configuration issue - sometimes it goes through, sometimes it doesn't. If it were a configuration issue, it would never go through (unless it was a resource that was near its margin). So that's why I wanted to insert a delay. It appears to me that the segfault is being caused by the file getting moved, deleted, or still being open when sendmail attempts to access it. I wish I had more to go on though. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 1:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Slightly unrelated, but I have been using the python script sendemail.py until recently when a user noticed that they were not being emailed some voicemails. I have since changed to using ssmtp (rather than nullmailer as my mail server requires a login, TLS on tcp/587). One thing I noticed, and it may be me that caused it, but the sendemail.py script had some indentation using (4) spaces and some using tabs. But the main problem is that the script keeps warning about the MimeWriter class being deprecated and has no error reporting or recovery, it seems. For me, it was easier to install ssmtp than rewrite the python script. If you don't need a full mail server, I would recommend using a null mailer like ssmtp or nullmailer. John On 26/01/12 15:13, Rob Morin wrote: I tried to modify the eximcompat.sh script to work with sendmail. I'm not sure why it didn't work. It might be that, for starters, I don't have exim installed, I have sendmail. I haven't tried the python script. Is that the one you use? Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, January 26, 2012 8:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/6634c7b0/attachment-0001.html From elliott at zoogmedia.com Fri Jan 27 08:49:30 2012 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 27 Jan 2012 05:49:30 +0000 Subject: [Freeswitch-users] Options ping In-Reply-To: <89a8f0cef8dc002798a684ad051ff204.squirrel@fulvetta.riseup.net> References: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com>, <89a8f0cef8dc002798a684ad051ff204.squirrel@fulvetta.riseup.net> Message-ID: <461821C20115054F9056ED0E52EAC4DAE4A7FE@CH1PRD0710MB380.namprd07.prod.outlook.com> Do you know the rate pings are sent? can this be changed? ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of georg at riseup.net [georg at riseup.net] Sent: Thursday, January 26, 2012 9:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Options ping Hi Elliot, You could use to exactly address your issue as described in [1]. Greetings, Georg [1] http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From vetali100 at gmail.com Fri Jan 27 10:08:49 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Thu, 26 Jan 2012 23:08:49 -0800 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: <4F201657.9040005@softnet.si> References: <4F201657.9040005@softnet.si> Message-ID: Could you please elaborate more on your topology... Sip phone A (1001) --> FS --> what is here..? Where do you want it to call? Do you have an external profile with a gateway configured? 2012/1/25 Miha Zoubek > Hi, > > I have phone registered on FS and phone is set to make call FW. I > noticed that FS do not make a call FW the external number but looks on > FS if the phone is registered on FS (because the calling number is not > on FS the call is being rejected). > How to tell FS that it should make a call fw to a number that is not on FS. > > Thanks! > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120126/ad397f02/attachment.html From paul at iamfine.com Fri Jan 27 10:11:42 2012 From: paul at iamfine.com (Paul) Date: Thu, 26 Jan 2012 23:11:42 -0800 (PST) Subject: [Freeswitch-users] mod_xml_cdr not saving file correctly Message-ID: <1327648302533-7229340.post@n2.nabble.com> Using latest build I have encountered a very odd behavior and want to know if it is a bug or not. If not - I would appreciate any ideas on how to work around it -- I have a lua script that processes my inbound calls, and uses mod_xml_cdr to do a http curl to a web api and also save a copy to the local /log/xml_cdr directory -- When the call comes in - sometimes the xml_cdr works as advertised (posts and saves). This is what the cdr looks like ..... CS_REPORTING inbound 11 0=1;1=1;35=1;36=1;38=1;41=1;51=1 1=1;2=1;3=1;4=1;5=1;6=1 and sometimes it saves as follows, which breaks the http curl post. %3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3E8bc9c803-d459-4228-bf9d-87419e57c73b%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E66%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri I have taken the above snippets directly from the xml_cdr files that are saved both calls come in on same gateway and are answered with the same script THE only difference is the caller ID in the first case it is 10 digits NPA-NXX in the second i convert it to 11 digits before executing the bulk of the script with a 1NPA-NXX both numbers are used as strings (confirmed with the lua type(var)) Any ideas welcome - Paul -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-correctly-tp7229340p7229340.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at earthspike.net Fri Jan 27 10:16:26 2012 From: freeswitch at earthspike.net (John) Date: Fri, 27 Jan 2012 07:16:26 +0000 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <19c501ccdca1$beb1a3e0$3c14eba0$@blie-ent.com> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> <4F21E058.9000004@earthspike.net> <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> <4F2200A4.3010300@earthspike.net> <19c501ccdca1$beb1a3e0$3c14eba0$@blie-ent.com> Message-ID: <4F224F4A.3050408@earthspike.net> You need It's the same as the nullmailer case documented in the wiki. Ie, you need to override the default "-t" setting. John On 27/01/12 03:14, Rob Morin wrote: > > Okay, this is not expected. > > > > I switched over to ssmtp, since from what I can tell, that attaching > of the voicemail file occurs within Freeswitch. > > > > It isn't sending anything, although, again, I can send it from the > command line. > > > > When I run freeswitch from the command line and leave a voicemail, it > gives the following errors: > > > > ssmtp: recipients with -t option not supported > > ssmtp: recipients with -t option not supported > > > > The only problem is, my switch.conf file looks like this, > > > > > > > > > > > > > > > > > > So, where is it getting the "-t " ??? > > > > Just to be safe, I removed the line _and_ rebooted. But it is still there? > > Thank you, > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 8:41 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Rob, > > sendmail is running as a service, so the segfault could be being > thrown by the sendmail service, not necessarily by the > command-line-app-version that adds mails to the queue (and then > terminates). Most mailers work this way: separate processes to add > the mail to the queue, process the queue, accept connections on > various sockets. FreeSWITCH is written to wait until the > queue-insertion invocation of sendmail is completed. Thankfully, it > doesn't wait until the message is actually sent on to its next > location as, in some cases, that could take hours if the remote host > is down. In your log below, it looks like sendmail [25200] has > happily accepted the message from FreeSWITCH and queued it for > delivery. sendmail [25207] is the process that has then barfed, after > the message has been queued, which suggests that it is nothing to do > with FreeSWITCH itself. Perhaps there might be a copy of the message > from the crashing process in the message queues in /var/spool/mail (or > similar) which you could look at to see if there is anything weird in > it? But the bottom line is that it looks like you have pure sendmail > problem. Unfortunately, that's where my knowledge stops as I run > Postfix as my main mailer and ssmtp as a dumb relay. Maybe someone on > a CentOS or sendmail list can help you with the sendmail issue? Or, > as I mentioned earlier, if you don't need a full mailer, you could > remove sendmail and install a simpler mailer such as ssmtp (as I see > you have a relay that needs TLS and for which authentication appears > to be failing). > > John > > PS. The parallel test is to wrap the original command in parentheses > and duplicate it several times on a single command line, each time > followed by an ampersand to put it into the background, ie: > > rmorin at blie-fs$ ( cat .... | sendmail ...... ... ) & ( cat ..... | > sendmail .... ) & .... and so on > > [It maybe that your sendmail queues faster than your shell can invoke > sendmail, so it isn't really in parallel, but then that should be > similar for FreeSWITCH too, so it would be a 'rapid-fire' test instead > of a parallel test.] > > On 27/01/12 00:46, Rob Morin wrote: > > John, > > > > Thank you for your help. > > > > I ran the command line as you described, and everything worked fine. > > > > I should also mention, that some of my extensions do not have the > notifications turned on. And even for them, getting the message > through is hit or miss -- sometimes it segfaults, sometimes it > doesn't. So I don't necessarily think it's a "two message" issue. > Also, in looking at the logs, sometimes I see the segfault appear in > the middle of the maillog logging for the other message. So it doesn't > look like FS is waiting for the sendmail command to return before > executing the next command. Here's what I mean... > > > > Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: > Authentication-Warning: blie-fs.blie-ent.com: freeswitch set sender to > voicemail at blie-ent.com using -f > > Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: > from=voicemail at blie-ent.com , > size=18662, class=0, nrcpts=1, > msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com> > , > relay=freeswitch at localhost > > Jan 26 14:28:51 blie-fs sendmail[25201]: q0QESpVj025201: > from= , > size=19026, class=0, nrcpts=1, > msgid=<201201261428.q0QESoi9025200 at blie-fs.blie-ent.com> > , > proto=ESMTP, daemon=MTA, relay=localhost [127.0.0.1] > > Jan 26 14:28:51 blie-fs sendmail[25200]: q0QESoi9025200: > to=rmorin at blie-ent.com , > ctladdr=voicemail at blie-ent.com > (501/501), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=48662, > relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (q0QESpVj025201 > Message accepted for delivery) > > Jan 26 14:28:51 blie-fs kernel: [41082.268638] sendmail[25207]: > segfault at 7fff96dffed8 rip 7f158ee0b7fb rsp 7fff96dffea0 error 6 > > Jan 26 14:28:51 blie-fs sendmail[25203]: STARTTLS=client, > relay=smtp.sendgrid.net., version=TLSv1/SSLv3, verify=FAIL, > cipher=DHE-RSA-AES256-SHA, bits=256/256 > > Jan 26 14:28:51 blie-fs sendmail[25203]: q0QESpVj025201: > to= , delay=00:00:00, > xdelay=00:00:00, mailer=relay, pri=139026, relay=smtp.sendgrid.net. > [174.36.32.204], dsn=2.0.0, stat=Sent (Delivery in progress) > > > > On this instance, the notification didn't get through, but the message > did. > > > > All of that said, I'm not sure how to run the 'parallel' test, though. > (I did run the single test many times in quick succession with no > failures.) > > > > My ulimit --s is set at 8192. And I don't have any other arguments, > other than '-t'. > > > > Thank you, > > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 6:23 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Rob, > > ssmtp sends whatever it's given: FreeSWITCH puts the message together > (in src/switch_utils.c:switch_simple_email()) to make a fully > MIME-compliant message with the attachments read out of their files > and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, > sendemail.py or whatever) program for it to send/relay it. FreeSWITCH > waits for the 'cat | sendmail' program to return before continuing > execution and deleting files, so file deletion is unlikely to be the > issue. Like you say, it sounds like the problem is not with > configuration but it's also very unlikely to be with file deletion. > > Can you reproduce the fault by running sendmail from the command line > with a suitably formatted mail message[*] as input? This is pretty > much what FreeSWITCH does: > > cat message.txt | -f noreply at mydomain.com > recipient at mydomain.com > > > ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? > > [* 'suitably formatted message' could be one of the successful ones, > view source, then edit to trim off the transport headers] > > If that is successful, try to replicate your 2-message-scenario by > running two of the above commands in parallel with different input > files and see if that generates a segfault. [If you are a linux noob > as well as a FS noob I can spell this out for you.] > > John > > On 26/01/12 22:04, Rob Morin wrote: > > John, > > > > That's helpful, but ssmtp doesn't support attachments so it won't > support sending the voicemail files. > > > > Additionally, I'm somewhat stumped on the default 'sendmail' option. > The references in the Wiki are basically all 'configuration' issues -- > stack size, etc. The problem I have is that it isn't a configuration > issue -- sometimes it goes through, sometimes it doesn't. If it were a > configuration issue, it would never go through (unless it was a > resource that was near its margin). So that's why I wanted to insert > a delay. It appears to me that the segfault is being caused by the > file getting moved, deleted, or still being open when sendmail > attempts to access it. I wish I had more to go on though. > > > > Thank you, > > Rob > > > > *From:*John [mailto:freeswitch at earthspike.net] > *Sent:* Thursday, January 26, 2012 1:49 PM > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > Slightly unrelated, but I have been using the python script > sendemail.py until recently when a user noticed that they were not > being emailed some voicemails. I have since changed to using ssmtp > (rather than nullmailer as my mail server requires a login, TLS on > tcp/587). One thing I noticed, and it may be me that caused it, but > the sendemail.py script had some indentation using (4) spaces and some > using tabs. But the main problem is that the script keeps warning > about the MimeWriter class being deprecated and has no error reporting > or recovery, it seems. For me, it was easier to install ssmtp than > rewrite the python script. If you don't need a full mail server, I > would recommend using a null mailer like ssmtp or nullmailer. > > John > > On 26/01/12 15:13, Rob Morin wrote: > > I tried to modify the eximcompat.sh script to work with sendmail. I'm > not sure why it didn't work. It might be that, for starters, I don't > have exim installed, I have sendmail. > > > > I haven't tried the python script. Is that the one you use? > > > > Rob > > > > *From:*Michael Jerris [mailto:mike at jerris.com] > *Sent:* Thursday, January 26, 2012 8:09 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting > > > > delay won't help you. Why didn't the script work? I know we do this > and it works fine. Pretty sure we use a modified version of the > sample in tree. > > > > Mike > > > > On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: > > > > > > > I saw that, and tried everything I could. Still no luck. > > > > My stack limit is 8192, so that shouldn't be the issue. I'm running > CentOS x_64. > > > > I tried creating a script as the wiki suggests, so that I could make > changes without having to restart FS. But that didn't work either (I'm > still a noob). My goal was to add a sleep delay, in the case that FS > is calling the sendmail command before it's completely written and > released the email it's sending. Any idea how to make that work? > > Thank you, > > Rob > > > > *From:* Michael Jerris [mailto:mike at jerris.com] > > *Sent:* Wednesday, January 25, 2012 1:57 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Sendmail segfaulting before second > message is sent > > > > http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings > > > > On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: > > > > > > > > For some extensions, I've got FreeSWITCH configured to send both the > voicemail to an email address and a notification to a cell phone that > there is a message. > > > > Sendmail is segfaulting (error 6) before the second email message is > sent. Sometimes the notification message is going out, sometimes the > email message goes. > > > > If I am just sending an email, it is **generally** successful. > Although sometimes it fails, so I really need to fix this. > > > > Does anyone have any suggestions as to how I can troubleshoot this? > > (Running CentOS 5.7, sendmail 8.13). > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/b6fe06f0/attachment-0001.html From rhow at exemail.com.au Fri Jan 27 10:17:27 2012 From: rhow at exemail.com.au (Ryan How) Date: Fri, 27 Jan 2012 15:17:27 +0800 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <7d4c72305700590f9b3a813729aec63b.squirrel@fulvetta.riseup.net> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> <11171047-CD59-4868-BAD7-844F068FD3C2@jerris.com> <7d4c72305700590f9b3a813729aec63b.squirrel@fulvetta.riseup.net> Message-ID: <4F224F87.1000101@exemail.com.au> Hi, I had an idea. I read freeswitch can pause if doing a stun lookup each call. Maybe this is happening and that is why you get your variable sometimes ring straight away, sometimes not ? Or maybe it is completely unrelated, just thinking :). Also if you answer straight away does it cut off the first part of the call for the caller or does it actually connect even though the ringing hasn't come through yet? Ryan On 27/01/2012 12:52 AM, georg at riseup.net wrote: >> seems you have truncated anything useful in this paste. > You mean the "XXX"? Or is the paste "to short" or the loglevel not "deep" > enough? > > Thanks, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Fri Jan 27 10:49:46 2012 From: miha at softnet.si (Miha Zoubek) Date: Fri, 27 Jan 2012 08:49:46 +0100 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: References: <4F201657.9040005@softnet.si> Message-ID: <4F22571A.9010009@softnet.si> Hi @Vitalie, what is here..? here is trunk to sbc with registration false option (without password, just ip). Outside call I am doing like this : Thanks! On 1/27/2012 8:08 AM, Vitalie Colosov wrote: > Could you please elaborate more on your topology... > > Sip phone A (1001) --> FS --> what is here..? Where do you want it to > call? Do you have an external profile with a gateway configured? > > > 2012/1/25 Miha Zoubek > > > Hi, > > I have phone registered on FS and phone is set to make call FW. I > noticed that FS do not make a call FW the external number but looks on > FS if the phone is registered on FS (because the calling number is not > on FS the call is being rejected). > How to tell FS that it should make a call fw to a number that is > not on FS. > > Thanks! > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/44bd62cb/attachment.html From vetali100 at gmail.com Fri Jan 27 11:24:28 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Fri, 27 Jan 2012 00:24:28 -0800 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: <4F22571A.9010009@softnet.si> References: <4F201657.9040005@softnet.si> <4F22571A.9010009@softnet.si> Message-ID: And when you say "phone is set to make call FW" - are you calling to this phone A from another phone which is connected to FS? (lets say phone B), and Phone A is doing SIP redirect to the external number which is set by you in the Phone A's settings? If so, I assume the call will enter FS, and then it depends of your dialplan... You can setup it in the way that if the number does not exist, or based on some prefix, it should bridge to external number via sofia/external/386$1 at SBC_IP, but this should be done explicitly, FS does not know that you want to forward it to the outside world... For example: Let me know if this bring more light to your issue. Vitalie 2012/1/26 Miha Zoubek > Hi @Vitalie, > > what is here..? here is trunk to sbc with registration false option > (without password, just ip). Outside call I am doing like this : > > > > Thanks! > > > > On 1/27/2012 8:08 AM, Vitalie Colosov wrote: > > Could you please elaborate more on your topology... > > Sip phone A (1001) --> FS --> what is here..? Where do you want it to > call? Do you have an external profile with a gateway configured? > > > 2012/1/25 Miha Zoubek > >> Hi, >> >> I have phone registered on FS and phone is set to make call FW. I >> noticed that FS do not make a call FW the external number but looks on >> FS if the phone is registered on FS (because the calling number is not >> on FS the call is being rejected). >> How to tell FS that it should make a call fw to a number that is not on >> FS. >> >> Thanks! >> >> Regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/17ae1e03/attachment.html From georg at riseup.net Fri Jan 27 12:16:31 2012 From: georg at riseup.net (georg at riseup.net) Date: Fri, 27 Jan 2012 10:16:31 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: <4F224F87.1000101@exemail.com.au> References: <5f4357e714e3188f4a631c0eaa0a34ea.squirrel@fulvetta.riseup.net> <4AEE0C8F-87D2-4B9A-96DA-82805CF11781@jerris.com> <94b5d84313e52057a974867bf0e42aa9.squirrel@fulvetta.riseup.net> <11171047-CD59-4868-BAD7-844F068FD3C2@jerris.com> <7d4c72305700590f9b3a813729aec63b.squirrel@fulvetta.riseup.net> <4F224F87.1000101@exemail.com.au> Message-ID: <76be04b9100c67c6f1d58b76072d8382.squirrel@fulvetta.riseup.net> Hi, > I had an idea. I read freeswitch can pause if doing a stun lookup each > call. Maybe this is happening and that is why you get your variable > sometimes ring straight away, sometimes not ? Or maybe it is completely > unrelated, just thinking :). I'm not using STUN as far as I know... > Also if you answer straight away does it > cut off the first part of the call for the caller or does it actually > connect even though the ringing hasn't come through yet? Have to check this. Thanks, Georg From georg at riseup.net Fri Jan 27 12:19:08 2012 From: georg at riseup.net (georg at riseup.net) Date: Fri, 27 Jan 2012 10:19:08 +0100 Subject: [Freeswitch-users] Options ping In-Reply-To: <461821C20115054F9056ED0E52EAC4DAE4A7FE@CH1PRD0710MB380.namprd07.prod.outlook.com> References: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com>, <89a8f0cef8dc002798a684ad051ff204.squirrel@fulvetta.riseup.net> <461821C20115054F9056ED0E52EAC4DAE4A7FE@CH1PRD0710MB380.namprd07.prod.outlook.com> Message-ID: Hello, > Do you know the rate pings are sent? can this be changed? By rate you mean interval? No, sorry...but there are for sure people on this list who will know?! Greetings, Georg From all.eforums at gmail.com Fri Jan 27 12:49:27 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Fri, 27 Jan 2012 04:49:27 -0500 Subject: [Freeswitch-users] Skypopen on debian squeeze - All sorts of fonts and alsa/oss driver and binary issues In-Reply-To: References: Message-ID: On Thu, Jan 26, 2012 at 9:15 AM, A E [Gmail] wrote: > Hello All, > > anyone tried to get Skypopen to work in Debian? I am not really sure what > the real issue is as I get multiple errors when trying to start but the > biggest one I guess is the one claiming that the skype binary cannot be > executed. > > Will provide further info as requested as I really don't know what the > issue really is. Does the Skype client need a real sound card/driver > present? I doubt it. Does it really need a real screen attached to it? I > doubt it since I see some magic being done with the Xserver to create fake > displays and what not. > > Anyway, here's what I get when I try to start the start_skype_clients.sh > > *ERROR: Module snd_pcm_oss does not exist in /proc/modules* > *ERROR: Module snd_mixer_oss does not exist in /proc/modules* > *ERROR: Module snd_seq_oss does not exist in /proc/modules* > *mknod: `/dev/dsp': File exists* > *insmod: error inserting > '/opt/freeswitch/skypopen/sound-drv/skypopen.ko': -1 File exists* > *SELinux: Disabled on system, not enabling in X server* > *[dix] Could not init font path element > /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list!* > *bash: /opt/freeswitch/skypopen/clientsymlnk/skype101: cannot execute > binary file* > > Any ideas? Giovanni? :) > > Thanks in advance > Ok this "mystery" is solved as I only just realised that the package being downloaded is pre-compiled binary and not source that was further being compiled. The skypopen library was but not the actual Skype binary. So there you go! Now, is this a stupid question or can I get a hold of the actual source of the Skype client so I can try and compile this on my platform? Thanks so much -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/6f7148b0/attachment.html From georg at riseup.net Fri Jan 27 13:19:47 2012 From: georg at riseup.net (georg at riseup.net) Date: Fri, 27 Jan 2012 11:19:47 +0100 Subject: [Freeswitch-users] Line echo cancellation for voice Message-ID: <7ff2441a6e3b28173cf7c8a60cc6c8d3.squirrel@fulvetta.riseup.net> Hello all, Could someone tell me, how to enable this: http://docs.freeswitch.org/echo_can_page.html ? Thanks, Georg From Stefan.Weigel at allianz-warranty.com Fri Jan 27 13:32:14 2012 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Fri, 27 Jan 2012 11:32:14 +0100 Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing In-Reply-To: <02fe01ccdc4b$42985840$c7c908c0$@com> References: <5003D7D3E06F514E8C682F18D223265C05121CD349@AZWSMS03.azwarranty.int> <02fe01ccdc4b$42985840$c7c908c0$@com> Message-ID: <5003D7D3E06F514E8C682F18D223265C05121CD359@AZWSMS03.azwarranty.int> Hi Bote, I already know about this. I would like to know if it's possible to use moh and ringtones in combination if an agent phone is ringing. Best regards Stefan Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Bote Man Gesendet: Donnerstag, 26. Januar 2012 17:55 An: 'FreeSWITCH Users Help' Betreff: Re: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing If you do not need the music on hold, create the desired ringback tone as a sound file and install that in place of the moh sound. Because the moh sound repeats forever you only need to create a file long enough for the ringback tone and silence period. I do not know a method to change from moh to generated ringback because I am new to FreeSWITCH and still learning. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Weigel, Stefan Sent: Thursday, 26 January, 2012 06:03 To: 'freeswitch-users at lists.freeswitch.org' Subject: [Freeswitch-users] mod_callcenter - ring tones when phone is ringing Hi all, we're using successfully mod_callcenter. If a caller enters the queue, he/she is getting default moh sound. The sound is also playing when the call is passed to an agent and the phone is ringing. Is it possible to play a normal ringtone when a call is passed to an agent ? Thanks and best regards Stefan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/cf226d46/attachment.html From dgarcia at anew.com.ve Fri Jan 27 15:42:08 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 27 Jan 2012 08:12:08 -0430 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> Message-ID: <4F229BA0.9030400@anew.com.ve> Hi Brian, Take a look to this links: http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html On 1/26/2012 5:45 PM, Brian wrote: > > Hi List, > > I'm quite new to FreeSwitch and I'm trying to replace an old asterisk > installation with a new FreeSwitch. There is however one feature that > is available with Asterisk and I haven't managed to figure it out how > to do it with FreeSwitch. That is about the "s" extension in Asterisk > dial plan. In the old Asterisk installation, when I pick up an > analogic phone (plugged into an FXS slot of a Digium card) the "s" > extension gets run and I can play early media to the phone. Does > anyone know if it is possible to achieve the same thing with > FreeSwitch/FreeTDM? > > Best regards, > > Brian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/2fded37c/attachment-0001.html From david.villasmil.work at gmail.com Fri Jan 27 17:07:54 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 27 Jan 2012 15:07:54 +0100 Subject: [Freeswitch-users] FreeSWITCH Viking Message-ID: Hello Guys, I updated the wiki with the following: CLI-based/Retail I have now made a retail version for FreeSWITCH-Billing. It works like this: - A specific context is created for this application. - Any calls which comes in on said context is handled by the retail script. - The script Looks up the incomming caller ID Number in a DB table. - The CLI is associated with a "Master" acount via the Master account's ID. - This Master account has a balance, a rate table and a username/password (for later use, this user will be able to check his traffic, balance, etc via web) - If the account has a positive balance, the rate table is used to apply a price per minute. - The Max talk time is calculated and set by the script and the call is sent out using the route tables (already existing in the wholesale side). - Once the call is finished, the CDR is posted using mod_xml_cdr on a web server and the receiving php inserts the cdr into the cdr table, also calculating and setting the cost of the call. TO DO: - web pages for the client to check out his balance, clis, etc. I will upload this into the github this weekend! Have fun! note: any suggestion is VERY welcome! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/4df3a1de/attachment.html From elliott at zoogmedia.com Fri Jan 27 17:11:26 2012 From: elliott at zoogmedia.com (Elliott Vogel) Date: Fri, 27 Jan 2012 14:11:26 +0000 Subject: [Freeswitch-users] Options ping In-Reply-To: References: <461821C20115054F9056ED0E52EAC4DAE493FF@BY2PRD0710MB378.namprd07.prod.outlook.com>, <89a8f0cef8dc002798a684ad051ff204.squirrel@fulvetta.riseup.net> <461821C20115054F9056ED0E52EAC4DAE4A7FE@CH1PRD0710MB380.namprd07.prod.outlook.com>, Message-ID: <461821C20115054F9056ED0E52EAC4DAE4BABD@BY2PRD0710MB378.namprd07.prod.outlook.com> Yes, interval... The ping diesnt seem to be keeping the firewall open. the only way I can keep the phones working if i set the register to 120. ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of georg at riseup.net [georg at riseup.net] Sent: Friday, January 27, 2012 3:19 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Options ping Hello, > Do you know the rate pings are sent? can this be changed? By rate you mean interval? No, sorry...but there are for sure people on this list who will know?! Greetings, Georg _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at neotiq.com Fri Jan 27 17:38:35 2012 From: brian at neotiq.com (Brian) Date: Fri, 27 Jan 2012 15:38:35 +0100 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <4F229BA0.9030400@anew.com.ve> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> Message-ID: <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> Hi Saugort, Thank you for these links. However that is not really what I?m looking for. What I am trying to do is to play a sound file to an analogic phone as soon as the later goes off-hook. What I called ?eary media? is not really the ringtone as we often see in SIP protocol. Sorry for the confusion. Any help would be greatly appreciated, Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 13:42 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Take a look to this links: http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as- early-media-td2460205.html http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581 .html On 1/26/2012 5:45 PM, Brian wrote: Hi List, I?m quite new to FreeSwitch and I?m trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven?t managed to figure it out how to do it with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the ?s? extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/505f4516/attachment.html From avi at avimarcus.net Fri Jan 27 17:41:50 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 27 Jan 2012 16:41:50 +0200 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> Message-ID: I think you want a hot-dial..? Some phones/ATAs support it. I don't think it's an FS/Asterisk feature. e.g. most phones don't even contact the sip server until the person "finished dialing". -Avi On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote: > Hi Saugort,**** > > ** ** > > Thank you for these links. However that is not really what I?m looking > for. What I am trying to do is to play a sound file to an analogic phone as > soon as the later goes off-hook. What I called ?eary media? is not really > the ringtone as we often see in SIP protocol. Sorry for the confusion.**** > > ** ** > > Any help would be greatly appreciated,**** > > Regards,**** > > Brian**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort > Dario Garcia Tovar > *Envoy? :* vendredi 27 janvier 2012 13:42 > *? :* freeswitch-users at lists.freeswitch.org > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk** > ** > > ** ** > > Hi Brian, > > Take a look to this links: > > http://wiki.freeswitch.org/wiki/Early_Media > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > > > http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html > > > http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html > > > > On 1/26/2012 5:45 PM, Brian wrote: **** > > Hi List,**** > > **** > > I?m quite new to FreeSwitch and I?m trying to replace an old asterisk > installation with a new FreeSwitch. There is however one feature that is > available with Asterisk and I haven?t managed to figure it out how to do it > with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In > the old Asterisk installation, when I pick up an analogic phone (plugged > into an FXS slot of a Digium card) the ?s? extension gets run and I can > play early media to the phone. Does anyone know if it is possible to > achieve the same thing with FreeSwitch/FreeTDM? **** > > **** > > Best regards,**** > > **** > > Brian**** > > **** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > ** ** > > FreeSWITCH-users mailing list**** > > FreeSWITCH-users at lists.freeswitch.org**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > > http://www.freeswitch.org**** > > > > > **** > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12* > *** > > ** ** > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/3a7b087a/attachment-0001.html From brian at neotiq.com Fri Jan 27 18:19:42 2012 From: brian at neotiq.com (Brian) Date: Fri, 27 Jan 2012 16:19:42 +0100 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> Message-ID: <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> Hi Avi, Maybe that?s the term I should use, hot-dial. I?m talking about analogic phones, not VoIP phones. If it?s an analogic phone plugged into an FXS socket, when you pick up the phone, the PBX is capable of detecting the off-hook event. That feature must be supported by the analogic card?s driver. In asterisk?s case, that is the option named ?immediate=true|false? in the Zapata.conf file, as stated here: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Best regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Avi Marcus Envoy? : vendredi 27 janvier 2012 15:42 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk I think you want a hot-dial..? Some phones/ATAs support it. I don't think it's an FS/Asterisk feature. e.g. most phones don't even contact the sip server until the person "finished dialing". -Avi On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote: Hi Saugort, Thank you for these links. However that is not really what I?m looking for. What I am trying to do is to play a sound file to an analogic phone as soon as the later goes off-hook. What I called ?eary media? is not really the ringtone as we often see in SIP protocol. Sorry for the confusion. Any help would be greatly appreciated, Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 13:42 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Take a look to this links: http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as- early-media-td2460205.html http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581 .html On 1/26/2012 5:45 PM, Brian wrote: Hi List, I?m quite new to FreeSwitch and I?m trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven?t managed to figure it out how to do it with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the ?s? extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/3a6e869e/attachment.html From kris at kriskinc.com Fri Jan 27 18:21:47 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Fri, 27 Jan 2012 10:21:47 -0500 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> Message-ID: IIRC Asterisk uses the "s" extension when a call enters a context without any specific destination information (i.e. a phone on an FXS port going off-hook). I'm not sure about the OpenZAP/Sangoma/etc stuff in FreeSWITCH but the dialplan equivalent of Asterisk "s" would be an extension with no condition: On Thu, Jan 26, 2012 at 5:15 PM, Brian wrote: > Hi List, > > > > I?m quite new to FreeSwitch and I?m trying to replace an old asterisk > installation with a new FreeSwitch. There is however one feature that is > available with Asterisk and I haven?t managed to figure it out how to do it > with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In > the old Asterisk installation, when I pick up an analogic phone (plugged > into an FXS slot of a Digium card) ?the ?s? extension gets run and I can > play early media to the phone. Does anyone know if it is possible to achieve > the same thing with FreeSwitch/FreeTDM? > > > > Best regards, > > > > Brian > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From rmorin at blie-ent.com Fri Jan 27 18:41:37 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Fri, 27 Jan 2012 10:41:37 -0500 Subject: [Freeswitch-users] Sendmail segfaulting In-Reply-To: <4F224F4A.3050408@earthspike.net> References: <16d701ccdbdc$b9e1e8e0$2da5baa0$@blie-ent.com> <678F588B-3C2C-49E6-9C62-C5E2697B8398@jerris.com> <178001ccdc3d$1c4bfec0$54e3fc40$@blie-ent.com> <4F21A009.8080400@earthspike.net> <193001ccdc76$830b28a0$892179e0$@blie-ent.com> <4F21E058.9000004@earthspike.net> <197b01ccdc8d$2c2bf700$8483e500$@blie-ent.com> <4F2200A4.3010300@earthspike.net> <19c501ccdca1$beb1a3e0$3c14eba0$@blie-ent.com> <4F224F4A.3050408@earthspike.net> Message-ID: <1a7d01ccdd0a$283c8c90$78b5a5b0$@blie-ent.com> John, Thank you. That did it (and was the reason why it wouldn't work when I attempted it before.) Rob From: John [mailto:freeswitch at earthspike.net] Sent: Friday, January 27, 2012 2:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting You need It's the same as the nullmailer case documented in the wiki. Ie, you need to override the default "-t" setting. John On 27/01/12 03:14, Rob Morin wrote: Okay, this is not expected. I switched over to ssmtp, since from what I can tell, that attaching of the voicemail file occurs within Freeswitch. It isn't sending anything, although, again, I can send it from the command line. When I run freeswitch from the command line and leave a voicemail, it gives the following errors: ssmtp: recipients with -t option not supported ssmtp: recipients with -t option not supported The only problem is, my switch.conf file looks like this, So, where is it getting the "-t " ??? Just to be safe, I removed the line and rebooted. But it is still there? Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 8:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Rob, sendmail is running as a service, so the segfault could be being thrown by the sendmail service, not necessarily by the command-line-app-version that adds mails to the queue (and then terminates). Most mailers work this way: separate processes to add the mail to the queue, process the queue, accept connections on various sockets. FreeSWITCH is written to wait until the queue-insertion invocation of sendmail is completed. Thankfully, it doesn't wait until the message is actually sent on to its next location as, in some cases, that could take hours if the remote host is down. In your log below, it looks like sendmail [25200] has happily accepted the message from FreeSWITCH and queued it for delivery. sendmail [25207] is the process that has then barfed, after the message has been queued, which suggests that it is nothing to do with FreeSWITCH itself. Perhaps there might be a copy of the message from the crashing process in the message queues in /var/spool/mail (or similar) which you could look at to see if there is anything weird in it? But the bottom line is that it looks like you have pure sendmail problem. Unfortunately, that's where my knowledge stops as I run Postfix as my main mailer and ssmtp as a dumb relay. Maybe someone on a CentOS or sendmail list can help you with the sendmail issue? Or, as I mentioned earlier, if you don't need a full mailer, you could remove sendmail and install a simpler mailer such as ssmtp (as I see you have a relay that needs TLS and for which authentication appears to be failing). John PS. The parallel test is to wrap the original command in parentheses and duplicate it several times on a single command line, each time followed by an ampersand to put it into the background, ie: rmorin at blie-fs$ ( cat .... | sendmail ...... ... ) & ( cat ..... | sendmail .... ) & .... and so on [It maybe that your sendmail queues faster than your shell can invoke sendmail, so it isn't really in parallel, but then that should be similar for FreeSWITCH too, so it would be a 'rapid-fire' test instead of a parallel test.] On 27/01/12 00:46, Rob Morin wrote: John, Thank you for your help. I ran the command line as you described, and everything worked fine. I should also mention, that some of my extensions do not have the notifications turned on. And even for them, getting the message through is hit or miss - sometimes it segfaults, sometimes it doesn't. So I don't necessarily think it's a "two message" issue. Also, in looking at the logs, sometimes I see the segfault appear in the middle of the maillog logging for the other message. So it doesn't look like FS is waiting for the sendmail command to return before executing the next command. Here's what I mean. Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: Authentication-Warning: blie-fs.blie-ent.com: freeswitch set sender to voicemail at blie-ent.com using -f Jan 26 14:28:50 blie-fs sendmail[25200]: q0QESoi9025200: from=voicemail at blie-ent.com, size=18662, class=0, nrcpts=1, msgid= <201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, relay=freeswitch at localhost Jan 26 14:28:51 blie-fs sendmail[25201]: q0QESpVj025201: from= , size=19026, class=0, nrcpts=1, msgid= <201201261428.q0QESoi9025200 at blie-fs.blie-ent.com>, proto=ESMTP, daemon=MTA, relay=localhost [127.0.0.1] Jan 26 14:28:51 blie-fs sendmail[25200]: q0QESoi9025200: to=rmorin at blie-ent.com, ctladdr=voicemail at blie-ent.com (501/501), delay=00:00:01, xdelay=00:00:00, mailer=relay, pri=48662, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (q0QESpVj025201 Message accepted for delivery) Jan 26 14:28:51 blie-fs kernel: [41082.268638] sendmail[25207]: segfault at 7fff96dffed8 rip 7f158ee0b7fb rsp 7fff96dffea0 error 6 Jan 26 14:28:51 blie-fs sendmail[25203]: STARTTLS=client, relay=smtp.sendgrid.net., version=TLSv1/SSLv3, verify=FAIL, cipher=DHE-RSA-AES256-SHA, bits=256/256 Jan 26 14:28:51 blie-fs sendmail[25203]: q0QESpVj025201: to= , delay=00:00:00, xdelay=00:00:00, mailer=relay, pri=139026, relay=smtp.sendgrid.net. [174.36.32.204], dsn=2.0.0, stat=Sent (Delivery in progress) On this instance, the notification didn't get through, but the message did. All of that said, I'm not sure how to run the 'parallel' test, though. (I did run the single test many times in quick succession with no failures.) My ulimit -s is set at 8192. And I don't have any other arguments, other than '-t'. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 6:23 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting Rob, ssmtp sends whatever it's given: FreeSWITCH puts the message together (in src/switch_utils.c:switch_simple_email()) to make a fully MIME-compliant message with the attachments read out of their files and then pipes it into the 'sendmail' (whether ssmtp, nullmailer, sendemail.py or whatever) program for it to send/relay it. FreeSWITCH waits for the 'cat | sendmail' program to return before continuing execution and deleting files, so file deletion is unlikely to be the issue. Like you say, it sounds like the problem is not with configuration but it's also very unlikely to be with file deletion. Can you reproduce the fault by running sendmail from the command line with a suitably formatted mail message[*] as input? This is pretty much what FreeSWITCH does: cat message.txt | -f noreply at mydomain.com recipient at mydomain.com ... with <..> from what you have in conf/autoload_configs/switch.conf.xml? [* 'suitably formatted message' could be one of the successful ones, view source, then edit to trim off the transport headers] If that is successful, try to replicate your 2-message-scenario by running two of the above commands in parallel with different input files and see if that generates a segfault. [If you are a linux noob as well as a FS noob I can spell this out for you.] John On 26/01/12 22:04, Rob Morin wrote: John, That's helpful, but ssmtp doesn't support attachments so it won't support sending the voicemail files. Additionally, I'm somewhat stumped on the default 'sendmail' option. The references in the Wiki are basically all 'configuration' issues - stack size, etc. The problem I have is that it isn't a configuration issue - sometimes it goes through, sometimes it doesn't. If it were a configuration issue, it would never go through (unless it was a resource that was near its margin). So that's why I wanted to insert a delay. It appears to me that the segfault is being caused by the file getting moved, deleted, or still being open when sendmail attempts to access it. I wish I had more to go on though. Thank you, Rob From: John [mailto:freeswitch at earthspike.net] Sent: Thursday, January 26, 2012 1:49 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Sendmail segfaulting Slightly unrelated, but I have been using the python script sendemail.py until recently when a user noticed that they were not being emailed some voicemails. I have since changed to using ssmtp (rather than nullmailer as my mail server requires a login, TLS on tcp/587). One thing I noticed, and it may be me that caused it, but the sendemail.py script had some indentation using (4) spaces and some using tabs. But the main problem is that the script keeps warning about the MimeWriter class being deprecated and has no error reporting or recovery, it seems. For me, it was easier to install ssmtp than rewrite the python script. If you don't need a full mail server, I would recommend using a null mailer like ssmtp or nullmailer. John On 26/01/12 15:13, Rob Morin wrote: I tried to modify the eximcompat.sh script to work with sendmail. I'm not sure why it didn't work. It might be that, for starters, I don't have exim installed, I have sendmail. I haven't tried the python script. Is that the one you use? Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Thursday, January 26, 2012 8:09 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting delay won't help you. Why didn't the script work? I know we do this and it works fine. Pretty sure we use a modified version of the sample in tree. Mike On Jan 25, 2012, at 10:43 PM, Rob Morin wrote: I saw that, and tried everything I could. Still no luck. My stack limit is 8192, so that shouldn't be the issue. I'm running CentOS x_64. I tried creating a script as the wiki suggests, so that I could make changes without having to restart FS. But that didn't work either (I'm still a noob). My goal was to add a sleep delay, in the case that FS is calling the sendmail command before it's completely written and released the email it's sending. Any idea how to make that work? Thank you, Rob From: Michael Jerris [mailto:mike at jerris.com] Sent: Wednesday, January 25, 2012 1:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sendmail segfaulting before second message is sent http://wiki.freeswitch.org/wiki/Mod_voicemail#Exim4_settings On Jan 25, 2012, at 1:31 PM, Rob Morin wrote: For some extensions, I've got FreeSWITCH configured to send both the voicemail to an email address and a notification to a cell phone that there is a message. Sendmail is segfaulting (error 6) before the second email message is sent. Sometimes the notification message is going out, sometimes the email message goes. If I am just sending an email, it is *generally* successful. Although sometimes it fails, so I really need to fix this. Does anyone have any suggestions as to how I can troubleshoot this? (Running CentOS 5.7, sendmail 8.13). _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/b02b6a94/attachment-0001.html From douglas at hubler.us Fri Jan 27 19:07:28 2012 From: douglas at hubler.us (Douglas Hubler) Date: Fri, 27 Jan 2012 11:07:28 -0500 Subject: [Freeswitch-users] sipXecs Conference March 5-6th Message-ID: I wanted to invite everyone interested in learning about the upcoming release of sipXecs v4.6 to come to SIPfoundry's Co-Lab conference on March 5-6th in Fort Collins, CO USA. For those that do not know about sipXecs, it's an open source project that combines SIP based services together with FreeSWITCH, OpenFire and OpenACD into a complete unified communications system. New to the upcoming release 4.6, is sipXecs's tremendous platform capabilities to build and launch your own communication applications. This platform allows you to create services that are secure, redundant, load balancing and easily managed in any cluster. Customize the cluster to run whatever services you want. sipXecs takes a different approach to accomplishing this by letting you chose what technology stack and language to build your apps on. sipXecs just gives you a platform to compile, install and configure your services and integrate them other services down to the finest level of control. Once you're done building your component, you can share it with a greater SIPfoundry community all while maintaining your own source code repository. I've worked on sipXecs since version sipXecs v1.0 and I've never been more excited about a release than sipXecs v4.6. If I've just peeked your interest, please email me at dhubler at ezuce.com and I can provide more details. You can also email me on the sipxecs user's mailing list : sipx-users at list.sipfoundry.org http://www.sipfoundry.org/sipx-colab Douglas Hubler sipXecs developer From dgarcia at anew.com.ve Fri Jan 27 19:23:21 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 27 Jan 2012 11:53:21 -0430 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> Message-ID: <4F22CF79.5090801@anew.com.ve> Hi Brian, Could you explain a little bit more what your want do do? Are you trying to do this: the analogic phone is a FS extension, connected to an ATA device (or it is connected to FXS card in FS server?). When someone take the handset, automatically connect to another extension, make pstn call or play a message? Or your are trying to do this: An incoming call from a PSTN line connected to a TDM card in your FS server, you want to automatically connect to another extension, make pstn call or play a message? On 1/27/2012 10:49 AM, Brian wrote: > > Hi Avi, > > Maybe that's the term I should use, hot-dial. I'm talking about > analogic phones, not VoIP phones. If it's an analogic phone plugged > into an FXS socket, when you pick up the phone, the PBX is capable of > detecting the off-hook event. That feature must be supported by the > analogic card's driver. In asterisk's case, that is the option named > "immediate=true|false" in the Zapata.conf file, as stated here: > http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf > > Best regards, > > Brian > > *De :*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *De la part de* > Avi Marcus > *Envoy? :* vendredi 27 janvier 2012 15:42 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk > > I think you want a hot-dial..? Some phones/ATAs support it. I don't > think it's an FS/Asterisk feature. > > e.g. most phones don't even contact the sip server until the person > "finished dialing". > > > -Avi > > > > On Fri, Jan 27, 2012 at 4:38 PM, Brian > wrote: > > Hi Saugort, > > Thank you for these links. However that is not really what I'm looking > for. What I am trying to do is to play a sound file to an analogic > phone as soon as the later goes off-hook. What I called "eary media" > is not really the ringtone as we often see in SIP protocol. Sorry for > the confusion. > > Any help would be greatly appreciated, > > Regards, > > Brian > > *De :*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *De la part > de* Saugort Dario Garcia Tovar > *Envoy? :* vendredi 27 janvier 2012 13:42 > *? :* freeswitch-users at lists.freeswitch.org > > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk > > Hi Brian, > > Take a look to this links: > > http://wiki.freeswitch.org/wiki/Early_Media > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > > http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html > > http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html > > > > On 1/26/2012 5:45 PM, Brian wrote: > > Hi List, > > I'm quite new to FreeSwitch and I'm trying to replace an old asterisk > installation with a new FreeSwitch. There is however one feature that > is available with Asterisk and I haven't managed to figure it out how > to do it with FreeSwitch. That is about the "s" extension in Asterisk > dial plan. In the old Asterisk installation, when I pick up an > analogic phone (plugged into an FXS slot of a Digium card) the "s" > extension gets run and I can play early media to the phone. Does > anyone know if it is possible to achieve the same thing with > FreeSwitch/FreeTDM? > > Best regards, > > Brian > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/e10924e4/attachment-0001.html From jeff at jefflenk.com Fri Jan 27 19:50:15 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 27 Jan 2012 08:50:15 -0800 (PST) Subject: [Freeswitch-users] Line echo cancellation for voice In-Reply-To: <7ff2441a6e3b28173cf7c8a60cc6c8d3.squirrel@fulvetta.riseup.net> References: <7ff2441a6e3b28173cf7c8a60cc6c8d3.squirrel@fulvetta.riseup.net> Message-ID: <1327683015704-7230417.post@n2.nabble.com> This doc comes from spandsp. I dont believe that functionality is exposed in fs. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Line-echo-cancellation-for-voice-tp7229557p7230417.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at neotiq.com Fri Jan 27 20:05:06 2012 From: brian at neotiq.com (Brian) Date: Fri, 27 Jan 2012 18:05:06 +0100 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <4F22CF79.5090801@anew.com.ve> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> Message-ID: <007c01ccdd15$d2354560$769fd020$@neotiq.com> Hi Saugort, I?m trying to do the first one. When someone take the handset off-hook, before he has pressed any key, play some messages to him. That can be used to play some warning if the outgoing SIP trunk is down, for example. That feature is called batPhone by some people because it is seen in a film about Batman J Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 17:23 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Could you explain a little bit more what your want do do? Are you trying to do this: the analogic phone is a FS extension, connected to an ATA device (or it is connected to FXS card in FS server?). When someone take the handset, automatically connect to another extension, make pstn call or play a message? Or your are trying to do this: An incoming call from a PSTN line connected to a TDM card in your FS server, you want to automatically connect to another extension, make pstn call or play a message? On 1/27/2012 10:49 AM, Brian wrote: Hi Avi, Maybe that?s the term I should use, hot-dial. I?m talking about analogic phones, not VoIP phones. If it?s an analogic phone plugged into an FXS socket, when you pick up the phone, the PBX is capable of detecting the off-hook event. That feature must be supported by the analogic card?s driver. In asterisk?s case, that is the option named ?immediate=true|false? in the Zapata.conf file, as stated here: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Best regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Avi Marcus Envoy? : vendredi 27 janvier 2012 15:42 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk I think you want a hot-dial..? Some phones/ATAs support it. I don't think it's an FS/Asterisk feature. e.g. most phones don't even contact the sip server until the person "finished dialing". -Avi On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote: Hi Saugort, Thank you for these links. However that is not really what I?m looking for. What I am trying to do is to play a sound file to an analogic phone as soon as the later goes off-hook. What I called ?eary media? is not really the ringtone as we often see in SIP protocol. Sorry for the confusion. Any help would be greatly appreciated, Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 13:42 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Take a look to this links: http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as- early-media-td2460205.html http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581 .html On 1/26/2012 5:45 PM, Brian wrote: Hi List, I?m quite new to FreeSwitch and I?m trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven?t managed to figure it out how to do it with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the ?s? extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/b7b6a604/attachment-0001.html From oseslija at gmail.com Sat Jan 28 00:16:48 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 27 Jan 2012 22:16:48 +0100 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <007c01ccdd15$d2354560$769fd020$@neotiq.com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> <007c01ccdd15$d2354560$769fd020$@neotiq.com> Message-ID: You can do this easy in Linksys/Cisco phones/ATAs with dialplan manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk ext which does the IVR or whatever. On Fri, Jan 27, 2012 at 6:05 PM, Brian wrote: > Hi Saugort,**** > > ** ** > > I?m trying to do the first one. When someone take the handset off-hook, > before he has pressed any key, play some messages to him. That can be used > to play some warning if the outgoing SIP trunk is down, for example. That > feature is called batPhoneby some people because it is seen in a film about Batman > J**** > > ** ** > > Regards,**** > > Brian**** > > ** ** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort > Dario Garcia Tovar > *Envoy? :* vendredi 27 janvier 2012 17:23 > > *? :* freeswitch-users at lists.freeswitch.org > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk** > ** > > ** ** > > Hi Brian, > > Could you explain a little bit more what your want do do? > > Are you trying to do this: > the analogic phone is a FS extension, connected to an ATA device (or it is > connected to FXS card in FS server?). When someone take the handset, > automatically connect to another extension, make pstn call or play a > message? > > Or your are trying to do this: > An incoming call from a PSTN line connected to a TDM card in your FS > server, you want to automatically connect to another extension, make pstn > call or play a message? > > On 1/27/2012 10:49 AM, Brian wrote: **** > > Hi Avi,**** > > **** > > Maybe that?s the term I should use, hot-dial. I?m talking about analogic > phones, not VoIP phones. If it?s an analogic phone plugged into an FXS > socket, when you pick up the phone, the PBX is capable of detecting the > off-hook event. That feature must be supported by the analogic card?s > driver. In asterisk?s case, that is the option named ?immediate=true|false? > in the Zapata.conf file, as stated here: > http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf**** > > **** > > Best regards,**** > > Brian**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *De la part de* Avi Marcus > *Envoy? :* vendredi 27 janvier 2012 15:42 > *? :* FreeSWITCH Users Help > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk** > ** > > **** > > I think you want a hot-dial..? Some phones/ATAs support it. I don't think > it's an FS/Asterisk feature.