Re: [Freeswitch-users] compilation error of skypiax_protocol.c
Hi João, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. Here's how I normally update FreeSwitch: *make clean svn up ./bootstrap.sh ./configure make install * If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. Regards, -Jingwei 2009/12/8 João Mesquita jmesqu...@freeswitch.org Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang jingwei.y...@gmail.comwrote: I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ‘command_search’: at_interpreter.c:5299: error: ‘COMMAND_TRIE_LEN’ undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ‘command_trie’ undeclared (first use in this function) at_interpreter.c: In function ‘at_interpreter’: at_interpreter.c:5424: error: ‘at_commands’ undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang jingwei.y...@gmail.comwrote: No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene mrene_li...@avgs.cawrote: Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.cawrote: Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... *cc1: warnings being treated as errors* skypiax_protocol.c: In function ‘X11_errors_handler’: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_send_message’: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_do_skypeapi_thread_func’: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed declarations and code make[5]: *** [skypiax_protocol.o] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_skypiax-install] Error 1 make[2]: *** [install-recursive] Error 1 I personally checked the file and it shouldn't be
Re: [Freeswitch-users] OpenZap issues with incoming and outgoing calls
Problem solved. It's due to the lack of definition in tones.conf. In case anyone else need it, here's the tone plan for Singapore. [sg] generate-dial = v=-7;%(1000,0,425) detect-dial = 425 generate-ring = v=-7;%(2000,4000,425) detect-ring = 425 generate-busy = v=-7;%(750,750,425) detect-busy = 425 generate-attn = v=0;%(100,100,1400,2060,2450,2600) detect-attn = 1400,2060,2450,2600 generate-callwaiting-sas = v=0;%(300,0,440) detect-callwaiting-sas = 440 generate-callwaiting-cas = v=0;%(80,0,2750,2130) detect-callwaiting-cas = 2750,2130 detect-fail1 = 913.8 detect-fail2 = 1370.6 detect-fail3 = 1776.7 On Thu, Dec 3, 2009 at 5:29 PM, Jingwei Yang jingwei.y...@gmail.com wrote: Hello All, I have a Digium TDM400P pci card with two FXO ports installed on my linux box. I've connected an external telephone line to the first FXO port. But I can't either make outgoing calls or receive incoming ones. Here are my setups, please let me know where goes wrong. * /etc/zaptel.conf* loadzone = sg defaultzone=sg fxsks=1,2 */usr/local/freeswitch/conf/zt.conf* remains unchanged [defaults] codec_ms = 20 wink_ms = 150 flash_ms = 750 echo_cancel_level = 64 rxgain = 0.0 txgain = 0.0 */usr/local/freeswitch/conf/openzap.conf* [span zt] name = OpenZAP number = 1 fxo-channel = 1 [span zt] name = OpenZAP number = 2 fxo-channel = 2 */usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml* configuration name=openzap.conf description=OpenZAP Configuration settings param name=debug value=0/ /settings !-- one entry here per openzap span -- analog_spans span id=1 param name=tonegroup value=sg/ param name=digit-timeout value=2000/ param name=max-digits value=11/ param name=dialplan value=XML/ param name=context value=default/ /span span id=2 param name=tonegroup value=sg/ param name=digit-timeout value=2000/ param name=max-digits value=1/ param name=dialplan value=XML/ param name=context value=default/ /span /analog_spans /configuration I defined an extension in dialplan/default.xml to receive bridge incoming calls to my skype instance. Frankly speaking, I'm not sure whether this definition is correct. How should I define the expression? When I dial the telephone number, the FS console has no response and I hear nother but busy tones. extension name=incoming_fxo condition field=destination_number expression=^(1)$ action application=bridge data=skypiax/ANY/my_skype_account/ /condition /extension For outgoing calls, I tried something like this: originate openzap/1/1/ echo, while is my handphone number. Again, my handphone has no response. Hopefully I've explained my situation clearly. Please kindly enlighten where the problem might be. Thanks, -Jingwei p.s. here is the outgoing log trace for your reference. freeswi...@localhost.localdomain originate openzap/1/1/ echo 2009-12-03 17:21:04.664276 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.664276 [DEBUG] mod_openzap.c:366 Set codec PCMU 20ms 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1191 Connect outbound channel OpenZAP/1:1/ 2009-12-03 17:21:04.665278 [NOTICE] switch_channel.c:613 New Channel OpenZAP/1:1/ [6f843194-18ce-4525-862f-f5f4e96db5eb] 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:1203 (OpenZAP/1:1/) State Change CS_NEW - CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/ [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:59 Changing state on 1:1 from DOWN to DIALING 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:279 ANALOG CHANNEL thread starting. 2009-12-03 17:21:04.665278 [INFO] ozmod_zt.c:636 Setting echo cancel to 64 taps for 1:1 2009-12-03 17:21:04.665278 [DEBUG] ozmod_analog.c:450 Executing state handler on 1:1 for DIALING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/) Running State Change CS_INIT 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/) State INIT 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:390 (OpenZAP/1:1/) State Change CS_INIT - CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_session.c:999 Send signal OpenZAP/1:1/ [BREAK] 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/1:1/) State INIT going to sleep 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/1:1/) Running State Change CS_ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/1:1/) State ROUTING 2009-12-03 17:21:04.665278 [DEBUG] mod_openzap.c:413 OpenZAP/1:1/ CHANNEL ROUTING 2009-12-03 17:21:04.665278 [DEBUG] switch_ivr_originate.c:66 (OpenZAP/1:1/) State Change CS_ROUTING -