**** > > e.g. most phones don't even contact the sip server until the person > "finished dialing".**** > > > **** > > -Avi**** > > > > > **** > > On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote:**** > > Hi Saugort,**** > > **** > > Thank you for these links. However that is not really what I?m looking > for. What I am trying to do is to play a sound file to an analogic phone as > soon as the later goes off-hook. What I called ?eary media? is not really > the ringtone as we often see in SIP protocol. Sorry for the confusion.**** > > **** > > Any help would be greatly appreciated,**** > > Regards,**** > > Brian**** > > **** > > *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort > Dario Garcia Tovar > *Envoy? :* vendredi 27 janvier 2012 13:42 > *? :* freeswitch-users at lists.freeswitch.org > *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk** > ** > > **** > > Hi Brian, > > Take a look to this links: > > http://wiki.freeswitch.org/wiki/Early_Media > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > > > http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html > > > http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html > > > > On 1/26/2012 5:45 PM, Brian wrote: **** > > Hi List,**** > > **** > > I?m quite new to FreeSwitch and I?m trying to replace an old asterisk > installation with a new FreeSwitch. There is however one feature that is > available with Asterisk and I haven?t managed to figure it out how to do it > with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In > the old Asterisk installation, when I pick up an analogic phone (plugged > into an FXS slot of a Digium card) the ?s? extension gets run and I can > play early media to the phone. Does anyone know if it is possible to > achieve the same thing with FreeSwitch/FreeTDM? **** > > **** > > Best regards,**** > > **** > > Brian**** > > **** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > **** > > **** > > **** > > **** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > **** > > FreeSWITCH-users mailing list**** > > FreeSWITCH-users at lists.freeswitch.org**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > > http://www.freeswitch.org**** > > > > > **** > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12* > *** > > **** > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > ** ** > > FreeSWITCH-users mailing list**** > > FreeSWITCH-users at lists.freeswitch.org**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > > http://www.freeswitch.org**** > > > > > **** > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12* > *** > > ** ** > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/67f21f84/attachment-0001.html From aksrini at hotmail.com Sat Jan 28 01:01:21 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 27 Jan 2012 14:01:21 -0800 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed Message-ID: Hi, I like to add a gateway on the fly. FS applcation which is in mod_managed will read the gateway configuration from the db and will add/update the conf\sip_profiles\external\MygateWay.xml config file. How to execute this command "sofia profile external rescan reloadxml" to reload the gateway using the mod_managed application without using cli. RegardsSrini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120127/5bdc119d/attachment.html From henrikaagaardsorensen at gmail.com Sat Jan 28 14:49:37 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sat, 28 Jan 2012 12:49:37 +0100 Subject: [Freeswitch-users] Can FreeSWITCH go into idle mode or something? Message-ID: I have a completely clean installation of FreeSWITCH on a brand new CentOS 6. Everything works as it should. But testing over a longer period, I see some issues when after 2-3 days without any device registrations or contact to the server in any way, the server doesn't accept registrations. anymore I get timeouts on all devices. A wireshark monitor does see the incoming SIP requests. But nothing happens. A quick restart of the server fixes everything. The FS log only contains: 2012-01-26 10:40:32.184037 [WARNING] switch_scheduler.c:114 Task was executed late by 2 seconds 1 heartbeat (core) Can FS somehow enter an idle period, if no activity has been made, or what could cause this sort of problem? From avi at avimarcus.net Sat Jan 28 19:27:16 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 28 Jan 2012 18:27:16 +0200 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> <007c01ccdd15$d2354560$769fd020$@neotiq.com> Message-ID: Brian, your phones are plugged directly into an FXS card in your FreeSWITCH server? What kind of card is it? This is something that the specific hardware/driver has to support, knowing which one you are using will be essential to giving you an answer.. -Avi On Fri, Jan 27, 2012 at 11:16 PM, Ognjen Seslija wrote: > You can do this easy in Linksys/Cisco phones/ATAs with dialplan > manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk > ext which does the IVR or whatever. > > > On Fri, Jan 27, 2012 at 6:05 PM, Brian wrote: > >> Hi Saugort,**** >> >> ** ** >> >> I?m trying to do the first one. When someone take the handset off-hook, >> before he has pressed any key, play some messages to him. That can be used >> to play some warning if the outgoing SIP trunk is down, for example. That >> feature is called batPhoneby some people because it is seen in a film about Batman >> J**** >> >> ** ** >> >> Regards,**** >> >> Brian**** >> >> ** ** >> >> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort >> Dario Garcia Tovar >> *Envoy? :* vendredi 27 janvier 2012 17:23 >> >> *? :* freeswitch-users at lists.freeswitch.org >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* >> *** >> >> ** ** >> >> Hi Brian, >> >> Could you explain a little bit more what your want do do? >> >> Are you trying to do this: >> the analogic phone is a FS extension, connected to an ATA device (or it >> is connected to FXS card in FS server?). When someone take the handset, >> automatically connect to another extension, make pstn call or play a >> message? >> >> Or your are trying to do this: >> An incoming call from a PSTN line connected to a TDM card in your FS >> server, you want to automatically connect to another extension, make pstn >> call or play a message? >> >> On 1/27/2012 10:49 AM, Brian wrote: **** >> >> Hi Avi,**** >> >> **** >> >> Maybe that?s the term I should use, hot-dial. I?m talking about analogic >> phones, not VoIP phones. If it?s an analogic phone plugged into an FXS >> socket, when you pick up the phone, the PBX is capable of detecting the >> off-hook event. That feature must be supported by the analogic card?s >> driver. In asterisk?s case, that is the option named ?immediate=true|false? >> in the Zapata.conf file, as stated here: >> http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf**** >> >> **** >> >> Best regards,**** >> >> Brian**** >> >> **** >> >> *De :* freeswitch-users-bounces at lists.freeswitch.org [ >> mailto:freeswitch-users-bounces at lists.freeswitch.org] >> *De la part de* Avi Marcus >> *Envoy? :* vendredi 27 janvier 2012 15:42 >> *? :* FreeSWITCH Users Help >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* >> *** >> >> **** >> >> I think you want a hot-dial..? Some phones/ATAs support it. I don't think >> it's an FS/Asterisk feature.**** >> >> e.g. most phones don't even contact the sip server until the person >> "finished dialing".**** >> >> >> **** >> >> -Avi**** >> >> >> >> >> **** >> >> On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote:**** >> >> Hi Saugort,**** >> >> **** >> >> Thank you for these links. However that is not really what I?m looking >> for. What I am trying to do is to play a sound file to an analogic phone as >> soon as the later goes off-hook. What I called ?eary media? is not really >> the ringtone as we often see in SIP protocol. Sorry for the confusion.*** >> * >> >> **** >> >> Any help would be greatly appreciated,**** >> >> Regards,**** >> >> Brian**** >> >> **** >> >> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort >> Dario Garcia Tovar >> *Envoy? :* vendredi 27 janvier 2012 13:42 >> *? :* freeswitch-users at lists.freeswitch.org >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* >> *** >> >> **** >> >> Hi Brian, >> >> Take a look to this links: >> >> http://wiki.freeswitch.org/wiki/Early_Media >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready >> >> >> http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html >> >> >> http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html >> >> >> >> On 1/26/2012 5:45 PM, Brian wrote: **** >> >> Hi List,**** >> >> **** >> >> I?m quite new to FreeSwitch and I?m trying to replace an old asterisk >> installation with a new FreeSwitch. There is however one feature that is >> available with Asterisk and I haven?t managed to figure it out how to do it >> with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In >> the old Asterisk installation, when I pick up an analogic phone (plugged >> into an FXS slot of a Digium card) the ?s? extension gets run and I can >> play early media to the phone. Does anyone know if it is possible to >> achieve the same thing with FreeSwitch/FreeTDM? **** >> >> **** >> >> Best regards,**** >> >> **** >> >> Brian**** >> >> **** >> >> >> >> >> **** >> >> _________________________________________________________________________**** >> >> Professional FreeSWITCH Consulting Services:**** >> >> consulting at freeswitch.org**** >> >> http://www.freeswitchsolutions.com**** >> >> **** >> >> **** >> >> **** >> >> **** >> >> Official FreeSWITCH Sites**** >> >> http://www.freeswitch.org**** >> >> http://wiki.freeswitch.org**** >> >> http://www.cluecon.com**** >> >> **** >> >> FreeSWITCH-users mailing list**** >> >> FreeSWITCH-users at lists.freeswitch.org**** >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** >> >> http://www.freeswitch.org**** >> >> >> >> >> **** >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 >> **** >> >> **** >> >> -- >> Atentamente, >> *Dario Garc?a* >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> **** >> >> >> >> >> **** >> >> _________________________________________________________________________**** >> >> Professional FreeSWITCH Consulting Services:**** >> >> consulting at freeswitch.org**** >> >> http://www.freeswitchsolutions.com**** >> >> ** ** >> >> **** >> >> **** >> >> ** ** >> >> Official FreeSWITCH Sites**** >> >> http://www.freeswitch.org**** >> >> http://wiki.freeswitch.org**** >> >> http://www.cluecon.com**** >> >> ** ** >> >> FreeSWITCH-users mailing list**** >> >> FreeSWITCH-users at lists.freeswitch.org**** >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** >> >> http://www.freeswitch.org**** >> >> >> >> >> **** >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12 >> **** >> >> ** ** >> >> -- >> Atentamente, >> *Dario Garc?a* >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/8838ed55/attachment-0001.html From all.eforums at gmail.com Sat Jan 28 20:02:32 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Sat, 28 Jan 2012 12:02:32 -0500 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module Message-ID: Dear All, As stated in the subject, has anyone tried to configure the linux client downloaded during the Skypopen installation in a way such that the Skype client can run in a different machine than the one running Freeswitch w/mod_skypopen loaded? Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/057a4e19/attachment.html From curriegrad2004 at gmail.com Sat Jan 28 21:10:29 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 28 Jan 2012 10:10:29 -0800 Subject: [Freeswitch-users] Can FreeSWITCH go into idle mode or something? In-Reply-To: References: Message-ID: Are you running FS under a Virtual Machine? Doubt the timing resolution is an issue here as CentOS 6 comes with a tickless kernel by default. 2012/1/28 Henrik Aagaard S?rensen : > I have a completely clean installation of FreeSWITCH on a brand new > CentOS 6. Everything works as it should. > > But testing over a longer period, I see some issues when after 2-3 > days without any device registrations or contact to the server in any > way, the server doesn't accept registrations. anymore I get timeouts > on all devices. A wireshark monitor does see the incoming SIP > requests. But nothing happens. A quick restart of the server fixes > everything. > > The FS log only contains: > 2012-01-26 10:40:32.184037 [WARNING] switch_scheduler.c:114 Task was > executed late by 2 seconds 1 heartbeat (core) > > Can FS somehow enter an idle period, if no activity has been made, or > what could cause this sort of problem? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Sat Jan 28 21:13:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 28 Jan 2012 13:13:47 -0500 Subject: [Freeswitch-users] Can FreeSWITCH go into idle mode or something? In-Reply-To: References: Message-ID: I know of some VPS providers that will basically suspend the instance when no activity is detected. That could very well be the issue. On Jan 28, 2012 1:11 PM, "curriegrad2004" wrote: > Are you running FS under a Virtual Machine? Doubt the timing > resolution is an issue here as CentOS 6 comes with a tickless kernel > by default. > > 2012/1/28 Henrik Aagaard S?rensen : > > I have a completely clean installation of FreeSWITCH on a brand new > > CentOS 6. Everything works as it should. > > > > But testing over a longer period, I see some issues when after 2-3 > > days without any device registrations or contact to the server in any > > way, the server doesn't accept registrations. anymore I get timeouts > > on all devices. A wireshark monitor does see the incoming SIP > > requests. But nothing happens. A quick restart of the server fixes > > everything. > > > > The FS log only contains: > > 2012-01-26 10:40:32.184037 [WARNING] switch_scheduler.c:114 Task was > > executed late by 2 seconds 1 heartbeat (core) > > > > Can FS somehow enter an idle period, if no activity has been made, or > > what could cause this sort of problem? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/7f392b15/attachment.html From curriegrad2004 at gmail.com Sat Jan 28 21:18:38 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 28 Jan 2012 10:18:38 -0800 Subject: [Freeswitch-users] Can FreeSWITCH go into idle mode or something? In-Reply-To: References: Message-ID: VMware workstation also behaves like this when the priority settings are incorrect in certain versions of the virtualization software. On Sat, Jan 28, 2012 at 10:13 AM, Brian Foster wrote: > I know of some VPS providers that will basically suspend the instance when > no activity is detected. That could very well be the issue. > > On Jan 28, 2012 1:11 PM, "curriegrad2004" wrote: >> >> Are you running FS under a Virtual Machine? Doubt the timing >> resolution is an issue here as CentOS 6 comes with a tickless kernel >> by default. >> >> 2012/1/28 Henrik Aagaard S?rensen : >> > I have a completely clean installation of FreeSWITCH on a brand new >> > CentOS 6. Everything works as it should. >> > >> > But testing over a longer period, I see some issues when after 2-3 >> > days without any device registrations or contact to the server in any >> > way, the server doesn't accept registrations. anymore I get timeouts >> > on all devices. A wireshark monitor does see the incoming SIP >> > requests. But nothing happens. A quick restart of the server fixes >> > everything. >> > >> > The FS log only contains: >> > 2012-01-26 10:40:32.184037 [WARNING] switch_scheduler.c:114 Task was >> > executed late by 2 seconds 1 heartbeat (core) >> > >> > Can FS somehow enter an idle period, if no activity has been made, or >> > what could cause this sort of problem? >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From georg at riseup.net Sat Jan 28 22:37:02 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 28 Jan 2012 20:37:02 +0100 Subject: [Freeswitch-users] mod_fifo: agent not ringing Message-ID: <3a92f8833f8c048799437b1e183a5e86.squirrel@fulvetta.riseup.net> He all, I'm still quite new to FS, so sorry if this is just a stupid question. I'm trying to setup a queue with mod_fifo. Really simple, just with two phones. Idea is that people call a number, hear moh and the two phones are ringing. If picked up, the call is bridged. So I tried this: fifo.con.xml {call_timeout=30,fifo_member_wait=nowait}user/10 at 192.168.1.93 {call_timeout=30,fifo_member_wait=nowait}user/15 at 192.168.1.93 In my public.xml dialplan I put: A caller now hears the music, but the phones aren't ringing (and there are no error messages about this). I changed the @192.168.1.93 from @$${domain}, cause FS tried to call for example 12 at a different ip then my phones are registered. This is, because I've got two ips. One from my provider, receiving and sending calls there, and 192.168.1.93, where my phones are registered. Before I changed the @setting, FS gave me an error that the phone 12 at my-provider-ip is absent, which is true. So I thought I just have to change the domain setting. What I'm missing here? Thanks, Georg From jdiaz at coinfru.com Sat Jan 28 23:06:50 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Sat, 28 Jan 2012 21:06:50 +0100 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM Message-ID: Is there any way to have a SIP Trunk. I mean to have for example 32 channels merged in one? or something like this? When i try to find SIP trunk on internet i just see options for TDM gateway or similar but not really a multiplexed trunk. Can we do something with freeswitch? Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/ca27b2c4/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/ca27b2c4/attachment-0001.jpe From sos at sokhapkin.dyndns.org Sat Jan 28 23:13:54 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 28 Jan 2012 15:13:54 -0500 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: Message-ID: <6180011.valLcmJJfL@sos> SIP (and RTP) have no concept of trunking/audio frames multiplexing unlike IAX2 and TDM. On Saturday 28 January 2012 21:06:50 Josue Diaz Cruz wrote: > Is there any way to have a SIP Trunk. I mean to have for example 32 channels > merged in one? or something like this? When i try to find SIP trunk on > internet i just see options for TDM gateway or similar but not really a > multiplexed trunk. > > Can we do something with freeswitch? > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > jdiaz at coinfru.com > > > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.com From fraserredmond at gmail.com Sat Jan 28 23:17:17 2012 From: fraserredmond at gmail.com (Fraser Redmond) Date: Sat, 28 Jan 2012 15:17:17 -0500 Subject: [Freeswitch-users] Whats the best way to insert audio into a call from event socket Message-ID: I want to insert a queue of voice recordings into a call, and be able to stop them all. Doing both via the event socket. Currently I'm doing: api uuid_broadcast $uuid fileName1.wav both api uuid_broadcast $uuid fileName2.wav both And stopping the audio with: api uuid_break $uuid all That works quite well, except for two problems: 1) I can't use it because when the audio is playing the parties can't hear each other talk. 2) There is a slight delay of about half a second before the audio starts (though I'm guessing this can't be helped.) Previously I'd used: api uuid_displace $uuid start fileName1.wav 0 mux But I don't think I'd been able to queue up more than one file with that. There's a few other possibilities that I've thought of, including going into a js/lua script and looping inside that, or changing the call into a conference, and using the conference commands. Rather than mess around trying lots of ideas, I'd rather focus on the one best approach - any ideas? Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/6f7f6ded/attachment.html From georg at riseup.net Sat Jan 28 23:18:31 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 28 Jan 2012 21:18:31 +0100 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: <6180011.valLcmJJfL@sos> References: <6180011.valLcmJJfL@sos> Message-ID: Hi, > SIP (and RTP) have no concept of trunking/audio frames multiplexing unlike > IAX2 and TDM. Are you sure with this? I've got a sip trunk trough a german provider with eight channels. Greetings, Georg From curriegrad2004 at gmail.com Sat Jan 28 23:22:03 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 28 Jan 2012 12:22:03 -0800 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: <6180011.valLcmJJfL@sos> Message-ID: SIP only has gateways, no trunks. When you're dealing with VoIP, treat SIP like data, not like voice channels. On Sat, Jan 28, 2012 at 12:18 PM, wrote: > Hi, > >> SIP (and RTP) have no concept of trunking/audio frames multiplexing unlike >> IAX2 and TDM. > > Are you sure with this? I've got a sip trunk trough a german provider with > eight channels. > > Greetings, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From georg at riseup.net Sat Jan 28 23:26:24 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 28 Jan 2012 21:26:24 +0100 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: <6180011.valLcmJJfL@sos> Message-ID: > SIP only has gateways, no trunks. When you're dealing with VoIP, treat > SIP like data, not like voice channels. Allright, but makes this a difference in practice? From freeswitch at earthspike.net Sat Jan 28 23:42:32 2012 From: freeswitch at earthspike.net (John) Date: Sat, 28 Jan 2012 20:42:32 +0000 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: <6180011.valLcmJJfL@sos> Message-ID: <4F245DB8.3000007@earthspike.net> It does make a considerable difference if you are running your RTP through IP encryption (eg IPSec VPN) then the per-packet overhead becomes a significant factor. Companies such as NET (www.net.com) provide switches (VX400, VX900, etc) that run bespoke SIP-like and RTP-like protocols that merge RTP calls between 2 WAN nodes into single packets and this can reduce the per-packet overhead of encryption quite significantly. The cost is a small additional latency (eg 20-40ms) while the packets from various calls are aggregated. John PS. I have no affiliation with NET. On 28/01/12 20:26, georg at riseup.net wrote: >> SIP only has gateways, no trunks. When you're dealing with VoIP, treat >> SIP like data, not like voice channels. > Allright, but makes this a difference in practice? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From sos at sokhapkin.dyndns.org Sat Jan 28 23:44:44 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Sat, 28 Jan 2012 15:44:44 -0500 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: <6180011.valLcmJJfL@sos> Message-ID: <1350919.7nuRjOopdM@sos> "Channels" and "audio frames multiplexing" to preserve bandwidth are different things. On Saturday 28 January 2012 21:18:31 georg at riseup.net wrote: > Hi, > > > SIP (and RTP) have no concept of trunking/audio frames multiplexing > > unlike IAX2 and TDM. > > Are you sure with this? I've got a sip trunk trough a german provider with > eight channels. > > Greetings, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From georg at riseup.net Sun Jan 29 00:32:03 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 28 Jan 2012 22:32:03 +0100 Subject: [Freeswitch-users] mod_fifo: agent not ringing In-Reply-To: <3a92f8833f8c048799437b1e183a5e86.squirrel@fulvetta.riseup.net> References: <3a92f8833f8c048799437b1e183a5e86.squirrel@fulvetta.riseup.net> Message-ID: <32080f6b0de03dc90cb21b8a33ae357d.squirrel@fulvetta.riseup.net> > A caller now hears the music, but the phones aren't ringing (and there are > no error messages about this). Solved this by myself. My mistake was using different values for @xxx. Georg From georg at riseup.net Sun Jan 29 00:40:21 2012 From: georg at riseup.net (georg at riseup.net) Date: Sat, 28 Jan 2012 22:40:21 +0100 Subject: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter Message-ID: He there, I would like to implement a quite simple queue, but I don't know if mod_fifo is able to handle this. Would be great if someone could clarify this: - One phone number, two phones - If both or one is idle, call should be routed directly to the phone(s), no moh should be played (just ringtones) - if both in use, call should be routed into a queue, caller should hear moh - if after this a phone gets idle, call should be routed to the phone, moh should stopped and ringtones be played to the caller - if a call comes from the queue, and both phones are idle, both phones should ring at the same time (I tried this with mod_fifo, is this possible?) Thanks, Georg From georg at riseup.net Sun Jan 29 02:02:49 2012 From: georg at riseup.net (georg at riseup.net) Date: Sun, 29 Jan 2012 00:02:49 +0100 Subject: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter In-Reply-To: References: Message-ID: <66d00fb19143721401461709b9c1d7f8.squirrel@fulvetta.riseup.net> > - if a call comes from the queue, and both phones are idle, both phones > should ring at the same time (I tried this with mod_fifo, is this > possible?) I achieved this now using call groups. Georg From th982a at googlemail.com Sun Jan 29 02:46:55 2012 From: th982a at googlemail.com (Tamer Higazi) Date: Sun, 29 Jan 2012 00:46:55 +0100 Subject: [Freeswitch-users] building mod_opal fails on gentoo linux Message-ID: <4F2488EF.7080106@googlemail.com> Hi people! I have successfully built the latest freeswitch_snapshot but I am not capable to build "mod_opal". Can somebody tell me, what I did wrong? ptlib and opal are installed on my system. thanks for any support. here is the output: making all mod_opal Compiling /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp... Compiling /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp ... /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:564:26: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, void*&, unsigned int&, OpalConnection::StringOptions*&)' /usr/include/opal/opal/localep.h:249:5: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*) /usr/include/opal/opal/localep.h:242:1: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1295:25: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)': /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1417:25: error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' make[5]: *** [mod_opal.lo] Fehler 1 make[4]: *** [all] Fehler 1 make[3]: *** [mod_opal-all] Fehler 1 make[2]: *** [all-recursive] Fehler 1 make[1]: *** [all-recursive] Fehler 1 make: *** [all] Fehler 2 and my system: tamer at office /storage/downloads/freeswitch-snapshot $ emerge --info opal ptlib Portage 2.1.10.44 (default/linux/amd64/10.0/desktop/gnome, gcc-4.5.3, glibc-2.13-r4, 3.0.6-gentoo x86_64) ================================================================= System Settings ================================================================= System uname: Linux-3.0.6-gentoo-x86_64-Intel-R-_Core-TM-2_CPU_6600_ at _2.40GHz-with-gentoo-2.0.3 Timestamp of tree: Thu, 26 Jan 2012 23:30:01 +0000 app-shells/bash: 4.1_p9 dev-java/java-config: 2.1.11-r3 dev-lang/python: 2.7.2-r3, 3.1.4-r3 dev-util/cmake: 2.8.6-r4 dev-util/pkgconfig: 0.26 sys-apps/baselayout: 2.0.3 sys-apps/openrc: 0.9.8.2 sys-apps/sandbox: 2.5 sys-devel/autoconf: 2.13, 2.68 sys-devel/automake: 1.10.3, 1.11.1 sys-devel/binutils: 2.21.1-r1 sys-devel/gcc: 4.4.5, 4.5.3-r1 sys-devel/gcc-config: 1.4.1-r1 sys-devel/libtool: 2.4-r1 sys-devel/make: 3.82-r1 sys-kernel/linux-headers: 3.1 (virtual/os-headers) sys-libs/glibc: 2.13-r4 Repositories: gentoo freeswitch voyageur proaudio calculate x-liquidx ACCEPT_KEYWORDS="amd64" ACCEPT_LICENSE="*" CBUILD="x86_64-pc-linux-gnu" CFLAGS="-march=core2 -O3 -pipe" CHOST="x86_64-pc-linux-gnu" CONFIG_PROTECT="/etc /opt/openfire/resources/security/ /usr/lib64/fax /usr/share/config /usr/share/gnupg/qualified.txt /var/spool/fax/etc" CONFIG_PROTECT_MASK="/etc/ca-certificates.conf /etc/dconf /etc/env.d /etc/env.d/java/ /etc/fonts/fonts.conf /etc/gconf /etc/gentoo-release /etc/revdep-rebuild /etc/sandbox.d /etc/terminfo /etc/texmf/language.dat.d /etc/texmf/language.def.d /etc/texmf/updmap.d /etc/texmf/web2c" CXXFLAGS="-march=core2 -O3 -pipe" DISTDIR="/usr/portage/distfiles" EMERGE_DEFAULT_OPTS="--autounmask=n" FEATURES="assume-digests binpkg-logs distlocks ebuild-locks fixlafiles news parallel-fetch protect-owned sandbox sfperms strict unknown-features-warn unmerge-logs unmerge-orphans userfetch" FFLAGS="" GENTOO_MIRRORS="http://gentoo.mneisen.org/ http://gentoo.mirror.dkm.cz/pub/gentoo/ http://mirrors.linuxant.fr/distfiles.gentoo.org/ http://mirror.netcologne.de/gentoo/" LANG="de_DE.UTF-8" LDFLAGS="-Wl,-O1 -Wl,--as-needed" LINGUAS="de en ar" MAKEOPTS="-j3" PKGDIR="/usr/portage/packages" PORTAGE_CONFIGROOT="/" PORTAGE_RSYNC_OPTS="--recursive --links --safe-links --perms --times --compress --force --whole-file --delete --stats --timeout=180 --exclude=/distfiles --exclude=/local --exclude=/packages" PORTAGE_TMPDIR="/var/tmp" PORTDIR="/usr/portage" PORTDIR_OVERLAY="/var/lib/layman/freeswitch /var/lib/layman/voyageur /var/lib/layman/pro-audio /var/lib/layman/calculate /var/lib/layman/liquidx" SYNC="rsync://rsync.gentoo.org/gentoo-portage" USE="X a52 aac acl acpi alsa amd64 apm avahi berkdb bluetooth branding bzip2 cairo cdda cdr cli colord consolekit cracklib crypt cups custom-cflags cxx dbus disk-partition dri dts dvd dvdr eds emboss encode evo exif fam firefox flac fontconfig fortran gdbm gdu gif gnome gnome-keyring gnome-online-accounts gpm gstreamer gtk gtk3 iconv ipv6 java jpeg kde kerberos lcms ldap libnotify mad mmx mng modules mp3 mp4 mpeg mudflap multilib nautilus ncurses nls nptl nptlonly ogg opengl openmp pam pango pcre pdf png policykit ppds pppd pulseaudio python qt3 qt3support qt4 readline scanner sdl session socialweb spell sqlite sse sse2 ssl startup-notification svg sysfs system-sqlite tcpd tiff truetype udev unicode unlock-notify usb vorbis x264 xcb xinerama xml xorg xulrunner xv xvid zlib" ALSA_CARDS="ali5451 als4000 atiixp atiixp-modem bt87x ca0106 cmipci emu10k1x ens1370 ens1371 es1938 es1968 fm801 hda-intel intel8x0 intel8x0m maestro3 trident usb-audio via82xx via82xx-modem ymfpci" ALSA_PCM_PLUGINS="adpcm alaw asym copy dmix dshare dsnoop empty extplug file hooks iec958 ioplug ladspa lfloat linear meter mmap_emul mulaw multi null plug rate route share shm softvol" APACHE2_MODULES="actions alias auth_basic authn_alias authn_anon authn_dbm authn_default authn_file authz_dbm authz_default authz_groupfile authz_host authz_owner authz_user autoindex cache cgi%* cgid%* dav dav_fs dav_lock deflate dir disk_cache env expires ext_filter file_cache filter headers include info log_config logio mem_cache mime mime_magic negotiation rewrite setenvif speling status unique_id userdir usertrack vhost_alias" CALLIGRA_FEATURES="kexi words flow plan stage tables krita karbon braindump" CAMERAS="ptp2" COLLECTD_PLUGINS="df interface irq load memory rrdtool swap syslog" ELIBC="glibc" GPSD_PROTOCOLS="ashtech aivdm earthmate evermore fv18 garmin garmintxt gpsclock itrax mtk3301 nmea ntrip navcom oceanserver oldstyle oncore rtcm104v2 rtcm104v3 sirf superstar2 timing tsip tripmate tnt ubx" INPUT_DEVICES="evdev synaptics" KERNEL="linux" LCD_DEVICES="bayrad cfontz cfontz633 glk hd44780 lb216 lcdm001 mtxorb ncurses text" LINGUAS="de en ar" PHP_TARGETS="php5-3" RUBY_TARGETS="ruby18" USERLAND="GNU" VIDEO_CARDS="nvidia nv" XTABLES_ADDONS="quota2 psd pknock lscan length2 ipv4options ipset ipp2p iface geoip fuzzy condition tee tarpit sysrq steal rawnat logmark ipmark dhcpmac delude chaos account" Unset: CPPFLAGS, CTARGET, INSTALL_MASK, LC_ALL, PORTAGE_BUNZIP2_COMMAND, PORTAGE_COMPRESS, PORTAGE_COMPRESS_FLAGS, PORTAGE_RSYNC_EXTRA_OPTS ================================================================= Package Settings ================================================================= net-libs/opal-3.6.8-r2 was built with the following: USE="audio celt dtmf fax ffmpeg h224 h281 h323 iax ipv6 java ldap (multilib) plugins sip sipim srtp ssl theora video wav x264 xml -capi -debug -doc -examples -ilbc -ivr -ixj -lid -sbc -static-libs -stats -swig -vpb -vxml -x264-static" CFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden -D__STDC_CONSTANT_MACROS" CXXFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden -D__STDC_CONSTANT_MACROS" net-libs/ptlib-2.6.7-r1 was built with the following: USE="alsa asn audio dtmf ffmpeg http ipv6 jabber ldap (multilib) sasl sdl soap ssl stun v4l video vxml wav xml xmlrpc -debug -doc (-esd) -examples -ftp -mail -odbc -oss -pch -qos -remote -serial -shmvideo -snmp -socks -static-libs -telnet -tts" From krice at freeswitch.org Sun Jan 29 03:41:13 2012 From: krice at freeswitch.org (Ken Rice) Date: Sat, 28 Jan 2012 18:41:13 -0600 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: <4F245DB8.3000007@earthspike.net> Message-ID: There is no such thing as trunking in sip as in trunking like iax2... There are much easier ways to deal with this in sip... For instance set your ptime between freeswitch boxes to be 40ms, this effectively cuts the RTP/PPS overhead in 1/2... Set it to 60ms cut it by 66%.... This works by increasing the amount of audio sent in 1 pack from 20ms to 40 or 60ms... Merging RTP between calls after that point is pretty much a mute point as you already are achieving a large reduction in RTP overhead... K On 1/28/12 2:42 PM, "John" wrote: > It does make a considerable difference if you are running your RTP > through IP encryption (eg IPSec VPN) then the per-packet overhead > becomes a significant factor. Companies such as NET (www.net.com) > provide switches (VX400, VX900, etc) that run bespoke SIP-like and > RTP-like protocols that merge RTP calls between 2 WAN nodes into single > packets and this can reduce the per-packet overhead of encryption quite > significantly. The cost is a small additional latency (eg 20-40ms) > while the packets from various calls are aggregated. > > John > > PS. I have no affiliation with NET. > > On 28/01/12 20:26, georg at riseup.net wrote: >>> SIP only has gateways, no trunks. When you're dealing with VoIP, treat >>> SIP like data, not like voice channels. >> Allright, but makes this a difference in practice? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bote_radio at botecomm.com Sun Jan 29 08:32:01 2012 From: bote_radio at botecomm.com (Bote Man) Date: Sun, 29 Jan 2012 00:32:01 -0500 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> <007c01ccdd15$d2354560$769fd020$@neotiq.com> Message-ID: <039d01ccde47$551a8f30$ff4fad90$@com> It does not matter if the FXS port is a card in the FS server or an analog gateway/ATA. The device to which the analog phone is connected must provide the first sound that is heard (dial tone), or else connect immediately via FS to the source of the desired sound file. Some people call this "hot line", old telephone heads in the U.S. call this "ringdown private line", Cisco turns this name around and calls it PLAR (private line automatic ringdown) in their gateway devices. With no special processing or commands you will hear the happy sound of the dial tone, as always. I agree with the answer that says to call automatically an IVR application that plays the desired file(s) and processes the dialed digits in response. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, 28 January, 2012 11:27 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Brian, your phones are plugged directly into an FXS card in your FreeSWITCH server? What kind of card is it? This is something that the specific hardware/driver has to support, knowing which one you are using will be essential to giving you an answer.. -Avi On Fri, Jan 27, 2012 at 11:16 PM, Ognjen Seslija wrote: You can do this easy in Linksys/Cisco phones/ATAs with dialplan manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk ext which does the IVR or whatever. On Fri, Jan 27, 2012 at 6:05 PM, Brian wrote: Hi Saugort, I?m trying to do the first one. When someone take the handset off-hook, before he has pressed any key, play some messages to him. That can be used to play some warning if the outgoing SIP trunk is down, for example. That feature is called batPhone by some people because it is seen in a film about Batman J Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 17:23 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Could you explain a little bit more what your want do do? Are you trying to do this: the analogic phone is a FS extension, connected to an ATA device (or it is connected to FXS card in FS server?). When someone take the handset, automatically connect to another extension, make pstn call or play a message? Or your are trying to do this: An incoming call from a PSTN line connected to a TDM card in your FS server, you want to automatically connect to another extension, make pstn call or play a message? On 1/27/2012 10:49 AM, Brian wrote: Hi Avi, Maybe that?s the term I should use, hot-dial. I?m talking about analogic phones, not VoIP phones. If it?s an analogic phone plugged into an FXS socket, when you pick up the phone, the PBX is capable of detecting the off-hook event. That feature must be supported by the analogic card?s driver. In asterisk?s case, that is the option named ?immediate=true|false? in the Zapata.conf file, as stated here: http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Best regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Avi Marcus Envoy? : vendredi 27 janvier 2012 15:42 ? : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk I think you want a hot-dial..? Some phones/ATAs support it. I don't think it's an FS/Asterisk feature. e.g. most phones don't even contact the sip server until the person "finished dialing". -Avi On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote: Hi Saugort, Thank you for these links. However that is not really what I?m looking for. What I am trying to do is to play a sound file to an analogic phone as soon as the later goes off-hook. What I called ?eary media? is not really the ringtone as we often see in SIP protocol. Sorry for the confusion. Any help would be greatly appreciated, Regards, Brian De : freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de Saugort Dario Garcia Tovar Envoy? : vendredi 27 janvier 2012 13:42 ? : freeswitch-users at lists.freeswitch.org Objet : Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk Hi Brian, Take a look to this links: http://wiki.freeswitch.org/wiki/Early_Media http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file -as-early-media-td2460205.html http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td547 9581.html On 1/26/2012 5:45 PM, Brian wrote: Hi List, I?m quite new to FreeSwitch and I?m trying to replace an old asterisk installation with a new FreeSwitch. There is however one feature that is available with Asterisk and I haven?t managed to figure it out how to do it with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In the old Asterisk installation, when I pick up an analogic phone (plugged into an FXS slot of a Digium card) the ?s? extension gets run and I can play early media to the phone. Does anyone know if it is possible to achieve the same thing with FreeSwitch/FreeTDM? Best regards, Brian ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/5b2f6614/attachment-0001.html From dujinfang at gmail.com Sun Jan 29 10:09:47 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Jan 2012 15:09:47 +0800 Subject: [Freeswitch-users] Whats the best way to insert audio into a call from event socket In-Reply-To: References: Message-ID: <8532801A6C8A46D1B832FCE1271EB111@gmail.com> You can chain files with file_string:// so it should work with uuid_displace http://wiki.freeswitch.org/wiki/Mod_file_string -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Sunday, January 29, 2012 at 4:17 AM, Fraser Redmond wrote: > I want to insert a queue of voice recordings into a call, and be able to stop them all. Doing both via the event socket. > > Currently I'm doing: > api uuid_broadcast $uuid fileName1.wav both > api uuid_broadcast $uuid fileName2.wav both > > And stopping the audio with:api uuid_break $uuid all > > That works quite well, except for two problems: > 1) I can't use it because when the audio is playing the parties can't hear each other talk. > 2) There is a slight delay of about half a second before the audio starts (though I'm guessing this can't be helped.) > > Previously I'd used: > api uuid_displace $uuid start fileName1.wav 0 mux > > But I don't think I'd been able to queue up more than one file with that. > > There's a few other possibilities that I've thought of, including going into a js/lua script and looping inside that, or changing the call into a conference, and using the conference commands. > > Rather than mess around trying lots of ideas, I'd rather focus on the one best approach - any ideas? > > Cheers, > Fraser > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/7faa82d1/attachment.html From dujinfang at gmail.com Sun Jan 29 10:14:49 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Jan 2012 15:14:49 +0800 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: References: Message-ID: skype clients should run with FS on the same box. You could run both FS and skype clients on another box just as a gateway if you want scale. We had run multi- FS instances on one box for scale/stable purpose, but you can get the idea from http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Sunday, January 29, 2012 at 1:02 AM, A E [Gmail] wrote: > Dear All, > > As stated in the subject, has anyone tried to configure the linux client downloaded during the Skypopen installation in a way such that the Skype client can run in a different machine than the one running Freeswitch w/mod_skypopen loaded? > > Thx > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/d538ca65/attachment.html From dujinfang at gmail.com Sun Jan 29 10:36:03 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Jan 2012 15:36:03 +0800 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> Message-ID: <7B9F69B4658B4B54B245BB5E37907E20@gmail.com> -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Tuesday, January 24, 2012 at 6:20 AM, Stephen Wilde wrote: > Can be useful to run Freeswitch with "-nosql" option? > What I lose in this case? > Just try and see. Not much if you don't use help, show channels, show calls etc, and presence. > > On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde wrote: > > I'm already using the ramdisk. > > The problem happens when I have a provider that give congestion. > > In this case Freeswitch receives many tries but few connected calls and the number of session per second is high. > > To avoid the "event system overloading" (avoiding to lower the global session per second 'sps' parameter) I have insert in dialplan: > > > > > > > > > > > > > > In this way I have limited the session rate for the congestioned destination where I have so many tries. > > > > My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage of disk is near zero (dstat) why I cannot handle this session rate? > > > > > > > > On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale wrote: > > > you can tell that by taking system vitals its hard to tell from the small amount of data. > > > I do know that to get those errors, you have to push the core so hard that the sql stmts queuing up for transactions are getting too large for the rate at which they are written to the DB. Try a ramdisk like Michael suggested. > > > > > > > > > On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: > > > > I'm using ram disk for the FS database "freeswitch/db". > > > > Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect the performance? > > > > > > > > > > > > On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale wrote: > > > > > the box can't handle the load, the disk io from the sql stmts is backing up the events. > > > > > get a nicer box with the money saved from free softswitch =p > > > > > > > > > > > > > > > On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: > > > > > > Did you bind any callbacks to events that might be taking a long time to process? > > > > > > > > > > > > > > > > > > > > > > > > On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: > > > > > > > I saw "increase the event system capacity" because in the log there was a row: > > > > > > > > > > > > > > [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things down. > > > > > > > > > > > > > > Where it seems that all event dispatch threads are "busy" but I see that the cpu has many idle cycles so why not increase the number of dispatch threads? > > > > > > > Or I'm wrong? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > Anthony Minessale II > > > > > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > > > ClueCon http://www.cluecon.com/ > > > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > > > > > AIM: anthm > > > > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > > > > > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > > > > > > > > > FreeSWITCH Developer Conference > > > > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > > > > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > > > > > pstn:+19193869900 (tel:%2B19193869900) > > > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://wiki.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > Anthony Minessale II > > > > > > FreeSWITCH http://www.freeswitch.org/ > > > ClueCon http://www.cluecon.com/ > > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > > > AIM: anthm > > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com) > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com) > > > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > > > > > FreeSWITCH Developer Conference > > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org) > > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org) > > > pstn:+19193869900 (tel:%2B19193869900) > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/b54ae622/attachment-0001.html From dujinfang at gmail.com Sun Jan 29 10:36:20 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Jan 2012 15:36:20 +0800 Subject: [Freeswitch-users] NDLB-connectile-dysfunction does not work In-Reply-To: <90CD446A38F84C33BAAD4FE6592259D4@omni1.local> References: <90CD446A38F84C33BAAD4FE6592259D4@omni1.local> Message-ID: <0661B5C00A504A079608E106D6E9CDB8@gmail.com> -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Monday, January 23, 2012 at 6:40 PM, Anestis Mavro wrote: > > Hello, > > > > > > I have tried to add > > > > > > > > > > to the user according to the Wiki page about NAT traversal, but it is not working as expected. > > > I see the answer to the registration request going to the correct (Ethernet) IP address, but the port is wrong. It still uses the port from within the SIP packet, instead of the Ethernet port. > > > > > I have tried also to add at the same time > > > > > > > I guess this one only works in profile settings. > without success > > > > > I am using XML curl for the directory and I confirmed that both settings are in the xml. > > > > > > Is there any other additional setting missing? > > > > > > Thank you > > > > > > > > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/52d666b3/attachment.html From dujinfang at gmail.com Sun Jan 29 14:34:51 2012 From: dujinfang at gmail.com (Seven Du) Date: Sun, 29 Jan 2012 19:34:51 +0800 Subject: [Freeswitch-users] Extending the field size of core's db_data table? In-Reply-To: References: Message-ID: <81FE1555B2164D17B698567AA1FF788B@gmail.com> I think you could just ALTER TABLE ... or re-create the table maybe sqlite doesn't support alter table? -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Tuesday, December 20, 2011 at 2:35 PM, Yehavi Bourvine wrote: > Hello, > > We use the DB api for storing and caching various data. We need to store data that is longer than 255 characters, which is the current limit on the field size there. > > The backend for this API is MySQL (via ODBC). Can I just increase the field size in MySQL, or is there some dependency in FreeSwitch on this size? > > Thanks, __Yehavi: > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/f5df09fb/attachment.html From yehavi.bourvine at gmail.com Sun Jan 29 14:56:45 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sun, 29 Jan 2012 13:56:45 +0200 Subject: [Freeswitch-users] Extending the field size of core's db_data table? In-Reply-To: <81FE1555B2164D17B698567AA1FF788B@gmail.com> References: <81FE1555B2164D17B698567AA1FF788B@gmail.com> Message-ID: I know I can do ALTER TABLE, but I've asked what is the implication about FreeSWITCH software. Anyway, I managed to bypass the problem by other means. Thanks, __Yehavi: 2012/1/29 Seven Du > I think you could just ALTER TABLE ... or re-create the table maybe > sqlite doesn't support alter table? > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > On Tuesday, December 20, 2011 at 2:35 PM, Yehavi Bourvine wrote: > > Hello, > > We use the DB api for storing and caching various data. We need to store > data that is longer than 255 characters, which is the current limit on the > field size there. > > The backend for this API is MySQL (via ODBC). Can I just increase the > field size in MySQL, or is there some dependency in FreeSwitch on this size? > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/17ef5b0f/attachment-0001.html From wstephen80 at gmail.com Sun Jan 29 20:34:37 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Sun, 29 Jan 2012 18:34:37 +0100 Subject: [Freeswitch-users] Event system overloading. Taking a 10 second break In-Reply-To: <7B9F69B4658B4B54B245BB5E37907E20@gmail.com> References: <6FF0ACAB-94E7-42C4-9C98-AA3137B802A9@freeswitch.org> <409622FA-4A80-4155-8DBB-3073594CAF28@freeswitch.org> <7B9F69B4658B4B54B245BB5E37907E20@gmail.com> Message-ID: Ok, thank you for your advice. I don't use show channels or presence, I'll try. Stephen On Sun, Jan 29, 2012 at 8:36 AM, Seven Du wrote: > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > On Tuesday, January 24, 2012 at 6:20 AM, Stephen Wilde wrote: > > Can be useful to run Freeswitch with "-nosql" option? > What I lose in this case? > > > Just try and see. > > Not much if you don't use help, show channels, show calls etc, and > presence. > > > > > On Sat, Jan 21, 2012 at 10:04 PM, Stephen Wilde wrote: > > I'm already using the ramdisk. > The problem happens when I have a provider that give congestion. > In this case Freeswitch receives many tries but few connected calls and > the number of session per second is high. > To avoid the "event system overloading" (avoiding to lower the global > session per second 'sps' parameter) I have insert in dialplan: > > break="never"> > > > > In this way I have limited the session rate for the congestioned > destination where I have so many tries. > > My dubt remain: I have ramdisk, I have many idle cycles on cpu, the usage > of disk is near zero (dstat) why I cannot handle this session rate? > > > > On Sat, Jan 21, 2012 at 12:30 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > you can tell that by taking system vitals its hard to tell from the small > amount of data. > I do know that to get those errors, you have to push the core so hard that > the sql stmts queuing up for transactions are getting too large for the > rate at which they are written to the DB. Try a ramdisk like Michael > suggested. > > > On Fri, Jan 20, 2012 at 3:11 PM, Stephen Wilde wrote: > > I'm using ram disk for the FS database "freeswitch/db". > Can the disks (a dedicated couple of 15k rpm scsi 6gb/s in raid 1) affect > the performance? > > > On Fri, Jan 20, 2012 at 9:45 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > the box can't handle the load, the disk io from the sql stmts is backing > up the events. > get a nicer box with the money saved from free softswitch =p > > > On Fri, Jan 20, 2012 at 1:22 PM, Christopher Rienzo wrote: > > Did you bind any callbacks to events that might be taking a long time to > process? > > > > On Fri, Jan 20, 2012 at 1:54 PM, Stephen Wilde wrote: > > I saw "increase the event system capacity" because in the log there was a > row: > > [CRIT] switch_event.c:360 Out of event dispatch threads! Slowing things > down. > > Where it seems that all event dispatch threads are "busy" but I see that the > cpu has many idle cycles so why not increase the number of dispatch threads? > Or I'm wrong? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/92b0451c/attachment-0001.html From justlikeef at gmail.com Sun Jan 29 20:35:31 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Sun, 29 Jan 2012 12:35:31 -0500 Subject: [Freeswitch-users] building mod_opal fails on gentoo linux In-Reply-To: <4F2488EF.7080106@googlemail.com> References: <4F2488EF.7080106@googlemail.com> Message-ID: <201201291235.32142.justlikeef@gmail.com> I beleive you are using a too new version of mod_opal. Make sure you are using the version specified in the wiki. On Saturday 28 January 2012 18:46:55 Tamer Higazi wrote: > Hi people! > I have successfully built the latest freeswitch_snapshot but I am not > capable to build "mod_opal". Can somebody tell me, what I did wrong? > > ptlib and opal are installed on my system. > > thanks for any support. > > > here is the output: > > making all mod_opal > Compiling > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp... > Compiling > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp > ... > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: > In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, > void*, unsigned int, OpalConnection::StringOptions*, > switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:564:26: > error: no matching function for call to > 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, > void*&, unsigned int&, OpalConnection::StringOptions*&)' > /usr/include/opal/opal/localep.h:249:5: note: candidates are: > OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, > void*) > /usr/include/opal/opal/localep.h:242:1: note: > OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&) > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: > In member function 'switch_status_t > FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)': > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1295:25: > error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp: > In member function 'switch_status_t FSMediaStream::write_frame(const > switch_frame_t*, switch_io_flag_t)': > /storage/downloads/freeswitch-snapshot/src/mod/endpoints/mod_opal/mod_opal.cpp:1417:25: > error: 'class OpalMediaPatch' has no member named 'OnStartMediaPatch' > make[5]: *** [mod_opal.lo] Fehler 1 > make[4]: *** [all] Fehler 1 > make[3]: *** [mod_opal-all] Fehler 1 > make[2]: *** [all-recursive] Fehler 1 > make[1]: *** [all-recursive] Fehler 1 > make: *** [all] Fehler 2 > > > > and my system: > > tamer at office /storage/downloads/freeswitch-snapshot $ emerge --info opal > ptlib > Portage 2.1.10.44 (default/linux/amd64/10.0/desktop/gnome, gcc-4.5.3, > glibc-2.13-r4, 3.0.6-gentoo x86_64) > ================================================================= > System Settings > ================================================================= > System uname: > Linux-3.0.6-gentoo-x86_64-Intel-R-_Core-TM-2_CPU_6600_ at _2.40GHz-with-gentoo-2.0.3 > Timestamp of tree: Thu, 26 Jan 2012 23:30:01 +0000 > app-shells/bash: 4.1_p9 > dev-java/java-config: 2.1.11-r3 > dev-lang/python: 2.7.2-r3, 3.1.4-r3 > dev-util/cmake: 2.8.6-r4 > dev-util/pkgconfig: 0.26 > sys-apps/baselayout: 2.0.3 > sys-apps/openrc: 0.9.8.2 > sys-apps/sandbox: 2.5 > sys-devel/autoconf: 2.13, 2.68 > sys-devel/automake: 1.10.3, 1.11.1 > sys-devel/binutils: 2.21.1-r1 > sys-devel/gcc: 4.4.5, 4.5.3-r1 > sys-devel/gcc-config: 1.4.1-r1 > sys-devel/libtool: 2.4-r1 > sys-devel/make: 3.82-r1 > sys-kernel/linux-headers: 3.1 (virtual/os-headers) > sys-libs/glibc: 2.13-r4 > Repositories: gentoo freeswitch voyageur proaudio calculate x-liquidx > ACCEPT_KEYWORDS="amd64" > ACCEPT_LICENSE="*" > CBUILD="x86_64-pc-linux-gnu" > CFLAGS="-march=core2 -O3 -pipe" > CHOST="x86_64-pc-linux-gnu" > CONFIG_PROTECT="/etc /opt/openfire/resources/security/ /usr/lib64/fax > /usr/share/config /usr/share/gnupg/qualified.txt /var/spool/fax/etc" > CONFIG_PROTECT_MASK="/etc/ca-certificates.conf /etc/dconf /etc/env.d > /etc/env.d/java/ /etc/fonts/fonts.conf /etc/gconf /etc/gentoo-release > /etc/revdep-rebuild /etc/sandbox.d /etc/terminfo > /etc/texmf/language.dat.d /etc/texmf/language.def.d /etc/texmf/updmap.d > /etc/texmf/web2c" > CXXFLAGS="-march=core2 -O3 -pipe" > DISTDIR="/usr/portage/distfiles" > EMERGE_DEFAULT_OPTS="--autounmask=n" > FEATURES="assume-digests binpkg-logs distlocks ebuild-locks fixlafiles > news parallel-fetch protect-owned sandbox sfperms strict > unknown-features-warn unmerge-logs unmerge-orphans userfetch" > FFLAGS="" > GENTOO_MIRRORS="http://gentoo.mneisen.org/ > http://gentoo.mirror.dkm.cz/pub/gentoo/ > http://mirrors.linuxant.fr/distfiles.gentoo.org/ > http://mirror.netcologne.de/gentoo/" > LANG="de_DE.UTF-8" > LDFLAGS="-Wl,-O1 -Wl,--as-needed" > LINGUAS="de en ar" > MAKEOPTS="-j3" > PKGDIR="/usr/portage/packages" > PORTAGE_CONFIGROOT="/" > PORTAGE_RSYNC_OPTS="--recursive --links --safe-links --perms --times > --compress --force --whole-file --delete --stats --timeout=180 > --exclude=/distfiles --exclude=/local --exclude=/packages" > PORTAGE_TMPDIR="/var/tmp" > PORTDIR="/usr/portage" > PORTDIR_OVERLAY="/var/lib/layman/freeswitch /var/lib/layman/voyageur > /var/lib/layman/pro-audio /var/lib/layman/calculate /var/lib/layman/liquidx" > SYNC="rsync://rsync.gentoo.org/gentoo-portage" > USE="X a52 aac acl acpi alsa amd64 apm avahi berkdb bluetooth branding > bzip2 cairo cdda cdr cli colord consolekit cracklib crypt cups > custom-cflags cxx dbus disk-partition dri dts dvd dvdr eds emboss encode > evo exif fam firefox flac fontconfig fortran gdbm gdu gif gnome > gnome-keyring gnome-online-accounts gpm gstreamer gtk gtk3 iconv ipv6 > java jpeg kde kerberos lcms ldap libnotify mad mmx mng modules mp3 mp4 > mpeg mudflap multilib nautilus ncurses nls nptl nptlonly ogg opengl > openmp pam pango pcre pdf png policykit ppds pppd pulseaudio python qt3 > qt3support qt4 readline scanner sdl session socialweb spell sqlite sse > sse2 ssl startup-notification svg sysfs system-sqlite tcpd tiff truetype > udev unicode unlock-notify usb vorbis x264 xcb xinerama xml xorg > xulrunner xv xvid zlib" ALSA_CARDS="ali5451 als4000 atiixp atiixp-modem > bt87x ca0106 cmipci emu10k1x ens1370 ens1371 es1938 es1968 fm801 > hda-intel intel8x0 intel8x0m maestro3 trident usb-audio via82xx > via82xx-modem ymfpci" ALSA_PCM_PLUGINS="adpcm alaw asym copy dmix dshare > dsnoop empty extplug file hooks iec958 ioplug ladspa lfloat linear meter > mmap_emul mulaw multi null plug rate route share shm softvol" > APACHE2_MODULES="actions alias auth_basic authn_alias authn_anon > authn_dbm authn_default authn_file authz_dbm authz_default > authz_groupfile authz_host authz_owner authz_user autoindex cache cgi%* > cgid%* dav dav_fs dav_lock deflate dir disk_cache env expires ext_filter > file_cache filter headers include info log_config logio mem_cache mime > mime_magic negotiation rewrite setenvif speling status unique_id userdir > usertrack vhost_alias" CALLIGRA_FEATURES="kexi words flow plan stage > tables krita karbon braindump" CAMERAS="ptp2" COLLECTD_PLUGINS="df > interface irq load memory rrdtool swap syslog" ELIBC="glibc" > GPSD_PROTOCOLS="ashtech aivdm earthmate evermore fv18 garmin garmintxt > gpsclock itrax mtk3301 nmea ntrip navcom oceanserver oldstyle oncore > rtcm104v2 rtcm104v3 sirf superstar2 timing tsip tripmate tnt ubx" > INPUT_DEVICES="evdev synaptics" KERNEL="linux" LCD_DEVICES="bayrad > cfontz cfontz633 glk hd44780 lb216 lcdm001 mtxorb ncurses text" > LINGUAS="de en ar" PHP_TARGETS="php5-3" RUBY_TARGETS="ruby18" > USERLAND="GNU" VIDEO_CARDS="nvidia nv" XTABLES_ADDONS="quota2 psd pknock > lscan length2 ipv4options ipset ipp2p iface geoip fuzzy condition tee > tarpit sysrq steal rawnat logmark ipmark dhcpmac delude chaos account" > Unset: CPPFLAGS, CTARGET, INSTALL_MASK, LC_ALL, > PORTAGE_BUNZIP2_COMMAND, PORTAGE_COMPRESS, PORTAGE_COMPRESS_FLAGS, > PORTAGE_RSYNC_EXTRA_OPTS > > ================================================================= > Package Settings > ================================================================= > > net-libs/opal-3.6.8-r2 was built with the following: > USE="audio celt dtmf fax ffmpeg h224 h281 h323 iax ipv6 java ldap > (multilib) plugins sip sipim srtp ssl theora video wav x264 xml -capi > -debug -doc -examples -ilbc -ivr -ixj -lid -sbc -static-libs -stats > -swig -vpb -vxml -x264-static" > CFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden > -D__STDC_CONSTANT_MACROS" > CXXFLAGS="-march=core2 -O3 -pipe -fno-visibility-inlines-hidden > -D__STDC_CONSTANT_MACROS" > > > net-libs/ptlib-2.6.7-r1 was built with the following: > USE="alsa asn audio dtmf ffmpeg http ipv6 jabber ldap (multilib) sasl > sdl soap ssl stun v4l video vxml wav xml xmlrpc -debug -doc (-esd) > -examples -ftp -mail -odbc -oss -pch -qos -remote -serial -shmvideo > -snmp -socks -static-libs -telnet -tts" > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/f8c0ae53/attachment-0001.html From captdeaf at gmail.com Sun Jan 29 02:24:05 2012 From: captdeaf at gmail.com (Greg Millam) Date: Sat, 28 Jan 2012 15:24:05 -0800 Subject: [Freeswitch-users] mod_fsv and h263 Message-ID: Hi folks - I'm using mod_fsv to record incoming video to an .fsv file, then using ffmpeg to convert it. It works perfectly fine when the video codec is h264 or h263+. But when the video codec is h263, it creates corrupted video. ffmpeg creates a gibberish movie, and reports a large number of errors. (Pasted below). However, when play_fsv is used to play that same .fsv file back via freeswitch to my videophone, the video is clear and fine. The issue appears to be in mod_fsv's recording, fsvdec, or similar. ffmpeg is compiled with fsvdec.c Output when using h264 or h263+ is typical ffmpeg success output, and the generated .mp4 is nice and clear. Warning output from h263 spans a large number of pages, but are variations on: [h263 @ 0x3143200] illegal ac vlc code at 1x1 [h263 @ 0x3143200] Error at MB: 24 [h263 @ 0x3143200] illegal ac vlc code at 14x1 [h263 @ 0x3143200] Error at MB: 37 [h263 @ 0x3143200] concealing 396 DC, 396 AC, 396 MV errors [h263 @ 0x3143200] illegal ac vlc code at 21x15 [h263 @ 0x3143200] Error at MB: 366 [h263 @ 0x3143200] concealing 118 DC, 118 AC, 118 MV errors [h263 @ 0x3143200] illegal dc 0 at 9 16 Last message repeated 1 times ... mod_fsv documentation on the wiki states that mod_fsv is endian dependent, yet all processing is happening on the same machine. (And the fact it works just fine for h264 and h263+ is just baffling). Any thoughts? Thank you! - Greg Millam -- If we do not endeavor to take a step every day, the journey can become so long as to be infinite. From nicholas at hellohunter.com Sun Jan 29 02:49:38 2012 From: nicholas at hellohunter.com (Nicholas Blasgen) Date: Sat, 28 Jan 2012 15:49:38 -0800 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: Message-ID: But back to the main point of your question, yes, you should be able to find many providers who will offer you quite a few simultaneous outbound or inbound channels. There are too many to list so I'll just point you to VoIP-Info: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+B2B Nicholas Blasgen Developer Predictive Dialer Limited +1 (724) 252-7436 (cell) Skype: nblasgen www.hellohunter.com: Providing wholesale and white labeled predictive dialing and voice broadcasting services. On Sat, Jan 28, 2012 at 12:06 PM, Josue Diaz Cruz wrote: > ** > Is there any way to have a SIP Trunk. I mean to have for example 32 > channels merged in one? or something like this? When i try to find SIP > trunk on internet i just see options for TDM gateway or similar but not > really a multiplexed trunk. > > Can we do something with freeswitch? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120128/a380fa74/attachment.html From jcdmacleod at me.com Sun Jan 29 21:43:42 2012 From: jcdmacleod at me.com (John Macleod) Date: Sun, 29 Jan 2012 19:43:42 +0100 Subject: [Freeswitch-users] Inbound ANI Message-ID: I have a toll-free number coming into FreeSwitch via SIP. In the log, I see the ANI of the caller (I have sanitised this to protect the innocent): 2012-01-29 10:23:04.968117 [INFO] mod_dialplan_xml.c:481 Processing unknown <0000000000>->18001231234 in context public 2012-01-29 10:23:04.968117 [INFO] switch_core_session.c:1332 sofia/external/614xxxxxxx at x.x.x.x setting session heartbeat to 60 second(s). 2012-01-29 10:23:05.008105 [INFO] switch_core_session.c:2135 Sending early media 2012-01-29 10:23:05.008105 [NOTICE] mod_sofia.c:2576 Pre-Answer sofia/external/614xxxxxxx at x.x.x.x! 2012-01-29 10:23:07.328098 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/614xxxxxxx at x.x.x.x] has been answered 2012-01-29 10:23:12.108110 [NOTICE] mod_dptools.c:1121 Hangup sofia/external/614xxxxxxx at x.x.x.x [CS_RESET] [NORMAL_CLEARING] 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1398 Session 7 (sofia/external/614xxxxxxx at x.x.x.x) Ended 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/614xxxxxxx at x.x.x.x [CS_DESTROY] This is just a Plivo app that plays a greeting and disconnects. However, in the CDRs, I do not get the ANI, I get 0000000000 and the DNIS, I would like the ANI and DNIS. "unknown","0000000000","18001231234","public","2012-01-29 10:23:04","2012-01-29 10:23:07","2012-01-29 10:23:12","8","5","NORMAL_CLEARING","49209654-4aa6-11e1-ac59-9949a61da724","","","PCMU","PCMU" Is there a parameter I am missing to pass this variable from sofia? Thanks, John From all.eforums at gmail.com Mon Jan 30 07:46:01 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Sun, 29 Jan 2012 23:46:01 -0500 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: References: Message-ID: On Sun, Jan 29, 2012 at 2:14 AM, Seven Du wrote: > skype clients should run with FS on the same box. > > You could run both FS and skype clients on another box just as a gateway > if you want scale. We had run multi- FS instances on one box for > scale/stable purpose, but you can get the idea from > http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > Thanks Seven. I had a good read of that testimonial and it's basically saying what you just said here in a line. The question now is are the design docs for that setup proprietary or kinda sorta can be GPL'ed or "shared" in some form? :) If not, then just to clarify, the FS-Skype servers load their own mod_skypopen and is just a gateway to send calls to once it's determined by the FS in the center acting like a soft-switch/call-control platform? Is it completely possible to run/build an FS instance with just mod_skyopen or mod_dingaling (and anything else that's fundamentally needed by FS) loaded? I mean I could Google this but it might help if you could tell me what your modules.conf.xml looks like for Skype only instance? Thanks so much in advance, and apologies if I'm being unreasonable in requesting for the info :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120129/9e60fffa/attachment.html From paul at iamfine.com Mon Jan 30 11:16:02 2012 From: paul at iamfine.com (Paul) Date: Mon, 30 Jan 2012 00:16:02 -0800 (PST) Subject: [Freeswitch-users] mod_xml_cdr not saving file correctly -- PLease help Message-ID: <1327911362882-7235914.post@n2.nabble.com> I posted recently but my description may have been to complex for others to replicate, so I have boiled the problem down to the simplest of configurations and still need some help Running version is FreeSWITCH Version 1.0.head (git-f25c5aa 2012-01-29 17-37-56 -0600) Calls are answered with a LUA script that has been working well on another server 1) any call coming in on a softphone (with caller id of 1000) that is registered to the server will post to url and save correctly to the log directory 2)Calls with a 10 digit US caller_id coming in on public dialplan to same script receive an error 500 when posting the cdr using mod_xml_cdr and post a garbled file in the errors directory (see below) 3) calls coming in on public dialplan with a UK caller id post to url and save correctly encode is set to true in the mod_xml conf file It appears that there may be 2 issues here 1) mod_xml_cdr (with encode=true) is trying to send file (and save in error log with bad formatting) 2) when it writes the file to the main log directory it is saved correctly I end up with a failed post and 2 cdr's that are saved differently. see examples below THIS IS THE FILE IN THE MAIN LOG DIRECTORY /usr/local/freeswitch/log/xml_cdr/a_f4eb00ab-a364-4c95-a86d-df6eafa3bf98.cdr.xml CS_REPORTING inbound 11 0=1;1=1;35=1;36=1;38=1;41=1;51=1 1=1;2=1;3=1;4=1;5=1;6=1 inbound f4eb00ab-a364-4c95-a86d-df6eafa3bf98 AND THIS IS THEY WAY THE FILE IS SAVED IN THE ERROR LOG %3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3Ef4eb00ab-a364-4c95-a86d-df6eafa3bf98%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E6%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri%3E4088924027%254072.249.14.242%3C/sip_from_uri%3E%0A%20%20%20%20%3Csip_from_host%3E72.249.14.242%3C/sip_from_host%3E%0A%20%20%20%20%3Csip_from_user_stripped%3E4088924027%3C/sip_from_user_stripped%3E%0A%20%20%20%20%3Csofia_profile_name%3Eexternal%3C/sofia_profile_name%3E%0A%20%20%20%20%3Csip_req_user%3E8132007296%3C/sip_req_user%3E%0A%20%20%20%20%3Csip_req_port%3E5080%3C/sip_req_port%3E%0A%20%20%20%20%3Csip_req_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_req_uri%3E%0A%20%20%20%20%3Csip_req_host%3E192.168.100.82%3C/sip_req_host%3E%0A%20%20%20%20%3Csip_to_user%3E8132007296%3C/sip_to_user%3E%0A%20%20%20%20%3Csip_to_port%3E5080%3C/sip_to_port%3E%0A%20%20%20%20%3Csip_to_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_to_uri%3E%0A%20%20%20%20%3Csip_to_host%3E192.168.100.82%3C/sip_to_host%3E%0A%20%20%20%20%3Csip_contact_user%3E4088924027%3C/sip_contact_user%3E%0A%20%20%20%20%3Csip_contact_uri%3E4088924027%254072.249.14.242%3C/sip_contact_uri%3E%0A%20%20%20%20%3Csip_contact_host%3E72.249.14.242%3C/sip_contact_host%3E%0A%20%20%20%20% -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-correctly-PLease-help-tp7235914p7235914.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Mon Jan 30 12:26:00 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 30 Jan 2012 10:26:00 +0100 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: References: <4F201657.9040005@softnet.si> <4F22571A.9010009@softnet.si> Message-ID: <4F266228.9050008@softnet.si> Hi @Vitalie, I guess this will do it, but I just have one questione. As call FW number is not in destination_number variable I need a help how to make a bridge with a contact variable (Contact: ;reg-id=1) as in this variblae CFW number is hidden on which call should transfer. Thanks! Miha On 1/27/2012 9:24 AM, Vitalie Colosov wrote: > And when you say "phone is set to make call FW" - are you calling to > this phone A from another phone which is connected to FS? (lets say > phone B), and Phone A is doing SIP redirect to the external number > which is set by you in the Phone A's settings? > > If so, I assume the call will enter FS, and then it depends of your > dialplan... You can setup it in the way that if the number does not > exist, or based on some prefix, it should bridge to external number > via sofia/external/386$1 at SBC_IP, but this should be done explicitly, > FS does not know that you want to forward it to the outside world... > > For example: > > > expression="false"> > data="sofia/external/386${destination_number}@SBC_IP"/> > > > > Let me know if this bring more light to your issue. > > Vitalie > > > 2012/1/26 Miha Zoubek > > > Hi @Vitalie, > > what is here..? here is trunk to sbc with registration false > option (without password, just ip). Outside call I am doing like > this : > > > > Thanks! > > > > On 1/27/2012 8:08 AM, Vitalie Colosov wrote: >> Could you please elaborate more on your topology... >> >> Sip phone A (1001) --> FS --> what is here..? Where do you want >> it to call? Do you have an external profile with a gateway >> configured? >> >> >> 2012/1/25 Miha Zoubek > >> >> Hi, >> >> I have phone registered on FS and phone is set to make call FW. I >> noticed that FS do not make a call FW the external number but >> looks on >> FS if the phone is registered on FS (because the calling >> number is not >> on FS the call is being rejected). >> How to tell FS that it should make a call fw to a number that >> is not on FS. >> >> Thanks! >> >> Regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/5f28286c/attachment-0001.html From brian at neotiq.com Mon Jan 30 12:29:00 2012 From: brian at neotiq.com (brian at neotiq.com) Date: Mon, 30 Jan 2012 10:29:00 +0100 (CET) Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> <007c01ccdd15$d2354560$769fd020$@neotiq.com> Message-ID: <2012372837.35999.1327915740886.JavaMail.open-xchange@email.1and1.fr> Hi Avi, I'm testing on a Digium TDM400P card plugged directly to FreeSwitch Server. I know that this feature is supported by the driver because it works with Asterisk. Brian Le 28 janvier 2012 ? 17:27, Avi Marcus a ?crit : > Brian, your phones are plugged directly into an FXS card in your FreeSWITCH > server? What kind of card is it? This is something that the specific > hardware/driver has to support, knowing which one you are using will be > essential to giving you an answer.. > > -Avi > > > On Fri, Jan 27, 2012 at 11:16 PM, Ognjen Seslija wrote: > > > You can do this easy in Linksys/Cisco phones/ATAs with dialplan > > manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk > > ext which does the IVR or whatever. > > > > > > On Fri, Jan 27, 2012 at 6:05 PM, Brian wrote: > > > >> Hi Saugort,**** > >> > >> ** ** > >> > >> I?m trying to do the first one. When someone take the handset off-hook, > >> before he has pressed any key, play some messages to him. That can be used > >> to play some warning if the outgoing SIP trunk is down, for example. That > >> feature is called batPhoneby some people because it is seen in a film about Batman > >> J**** > >> > >> ** ** > >> > >> Regards,**** > >> > >> Brian**** > >> > >> ** ** > >> > >> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > >> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort > >> Dario Garcia Tovar > >> *Envoy? :* vendredi 27 janvier 2012 17:23 > >> > >> *? :* freeswitch-users at lists.freeswitch.org > >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* > >> *** > >> > >> ** ** > >> > >> Hi Brian, > >> > >> Could you explain a little bit more what your want do do? > >> > >> Are you trying to do this: > >> the analogic phone is a FS extension, connected to an ATA device (or it > >> is connected to FXS card in FS server?). When someone take the handset, > >> automatically connect to another extension, make pstn call or play a > >> message? > >> > >> Or your are trying to do this: > >> An incoming call from a PSTN line connected to a TDM card in your FS > >> server, you want to automatically connect to another extension, make pstn > >> call or play a message? > >> > >> On 1/27/2012 10:49 AM, Brian wrote: **** > >> > >> Hi Avi,**** > >> > >> **** > >> > >> Maybe that?s the term I should use, hot-dial. I?m talking about analogic > >> phones, not VoIP phones. If it?s an analogic phone plugged into an FXS > >> socket, when you pick up the phone, the PBX is capable of detecting the > >> off-hook event. That feature must be supported by the analogic card?s > >> driver. In asterisk?s case, that is the option named ?immediate=true|false? > >> in the Zapata.conf file, as stated here: > >> http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf**** > >> > >> **** > >> > >> Best regards,**** > >> > >> Brian**** > >> > >> **** > >> > >> *De :* freeswitch-users-bounces at lists.freeswitch.org [ > >> mailto:freeswitch-users-bounces at lists.freeswitch.org] > >> *De la part de* Avi Marcus > >> *Envoy? :* vendredi 27 janvier 2012 15:42 > >> *? :* FreeSWITCH Users Help > >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* > >> *** > >> > >> **** > >> > >> I think you want a hot-dial..? Some phones/ATAs support it. I don't think > >> it's an FS/Asterisk feature.**** > >> > >> e.g. most phones don't even contact the sip server until the person > >> "finished dialing".**** > >> > >> > >> **** > >> > >> -Avi**** > >> > >> > >> > >> > >> **** > >> > >> On Fri, Jan 27, 2012 at 4:38 PM, Brian wrote:**** > >> > >> Hi Saugort,**** > >> > >> **** > >> > >> Thank you for these links. However that is not really what I?m looking > >> for. What I am trying to do is to play a sound file to an analogic phone as > >> soon as the later goes off-hook. What I called ?eary media? is not really > >> the ringtone as we often see in SIP protocol. Sorry for the confusion.*** > >> * > >> > >> **** > >> > >> Any help would be greatly appreciated,**** > >> > >> Regards,**** > >> > >> Brian**** > >> > >> **** > >> > >> *De :* freeswitch-users-bounces at lists.freeswitch.org [mailto: > >> freeswitch-users-bounces at lists.freeswitch.org] *De la part de* Saugort > >> Dario Garcia Tovar > >> *Envoy? :* vendredi 27 janvier 2012 13:42 > >> *? :* freeswitch-users at lists.freeswitch.org > >> *Objet :* Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk* > >> *** > >> > >> **** > >> > >> Hi Brian, > >> > >> Take a look to this links: > >> > >> http://wiki.freeswitch.org/wiki/Early_Media > >> > >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready > >> > >> > >> http://freeswitch-users.2379917.n2.nabble.com/Waiting-for-playing-a-file-as-early-media-td2460205.html > >> > >> > >> http://freeswitch-users.2379917.n2.nabble.com/Early-Media-troubles-td5479581.html > >> > >> > >> > >> On 1/26/2012 5:45 PM, Brian wrote: **** > >> > >> Hi List,**** > >> > >> **** > >> > >> I?m quite new to FreeSwitch and I?m trying to replace an old asterisk > >> installation with a new FreeSwitch. There is however one feature that is > >> available with Asterisk and I haven?t managed to figure it out how to do it > >> with FreeSwitch. That is about the ?s? extension in Asterisk dial plan. In > >> the old Asterisk installation, when I pick up an analogic phone (plugged > >> into an FXS slot of a Digium card) the ?s? extension gets run and I can > >> play early media to the phone. Does anyone know if it is possible to > >> achieve the same thing with FreeSwitch/FreeTDM? **** > >> > >> **** > >> > >> Best regards,**** > >> > >> **** > >> > >> Brian**** > >> > >> **** > >> > >> > >> > >> > >> **** > >> > >> _________________________________________________________________________**** > >> > >> Professional FreeSWITCH Consulting Services:**** > >> > >> consulting at freeswitch.org**** > >> > >> http://www.freeswitchsolutions.com**** > >> > >> **** > >> > >> **** > >> > >> **** > >> > >> **** > >> > >> Official FreeSWITCH Sites**** > >> > >> http://www.freeswitch.org**** > >> > >> http://wiki.freeswitch.org**** > >> > >> http://www.cluecon.com**** > >> > >> **** > >> > >> FreeSWITCH-users mailing list**** > >> > >> FreeSWITCH-users at lists.freeswitch.org**** > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > >> > >> http://www.freeswitch.org**** > >> > >> > >> > >> > >> **** > >> > >> No virus found in this message. > >> Checked by AVG - www.avg.com > >> Version: 2012.0.1901 / Virus Database: 2109/4766 - Release Date: 01/25/12 > >> **** > >> > >> **** > >> > >> -- > >> Atentamente, > >> *Dario Garc?a* > >> Consultor. > >> > >> CCCT, Nivel C2, Sector Yarey, Mz, > >> Ofc. MZ03a. > >> Caracas-Venezuela. > >> Tel?fono: +58 212 9081842 > >> Cel: +58 412 2221515 > >> dgarcia at anew.com.ve > >> http://www.anew.com.ve**** > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org**** > >> > >> **** > >> > >> > >> > >> > >> **** > >> > >> _________________________________________________________________________**** > >> > >> Professional FreeSWITCH Consulting Services:**** > >> > >> consulting at freeswitch.org**** > >> > >> http://www.freeswitchsolutions.com**** > >> > >> ** ** > >> > >> **** > >> > >> **** > >> > >> ** ** > >> > >> Official FreeSWITCH Sites**** > >> > >> http://www.freeswitch.org**** > >> > >> http://wiki.freeswitch.org**** > >> > >> http://www.cluecon.com**** > >> > >> ** ** > >> > >> FreeSWITCH-users mailing list**** > >> > >> FreeSWITCH-users at lists.freeswitch.org**** > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > >> > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > >> > >> http://www.freeswitch.org**** > >> > >> > >> > >> > >> **** > >> > >> No virus found in this message. > >> Checked by AVG - www.avg.com > >> Version: 2012.0.1901 / Virus Database: 2109/4768 - Release Date: 01/26/12 > >> **** > >> > >> ** ** > >> > >> -- > >> Atentamente, > >> *Dario Garc?a* > >> Consultor. > >> > >> CCCT, Nivel C2, Sector Yarey, Mz, > >> Ofc. MZ03a. > >> Caracas-Venezuela. > >> Tel?fono: +58 212 9081842 > >> Cel: +58 412 2221515 > >> dgarcia at anew.com.ve > >> http://www.anew.com.ve**** > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/d52f87f1/attachment-0001.html From chris at ghosttelecom.com Mon Jan 30 13:55:02 2012 From: chris at ghosttelecom.com (Chris Martineau) Date: Mon, 30 Jan 2012 10:55:02 -0000 Subject: [Freeswitch-users] sofia require: timer missing from 200 ok Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> Hi, There seems to be something broken between git versions (f2cf68b 2011-11-20 18-40-41 -0500) and (58c3c3a 2011-11-22 18-22-57 -0600) in that the first works fine and issues require headers where necessary, but the second does not. They have duplicate configs so can only assume something is broken between the 2 versions. Remember reading something about this but cannot find it again. Is this a known issue? Is it still possible to pull down the older git? Many thanks Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/e049f6a5/attachment.html From freeswitch at earthspike.net Mon Jan 30 14:17:12 2012 From: freeswitch at earthspike.net (John) Date: Mon, 30 Jan 2012 11:17:12 +0000 Subject: [Freeswitch-users] sofia require: timer missing from 200 ok In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> References: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> Message-ID: <4F267C38.5070405@earthspike.net> Chris, Sounds like you need to raise a bug on jira.freeswitch.org. Sorry, I don't know how to use an older git. Someone else may be able to help with that, or you could try http://progit.org. John On 30/01/12 10:55, Chris Martineau wrote: > > > > *Hi,* > > * * > > *There seems to be something broken between git versions (f2cf68b > 2011-11-20 18-40-41 -0500) and (58c3c3a 2011-11-22 18-22-57 -0600) in > that the first works fine and issues require headers where necessary, > but the second does not. They have duplicate configs so can only > assume something is broken between the 2 versions. Remember reading > something about this but cannot find it again.* > > * * > > *Is this a known issue?* > > * * > > *Is it still possible to pull down the older git?* > > * * > > *Many thanks* > > * * > > *Chris* > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/32259abf/attachment.html From dujinfang at gmail.com Mon Jan 30 17:00:19 2012 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Jan 2012 22:00:19 +0800 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: References: Message-ID: <7FB3A2EEA03E4DEB8D617B131C6FB9A7@gmail.com> Yes, exactly, you could remove any unnecessary modules if you don't use them. to be clear, you could send calls from FS1 to FS2 and FS3 give you have proper dialplan set to route calls to the desired destination. that means you could just treat FS2 and FS3 as a third party sip gateway but they can routing calls to anywhere you want. FS1 (mod_sofia) ---------------------- (mod_sofia)FS2(mod_skypopen) -------------------skype \ \------------------ (mod_sofia)FS3(mod_dingaling) ---------------------gtalk That testimonial is completely following the wiki's sharing rule, but you can use the idea completely freely. On Monday, January 30, 2012 at 12:46 PM, A E [Gmail] wrote: > On Sun, Jan 29, 2012 at 2:14 AM, Seven Du wrote: > > skype clients should run with FS on the same box. > > > > You could run both FS and skype clients on another box just as a gateway if you want scale. We had run multi- FS instances on one box for scale/stable purpose, but you can get the idea from http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com > > > > -- > > About: http://about.me/dujinfang > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > Thanks Seven. I had a good read of that testimonial and it's basically saying what you just said here in a line. The question now is are the design docs for that setup proprietary or kinda sorta can be GPL'ed or "shared" in some form? :) > > If not, then just to clarify, the FS-Skype servers load their own mod_skypopen and is just a gateway to send calls to once it's determined by the FS in the center acting like a soft-switch/call-control platform? Is it completely possible to run/build an FS instance with just mod_skyopen or mod_dingaling (and anything else that's fundamentally needed by FS) loaded? I mean I could Google this but it might help if you could tell me what your modules.conf.xml looks like for Skype only instance? > > Thanks so much in advance, and apologies if I'm being unreasonable in requesting for the info :) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/93446a14/attachment.html From anthony.minessale at gmail.com Mon Jan 30 19:01:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 10:01:22 -0600 Subject: [Freeswitch-users] mod_xml_cdr not saving file correctly -- PLease help In-Reply-To: <1327911362882-7235914.post@n2.nabble.com> References: <1327911362882-7235914.post@n2.nabble.com> Message-ID: have you looked in the error log of the web server or checked the cgi handler to see what the problem is? The data you pasted looks identical to me. The encoded format is urlencoded: See this command like that decodes it: echo "%3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3Ef4eb00ab-a364-4c95-a86d-df6eafa3bf98%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E6%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri%3E4088924027%254072.249.14.242%3C/sip_from_uri%3E%0A%20%20%20%20%3Csip_from_host%3E72.249.14.242%3C/sip_from_host%3E%0A%20%20%20%20%3Csip_from_user_stripped%3E4088924027%3C/sip_from_user_stripped%3E%0A%20%20%20%20%3Csofia_profile_name%3Eexternal%3C/sofia_profile_name%3E%0A%20%20%20%20%3Csip_req_user%3E8132007296%3C/sip_req_user%3E%0A%20%20%20%20%3Csip_req_port%3E5080%3C/sip_req_port%3E%0A%20%20%20%20%3Csip_req_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_req_uri%3E%0A%20%20%20%20%3Csip_req_host%3E192.168.100.82%3C/sip_req_host%3E%0A%20%20%20%20%3Csip_to_user%3E8132007296%3C/sip_to_user%3E%0A%20%20%20%20%3Csip_to_port%3E5080%3C/sip_to_port%3E%0A%20%20%20%20%3Csip_to_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_to_uri%3E%0A%20%20%20%20%3Csip_to_host%3E192.168.100.82%3C/sip_to_host%3E%0A%20%20%20%20%3Csip_contact_user%3E4088924027%3C/sip_contact_user%3E%0A%20%20%20%20%3Csip_contact_uri%3E4088924027%254072.249.14.242%3C/sip_contact_uri%3E%0A%20%20%20%20%3Csip_contact_host%3E72.249.14.242%3C/sip_contact_host%3E%0A%20%20%20%20%" | perl -ne 's/%(..)