Re: [Freeswitch-users] Skype SIP Beta
Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org wrote: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Debugging reeswitch (especially TLS)
Hello, I have some black hole understading how to debug Freeswitch. In fs_cli I do sofia debug all 7 and indeed get a lot of debugging messages on the console; however, the logfiles get only Critical messages. Where do I define which messages go to the logfile? And in a related topic: I've set a Polycom to use TLS with Freeswitch. I see it contacts FS on TCP port 5061, do some exchange, and then quits and does not use TLS. How do I debug TLS from FS side? Thanks! __Yehavi: ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not like: require luasql.sqlite env = luasql.sqlite() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) After changing it to require luasql.sqlite3 env = luasql.sqlite3() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Hello I'd like to install OpenZAP so I can use a TDM card with Freeswitch, but I'm getting a software error althought the TDM card seems detected (lspci -v OK). Dahdi was successfully compiled from source code. Is it OK to just install Dahdi 2.2.0 without Asterisk before going ahead with OpenZAP? The reason I ask, is that in another forum, someone mentionned /etc/asterisk/chan_dahdi.conf. Here's the output from dahdi_cfg -vvv: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.2 Echo Canceller(s): Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) 1 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) Here's ls -l /dev/dahdi/: total 0 crw-rw 1 root root 196, 254 Dec 8 13:38 channel crw-rw 1 root root 196, 0 Dec 8 13:38 ctl crw-rw 1 root root 196, 255 Dec 8 13:38 pseudo crw-rw 1 root root 196, 253 Dec 8 13:38 timer Has someone succesfully installed Dahdi without Asterisk? Thank you. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694069.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Fred-145 codecompl...@free.fr said: Has someone succesfully installed Dahdi without Asterisk? Of course, and it's working like a charm. DAHDI is a driver. It doesn't care what software uses it. We're using DAHDI with a TE110P PRI T1 card. What is in /proc/dahdi? If it shows 1, what do you see if you cat /proc/dahdi/1? Did you correctly configure the files in /etc/dahdi? How did you configure ../freeswitch/conf/openzap.conf? Maybe it would be helpful to spend a few minutes browsing the wiki at http://wiki.freeswitch.org/wiki/OpenZAP -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green ke...@johnnyvoip.com wrote: It seems you can get a copy of either the binaries or the source by doing the following: Review execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. Return executed agreement to silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN Skype will email you the SILK binary once we receive the executed agreement. Check out documentation, FAQ, and discussion forum (URL TBD) Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli gmar...@celliax.org wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org wrote: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org Copy of Skype - SILK Codec License 27052009.pdf ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Thanks Russel for the tip. After more googling, I ended up figuring that /etc/dahdi/modules had to contain the list of drivers to load. For those interested, here's how to compile and install Dahdi (which doesn't need Asterisk at all, unlike some docs on the Net seem to imply due to references to /etc/asterisk/*.conf): 1. Download and unpack the Dahdi tarball 2. make all ; make install ; make config 3. cd /etc/dahdi/ 4. vi system.conf: #For France, single FXO module on TDM card loadzone= fr defaultzone = fr fxsks=1 5. vi modules: wcfxo wctdm dahdi 6. /etc/init.d/dahdi start 7. dahdi_cfg -vvv 8. ls -la /proc/dahdi/ Now, on to OpenZAP... Thanks again. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26694801.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris m...@jerris.com wrote: That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green ke...@johnnyvoip.com wrote: It seems you can get a copy of either the binaries or the source by doing the following: - Review execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. - Return executed agreement to silksupp...@skype.net silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN - Skype will email you the SILK binary once we receive the executed agreement. - Check out documentation, FAQ, and discussion forum (URL TBD) - Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.comhttp://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli gmar...@celliax.org gmar...@celliax.org wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org br...@freeswitch.org wrote: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.orghttp://www.freeswitch.org Copy of Skype - SILK Codec License 27052009.pdf ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Fred-145 codecompl...@free.fr said: 5. vi modules: wcfxo wctdm dahdi You only need one of the modules above, if you have one card. I don't see a dahdi module listed in the file here. 8. ls -la /proc/dahdi/ You should be able to cat the file in that directory for more information. -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Fred-145 codecompl...@free.fr said: For those interested, here's how to compile and install Dahdi It would be helpful to others if you add the results of your efforts to the wiki. -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Russell.Mosemann wrote: You only need one of the modules above, if you have one card. I don't see a dahdi module listed in the file here. Yup, turns out wcfxo is needed for the X10xP card, while wctdm is needed for Digium cards. As for dahdi, maybe wcfxo/wctdm loads the dahdi module automatically? Russell.Mosemann wrote: 8. ls -la /proc/dahdi/ You should be able to cat the file in that directory for more information. Yes indeed: # ls -al /proc/dahdi/ total 0 dr-xr-xr-x 2 root root 0 Dec 8 16:30 . dr-xr-xr-x 80 root root 0 Dec 8 13:37 .. -r--r--r-- 1 root root 0 Dec 8 16:30 1 # cat 1 Span 1: WCTDM/4 Wildcard TDM400P REV E/F Board 5 (MASTER) 1 WCTDM/4/0 FXSKS RED 2 WCTDM/4/1 3 WCTDM/4/2 4 WCTDM/4/3 Thanks for the tip. I'll see if I can update the wiki accordingly. -- View this message in context: http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26695674.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
They provide you with a 32 bit library, with the header files to link with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 8-Dec-09, at 9:39 AM, Kevin Green wrote: Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris m...@jerris.com wrote: That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green ke...@johnnyvoip.com wrote: It seems you can get a copy of either the binaries or the source by doing the following: Review execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. Return executed agreement to silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN Skype will email you the SILK binary once we receive the executed agreement. Check out documentation, FAQ, and discussion forum (URL TBD) Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli gmar...@celliax.org wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org wrote: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Copy of Skype - SILK Codec License 27052009.pdf ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
We have as of yet been unable to obtain source and we have been in very close contact with skype all the way up to the lead technical and business people on this project. We would of course welcome access to the source but we have as of yet not been able to get a copy Mike On Dec 8, 2009, at 9:39 AM, Kevin Green ke...@johnnyvoip.com wrote: Their site (https://developer.skype.com/silk) specifies that they will provide the source, which as you say may not be 64-Bit compatible but could likely be tweaked to work. I think you just need to be specific in that you want a source copy not a binary copy of the codec. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 9:27 AM, Michael Jerris m...@jerris.com wrote: That would binary only, not 64 bit Linux . On Dec 8, 2009, at 9:17 AM, Kevin Green ke...@johnnyvoip.com wrote: It seems you can get a copy of either the binaries or the source by doing the following: Review execute SILK Agreement - attached. NOTE - please add your Skype login to this form also. Return executed agreement to silksupp...@skype.net and mail hardcopy to: Neil Barrett-Bowen, 3rd Floor, 2 Stephen Street, London W1T 1AN Skype will email you the SILK binary once we receive the executed agreement. Check out documentation, FAQ, and discussion forum (URL TBD) Provide feedback to Skype. Regards, Kevin Green JohnnyVoIP http://www.johnnyvoip.com On Tue, Dec 8, 2009 at 5:14 AM, Giovanni Maruzzelli gmar...@celliax.org wrote: Or it can be LGPL, that's acceptable for FreeSWITCH for my understanding... On Tue, Dec 8, 2009 at 2:50 AM, Brian West br...@freeswitch.org wrote: We can ONLY hope someone will do this and BSD/MIT the library and NOT GPL it... if they GPL it then we'll have to have someone write it all over again... love the Open Source oil and water. /b On Dec 7, 2009, at 7:39 PM, Jason White wrote: it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org Copy of Skype - SILK Codec License 27052009.pdf ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@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@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] Force presence status manually
Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] 100% CPU Usage when Asterisk 1.6.0.19 Registers
Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
I have resubmitted our request for the source. /b On Dec 8, 2009, at 9:58 AM, Michael Jerris wrote: We have as of yet been unable to obtain source and we have been in very close contact with skype all the way up to the lead technical and business people on this project. We would of course welcome access to the source but we have as of yet not been able to get a copy Mike ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
The fun part comes when you try to link that 32bit .a file into a 64bit so file. :P /b On Dec 8, 2009, at 9:49 AM, Mathieu Rene wrote: They provide you with a 32 bit library, with the header files to link with it. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] continue_on_fail