/chr hex $1/eg; print ' CS_REPORTING inbound 11 0=1;1=1;35=1;36=1;38=1;41=1;51=1 1=1;2=1;3=1;4=1;5=1;6=1 inbound f4eb00ab-a364-4c95-a86d-df6eafa3bf98 6 192.168.100.82 72.249.14.242 5060 72.249.14.242 5060 udp 4088924027 4088924027%4072.249.14.242 72.249.14.242 4088924027 external 8132007296 5080 8132007296%40192.168.100.82%3A5080 192.168.100.82 8132007296 5080 8132007296%40192.168.100.82%3A5080 192.168.100.82 4088924027 4088924027%4072.249.14.242 72.249.14.242 % On Mon, Jan 30, 2012 at 2:16 AM, Paul wrote: > I posted recently but my description may have been to complex for others to > replicate, so I have boiled the problem down to the simplest of > configurations and still need some help > > Running version is FreeSWITCH Version 1.0.head (git-f25c5aa 2012-01-29 > 17-37-56 -0600) > > Calls are answered with a LUA script that has been working well on another > server > 1) any call coming in on a softphone (with caller id of 1000) that is > registered to the server will post to url and save correctly to the log > directory > 2)Calls with a 10 digit US caller_id coming in on public dialplan to same > script receive an error 500 when posting the cdr using mod_xml_cdr and post > a garbled file in the errors directory (see below) > 3) calls coming in on public dialplan with a UK caller id post to url and > save correctly > > encode is set to true in the mod_xml conf file > > It appears that there may be 2 issues here > 1) mod_xml_cdr (with encode=true) is trying to send file (and save in error > log with bad formatting) > 2) when it writes the file to the main log directory it is saved correctly > > I end up with a failed post and 2 cdr's that are saved differently. > > > see examples below > THIS IS THE FILE IN THE MAIN LOG DIRECTORY > /usr/local/freeswitch/log/xml_cdr/a_f4eb00ab-a364-4c95-a86d-df6eafa3bf98.cdr.xml > > > ? > ? ?CS_REPORTING > ? ?inbound > ? ?11 > ? ?0=1;1=1;35=1;36=1;38=1;41=1;51=1 > ? ?1=1;2=1;3=1;4=1;5=1;6=1 > ? > ? > ? ?inbound > ? ?f4eb00ab-a364-4c95-a86d-df6eafa3bf98 > > AND THIS IS THEY WAY THE FILE IS SAVED IN THE ERROR LOG > > %3C%3Fxml%20version%3D%221.0%22%3F%3E%0A%3Ccdr%3E%0A%20%20%3Cchannel_data%3E%0A%20%20%20%20%3Cstate%3ECS_REPORTING%3C/state%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cstate_number%3E11%3C/state_number%3E%0A%20%20%20%20%3Cflags%3E0%3D1%3B1%3D1%3B35%3D1%3B36%3D1%3B38%3D1%3B41%3D1%3B51%3D1%3C/flags%3E%0A%20%20%20%20%3Ccaps%3E1%3D1%3B2%3D1%3B3%3D1%3B4%3D1%3B5%3D1%3B6%3D1%3C/caps%3E%0A%20%20%3C/channel_data%3E%0A%20%20%3Cvariables%3E%0A%20%20%20%20%3Cdirection%3Einbound%3C/direction%3E%0A%20%20%20%20%3Cuuid%3Ef4eb00ab-a364-4c95-a86d-df6eafa3bf98%3C/uuid%3E%0A%20%20%20%20%3Csession_id%3E6%3C/session_id%3E%0A%20%20%20%20%3Csip_local_network_addr%3E192.168.100.82%3C/sip_local_network_addr%3E%0A%20%20%20%20%3Csip_network_ip%3E72.249.14.242%3C/sip_network_ip%3E%0A%20%20%20%20%3Csip_network_port%3E5060%3C/sip_network_port%3E%0A%20%20%20%20%3Csip_received_ip%3E72.249.14.242%3C/sip_received_ip%3E%0A%20%20%20%20%3Csip_received_port%3E5060%3C/sip_received_port%3E%0A%20%20%20%20%3Csip_via_protocol%3Eudp%3C/sip_via_protocol%3E%0A%20%20%20%20%3Csip_from_user%3E4088924027%3C/sip_from_user%3E%0A%20%20%20%20%3Csip_from_uri%3E4088924027%254072.249.14.242%3C/sip_from_uri%3E%0A%20%20%20%20%3Csip_from_host%3E72.249.14.242%3C/sip_from_host%3E%0A%20%20%20%20%3Csip_from_user_stripped%3E4088924027%3C/sip_from_user_stripped%3E%0A%20%20%20%20%3Csofia_profile_name%3Eexternal%3C/sofia_profile_name%3E%0A%20%20%20%20%3Csip_req_user%3E8132007296%3C/sip_req_user%3E%0A%20%20%20%20%3Csip_req_port%3E5080%3C/sip_req_port%3E%0A%20%20%20%20%3Csip_req_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_req_uri%3E%0A%20%20%20%20%3Csip_req_host%3E192.168.100.82%3C/sip_req_host%3E%0A%20%20%20%20%3Csip_to_user%3E8132007296%3C/sip_to_user%3E%0A%20%20%20%20%3Csip_to_port%3E5080%3C/sip_to_port%3E%0A%20%20%20%20%3Csip_to_uri%3E8132007296%2540192.168.100.82%253A5080%3C/sip_to_uri%3E%0A%20%20%20%20%3Csip_to_host%3E192.168.100.82%3C/sip_to_host%3E%0A%20%20%20%20%3Csip_contact_user%3E4088924027%3C/sip_contact_user%3E%0A%20%20%20%20%3Csip_contact_uri%3E4088924027%254072.249.14.242%3C/sip_contact_uri%3E%0A%20%20%20%20%3Csip_contact_host%3E72.249.14.242%3C/sip_contact_host%3E%0A%20%20%20%20% > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-correctly-PLease-help-tp7235914p7235914.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mytemike72 at gmail.com Mon Jan 30 19:40:56 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 17:40:56 +0100 Subject: [Freeswitch-users] Method of checking if media is being played at a session Message-ID: Hi All, I am dealing with a little problem... To play asynchronous audio in my Lua script I use the api function "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio is being buffered if I call my function multiple times. When I want to abort all plays and flush the buffer I use "uuid_break {uuid} all". This all works excellent! My problem is I at certain points need to know if audio is being played (or awaits in the buffer) and need to pause execution untill all audio is beging played (or flushed). Is there a way of doing this in either Lua or apiFunctions? or a channel variable perhaps? I have looked and tryed but cannot seem to find a way doing this. Any hints (or solution ;-) appreciated! Regards, Michael Lutz From hynek.cihlar at gmail.com Mon Jan 30 19:56:01 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Mon, 30 Jan 2012 17:56:01 +0100 Subject: [Freeswitch-users] Method of checking if media is being played at a session In-Reply-To: References: Message-ID: Although I do it using ESL it may be helpful for your case. When issuing the broadcast request I create a 'correlation ID' variable with a generated unique ID (can be any suitable string) for each file to be broadcast. The ID is passed back to the ESL app after each particular file stops playing back. In the ESl language I issue the command "execute playback file_string://myfile.mp3{CorrelationID=myuniqueID}" and then wait for an event with the correct correlation ID. Hynek On Mon, Jan 30, 2012 at 5:40 PM, Michael Lutz wrote: > Hi All, > > I am dealing with a little problem... > > To play asynchronous audio in my Lua script I use the api function > "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio > is being buffered if I call my function multiple times. > When I want to abort all plays and flush the buffer I use "uuid_break > {uuid} all". This all works excellent! > > My problem is I at certain points need to know if audio is being > played (or awaits in the buffer) and need to pause execution untill > all audio is beging played (or flushed). > > Is there a way of doing this in either Lua or apiFunctions? or a > channel variable perhaps? I have looked and tryed but cannot seem to > find a way doing this. > > > Any hints (or solution ;-) appreciated! > > > Regards, > Michael Lutz > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/ab53af6d/attachment.html From msc at freeswitch.org Mon Jan 30 20:09:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 09:09:41 -0800 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital Message-ID: Hello FreeSWITCH Community, We wanted to let everyone know that Brian West is in St. Francis Hospital, Tulsa, OK. He is checked into room 4325. At this point it looks like he will be there for at least a week. I think it's safe to say that Brian is a dynamo not just for FreeSWITCH but for open source VoIP and telephony. My guess is that there's not a person who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't been aided in one way or another by Brian. Personally I can say that in the past five years Brian has helped me immeasurably. I think I have literally asked him thousands of questions and he has patiently answered them all. We would like to have everyone keep Brian in their thoughts and prayers. Please feel free to send him cards and well wishes at the hospital. Also, if you wish to send gifts and such to his home address you may do so: Brian West 714 Osage Ave McAlester, OK 74501-6638 For those who are interested in helping in more practical ways, such as helping to defray the considerable medical costs that no doubt will be incurred, we will have more information shortly. Please spread the word via Facebook, Twitter, and any other medium you have. Let's all let Brian know just how much we value him as a colleague and friend! -Michael Collins -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/7e822025/attachment.html From nbhatti at gmail.com Mon Jan 30 20:24:07 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 30 Jan 2012 20:24:07 +0300 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: This is sad new, what happened to him? On Mon, Jan 30, 2012 at 8:09 PM, Michael Collins wrote: > Hello FreeSWITCH Community, > > We wanted to let everyone know that Brian West is in St. Francis Hospital, > Tulsa, OK. He is checked into room 4325. At this point it looks like he will > be there for at least a week. > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH but > for open source VoIP and telephony. My guess is that there's not a person > who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't > been aided in one way or another by Brian. Personally I can say that in the > past five years Brian has helped me immeasurably. I think I have literally > asked him thousands of questions and he has patiently answered them all. > > We would like to have everyone keep Brian in their thoughts and prayers. > Please feel free to send him cards and well wishes at the hospital. Also, if > you wish to send gifts and such to his home address you may do so: > > Brian West > 714 Osage Ave > McAlester, OK > 74501-6638 > > For those who are interested in helping in more practical ways, such as > helping to defray the considerable medical costs that no doubt will be > incurred, we will have more information shortly. > > Please spread the word via Facebook, Twitter, and any other medium you have. > Let's all let Brian know just how much we value him as a colleague and > friend! > > -Michael Collins > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From basit.engg at gmail.com Mon Jan 30 20:40:09 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 30 Jan 2012 22:40:09 +0500 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: Brian, wish you good health and long life. -- Regards, Abdul Basit On Mon, Jan 30, 2012 at 10:24 PM, Muhammad Naseer Bhatti wrote: > This is sad new, what happened to him? > > On Mon, Jan 30, 2012 at 8:09 PM, Michael Collins > wrote: > > Hello FreeSWITCH Community, > > > > We wanted to let everyone know that Brian West is in St. Francis > Hospital, > > Tulsa, OK. He is checked into room 4325. At this point it looks like he > will > > be there for at least a week. > > > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH > but > > for open source VoIP and telephony. My guess is that there's not a person > > who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't > > been aided in one way or another by Brian. Personally I can say that in > the > > past five years Brian has helped me immeasurably. I think I have > literally > > asked him thousands of questions and he has patiently answered them all. > > > > We would like to have everyone keep Brian in their thoughts and prayers. > > Please feel free to send him cards and well wishes at the hospital. > Also, if > > you wish to send gifts and such to his home address you may do so: > > > > Brian West > > 714 Osage Ave > > McAlester, OK > > 74501-6638 > > > > For those who are interested in helping in more practical ways, such as > > helping to defray the considerable medical costs that no doubt will be > > incurred, we will have more information shortly. > > > > Please spread the word via Facebook, Twitter, and any other medium you > have. > > Let's all let Brian know just how much we value him as a colleague and > > friend! > > > > -Michael Collins > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/e08d34dc/attachment.html From shaheryarkh at googlemail.com Mon Jan 30 20:42:55 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 30 Jan 2012 22:42:55 +0500 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: Brian please don't die at least till we have human cloning possible and easily available everywhere. :P On serious note, May you get well soon, amen. Thank you. On Mon, Jan 30, 2012 at 10:24 PM, Muhammad Naseer Bhatti wrote: > This is sad new, what happened to him? > > On Mon, Jan 30, 2012 at 8:09 PM, Michael Collins > wrote: > > Hello FreeSWITCH Community, > > > > We wanted to let everyone know that Brian West is in St. Francis > Hospital, > > Tulsa, OK. He is checked into room 4325. At this point it looks like he > will > > be there for at least a week. > > > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH > but > > for open source VoIP and telephony. My guess is that there's not a person > > who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't > > been aided in one way or another by Brian. Personally I can say that in > the > > past five years Brian has helped me immeasurably. I think I have > literally > > asked him thousands of questions and he has patiently answered them all. > > > > We would like to have everyone keep Brian in their thoughts and prayers. > > Please feel free to send him cards and well wishes at the hospital. > Also, if > > you wish to send gifts and such to his home address you may do so: > > > > Brian West > > 714 Osage Ave > > McAlester, OK > > 74501-6638 > > > > For those who are interested in helping in more practical ways, such as > > helping to defray the considerable medical costs that no doubt will be > > incurred, we will have more information shortly. > > > > Please spread the word via Facebook, Twitter, and any other medium you > have. > > Let's all let Brian know just how much we value him as a colleague and > > friend! > > > > -Michael Collins > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/b57f7d36/attachment-0001.html From curriegrad2004 at gmail.com Mon Jan 30 20:43:36 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 30 Jan 2012 09:43:36 -0800 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: Turns out he's got a tumor in his heart and lungs. Hope he's going to be okay out of this. On 2012-01-30 9:25 AM, "Muhammad Naseer Bhatti" wrote: > This is sad new, what happened to him? > > On Mon, Jan 30, 2012 at 8:09 PM, Michael Collins > wrote: > > Hello FreeSWITCH Community, > > > > We wanted to let everyone know that Brian West is in St. Francis > Hospital, > > Tulsa, OK. He is checked into room 4325. At this point it looks like he > will > > be there for at least a week. > > > > I think it's safe to say that Brian is a dynamo not just for FreeSWITCH > but > > for open source VoIP and telephony. My guess is that there's not a person > > who's worked with Asterisk or FreeSWITCH in the past 10 years who hasn't > > been aided in one way or another by Brian. Personally I can say that in > the > > past five years Brian has helped me immeasurably. I think I have > literally > > asked him thousands of questions and he has patiently answered them all. > > > > We would like to have everyone keep Brian in their thoughts and prayers. > > Please feel free to send him cards and well wishes at the hospital. > Also, if > > you wish to send gifts and such to his home address you may do so: > > > > Brian West > > 714 Osage Ave > > McAlester, OK > > 74501-6638 > > > > For those who are interested in helping in more practical ways, such as > > helping to defray the considerable medical costs that no doubt will be > > incurred, we will have more information shortly. > > > > Please spread the word via Facebook, Twitter, and any other medium you > have. > > Let's all let Brian know just how much we value him as a colleague and > > friend! > > > > -Michael Collins > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/951a4a82/attachment.html From massimiliano.ravelli at gmail.com Mon Jan 30 20:45:25 2012 From: massimiliano.ravelli at gmail.com (Massimiliano Ravelli) Date: Mon, 30 Jan 2012 18:45:25 +0100 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: 2012/1/30 Michael Collins For those who are interested in helping in more practical ways, such as > helping to defray the considerable medical costs that no doubt will be > incurred, we will have more information shortly. > Let us know ! Massimiliano -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/5db96992/attachment.html From aksrini at hotmail.com Mon Jan 30 20:53:35 2012 From: aksrini at hotmail.com (Srini K) Date: Mon, 30 Jan 2012 09:53:35 -0800 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: References: Message-ID: HiHow to execute this command "sofia profile external rescan reloadxml" to reload the gateway using the mod_managed application without using cli. Thanks in advance. Regards Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/b87513dd/attachment.html From Prometheus001 at gmx.net Mon Jan 30 21:01:46 2012 From: Prometheus001 at gmx.net (Peter P GMX) Date: Mon, 30 Jan 2012 19:01:46 +0100 Subject: [Freeswitch-users] Not recognizing voice in mod_pocketsphinx In-Reply-To: References: Message-ID: <4F26DB0A.8070306@gmx.net> I am picking up this old thread as I have exactly the same problem and my log looks just the same. I am using mod_pocketsphing through event socket with a new freeswitch built from GIT. I also tested the Pizza demo with no success. The server is a Ubuntu 10_04. The module is loaded successfully: 2012-01-30 17:26:08.776531 [CONSOLE] switch_loadable_module.c:1299 Successfully Loaded [mod_pocketsphinx] 2012-01-30 17:26:08.776531 [NOTICE] switch_loadable_module.c:363 Adding ASR interface 'pocketsphinx' However it does neither detect speech via event socket nor in the pizza demo. And I cannot see any DETECTED_SPEECH events via event socket. In the logs I see the following: 2012-01-30 18:52:26.536529 [DEBUG] switch_ivr_play_say.c:1678 done playing file /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-Greeting.wav EXECUTE sofia/internal/200 at my.domain detect_speech(pocketsphinx pizza_order undefined) 2012-01-30 18:52:27.056538 [DEBUG] switch_core_media_bug.c:457 Attaching BUG to sofia/internal/200 at my.domain 2012-01-30 18:52:27.076532 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMA:8 EXECUTE sofia/internal/200 at my.domain detect_speech(resume) 2012-01-30 18:52:27.576530 [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16 at 8000hz 1 channels 20ms 2012-01-30 18:52:30.296528 [DEBUG] switch_ivr_play_say.c:1678 done playing file /usr/local/freeswitch/sounds/en/us/callie/pizza/GP-DeliveryorTakeout.wav EXECUTE sofia/internal/200 at my.domain detect_speech(resume) So there is no error message. Googling around I expected I should see something like this: SpeechTools.jm:150 console_log() ----XML: TAKEOUT TAKEOUT But there is none of this type in the logs. What could I do to investigate further? * maybe there is an older version of pocketsphinx on the system, which conflicts? (system was set up in may 2011) * Is there a way to have more debugging informations within pocket sphinx? Any help is appreciated. Best regards Peter Am 06.09.2010 09:50, schrieb Thangappan.M: > Dear all, > > I have installed the mod_pocketsphinx module successfully and > enabled with FreeSWITCH. Copied the relevant files for pizza > application and placed into appropriate directory. Made a call to > "74992" and able to hear the voice. > When I said "TAKEOUT' it was not recognizing my words. > > In the freeswitch CLI it shown the following messages. > > EXECUTE sofia/internal/1000 at 192.168.1.222 > javascript(ps_pizza.js) > 2010-09-06 13:18:48.023673 [DEBUG] sofia_glue.c:2594 AUDIO RTP > [sofia/internal/1000 at 192.168.1.222 ] > 192.168.1.222 port 32590 -> 192.168.6.60 port 8000 codec: 8 ms: 20 > 2010-09-06 13:18:48.024705 [DEBUG] switch_rtp.c:1182 Starting timer > [soft] 160 bytes per 20ms > 2010-09-06 13:18:48.025710 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf > send payload to 101 > 2010-09-06 13:18:48.026799 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf > receive payload to 101 > 2010-09-06 13:18:48.026799 [DEBUG] mod_sofia.c:636 Local SDP > sofia/internal/1000 at 192.168.1.222 : > v=0 > o=FreeSWITCH 1283726738 1283726739 IN IP4 192.168.1.222 > s=FreeSWITCH > c=IN IP4 192.168.1.222 > t=0 0 > m=audio 32590 RTP/AVP 8 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > > 2010-09-06 13:18:48.028235 [DEBUG] switch_core_session.c:641 Send > signal sofia/internal/1000 at 192.168.1.222 > [BREAK] > 2010-09-06 13:18:48.028235 [DEBUG] sofia.c:4153 Channel > sofia/internal/1000 at 192.168.1.222 entering > state [completed][200] > 2010-09-06 13:18:48.029256 [NOTICE] mod_spidermonkey.c:2066 Channel > [sofia/internal/1000 at 192.168.1.222 ] has > been answered > 2010-09-06 13:18:48.030269 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-09-06 13:18:48.123555 [DEBUG] sofia.c:4153 Channel > sofia/internal/1000 at 192.168.1.222 entering > state [ready][200] > 2010-09-06 13:18:48.171705 [DEBUG] switch_rtp.c:2066 Correct ip/port > confirmed. > 2010-09-06 13:18:50.212417 [DEBUG] switch_ivr_play_say.c:1444 done > playing file > EXECUTE sofia/internal/1000 at 192.168.1.222 > detect_speech(pocketsphinx pizza_order undefined) > 2010-09-06 13:18:51.768030 [DEBUG] switch_core_media_bug.c:360 > Attaching BUG to sofia/internal/1000 at 192.168.1.222 > > EXECUTE sofia/internal/1000 at 192.168.1.222 > detect_speech(resume) > 2010-09-06 13:18:51.774074 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > 2010-09-06 13:18:54.492023 [DEBUG] switch_ivr_play_say.c:1444 done > playing file > EXECUTE sofia/internal/1000 at 192.168.1.222 > detect_speech(resume) > EXECUTE sofia/internal/1000 at 192.168.1.222 > detect_speech(resume) > 2010-09-06 13:19:01.534474 [DEBUG] switch_ivr_play_say.c:1152 Codec > Activated L16 at 8000hz 1 channels 20ms > > > Please help me..... > > -- > Regards, > Thangappan.M > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/b410b7b1/attachment-0001.html From all.eforums at gmail.com Mon Jan 30 21:20:04 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Mon, 30 Jan 2012 13:20:04 -0500 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: <7FB3A2EEA03E4DEB8D617B131C6FB9A7@gmail.com> References: <7FB3A2EEA03E4DEB8D617B131C6FB9A7@gmail.com> Message-ID: Ok cool. Thank you so much. I found this as some very slimmed down modules.conf.xml figuring FS running on an embedded platform might give me the best case from which to build on. It's here: https://dev.openwrt.org/browser/packages/net/freeswitch/files/etc.minimal/autoload_configs/modules.conf.xml?order=date Apart from adding mod_skypopen to this, does this look good? What else needs to be added to it? Thanks so much for your help :) On Mon, Jan 30, 2012 at 9:00 AM, Seven Du wrote: > Yes, exactly, you could remove any unnecessary modules if you don't use > them. > > to be clear, you could send calls from FS1 to FS2 and FS3 give you have > proper dialplan set to route calls to the desired destination. > that means you could just treat FS2 and FS3 as a third party sip gateway > but they can routing calls to anywhere you want. > > FS1 (mod_sofia) ---------------------- (mod_sofia)FS2(mod_skypopen) > -------------------skype > \ > \------------------ > (mod_sofia)FS3(mod_dingaling) ---------------------gtalk > > That testimonial is completely following the wiki's sharing rule, but you > can use the idea completely freely. > > On Monday, January 30, 2012 at 12:46 PM, A E [Gmail] wrote: > > On Sun, Jan 29, 2012 at 2:14 AM, Seven Du wrote: > > skype clients should run with FS on the same box. > > You could run both FS and skype clients on another box just as a gateway > if you want scale. We had run multi- FS instances on one box for > scale/stable purpose, but you can get the idea from > http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > Thanks Seven. I had a good read of that testimonial and it's basically > saying what you just said here in a line. The question now is are the > design docs for that setup proprietary or kinda sorta can be GPL'ed or > "shared" in some form? :) > > If not, then just to clarify, the FS-Skype servers load their own > mod_skypopen and is just a gateway to send calls to once it's determined by > the FS in the center acting like a soft-switch/call-control platform? Is it > completely possible to run/build an FS instance with just mod_skyopen or > mod_dingaling (and anything else that's fundamentally needed by FS) loaded? > I mean I could Google this but it might help if you could tell me what your > modules.conf.xml looks like for Skype only instance? > > Thanks so much in advance, and apologies if I'm being unreasonable in > requesting for the info :) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/b5ac4e28/attachment.html From mytemike72 at gmail.com Mon Jan 30 21:48:07 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 19:48:07 +0100 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: References: Message-ID: Hi Srini, You should be able to do it using the API. Like: FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string result = fsApi.ExecuteString("sofia profile external rescan reloadxml"); Regards, Michael Lutz. 2012/1/30 Srini K : > > Hi > How to execute this command "sofia profile external rescan reloadxml" to > reload the gateway using the mod_managed application without using cli. > Thanks in advance. > > Regards > Srini > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From paul at iamfine.com Mon Jan 30 21:54:14 2012 From: paul at iamfine.com (Paul) Date: Mon, 30 Jan 2012 10:54:14 -0800 (PST) Subject: [Freeswitch-users] mod_xml_cdr not saving file correctly -- Please help In-Reply-To: References: <1327911362882-7235914.post@n2.nabble.com> Message-ID: <1327949654239-7237617.post@n2.nabble.com> Yes we did check that that receiving web server it confirms that one of the cdr's posts correctly and the other fails with a 500 error This is what we see in CLI - that mod_xml_cdr can post when the call is not a US 10 digit caller id and Fails when it is a 10 digit caller id here are the 2 items from access.log on apache2 and there are no errors in the error.log IP ADDRESS REMOVED - - [30/Jan/2012:18:42:07 +0000] "POST /index.php/fs/cdr_save?uuid=a_325804a1-6703-4566-b0f1-899ea89dd983 HTTP/1.1" 200 609 "-" "freeswitch-xml/1.0" IP ADDRESS REMOVED - - [30/Jan/2012:18:51:12 +0000] "POST /index.php/fs/cdr_save?uuid=a_8525a70a-1674-4bb6-8cf0-07bf2aa37638 HTTP/1.1" 500 2777 "-" "freeswitch-xml/1.0" I ran tcpdump on the incoming server and i see that the data is coming in from freeswitch but in one case it is encoded and the other it is not (i think) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-correctly-PLease-help-tp7235914p7237617.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Mon Jan 30 21:57:11 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 12:57:11 -0600 Subject: [Freeswitch-users] mod_xml_cdr not saving file correctly -- Please help In-Reply-To: <1327949654239-7237617.post@n2.nabble.com> References: <1327911362882-7235914.post@n2.nabble.com> <1327949654239-7237617.post@n2.nabble.com> Message-ID: What I was asking, was if you can get more details. a 500 error on your server means your CGI died or errored out. So if you can find the exact error from the webserver error log you can see what happens in the script to make it fail. So you should be concentrating on the web server and CGI not on FS. On Mon, Jan 30, 2012 at 12:54 PM, Paul wrote: > Yes we did check that that receiving web server > it confirms that one of the cdr's posts correctly > and the other fails with a 500 error > > This is what we see in CLI - that mod_xml_cdr can post when the call is not > a US 10 digit caller id > and Fails when it is a 10 digit caller id > > here are the 2 items from access.log on apache2 > and there are no errors in the error.log > > IP ADDRESS REMOVED - - [30/Jan/2012:18:42:07 +0000] "POST > /index.php/fs/cdr_save?uuid=a_325804a1-6703-4566-b0f1-899ea89dd983 HTTP/1.1" > 200 609 "-" "freeswitch-xml/1.0" > IP ADDRESS REMOVED - - [30/Jan/2012:18:51:12 +0000] "POST > /index.php/fs/cdr_save?uuid=a_8525a70a-1674-4bb6-8cf0-07bf2aa37638 HTTP/1.1" > 500 2777 "-" "freeswitch-xml/1.0" > > I ran tcpdump on the incoming server and i see that the data is coming in > from freeswitch but in one case it is encoded and the other it is not (i > think) > > > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-xml-cdr-not-saving-file-correctly-PLease-help-tp7235914p7237617.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From paul at cupis.co.uk Mon Jan 30 22:26:57 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Mon, 30 Jan 2012 19:26:57 +0000 Subject: [Freeswitch-users] sofia require: timer missing from 200 ok In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> References: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> Message-ID: <4F26EF01.1000109@cupis.co.uk> On 30/01/12 10:55, Chris Martineau wrote: > There seems to be something broken between git versions (f2cf68b > 2011-11-20 18-40-41 -0500) and (58c3c3a 2011-11-22 18-22-57 -0600) in > that the first works fine and issues require headers where necessary, > but the second does not. They have duplicate configs so can only assume > something is broken between the 2 versions. Remember reading something > about this but cannot find it again. Possibly related to this? git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 commit 58c3c3a049991fedd39f62008f8eb8fca047e7c5 Author: Anthony Minessale Date: Tue Nov 22 18:22:57 2011 -0600 comment out optional Require header from re-invites for the sake of interop Mentioned in this thread from earlier in January: http://lists.freeswitch.org/pipermail/freeswitch-users/2012-January/079492.html > Is it still possible to pull down the older git? You can get any version from any point - they are all stored within the git repositary. To get the version immediately before the above patch (if that is the one that it causing you the issue): git checkout dc9bf68301118b0e70ab707d2543bbcd96709196 Regards, From anthony.minessale at gmail.com Mon Jan 30 22:49:13 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 13:49:13 -0600 Subject: [Freeswitch-users] sofia require: timer missing from 200 ok In-Reply-To: <4F26EF01.1000109@cupis.co.uk> References: <1D10AB188D6CCA46BB4369E3268E36EF458291@SVR01.ghosttelecom.local> <4F26EF01.1000109@cupis.co.uk> Message-ID: or git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 | patch -p1 -R to reverse that one change on newer build. On Mon, Jan 30, 2012 at 1:26 PM, Paul Cupis wrote: > On 30/01/12 10:55, Chris Martineau wrote: >> There seems to be something broken between ?git versions (f2cf68b >> 2011-11-20 18-40-41 -0500) and (58c3c3a 2011-11-22 18-22-57 -0600) in >> that the first works fine and issues require headers where necessary, >> but the second does not. They have duplicate configs so can only assume >> something is broken between the 2 versions. Remember reading something >> about this but cannot find it again. > > Possibly related to this? > > git log -1 -p 58c3c3a049991fedd39f62008f8eb8fca047e7c5 > > commit 58c3c3a049991fedd39f62008f8eb8fca047e7c5 > Author: Anthony Minessale > Date: ? Tue Nov 22 18:22:57 2011 -0600 > > ? ? comment out optional Require header from re-invites for the sake of > interop > > > Mentioned in this thread from earlier in January: > > http://lists.freeswitch.org/pipermail/freeswitch-users/2012-January/079492.html > >> Is it still possible to pull down the older git? > > You can get any version from any point - they are all stored within the > git repositary. > > To get the version immediately before the above patch (if that is the > one that it causing you the issue): > > git checkout dc9bf68301118b0e70ab707d2543bbcd96709196 > > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Mon Jan 30 23:33:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 12:33:02 -0800 Subject: [Freeswitch-users] UPDATE: Donating To Assist Brian West Message-ID: Good news, We've contacted PayPal and made sure that they are aware of the fact that Brian may be receive a number of donations. Sometimes this can be construed as "suspicious activity" so it's good to be prepared. PayPal has requested that we use payment type "Personal" and that the money is being sent as a "gift." Here's a quick screenshot: Let's all give Brian some PayPal love! Thanks, Michael P.S. - If you have any questions please email me off list. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/d185c999/attachment-0001.html From msc at freeswitch.org Mon Jan 30 23:34:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jan 2012 12:34:57 -0800 Subject: [Freeswitch-users] UPDATE: Donating To Assist Brian West In-Reply-To: References: Message-ID: I forgot the most important part! brian at bkw.org Use that email address to send your donations. -MC On Mon, Jan 30, 2012 at 12:33 PM, Michael Collins wrote: > Good news, > > We've contacted PayPal and made sure that they are aware of the fact that > Brian may be receive a number of donations. Sometimes this can be construed > as "suspicious activity" so it's good to be prepared. PayPal has requested > that we use payment type "Personal" and that the money is being sent as a > "gift." Here's a quick screenshot: > > > > Let's all give Brian some PayPal love! > > Thanks, > Michael > > P.S. - If you have any questions please email me off list. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/0361249d/attachment.html From anthony.minessale at gmail.com Mon Jan 30 23:47:54 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 14:47:54 -0600 Subject: [Freeswitch-users] mod_fsv and h263 In-Reply-To: References: Message-ID: Maybe there are some specific encoding params from the fmtp that are not preserved or there are some other factors that the phone takes for granted so it's probably a co-incidence it works on the phone since fsv is basically a mirror of an exact rtp stream and probably it's already equipped to play its own video. You could play with the mod_mp4 and see if its any different. On Sat, Jan 28, 2012 at 5:24 PM, Greg Millam wrote: > Hi folks - > > ?I'm using mod_fsv to record incoming video to an .fsv file, then > using ffmpeg to convert it. It works perfectly fine when the video > codec is h264 or h263+. But when the video codec is h263, it creates > corrupted video. ffmpeg creates a gibberish movie, and reports a large > number of errors. (Pasted below). > > ?However, when play_fsv is used to play that same .fsv file back via > freeswitch to my videophone, the video is clear and fine. The issue > appears to be in mod_fsv's recording, fsvdec, or similar. > > ffmpeg is compiled with fsvdec.c > > Output when using h264 or h263+ is typical ffmpeg success output, and > the generated .mp4 is nice and clear. > > Warning output from h263 spans a large number of pages, but are variations on: > > [h263 @ 0x3143200] illegal ac vlc code at 1x1 > [h263 @ 0x3143200] Error at MB: 24 > [h263 @ 0x3143200] illegal ac vlc code at 14x1 > [h263 @ 0x3143200] Error at MB: 37 > [h263 @ 0x3143200] concealing 396 DC, 396 AC, 396 MV errors > [h263 @ 0x3143200] illegal ac vlc code at 21x15 > [h263 @ 0x3143200] Error at MB: 366 > [h263 @ 0x3143200] concealing 118 DC, 118 AC, 118 MV errors > [h263 @ 0x3143200] illegal dc 0 at 9 16 > ? ?Last message repeated 1 times > > ... > > ?mod_fsv documentation on the wiki states that mod_fsv is endian > dependent, yet all processing is happening on the same machine. (And > the fact it works just fine for h264 and h263+ is just baffling). > > Any thoughts? > > Thank you! > > - Greg Millam > > -- > If we do not endeavor to take a step every day, the journey can become > so long as to be infinite. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From potxoka at gmail.com Tue Jan 31 00:21:04 2012 From: potxoka at gmail.com (Anto) Date: Mon, 30 Jan 2012 22:21:04 +0100 Subject: [Freeswitch-users] ZRTP Message-ID: Hi I tried to install new machines in a ZRTP with FreeSWITCH, but I have not found the ZRTP library. I haven?t found FreeSWITCH repositories, nor on the website of Zfone (http://zfoneproject.com). I do not like much the licensing (or mode of use) of this library, so searching I found these two projects: http://code.google.com/p/open-zrtp/ and http://www.gnutelephony.org/index.php/GNU_ZRTP. Can be used in FreeSWITCH? Thanks Best regards From krice at freeswitch.org Tue Jan 31 00:24:19 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 30 Jan 2012 15:24:19 -0600 Subject: [Freeswitch-users] ZRTP In-Reply-To: Message-ID: Anto, I cant say much yet... But stand by for some more info coming soon on FreeSWITCH and ZRTP support K On 1/30/12 3:21 PM, "Anto" wrote: > Hi > > I tried to install new machines in a ZRTP with FreeSWITCH, but I have > not found the ZRTP library. I haven?t found FreeSWITCH repositories, > nor on the website of Zfone (http://zfoneproject.com). I do not like > much the licensing (or mode of use) of this library, so searching I > found these two projects: http://code.google.com/p/open-zrtp/ and > http://www.gnutelephony.org/index.php/GNU_ZRTP. Can be used in > FreeSWITCH? Thanks > > Best regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fraserredmond at gmail.com Tue Jan 31 00:51:20 2012 From: fraserredmond at gmail.com (Fraser Redmond) Date: Mon, 30 Jan 2012 16:51:20 -0500 Subject: [Freeswitch-users] Whats the best way to insert audio into a call from event socket In-Reply-To: <8532801A6C8A46D1B832FCE1271EB111@gmail.com> References: <8532801A6C8A46D1B832FCE1271EB111@gmail.com> Message-ID: Thanks. So in general, is uuid_displace a better choice than uuid_broadcast? Cheers, Fraser On 29 January 2012 02:09, Seven Du wrote: > You can chain files with file_string:// so it should work with > uuid_displace http://wiki.freeswitch.org/wiki/Mod_file_string > > > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > On Sunday, January 29, 2012 at 4:17 AM, Fraser Redmond wrote: > > I want to insert a queue of voice recordings into a call, and be able to > stop them all. Doing both via the event socket. > > Currently I'm doing: > api uuid_broadcast $uuid fileName1.wav both > api uuid_broadcast $uuid fileName2.wav both > > And stopping the audio with: > api uuid_break $uuid all > > That works quite well, except for two problems: > 1) I can't use it because when the audio is playing the parties can't hear > each other talk. > 2) There is a slight delay of about half a second before the audio starts > (though I'm guessing this can't be helped.) > > Previously I'd used: > api uuid_displace $uuid start fileName1.wav 0 mux > > But I don't think I'd been able to queue up more than one file with that. > > There's a few other possibilities that I've thought of, including going > into a js/lua script and looping inside that, or changing the call into a > conference, and using the conference commands. > > Rather than mess around trying lots of ideas, I'd rather focus on the one > best approach - any ideas? > > Cheers, > Fraser > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/e81053a4/attachment.html From mytemike72 at gmail.com Tue Jan 31 00:54:39 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 22:54:39 +0100 Subject: [Freeswitch-users] Method of checking if media is being played at a session In-Reply-To: References: Message-ID: Hi Hynek (or anyone else ;-) Thank you for your response. I see you also use the ESL to invoke the playback? Is there a particulair for that other than I am using the UUID_BROADCAST using the api? I am trying to make it work through the ESL but it seems the initial events are not received directly after the the broadcast, but it takes a while. I will paste some example code of what I have done/am trying to do. Any help apreciated! In this example I have created a couple of functions (explained in code) and try t odo an async playback of an audio file, wait for the ESL to confirm the PLAYBACK_START and then call another function which waits for all PLAYBACK_STOP's (sum of plays) to return. Then I do session:hangup. But it never gets the ESL event. In theory this should work. But in some way the events do not come in right away. This code is testable. (only specify a valid filename in the streamFile) Code: http://pastebin.com/ri0L7wmx Regards, Michael Lutz. 2012/1/30 Hynek Cihlar : > Although I do it using ESL it may be helpful for your case. When issuing the > broadcast request I create a 'correlation ID' variable with a generated > unique ID (can be any suitable string) for each file to be broadcast. The ID > is passed back to the ESL app after each particular file stops playing back. > > In the ESl language I issue the command "execute > playback?file_string://myfile.mp3{CorrelationID=myuniqueID}" and then wait > for an event with the correct correlation ID. > > Hynek > > > > On Mon, Jan 30, 2012 at 5:40 PM, Michael Lutz wrote: >> >> Hi All, >> >> I am dealing with a little problem... >> >> To play asynchronous audio in my Lua script I use the api function >> "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio >> is being buffered if I call my function multiple times. >> When I want to abort all plays and flush the buffer I use "uuid_break >> {uuid} all". This all works excellent! >> >> My problem is I at certain points need to know if audio is being >> played (or awaits in the buffer) and need to pause execution untill >> all audio is beging played (or flushed). >> >> Is there a way of doing this in either Lua or apiFunctions? or a >> channel variable perhaps? I have looked and tryed but cannot seem to >> find a way doing this. >> >> >> Any hints (or solution ;-) appreciated! >> >> >> Regards, >> Michael Lutz >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Jan 31 01:12:16 2012 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Jan 2012 17:12:16 -0500 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: References: Message-ID: <4F9FB14B-3490-4103-9569-CFC51123158E@jerris.com> you would run that like any other fsapi command. On Jan 27, 2012, at 5:01 PM, Srini K wrote: > Hi, > I like to add a gateway on the fly. FS applcation which is in mod_managed will read the gateway configuration from the db and will add/update the conf\sip_profiles\external\MygateWay.xml config file. > How to execute this command "sofia profile external rescan reloadxml" to reload the gateway using the mod_managed application without using cli. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/a058efb6/attachment.html From mike at jerris.com Tue Jan 31 01:15:23 2012 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Jan 2012 17:15:23 -0500 Subject: [Freeswitch-users] Extending the field size of core's db_data table? In-Reply-To: References: <81FE1555B2164D17B698567AA1FF788B@gmail.com> Message-ID: <3C9DCFDC-A4FB-49D3-8103-FC1368E2EB35@jerris.com> I have not audited the code in that app, but easiest way to tell for sure is to just give it a try, I suspect there is no built in limit, but if you find one, let us know. MIke On Jan 29, 2012, at 6:56 AM, Yehavi Bourvine wrote: > I know I can do ALTER TABLE, but I've asked what is the implication about FreeSWITCH software. > > Anyway, I managed to bypass the problem by other means. > > Thanks, __Yehavi: > > > 2012/1/29 Seven Du > I think you could just ALTER TABLE ... or re-create the table maybe sqlite doesn't support alter table? > On Tuesday, December 20, 2011 at 2:35 PM, Yehavi Bourvine wrote: >> Hello, >> >> We use the DB api for storing and caching various data. We need to store data that is longer than 255 characters, which is the current limit on the field size there. >> >> The backend for this API is MySQL (via ODBC). Can I just increase the field size in MySQL, or is there some dependency in FreeSwitch on this size? >> >> Thanks, __Yehavi: >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/bfad55f0/attachment.html From mike at jerris.com Tue Jan 31 01:21:57 2012 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Jan 2012 17:21:57 -0500 Subject: [Freeswitch-users] Inbound ANI In-Reply-To: References: Message-ID: what does the invite packet look like, there are a few different settings for caller id in sip that tell it to look in various places. http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#CallerID_Related_options Mike On Jan 29, 2012, at 1:43 PM, John Macleod wrote: > I have a toll-free number coming into FreeSwitch via SIP. > > In the log, I see the ANI of the caller (I have sanitised this to protect the innocent): > > 2012-01-29 10:23:04.968117 [INFO] mod_dialplan_xml.c:481 Processing unknown <0000000000>->18001231234 in context public > 2012-01-29 10:23:04.968117 [INFO] switch_core_session.c:1332 sofia/external/614xxxxxxx at x.x.x.x setting session heartbeat to 60 second(s). > 2012-01-29 10:23:05.008105 [INFO] switch_core_session.c:2135 Sending early media > 2012-01-29 10:23:05.008105 [NOTICE] mod_sofia.c:2576 Pre-Answer sofia/external/614xxxxxxx at x.x.x.x! > 2012-01-29 10:23:07.328098 [NOTICE] mod_dptools.c:1135 Channel [sofia/external/614xxxxxxx at x.x.x.x] has been answered > 2012-01-29 10:23:12.108110 [NOTICE] mod_dptools.c:1121 Hangup sofia/external/614xxxxxxx at x.x.x.x [CS_RESET] [NORMAL_CLEARING] > 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1398 Session 7 (sofia/external/614xxxxxxx at x.x.x.x) Ended > 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/614xxxxxxx at x.x.x.x [CS_DESTROY] > > This is just a Plivo app that plays a greeting and disconnects. > > However, in the CDRs, I do not get the ANI, I get 0000000000 and the DNIS, I would like the ANI and DNIS. > > "unknown","0000000000","18001231234","public","2012-01-29 10:23:04","2012-01-29 10:23:07","2012-01-29 10:23:12","8","5","NORMAL_CLEARING","49209654-4aa6-11e1-ac59-9949a61da724","","","PCMU","PCMU" > > Is there a parameter I am missing to pass this variable from sofia? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/12591f0a/attachment.html From mytemike72 at gmail.com Tue Jan 31 01:26:13 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 23:26:13 +0100 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: <4F9FB14B-3490-4103-9569-CFC51123158E@jerris.com> References: <4F9FB14B-3490-4103-9569-CFC51123158E@jerris.com> Message-ID: Anybody, I got the impression before my replys don't get to the list. Could someone please confirm seeing this message with a reply? This because I replied with answer and api example 3 hours ago and Mike just responds with answer like mine wasn't there. (and have the feeling other responses never got here) Thanks, Michael Lutz 2012/1/30 Michael Jerris : > you would run that like any other fsapi command. > > On Jan 27, 2012, at 5:01 PM, Srini K wrote: > > Hi, > I like to add a gateway on the fly. FS applcation which is in mod_managed > will read the gateway configuration from the db and will add/update the > conf\sip_profiles\external\MygateWay.xml config file. > > How to execute this command "sofia profile external rescan reloadxml" to > reload the gateway using the mod_managed application without using cli. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mike at jerris.com Tue Jan 31 01:29:19 2012 From: mike at jerris.com (Michael Jerris) Date: Mon, 30 Jan 2012 17:29:19 -0500 Subject: [Freeswitch-users] Equivalent of "s" extension in Asterisk In-Reply-To: <039d01ccde47$551a8f30$ff4fad90$@com> References: <000001ccdc77$f29a28f0$d7ce7ad0$@neotiq.com> <4F229BA0.9030400@anew.com.ve> <005401ccdd01$5a79f6b0$0f6de410$@neotiq.com> <006501ccdd07$18c699c0$4a53cd40$@neotiq.com> <4F22CF79.5090801@anew.com.ve> <007c01ccdd15$d2354560$769fd020$@neotiq.com> <039d01ccde47$551a8f30$ff4fad90$@com> Message-ID: <87928A91-2EF8-47EA-B301-725777FD61E1@jerris.com> in your ftdm config for analog_spans, set the "hotline" param to the extension you want it to call. I don't think I see this anywhere on the wiki, would you mind adding it? Mike On Jan 29, 2012, at 12:32 AM, Bote Man wrote: > It does not matter if the FXS port is a card in the FS server or an analog gateway/ATA. The device to which the analog phone is connected must provide the first sound that is heard (dial tone), or else connect immediately via FS to the source of the desired sound file. > > Some people call this "hot line", old telephone heads in the U.S. call this "ringdown private line", Cisco turns this name around and calls it PLAR (private line automatic ringdown) in their gateway devices. > > With no special processing or commands you will hear the happy sound of the dial tone, as always. I agree with the answer that says to call automatically an IVR application that plays the desired file(s) and processes the dialed digits in response. > > Bote > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus > Sent: Saturday, 28 January, 2012 11:27 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Equivalent of "s" extension in Asterisk > > Brian, your phones are plugged directly into an FXS card in your FreeSWITCH server? What kind of card is it? This is something that the specific hardware/driver has to support, knowing which one you are using will be essential to giving you an answer.. > > -Avi > > > On Fri, Jan 27, 2012 at 11:16 PM, Ognjen Seslija wrote: > You can do this easy in Linksys/Cisco phones/ATAs with dialplan manipulation (i.e. (P0<:numbertocall>)). A number to call is a FS/Asterisk ext which does the IVR or whatever. > > > On Fri, Jan 27, 2012 at 6:05 PM, Brian wrote: > Hi Saugort, > > I?m trying to do the first one. When someone take the handset off-hook, before he has pressed any key, play some messages to him. That can be used to play some warning if the outgoing SIP trunk is down, for example. That feature is called batPhone by some people because it is seen in a film about Batman J > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/8ee7db84/attachment-0001.html From bobc at devassert.com Tue Jan 31 01:30:09 2012 From: bobc at devassert.com (Bob Coleman) Date: Tue, 31 Jan 2012 11:30:09 +1300 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: References: <4F9FB14B-3490-4103-9569-CFC51123158E@jerris.com> Message-ID: Hi Michael, I see your reply, 3 hours before, like you said Bob On Tue, Jan 31, 2012 at 11:26 AM, Michael Lutz wrote: > Anybody, > > I got the impression before my replys don't get to the list. Could > someone please confirm seeing this message with a reply? > This because I replied with answer and api example 3 hours ago and > Mike just responds with answer like mine wasn't there. (and have the > feeling other responses never got here) > > Thanks, > Michael Lutz > > 2012/1/30 Michael Jerris : >> you would run that like any other fsapi command. >> >> On Jan 27, 2012, at 5:01 PM, Srini K wrote: >> >> Hi, >> I like to add a gateway on the fly. FS applcation which is in mod_managed >> will read the gateway configuration from the db and will add/update the >> conf\sip_profiles\external\MygateWay.xml config file. >> >> How to execute this command "sofia profile external rescan reloadxml" to >> reload the gateway using the mod_managed application without using cli. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cjbujold at accra.ca Tue Jan 31 01:31:30 2012 From: cjbujold at accra.ca (Charles Bujold) Date: Mon, 30 Jan 2012 18:31:30 -0400 Subject: [Freeswitch-users] Newbie questions - Long time to connect Message-ID: <019201ccdf9e$e9d503a0$bd7f0ae0$@accra.ca> Installing a new Freeswitch - Ubuntu (AMD Quad FX, 8 GB RAM, Intel Network card ) and I am getting several issues that hopefully somebody can point me in the correct direction. 1) I'm Using VoiceMeUp (Canada) as provider and I keep getting Gateway being Registered, Fail Wait and Trying after the first call is received or outgoing. If I do not use the line it shows Registered. After every call it seems to take several minutes before I can make another call. 2) I cannot seem to be able to make more than 1 call at a time although the service can support 5 calls simultaneously. 3) When making a call I see the connection appear in Freeswitch but it can take over a minute before I hear a ring . Some calls work and other do not. 4) When calling myself using the outside number I can see the call originating and then being received again (almost instantaneous) by Freeswitch but it still takes close to a minute before it starts ringing and when I pick up it hangs up immediately. ( I presume that a time out occurred. My question is what could cause this type of slow response to connect or answer a call. Is there a timer setting that needs to be adjusted? I have a first server similar in configuration and it works properly. I presume that I missed configured something but can't seem to figure out what? Thanks CJB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120130/14a441f5/attachment.html From mytemike72 at gmail.com Tue Jan 31 01:33:30 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 23:33:30 +0100 Subject: [Freeswitch-users] Inbound ANI In-Reply-To: References: Message-ID: Hi John, It looks like it's passing calling_name and not calling_number. From the logs it shows your provider is sending the "000000000" as the calling name. Show channels will also just show you calling_name and not calling_number. If you require the actual number, you should get the right channel variable "calling_number". Regards, Michael Lutz 2012/1/30 Michael Jerris : > what does the invite packet look like, there are a few different settings > for caller id in sip that tell it to look in various places. > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#CallerID_Related_options > > Mike > > On Jan 29, 2012, at 1:43 PM, John Macleod wrote: > > I have a toll-free number coming into FreeSwitch via SIP. > > In the log, I see the ANI of the caller (I have sanitised this to protect > the innocent): > > 2012-01-29 10:23:04.968117 [INFO] mod_dialplan_xml.c:481 Processing unknown > <0000000000>->18001231234 in context public > 2012-01-29 10:23:04.968117 [INFO] switch_core_session.c:1332 > sofia/external/614xxxxxxx at x.x.x.x setting session heartbeat to 60 second(s). > 2012-01-29 10:23:05.008105 [INFO] switch_core_session.c:2135 Sending early > media > 2012-01-29 10:23:05.008105 [NOTICE] mod_sofia.c:2576 Pre-Answer > sofia/external/614xxxxxxx at x.x.x.x! > 2012-01-29 10:23:07.328098 [NOTICE] mod_dptools.c:1135 Channel > [sofia/external/614xxxxxxx at x.x.x.x] has been answered > 2012-01-29 10:23:12.108110 [NOTICE] mod_dptools.c:1121 Hangup > sofia/external/614xxxxxxx at x.x.x.x [CS_RESET] [NORMAL_CLEARING] > 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1398 Session 7 > (sofia/external/614xxxxxxx at x.x.x.x) Ended > 2012-01-29 10:23:12.108110 [NOTICE] switch_core_session.c:1400 Close Channel > sofia/external/614xxxxxxx at x.x.x.x [CS_DESTROY] > > This is just a Plivo app that plays a greeting and disconnects. > > However, in the CDRs, I do not get the ANI, I get 0000000000 and the DNIS, I > would like the ANI and DNIS. > > "unknown","0000000000","18001231234","public","2012-01-29 > 10:23:04","2012-01-29 10:23:07","2012-01-29 > 10:23:12","8","5","NORMAL_CLEARING","49209654-4aa6-11e1-ac59-9949a61da724","","","PCMU","PCMU" > > Is there a parameter I am missing to pass this variable from sofia? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Jan 31 01:34:41 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jan 2012 16:34:41 -0600 Subject: [Freeswitch-users] Extending the field size of core's db_data table? In-Reply-To: <3C9DCFDC-A4FB-49D3-8103-FC1368E2EB35@jerris.com> References: <81FE1555B2164D17B698567AA1FF788B@gmail.com> <3C9DCFDC-A4FB-49D3-8103-FC1368E2EB35@jerris.com> Message-ID: you can probably safely change it to a TEXT field. SQLite ignores col lens internally so it only matters to your db of choice. On Mon, Jan 30, 2012 at 4:15 PM, Michael Jerris wrote: > I have not audited the code in that app, but easiest way to tell for sure is > to just give it a try, I suspect there is no built in limit, but if you find > one, let us know. > > MIke > > On Jan 29, 2012, at 6:56 AM, Yehavi Bourvine wrote: > > I know I can do ALTER TABLE, but I've asked what is the implication about > FreeSWITCH software. > > Anyway, I managed to bypass the problem by other means. > > ??????????????????????? Thanks, __Yehavi: > > > 2012/1/29 Seven Du >> >> I think you could just ALTER TABLE ... or re-create the table maybe sqlite >> doesn't support alter table? >> On Tuesday, December 20, 2011 at 2:35 PM, Yehavi Bourvine wrote: >> >> Hello, >> >> ? We use the DB api for storing and caching various data. We need to store >> data that is longer than 255 characters, which is the current limit on the >> field size there. >> >> The backend for this API is MySQL (via ODBC). Can I just increase the >> field size in MySQL, or is there some dependency in FreeSwitch on this size? >> >> ??????????????????????????????? Thanks, __Yehavi: >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mytemike72 at gmail.com Tue Jan 31 01:35:20 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 30 Jan 2012 23:35:20 +0100 Subject: [Freeswitch-users] How to reload the gateway from the mod_managed In-Reply-To: References: <4F9FB14B-3490-4103-9569-CFC51123158E@jerris.com> Message-ID: Great, Thanks for the confirmation Bob, Sorry to have spoiled the thread. But Srini's question has been answerred anyway.. ;-) Michael. 2012/1/30 Bob Coleman : > Hi Michael, > > I see your reply, 3 hours before, like you said > > Bob > > On Tue, Jan 31, 2012 at 11:26 AM, Michael Lutz wrote: >> Anybody, >> >> I got the impression before my replys don't get to the list. Could >> someone please confirm seeing this message with a reply? >> This because I replied with answer and api example 3 hours ago and >> Mike just responds with answer like mine wasn't there. (and have the >> feeling other responses never got here) >> >> Thanks, >> Michael Lutz >> >> 2012/1/30 Michael Jerris : >>> you would run that like any other fsapi command. >>> >>> On Jan 27, 2012, at 5:01 PM, Srini K wrote: >>> >>> Hi, >>> I like to add a gateway on the fly. FS applcation which is in mod_managed >>> will read the gateway configuration from the db and will add/update the >>> conf\sip_profiles\external\MygateWay.xml config file. >>> >>> How to execute this command "sofia profile external rescan reloadxml" to >>> reload the gateway using the mod_managed application without using cli. >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From david.villasmil.work at gmail.com Tue Jan 31 01:45:48 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 30 Jan 2012 23:45:48 +0100 Subject: [Freeswitch-users] how to find out the error? Message-ID: Hello Guys, When I try to start FS I get the following: 2012-01-30 22:41:36.184340 [INFO] switch_event.c:631 Activate Eventing Engine. 