You definitely need to use the settings in combination for what you are trying to do. Can you explain a bit more what you want to do in what conditions and maybe we can suggest how to accomplish this. NORMAL_CLEARING is not a failure, so it can continue on after the bridge unless you specify otherwise. Mike On Dec 7, 2009, at 1:31 PM, Peter P GMX wrote: I have a Problem with continue_on_fail. I have setup a hunt group action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/ action application=bridge data=sofia/external/2...@10.11.12.243,sofia/external/2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245/ action application=bridge data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added action application=set data=hangup_after_bridge=true/ and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] continue_on_fail
set hangup_after_bridge=true Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 7-Dec-09, at 1:31 PM, Peter P GMX wrote: I have a Problem with continue_on_fail. I have setup a hunt group action application=set data=continue_on_fail=NO_ANSWER,USER_BUSY/ action application=bridge data=sofia/external/2...@10.11.12.243,sofia/external/ 2...@10.11.12.234,sofia/external/2...@10.11.12.188,sofia/external/1...@10.11.12.245 / action application=bridge data= (dialstring for fallback user ) I want the fallback user to be called whenever none of the previously called 3 gateway numbers picks up or if they are all busy. Therefore continue_on_fail=NO_ANSWER,USER_BUSY The fallback user is called, however if any of the previously called gateways picks up and then hangs up, the fallback user is called afterwards. Means: The fallback user is always called. I had expected, that continue_on_fail=NO_ANSWER,USER_BUSY would not fire the next bridge if it gets a NORMAL_CLEARING. Am I thinking wrongly about this? I have added action application=set data=hangup_after_bridge=true/ and this works, but I would like to specify more in detail the conditions when to follow the next hunt group entry. Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
If this issue continues after another update and re bootstrap/configure, please open up a bug on jira.freeswitch.org under build system, assign to me, and attach the config.log and config.status file from the root of your freeswitch src dir. Mike On Dec 7, 2009, at 2:39 PM, Anthony Minessale wrote: try rerunning the ./bootstrap.sh On Mon, Dec 7, 2009 at 12:49 PM, Jerry Richards jerry.richa...@teotech.com wrote: When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org
Re: [Freeswitch-users] esl for Mac OS X 10.4
Please re-test this with svn trunk of freeswitch and if it is still the case open up a bug on jira.freeswitch.org in the build system catagory assigned to me and attach the config.log and config.status from the libs/esl dir to the bug. Mike On Dec 7, 2009, at 1:34 PM, Kendall Stauffer wrote: Any direction on where to start would be appreciated. I am trying to get freepbx working with this, and everything works (I think) except esl From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, December 07, 2009 1:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] esl for Mac OS X 10.4 The build system for libesl and everything below that won't work 100% on the mac just yet. You have to make some changes to how its linked and you'll have to compile php yourself to get everything in there properly. The perl one however is much easier to fix. -SOLINK=-shared -Xlinker -x +SOLINK=-dynamiclib -Xlinker -x Thats all you usually fix for the mac. /b On Dec 7, 2009, at 11:27 AM, Kendall Stauffer wrote: I have downloaded and compiled freeswitch, and it runs fine, can compile everything without error including spandsp, but can’t get esl to compile. My version is earlier than the snow leopard that is mentioned in the general install docs, and I have tried it with and without the compiler flags in the freewswtch installation - MAC os X. I have also googled this, and don’t see what I am doing wrong. Anybody there that can help? applesrv:/usr/src/freeswitch-1.0.4/libs/esl root# make phpmod-install make MYLIB=../libesl.a SOLINK=-Xlinker -x CFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=-I/usr/src/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php g++ -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/mysql -liodbc -lmysqlclient -lldap -llber -lcurl -lm -lcurl -liodbc -o ESL.so -L. /usr/libexec/gcc/powerpc-apple-darwin8/4.0.1/ld: Undefined symbols: _main __convert_to_string __efree __emalloc __estrndup __zend_get_parameters_array_ex __zend_list_find __zval_copy_ctor _compiler_globals _convert_to_long _zend_error _zend_get_constant _zend_hash_find _zend_register_list_destructors_ex _zend_register_long_constant _zend_register_resource _zend_rsrc_list_get_rsrc_type _zend_wrong_param_count collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make: *** [phpmod] Error 2 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers
I'm using FreeSWITCH in front of Asterisk without any issue. Stick with the latest trunk. Can you set your loglevel to debug and pastebin your log? Here are some additional tips to help us help you :) http://wiki.freeswitch.org/wiki/Reporting_Bugs Rob On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason spen...@5ninesolutions.com wrote: Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] no hangup on B leg
We will really need debug logs and sip traces to be able to figure out what exactly is going on here. Mike On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote: Sorry no, apart from the fact that I was seeing the hangup. I’m wondering if this a bandwidth congestion issue. Is there anyway on a bridged call I could trap on dtmf like look for ‘*’ and force a hangup? I don’t seem to able to see this tone on the B leg though. Regards, From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 07 December 2009 19:12 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] no hangup on B leg On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton nik.middle...@noblesolutions.co.uk wrote: Hi all, I’ll slowly pulling my hair out on this one. I had FS successfully hanging up both legs on a bridge, now today, with nothing changed, I’m not seeing a hangup of the b leg at all. FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup just fine. Before when I had an issue with the B leg not closing the bridge, I was at least getting a hangup event, now it’s not being fired. Does anyone have an idea what might be causing this? Regards, Time for SIP traces and debug logs. Also, do you have any logs from when things seemed to be working so that you can compare? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] compilation error of skypiax_protocol.c
If you can off list provide me with remote login information to this box I can troubleshot the issue. Mike On Dec 8, 2009, at 4:09 AM, Jingwei Yang wrote: Hi João, thanks for the reply. But I don't quite get you.. Could you please elaborate a little bit? I tried installing libtiff and upgrading FS to the latest revision, but still the same error. Here's how I normally update FreeSwitch: make clean svn up ./bootstrap.sh ./configure make install If any step missing, please kindly let me know. In addition, my OS is CentOS 5.3 and my gcc is version 4.1.2. Regards, -Jingwei 2009/12/8 João Mesquita jmesqu...@freeswitch.org Maybe, just maybe isse that make target to reconf libtiff? Regards, JM On Thu, Dec 3, 2009 at 6:24 AM, Jingwei Yang jingwei.y...@gmail.com wrote: I installed libjpeg-7 following this website: http://www.linuxfromscratch.org/blfs/view/svn/general/libjpeg.html. And the previous error is replaced by a new one: gcc -DHAVE_CONFIG_H -I. -I. -I. -I.. -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -MT at_interpreter.lo -MD -MP -MF .deps/at_interpreter.Tpo -c at_interpreter.c -fPIC -DPIC -o at_interpreter.o at_interpreter.c: In function ‘command_search’: at_interpreter.c:5299: error: ‘COMMAND_TRIE_LEN’ undeclared (first use in this function) at_interpreter.c:5299: error: (Each undeclared identifier is reported only once at_interpreter.c:5299: error: for each function it appears in.) at_interpreter.c:5308: error: ‘command_trie’ undeclared (first use in this function) at_interpreter.c: In function ‘at_interpreter’: at_interpreter.c:5424: error: ‘at_commands’ undeclared (first use in this function) make[8]: *** [at_interpreter.lo] Error 1 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 However, I'm still able to start freeswitch and mod_skypiax and make skype calls with no problem. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:49 PM, Jingwei Yang jingwei.y...@gmail.com wrote: No, I didn't change or update the system libs. I just wanted to double check whether my system has this libjpeg library. ./configure was definitely executed before the source codes were rebuilt. Regards, -Jingwei On Thu, Dec 3, 2009 at 2:39 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Hi, That one is on your side. If you changed/updated system libs it might be worth doing another ./configure Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:33 AM, Jingwei Yang wrote: Hi Mathieu, thanks for the promptly reply. The error has been fixed. However, I encounter another one. gcc -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -DNDEBUG -std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -o make_at_dictionary make_at_dictionary.o -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc ./make_at_dictionary: error while loading shared libraries: libjpeg.so.7: cannot open shared object file: No such file or directory make[8]: *** [at_interpreter_dictionary.h] Error 127 make[7]: *** [all] Error 2 make[6]: *** [all-recursive] Error 1 make[5]: *** [../../../../libs/spandsp/src/libspandsp.la] Error 2 make[4]: *** [install] Error 1 make[3]: *** [mod_voipcodecs-install] Error 1 make[2]: *** [install-recursive] Error 1 Do you have idea about this one? Thanks! On Thu, Dec 3, 2009 at 2:09 PM, Mathieu Rene mrene_li...@avgs.ca wrote: Consider it fixed. Committed revision 15765. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mr...@avgs.ca On 3-Dec-09, at 1:02 AM, Jingwei Yang wrote: Hi Guys, I got a compilation error of skypiax_protocol.c with the latest version r15764. Compiling skypiax_protocol.c... cc1: warnings being treated as errors skypiax_protocol.c: In function ‘X11_errors_handler’: skypiax_protocol.c:1548: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_send_message’: skypiax_protocol.c:1582: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c: In function ‘skypiax_do_skypeapi_thread_func’: skypiax_protocol.c:1726: warning: ISO C90 forbids mixed declarations and code skypiax_protocol.c:1758: warning: ISO C90 forbids mixed
Re: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers
We could check it out for you if you want to contact me and give me ssh access. Or I can provide the instructions get it into the 100% cpu usage state then do the following without stopping FS. 1) run top -H and sort so all the FS threads are at the top and screen cap it so we can see which thread id is using the most cpu. 2) make sure you have gdb installed and issue this command from the build root ./support-d/fscore_pb gcore cpu_race_issue then we can compare the thread using the most cpu with the trace and locate your problem. On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason spen...@5ninesolutions.com wrote: Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel j...@consiglia.dk wrote: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I’m not as stupid as A.M II hints… The problem was that Lua did not “like”: require luasql.sqlite env = luasql.sqlite() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) After changing it to require luasql.sqlite3 env = luasql.sqlite3() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? *Jon Brüel* Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
I changed the name of key to ikey in trunk. Mike Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Brüel ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Force presence status manually
The best way to solve this is probably to share the db for presence and registration between those boxes. If you take a look at the default configs the settings should be commented there. Mike On Dec 8, 2009, at 11:02 AM, Peter P GMX wrote: Hello, is there a way to manually force a presence status update? In our scenario we have a Freeswitch cluster. As phones sometimes register on one and one time on another machine via the load balancer, we cannot dial via user/exten. Instead we dial each phone by it's register string via xml-curl. That way -when a phone is called - other phones who subscribed to this phone, do not receive a message to update their presence status. Is there a way to force the pesence status of a phone manually in the dialplan? We may then set the status before bridging and then reset it with a hangup hook. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
On Tue, Dec 8, 2009 at 3:46 AM, Jon Bruel j...@consiglia.dk wrote: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I’m not as stupid as A.M II hints… Tsk tsk! He didn't actually hint that you were stupid - all he said was that doing ODBC and configuring databases isn't something as simple as flipping on a switch. It takes a bit of knowledge, much of which is hard-earned through experience. Trying to get it all up and running by emailing the list every time something goes wrong is like trying to learning how to change the oil in your car and emailing the Audi-users list every time something doesn't go as expected: yeah, you can probably learn something, possibly you can get it working, but it's grossly inefficient. You'd be much better off paying someone a few euros to come out and give you a lesson because in the long run it would save you both time and money. Just my $0.02. (Don't know what that is in euros...) -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mutual Registration of servers
Otis wrote: div class=moz-text-flowed style=font-family: -moz-fixedMichael Collins wrote: On Sun, Dec 6, 2009 at 11:30 PM, Samuel Abekah-Mensah ab...@greatiam.com mailto:ab...@greatiam.com wrote: Pardon me if this has been addressed already. How does one go about having in the simplest instance 2 servers registering with each other on startup whereby the users registering would be able to call each other. The 2 servers are in different domains. Thanks. Are the two servers in different locations? Different LANs? Is NAT involved? Just checking. Really this is just a matter of loading the default config on each machine and then making some decisions about the dialplan: do you want prefix dialing so that you can have ext 1000 at both locations or do you want to have something like 1000~1099 at location A and 1100~1199 at location B? From there it's just a matter of creating the gateways on each machine and adding a dialplan entry to handle the routing. -MC Hello Michael Thanks Are the two servers in different locations? Yes Different LANs? Yes Is NAT involved? Yes but for my test Nat is not . The production setup I have in mind will certainly have Nat Each location will have their won set of extension but there could be some overlap. On server A a user would dial,. for example, 98 followed by the extension number of the user on server B and the call would then be routed to the extension on server B. And the same could be from Server B to a user on Server A MC Thanks . /div Please olks could someone let meknow if it is at possible. I have tried using the connecting to Asterisk without success, mimicked the link to a gateway unsuccessfully. Could someone please let me kno which .xml files to create etc. Thanks ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
Brian West br...@freeswitch.org said: The fun part comes when you try to link that 32bit .a file into a 64bit so file. That would require a dual-core processor. One core would be 32 bit and the other core would be 64 bit. ;-) -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
Well the fun part is you can't link them. :P /b On Dec 8, 2009, at 10:38 AM, russell.mosem...@cune.org russell.mosem...@cune.org wrote: That would require a dual-core processor. One core would be 32 bit and the other core would be 64 bit. ;-) -- Russell Mosemann ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