2012-01-30 22:41:36.203503 [DEBUG] switch_event.c:610 Create event dispatch thread 0 Cannot Initialize [[error near line 46]: unclosed FS---->1002(zoiper peer) using spandsp module. My actual test scenario is,I dial 1002 from 1001 with selecting some.tiff file,and fax should be send to 1002.but this scenario is not working in zoiper. I also try another scenario.By dialing 555 and try to set some T.38 variable and then bridge the call to 1002 for sending fax. but this scenario is also not working. pls help me resolved my problem soon. Thanks Amit Nakum -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/7bc977cc/attachment.html From gcd at i.ph Tue Jan 31 09:26:32 2012 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 31 Jan 2012 14:26:32 +0800 Subject: [Freeswitch-users] unable to sending and receiving fax between two peer In-Reply-To: References: Message-ID: hi amit, use txfax application instead of bridge. ...... however, don't dial 555 to send a fax. instead, dial 1002 first. when you make a connection, then make a blind transfer to extension 555. hope it works. -nandy On Tue, Jan 31, 2012 at 2:08 PM, amit nakum wrote: > Dear Friends > > > Pls Help me...... > > I am unable to send and receiving fax directly between two zoiper peer > i.e 1001(zoiper peer) --->FS---->1002(zoiper peer) using spandsp module. > > My actual test scenario is,I dial 1002 from 1001 with selecting some.tiff > file,and fax should be send to 1002.but this scenario is not working in > zoiper. > > I also try another scenario.By dialing 555 and try to set some T.38 > variable and then bridge the call to 1002 for sending fax. > > > > > > > > > > > > but this scenario is also not working. > > > pls help me resolved my problem soon. > > Thanks > > Amit Nakum > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/f6bdc1f5/attachment.html From engineerzuhairraza at gmail.com Tue Jan 31 10:22:34 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 31 Jan 2012 11:22:34 +0400 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: Get well soon Brian, Regards, Zohair Raza On Mon, Jan 30, 2012 at 9:45 PM, Massimiliano Ravelli < massimiliano.ravelli at gmail.com> wrote: > 2012/1/30 Michael Collins > > For those who are interested in helping in more practical ways, such as >> helping to defray the considerable medical costs that no doubt will be >> incurred, we will have more information shortly. >> > > Let us know ! > > Massimiliano > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/2f34854a/attachment.html From brian at freeswitch.org Tue Jan 31 12:34:43 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 31 Jan 2012 03:34:43 -0600 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: Message-ID: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> Thanks everyone for your kind words. I'm hanging in there trying to keep a level head about everything and have a positive outlook on everything. Thanks, Brian On Jan 31, 2012, at 1:22 AM, Zohair Raza wrote: > Get well soon Brian, > > Regards, > Zohair Raza > > On Mon, Jan 30, 2012 at 9:45 PM, Massimiliano Ravelli < > massimiliano.ravelli at gmail.com> wrote: > >> 2012/1/30 Michael Collins >> >> For those who are interested in helping in more practical ways, such as >>> helping to defray the considerable medical costs that no doubt will be >>> incurred, we will have more information shortly. >>> >> >> Let us know ! >> >> Massimiliano >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/bc2340fa/attachment-0001.html From gabe at gundy.org Tue Jan 31 12:39:57 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 31 Jan 2012 02:39:57 -0700 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> References: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> Message-ID: On Tue, Jan 31, 2012 at 2:34 AM, Brian West wrote: > Thanks everyone for your kind words. ?I'm hanging in there trying to keep a > level head about everything and have a positive outlook on everything. Just know that there are 100s of people hoping for your speedy recovery. Be well. Best, Gabe From gcd at i.ph Tue Jan 31 13:08:51 2012 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 31 Jan 2012 18:08:51 +0800 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> Message-ID: that's right Brian. get well soon. with our prayers. -Nandy On Tue, Jan 31, 2012 at 5:39 PM, Gabriel Gunderson wrote: > On Tue, Jan 31, 2012 at 2:34 AM, Brian West wrote: > > Thanks everyone for your kind words. I'm hanging in there trying to > keep a > > level head about everything and have a positive outlook on everything. > > Just know that there are 100s of people hoping for your speedy > recovery. Be well. > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/5f89b543/attachment.html From lists at telefaks.de Tue Jan 31 13:46:00 2012 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 31 Jan 2012 11:46:00 +0100 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> References: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> Message-ID: <4F27C668.2090808@telefaks.de> Get well soon Brian Peter Am 31.01.2012 10:34, schrieb Brian West: > Thanks everyone for your kind words. I'm hanging in there trying to > keep a level head about everything and have a positive outlook on > everything. > > Thanks, > Brian > > On Jan 31, 2012, at 1:22 AM, Zohair Raza wrote: > >> Get well soon Brian, >> >> Regards, >> Zohair Raza >> >> On Mon, Jan 30, 2012 at 9:45 PM, Massimiliano Ravelli < >> massimiliano.ravelli at gmail.com >> > wrote: >> >>> 2012/1/30 Michael Collins >> > >>> >>> For those who are interested in helping in more practical ways, such as >>>> helping to defray the considerable medical costs that no doubt will be >>>> incurred, we will have more information shortly. >>>> >>> >>> Let us know ! >>> >>> Massimiliano >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/f02af892/attachment.html From miha at softnet.si Tue Jan 31 14:57:47 2012 From: miha at softnet.si (Miha Zoubek) Date: Tue, 31 Jan 2012 12:57:47 +0100 Subject: [Freeswitch-users] Call FW, redirection Message-ID: <4F27D73B.8060105@softnet.si> Hi, I need a little help about understanding 302 redirect. I have set on my phone (registered on FS) redirection on number, that is not registered on FS. I have made a condition in my external dialplan. IF the number is on FS that dial to this number, if the number is not on FS, than FS should redirect to default dialplan and call outside number. I have problems with redirecting (302) to default dialplan. I have put in sip_profiles/external this and set in my public dialplan . How to put this variables for redirect BEFORE the redirect condition is hit. You can see in mypastebin what happens. http://pastebin.freeswitch.org/18257 Tahnks! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From steveu at coppice.org Tue Jan 31 18:14:15 2012 From: steveu at coppice.org (Steve Underwood) Date: Tue, 31 Jan 2012 23:14:15 +0800 Subject: [Freeswitch-users] CAMA Stack In-Reply-To: References: <4F1F90DF.30405@coppice.org> <4F2030B8.201@coppice.org> Message-ID: <4F280547.40101@coppice.org> On 01/31/2012 09:36 AM, Moises Silva wrote: > On Wed, Jan 25, 2012 at 11:41 AM, Steve Underwood > wrote: > > > > You omitted to offer your own reasonably priced services. Does Sangoma > already have a CAMA protocol? > > > Not really, I believe the number of customers inquiring about it until > now has been 2 ... including Mike ... > You know, people told me the same thing about MFC/R2 when I started implementing a free version. :-) Some of these things just appear to have no demand because they don't exist. Steve From mike at jerris.com Tue Jan 31 18:46:28 2012 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Jan 2012 10:46:28 -0500 Subject: [Freeswitch-users] Music on hold In-Reply-To: <4F275E57.000006.02768@DWP> References: <4F275E57.000006.02768@DWP> Message-ID: <7268633965498792998@unknownmsgid> no one is interested in your free emoticons. please stop posting about them to the list. On Jan 30, 2012, at 10:23 PM, Darcy wrote: When I receive a call and place it on hold, the calling party receives MOH as specified in . Also, I transfer this call to another extension, the caller receives MOH during the transfer, however, If I place an outbound call then place the party on hold, they just get silence. Am I missing a basic setting here. I also cannot transfer this call or park it. Using snom 3xx, 7xx and 8xx phones. Thanks, Darcy Primrose <02_splash_emoticon_03b_en.gif> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/9d76c403/attachment-0001.html From chris.chen2004 at gmail.com Tue Jan 31 18:50:26 2012 From: chris.chen2004 at gmail.com (Chris Chen) Date: Tue, 31 Jan 2012 10:50:26 -0500 Subject: [Freeswitch-users] Music on hold In-Reply-To: <3E87803C-7FE3-4BF6-BB35-6567A09E7DD0@freeswitch.org> References: <4F275E57.000006.02768@DWP> <3E87803C-7FE3-4BF6-BB35-6567A09E7DD0@freeswitch.org> Message-ID: Brian, you should take a break, you have all our best wishes. Thanks Chris On Jan 30, 2012 10:40 PM, "Brian West" wrote: > use dial plan and bridge_export (btw I'm bored out of MY MIND here in the > hospital.) > > /b > > On Jan 30, 2012, at 9:21 PM, Darcy wrote: > > When I receive a call and place it on hold, the calling party receives MOH > as specified in data="hold_music=/usr/local/freeswitch/sounds/music/8000/moh.WAV"/>. > > Also, I transfer this call to another extension, the caller receives MOH > during the transfer, however, If I place an outbound call then place the > party on hold, they just get silence. Am I missing a basic setting here. > I also cannot transfer this call or park it. Using snom 3xx, 7xx and 8xx > phones. > > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/009f1398/attachment.html From anthony.minessale at gmail.com Tue Jan 31 20:36:31 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Jan 2012 11:36:31 -0600 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: <4A81A8E2C3972444A68BB9D0093D783834096E@telisimo-mail-mx.tlisimo.dom> References: <6180011.valLcmJJfL@sos> <4A81A8E2C3972444A68BB9D0093D783834096E@telisimo-mail-mx.tlisimo.dom> Message-ID: I have pondered this topic for years and have never come to a conclusion I was happy with to move on it. Basically, on one hand, as Ken mentioned if you increase the ptime of every call, you get a lot of the same benefits of trunking at the cost of more audio lost if a packet disappears. This is not a big risk even at 60ms its only a minor loss of audio. The trunking approach does offer much less latency but its a matter of deciding if a complex implementation of trunking is worth the few ms of latency. If you look at trunking as a whole, its the idea of muxing in as many calls bound to the same destination into one stream to avoid overhead. One problem on the internet is that many devices very tightly obey a small MTU and even drop packets that exceed it. There are some ideas floating around but determining the acceptable MTU all the way across the internet is somewhat tricky. Assuming you can choose any MTU you want, trunking looks more attractive, the max allowed size of a UDP packet of 64K can contain several hundred calls even at PCMU. But this is unrealistic. We are most likely limited to the standard MTU in the neighborhood of 1500 and most guidelines suggest you only use a max percentage and you end up with 1200 bytes per packet for payload data. This only allows room for trunking 7 PCMU calls. Based on this conclusion it's obvious that only codecs that can compress the audio better are even practical in trunking. G.729 for instance, can hold dozens of calls since its very compressed. That makes me feel to even bother making trunking, it should probably revolve around some specific low bitrate codec. Then there is a matter of implementation. There are a few drafts on how to do SIP/RTP trunking but none are formally adopted and new sip drafts tend to be over engineered. I've had some ideas on it but the more I think of it, it pushes me towards making a dedicated protocol for it, and if I bother with that, I may as well make a full blown protocol that does everything else I always wanted from VoIP. So every time I think about this issue i go in an endless circle and end up just suggesting with Ken did and say use bigger ptimes between the boxes in question. On Mon, Jan 30, 2012 at 6:53 PM, Nowlin, Win wrote: > Josue, > >> Is there any way to have a SIP Trunk. I mean to have for example 32 > channels merged in one? or something like this? >>When i try to find SIP trunk on internet i just see options for TDM > gateway or similar but not really a multiplexed trunk. > > ? ? ? ?Let's define a SIP trunk for this discussion as "virtual > internet connection" between your switch or gateway's IP address and > your SIP provider's IP address. ? Over this connection your SIP provider > can send you any number of simultaneous conversations (basically > equivalent to "channels" or "time slots" in the TDM world). ?A SIP > "trunk" can have as many simultaneous conversations ("Channels") as you > wish, limited only by your internet bandwidth, how many "channels" you > wish to pay for, and any limits set by your SIP provider. ?Also the SIP > trunk can have as many different DID numbers as you wish or as limited > by your provider. ?In this scenario, your provider is multiplexing the > TDM sources into your SIP trunk. ?In its most basic form, a SIP provider > can "point" the SIP traffic directly at your Freeswitch's external IP > address and there you are!! ?Set up Freeswitch to be compatible with > your Provider's requirements and you are there. > > ? ? ? ?As far as TDM gateways are concerned, if you are using legacy > equipment that requires TDM service, there are several SIP-TDM gateways > available that can receive the SIP traffic from your Provider and > convert it back-and-forth between SIP and TDM. ?As far as the > suitability and hardware requirements for Freeswitch to perform that > function, since I am newly acquainted with Freeswitch, I leave that > discussion to those who are experienced. > > Win N. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Sergey Okhapkin > Sent: Saturday, January 28, 2012 12:14 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM > > SIP (and RTP) have no concept of trunking/audio frames multiplexing > unlike > IAX2 and TDM. > > On Saturday 28 January 2012 21:06:50 Josue Diaz Cruz wrote: >> Is there any way to have a SIP Trunk. I mean to have for example 32 > channels >> merged in one? or something like this? When i try to find SIP trunk on >> internet i just see options for TDM gateway or similar but not really > a >> multiplexed trunk. >> >> Can we do something with freeswitch? >> >> Josue Diaz Cruz >> >> Departamento Tecnico y Soporte >> >> ? jdiaz at coinfru.com >> >> >> >> C/ Balsicas 3 >> >> Alquerias | 30580 | Murcia >> >> ? www.coinfru.com > > ________________________________________________________________________ > _ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kevinw at centrallogic.com Tue Jan 31 21:14:58 2012 From: kevinw at centrallogic.com (Kevin White) Date: Tue, 31 Jan 2012 12:14:58 -0600 Subject: [Freeswitch-users] Installing with support for Digium card ... no freetdm in modules.conf? Message-ID: Hi all, I apologize for such a novice question, but I'm trying to get FreeSwitch installed correctly to support my Digium analog card, and then have Plivo run on top of it. According to FreeTDM site, it says to simply uncomment the mod_freetdm line in modules.conf of freeswitch. But I don't have that line in my modules.conf. I know I could just add it, but I'm not sure if the rest of the system actually has the required stuff in it to work that way. I look in the FreeSwitch 1.0.6 book and it has little to no actual information on doing anything with FreeTDM other than that it exists and replaces OpenZAP. No actual instructions to make it work. What's the "proper" way to get freeswitch installed and working with FreeTDM for digium analog card? Thanks! Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/c4d9ec6c/attachment.html From bdfoster at endigotech.com Tue Jan 31 21:30:12 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jan 2012 13:30:12 -0500 Subject: [Freeswitch-users] Installing with support for Digium card ... no freetdm in modules.conf? In-Reply-To: References: Message-ID: Hola, If you do not already have this line in modules.conf, insert this: ../../libs/freetdm/mod_freetdm Also, take a look at http://wiki.freeswitch.org/wiki/FreeTDM -BDF On Tue, Jan 31, 2012 at 1:14 PM, Kevin White wrote: > Hi all,**** > > ** ** > > I apologize for such a novice question, but I?m trying to get FreeSwitch > installed correctly to support my Digium analog card, and then have Plivo > run on top of it.**** > > > According to FreeTDM site, it says to simply uncomment the mod_freetdm > line in modules.conf of freeswitch. But I don?t have that line in my > modules.conf. I know I could just add it, but I?m not sure if the rest of > the system actually has the required stuff in it to work that way. **** > > ** ** > > I look in the FreeSwitch 1.0.6 book and it has little to no actual > information on doing anything with FreeTDM other than that it exists and > replaces OpenZAP. No actual instructions to make it work.**** > > ** ** > > What?s the ?proper? way to get freeswitch installed and working with > FreeTDM for digium analog card? **** > > ** ** > > Thanks!**** > > ** ** > > Kevin**** > > ** ** > ------------------------------ > > Kevin White**** > > Senior Software Engineer**** > > ** ** > > (801) 727-2362 office**** > > CentralLogic**** > > ** ** > > www.centrallogic.com **** > > **** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/1b6403e4/attachment-0001.html From msc at freeswitch.org Tue Jan 31 22:26:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 11:26:59 -0800 Subject: [Freeswitch-users] Outbound Caller ID on anoymous calls In-Reply-To: <1327504932620-7224077.post@n2.nabble.com> References: <1327494128468-7223625.post@n2.nabble.com> <1327504932620-7224077.post@n2.nabble.com> Message-ID: On Wed, Jan 25, 2012 at 7:22 AM, bennygeorge wrote: > hi there > > the problem is that I'm trying to set it up so that it uses the variables > set in vars.xml > > the dial plan is quite large and is curl based so don't really want to > increase the load on every call checking the cli on each call attempt > Why not? The "increased" load would be infinitesimal. Besides, checking for stuff like this in your curl app is exactly what that process is designed to allow. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/49cf9a44/attachment.html From msc at freeswitch.org Tue Jan 31 22:38:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 11:38:22 -0800 Subject: [Freeswitch-users] ACLs / changing to which IPs FS binds to In-Reply-To: <8d97bc5b095dee0208f316ff08702637.squirrel@fulvetta.riseup.net> References: <8d97bc5b095dee0208f316ff08702637.squirrel@fulvetta.riseup.net> Message-ID: Georg, Once you've created the ACLs in acl.conf.xml you then need to apply them in the SIP profiles. Look in conf/sip_profiles/internal.xml and you'll see that there are parameters for applying ACLs for various types of security. Specifically look for: apply-inbound-acl apply-register-acl -MC On Tue, Jan 24, 2012 at 4:33 PM, wrote: > Hi all, > > I've got a server running FS with five nets associated. There are just > two, from where I receive calls and my phones are registering. > > I would like to exclude all the nets by default from being allowed to > contact / register at FS, and only allow > > - one net 172.251.X.XXX > - one net 192.168.X.XXX > > I tried achieving this trough acl.conf, however, had no success. > I disabled NAT at startup trough -nonat. > > 'sofia status profile internal' is showing me a public ip of my server > next to "Pres Hosts" (but also one ip out of the mentioned 192.168.X.XXX > net, which is fine). > > In internal.xml, I set rtp-ip and sip-ip to this (correct) ip. > > I think my main mistake is that I don't understand how things are handled > in acl.conf. So far it looks like this: > > > > > > > > > > > > > > > Thanks in advance, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/e4de022e/attachment.html From msc at freeswitch.org Tue Jan 31 22:44:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 11:44:04 -0800 Subject: [Freeswitch-users] exit lua script in hangup hook In-Reply-To: References: Message-ID: Gerald, This is a feature/quirk of Lua. There is no explicit exit command. The script keeps running until there are no more commands. You'll need to restructure your loops so that when someone hangups it breaks out of the outermost loop and then has nothing left to execute. -MC On Wed, Jan 25, 2012 at 3:17 AM, Gerald Weber wrote: > Hello,**** > > ** ** > > i?m trying to exit my lua script (called from dialplan) to exit in the the > hanguphook:**** > > ** ** > > function on_hangup(s,status)**** > > freeswitch.consoleLog("NOTICE","---- on_hangup: "..status.."\n");* > *** > > error(); **** > > end**** > > ** ** > > freeswitch.consoleLog("NOTICE","---- ANSWER:\n");**** > > session:answer();**** > > freeswitch.consoleLog("NOTICE","---- SETHOOK: \n");**** > > session:setHangupHook("on_hangup");**** > > ** ** > > while (session:ready() == true) do**** > > freeswitch.consoleLog("NOTICE","---- START OF LOOP \n");**** > > freeswitch.consoleLog("NOTICE","---- STREAMFILE \n");**** > > > session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); > **** > > freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 1 -> HANGUP\n"); > **** > > end**** > > ** ** > > freeswitch.consoleLog("NOTICE","---- END OF SCRIPT 2 -> HANGUP\n");**** > > session:hangup()**** > > ** ** > > When I hangup during the streamFile Call, I geht the following output:*** > * > > ** ** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_channel.c:930 New Channel > sofia/internal/2001 at 192.168.20.73 [2b1cd9f6-4745-11e1-8c71-0d2938abf0b5]** > ** > > 2012-01-25 12:10:19.936097 [INFO] mod_dialplan_xml.c:481 Processing > B#-FS-2001 <2001>->3001 in context default**** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- ANSWER:**** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:599 Channel > [sofia/internal/2001 at 192.168.20.73] has been answered**** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- SETHOOK:**** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- START OF LOOP > **** > > 2012-01-25 12:10:19.936097 [NOTICE] switch_cpp.cpp:1227 ---- STREAMFILE*** > * > > 2012-01-25 12:10:21.116050 [NOTICE] sofia.c:624 Hangup sofia/internal/ > 2001 at 192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING]**** > > 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- on_hangup: > hangup**** > > 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF > SCRIPT 1 -> HANGUP**** > > 2012-01-25 12:10:21.116050 [NOTICE] switch_cpp.cpp:1227 ---- END OF > SCRIPT 2 -> HANGUP**** > > 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1398 Session 2 > (sofia/internal/2001 at 192.168.20.73) Ended**** > > 2012-01-25 12:10:21.116050 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/2001 at 192.168.20.73 [CS_DESTROY]**** > > ** ** > > Why does the hangup handler return and the rest of the script is executed ? > **** > > I event tried ?do return end;? , ?exit;? ,? exit();? instead of ?error(); > - none of them works.**** > > ** ** > > If i put the same logic into javascript, the hook works and the script is > terminated:**** > > ** ** > > function on_hangup(s,status)**** > > {**** > > console_log("NOTICE","---- hangup "+status+"\n");**** > > return "exit";**** > > }**** > > ** ** > > console_log("NOTICE","---- ANSWER\n");**** > > session.answer();**** > > console_log("NOTICE","---- ANSWER\n");**** > > session.setHangupHook(on_hangup);**** > > ** ** > > while(session.ready())**** > > {**** > > console_log("NOTICE","---- START OF LOOP\n");**** > > > session.streamFile("/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-on_hold_indefinitely.wav"); > **** > > console_log("NOTICE","---- END OF LOOP\n");**** > > }**** > > console_log("NOTICE","---- END OF SCRIPT\n");**** > > session.hangup();**** > > ** ** > > ** ** > > 2012-01-25 12:13:08.396041 [NOTICE] switch_channel.c:930 New Channel > sofia/internal/2001 at 192.168.20.73 [8f85a15c-4745-11e1-8c76-0d2938abf0b5]** > ** > > 2012-01-25 12:13:08.396041 [INFO] mod_dialplan_xml.c:481 Processing > B#-FS-2001 <2001>->3002 in context default**** > > 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER**** > > 2012-01-25 12:13:08.396041 [NOTICE] mod_spidermonkey.c:2068 Channel > [sofia/internal/2001 at 192.168.20.73] has been answered**** > > 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- ANSWER**** > > 2012-01-25 12:13:08.396041 [NOTICE] hangup.js:1 ---- START OF LOOP**** > > 2012-01-25 12:13:09.875996 [NOTICE] sofia.c:624 Hangup sofia/internal/ > 2001 at 192.168.20.73 [CS_EXECUTE] [NORMAL_CLEARING]**** > > 2012-01-25 12:13:09.875996 [NOTICE] hangup.js:2 ---- hangup hangup**** > > 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1398 Session 3 > (sofia/internal/2001 at 192.168.20.73) Ended**** > > 2012-01-25 12:13:09.875996 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/2001 at 192.168.20.73 [CS_DESTROY]**** > > ** ** > > ** ** > > Any suggestions ? Or do i miss something here ?**** > > ** ** > > The dialplan part:**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > (replace hangup.lua with hangup.js for the javascript version)**** > > Freeswitch Version:**** > > FreeSWITCH Version 1.0.head (git-9be51d5 2012-01-21 13-45-21 -0500)**** > > ** ** > > thx®ards,**** > > gw**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/3639fb11/attachment-0001.html From msc at freeswitch.org Tue Jan 31 22:50:40 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 11:50:40 -0800 Subject: [Freeswitch-users] Remote Extension In-Reply-To: References: Message-ID: Maybe you could get a sip trace on the 3cx dialog and compare it to the FS dialog and see what's different. You can put those up on pastebin.freeswitch.org and put the link on this thread. -MC On Wed, Jan 25, 2012 at 4:37 PM, Bob Coleman wrote: > Hi, > > I have a remote extension( Cisco SPA504G ) that connects to our > FreeSWITCH server across the internet. I am using a typical double nat > scenario as per wiki and the remote extension registers properly > against the right profile. > > The remote extension via a dialplan can dial any of the extensions > that are local to the FS boxes network successfully with no problems > > However when we dial out to the remote extension we get a "Cannot > locate any authentication on credentials to complete an authentication > request for realm '' > > I have tried using both the default force register options and also > setting up another directory realm as well. > > When looking at the log you see the originate attempting to connect to > the appropriate ip/port of the external phone, but after it "entering > state [calling]" it fails with the above error. > > Just to test I have installed the 3cx softphone client on the remote > network, and it can be dialled with no problems. > > The difference being that the 3cx seems to register a port in the > dynamic range (eg 52789) where as the cisco phone uses the 5090 port > > Has anyone struck that issue before > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/d819bca2/attachment.html From msc at freeswitch.org Tue Jan 31 22:59:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 11:59:26 -0800 Subject: [Freeswitch-users] Newbie questions - Long time to connect In-Reply-To: <019201ccdf9e$e9d503a0$bd7f0ae0$@accra.ca> References: <019201ccdf9e$e9d503a0$bd7f0ae0$@accra.ca> Message-ID: Can you get SIP traces on the working vs. not working boxes and compare them? If need be, put them on pastebin.freeswitch.org and the gang here will take a look. -MC On Mon, Jan 30, 2012 at 2:31 PM, Charles Bujold wrote: > Installing a new Freeswitch ? Ubuntu (AMD Quad FX, 8 GB RAM, Intel > Network card ) and I am getting several issues that hopefully somebody can > point me in the correct direction.**** > > ** ** > > ** ** > > **1) ** I?m Using VoiceMeUp (Canada) as provider and I keep getting > Gateway being Registered, Fail Wait and Trying after the first call is > received or outgoing. If I do not use the line it shows Registered. After > every call it seems to take several minutes before I can make another call. > **** > > ** ** > > **2) **I cannot seem to be able to make more than 1 call at a time > although the service can support 5 calls simultaneously.**** > > ** ** > > **3) **When making a call I see the connection appear in Freeswitch > but it can take over a minute before I hear a ring . Some calls work and > other do not.**** > > ** ** > > **4) **When calling myself using the outside number I can see the > call originating and then being received again (almost instantaneous) by > Freeswitch but it still takes close to a minute before it starts ringing > and when I pick up it hangs up immediately. ( I presume that a time out > occurred.**** > > ** ** > > ** ** > > My question is what could cause this type of slow response to connect or > answer a call. Is there a timer setting that needs to be adjusted?**** > > ** ** > > ** ** > > I have a first server similar in configuration and it works properly. I > presume that I missed configured something but can?t seem to figure out > what?**** > > ** ** > > Thanks**** > > ** ** > > CJB**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/fe484c8c/attachment.html From jeff at jefflenk.com Tue Jan 31 23:06:33 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 31 Jan 2012 12:06:33 -0800 (PST) Subject: [Freeswitch-users] exit lua script in hangup hook In-Reply-To: References: Message-ID: <1328040393457-7241003.post@n2.nabble.com> Just to let you know mercutioviz that anthm did make some source code changes last week that should have enabled this to work. Commit:09ad887948f7513725ca8b53bdfe721d9008e73b * FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/exit-lua-script-in-hangup-hook-tp7223506p7241003.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vipkilla at gmail.com Tue Jan 31 23:26:51 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 31 Jan 2012 15:26:51 -0500 Subject: [Freeswitch-users] valet parking with 2 FS servers Message-ID: Hi, Is there any way to share the valet parking lots across multiple FS servers? I have a setup where opensips distributes calls to two FS servers. When a call is parked and I run: CLI>valet_info The parked call only appears on one FS box. So if the call pickup goes to the opposite FS server, the call will not be picked up. Is there anyway to make both FS boxes aware of valet_info? Similar how to "sofia status profile internal reg" is aware using ODBC Thanks. From Ryan at ocens.com Tue Jan 31 23:46:32 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 31 Jan 2012 20:46:32 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E145D@CH1PRD0610MB355.namprd06.prod.outlook.com> Morning/Afternoon/Evening all... I've been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I've tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/84754fbd/attachment.html