Point taken Anthony. Naturally you are not going to work for me for free. But I'm a bit confused about the statement that I'm rude. That's not my purpose to be. And I certainly do hope that this is not just a question of a cultural clash between an elderly man with a Phd in black holes from a European background and a young American FS genius. But frankly, I did believe that focus regarding changes and new developments was somewhat guided by the input you get from the users list, including changes which makes the FS easier to access for newbies, but maybe I'm wrong. That's my last comment, hope we can continue the exchange of views in a good spirit. Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 8. december 2009 17:28 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db One last bit of free consulting advice for you: You are again being rude because you want us to work for you for free. The code is free sir, the support here is voluntary and based on our willingness to help and comments like that are all it takes to get us to ignore you completely. On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel j...@consiglia.dkmailto:j...@consiglia.dk wrote: I got the combination Lua with direct access to the core Sqlite database to work. Hurray, maybe I'm not as stupid as A.M II hints... The problem was that Lua did not like: require luasql.sqlite env = luasql.sqlite() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) After changing it to require luasql.sqlite3 env = luasql.sqlite3() con = assert(env:connect(/usr/local/freeswitch/db/core.db)) And seeing that there was a symlink in one of the right directories called with the name: sqlite3.so, it worked. Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? Jon Brüel Consiglia Telecommunications DK-2960 Rungsted Kyst Tel: +45 45 16 1000 Mob: +45 26 15 30 60 CVR: 27047882 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@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_miness...@hotmail.commailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.commailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.nethttp://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.orgmailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orgmailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] presense. freeswitch vs. spa962+932
Hello,
I tune in the presence freeswitch and linksys spa962 932 I had a few
qusteions:
1) if $ PROXY specified domain name and not ip the phone records. But
all the buttons on spa932 blinking orange indicating that no subscriptions.
phone logs like this:
Call-ID: 76e0f816-9617a...@192.168.0.100
User: 1...@192.168.50.10
Contact: user
sip:1...@192.168.0.100:1024;fs_nat=yes;fs_path=sip:1...@192.168.0.100:1024
Agent: Linksys/SPA962-6.1.3 (a)
Status: Registered (UDP-NAT) (unknown) EXP (2009-12-08 21:26:14)
Host: pbx0.test.lan
IP: 192.168.0.100
Port: 1024
Auth-User: 100
Auth-Realm: pbx0.test.lan
MWI-Account: 1...@192.168.50.10
while the phone is not a nat. spa932 shows that subscriptions
present.
2) how to see that now there is a basis of presence of fs_cli?
3) Can I configure two fs a mutually shared presence? This is done using
param name=presence-hosts value=$${domain}/?
Thabks.
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[Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here!
Greetings, The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH 1.0.5 pre-release version. Please check out the release announcementhttp://www.freeswitch.org/node/220. Let's all get updated as soon as possible. Also, please report bugs right away and follow up when the developers need further information. We have had to close out some bugs due to lack of information from the one reporting. Of course, those running SVN trunk are asked to do a make current as soon as reasonably possible. The devs love it when you are on the latest trunk. :) Thanks again for all of your help! Let's keep up the good work and we'll have 1.0.5 available in no time. -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] 100% CPU Usage when Asterisk 1.6.0.19 Registers
would not be able to even guess without some data to examine. On Tue, Dec 8, 2009 at 12:14 PM, Spencer Thomason spen...@5ninesolutions.com wrote: Hmm.. It doesn't seem to be a problem with Asterisk 1.6.0.13. Asterisk 1.6.0.15-18 doesn't work because of Asterisk bugs and I only noticed this after an upgrade to 1.6.0.19. We're using xen on all our machines with 250hz timers. Could that be a problem? When I get a change I'll try to recreate this with a few more virtual machines to try to debug it. Spencer On Dec 8, 2009, at 8:22 AM, Anthony Minessale wrote: We could check it out for you if you want to contact me and give me ssh access. Or I can provide the instructions get it into the 100% cpu usage state then do the following without stopping FS. 1) run top -H and sort so all the FS threads are at the top and screen cap it so we can see which thread id is using the most cpu. 2) make sure you have gdb installed and issue this command from the build root ./support-d/fscore_pb gcore cpu_race_issue then we can compare the thread using the most cpu with the trace and locate your problem. On Mon, Dec 7, 2009 at 3:34 PM, Spencer Thomason spen...@5ninesolutions.com wrote: Hi I'm trying to use Freeswitch as a SBC for a few Asterisk boxes. The Asterisk boxes are individual hosted PBXs but they are configured with identical software. This a x86_64 CentOS 5.4 system. I've tried 1.0.4 and the latest svn with the same results. Basically Freeswitch registers with outbound providers and I can send and receive test calls. Then without warning, i.e. the Asterisk boxes are all idle and there are no calls, the Freeswitch process starts using 100% of the cpu. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH 1.0.5 is (almost) here!
Let's see if we can beat Duke Nukem Forever! On Tue, Dec 8, 2009 at 12:19 PM, Michael Collins m...@freeswitch.org wrote: Greetings, The FreeSWITCH developers have uploaded the latest and greatest FreeSWITCH 1.0.5 pre-release version. Please check out the release announcementhttp://www.freeswitch.org/node/220. Let's all get updated as soon as possible. Also, please report bugs right away and follow up when the developers need further information. We have had to close out some bugs due to lack of information from the one reporting. Of course, those running SVN trunk are asked to do a make current as soon as reasonably possible. The devs love it when you are on the latest trunk. :) Thanks again for all of your help! Let's keep up the good work and we'll have 1.0.5 available in no time. -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
For those interested, here's how to compile and install Dahdi (which doesn't need Asterisk at all, unlike some docs on the Net seem to imply due to references to /etc/asterisk/*.conf): I understand that Some Debian based distro's have Dahdi in their repo's making it simple, but not many know that Digium runs its own repo for rpm based distros: http://packages.asterisk.org/ Can't get easier than that... jlc ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Choppy sound with PCMU
For reference, here is the AstLinux kernel config for the ALIX: http://astlinux.svn.sourceforge.net/viewvc/astlinux/trunk/target/device/alix/linux.config?view=markup We've got what I consider to be excellent support for the ALIX - most of the developers use them and they are very popular in the community. On Mon, Dec 7, 2009 at 11:00 AM, Anthony Minessale anthony.miness...@gmail.com wrote: Did you do each thing alone too to tell the difference? -hp alone, disable monotonic alone (i did not see you mention the disable monotonic) as for your 4ms thing, yes we require high resolution timing, if we ask to sleep 1000 microseconds that is what we need it to sleep for or at least as close as possible, and the main reason that thread is never sleeping is because you can't actually count on it to run every 1ms but you mostly can. Hence the whole philosophy on only making 1 thread run hot all the time to ensure that the rest don't have to repeat the same algorithm. We focus on high end performance this was the point of your experimentation because we will need to use a compile time defines and other logic to make it more efficient on your platform, a platform which we are not using. I am curious what would happen if you install Kristian's astlinux on one of your devices, i think you should also compare the kernel versions. What OS are you running anyway? Here are some more things to try (running plain trunk with no mods) do these systematically each alone and all together with/without -hp or disable monotonic etc to see what different combos create comment out this line (line 10) #define DISABLE_1MS_COND rebuild, this tells it to run a conditional at 1ms in the same timer thread which will make all the switch_cond_next share a 1ms conditional instead of doing microsleeps next some kernels/devices work better using select(0) for sleep where others work better using usleep. comment out line 109 apr_sleep(t); and try usleep(t) also mac works better using nanosleep so you could try changing it so it uses the code starting at 101 instead. also your claim about JS should be investigated because I do not think it should be the case. but you may want to move this to a jira http://jira.freeswitch.org As for the asterisk comparison, not sure how to answer you, that's your decision. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Wrong RFC2833 in SDP on NEC phones
Dear list, Some Nec phones sends DTMF RFC2833 with payload 101 during the call, but have negotiated a different one on SDP. When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 we notice this phone sends the following INVITE packet and RTP packets: http://pastebin.freeswitch.org/11433 Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap Is there any parameter to tweak FS in such a way to force understand 101 packets as DTMF? Thank you in advance! Fernando Testa PS: On pcap you have the following IPs: FS at 10.91.10.210 Nec Pbx 10.91.10.22 phone 10.91.10.85 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones
Best option for you is to use 96 in the sofia profile you're using to talk to these broken devices. /b On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote: Dear list, Some Nec phones sends DTMF RFC2833 with payload 101 during the call, but have negotiated a different one on SDP. When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1 we notice this phone sends the following INVITE packet and RTP packets: http://pastebin.freeswitch.org/11433 Whole wireshark capture file is on http://gregianin.org/teste_voice_rfc2833.pcap Is there any parameter to tweak FS in such a way to force understand 101 packets as DTMF? Thank you in advance! Fernando Testa PS: On pcap you have the following IPs: FS at 10.91.10.210 Nec Pbx 10.91.10.22 phone 10.91.10.85 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
Hey you guys, I know this isn't the right place for this, but I have been working with asterisk for 5 years now, and just got freeswitch working (on windows, not os x yet). All I can say is AWESOME --- thanks so much From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Tuesday, December 08, 2009 9:34 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Lua and database access to core_db And you didn't open a Jira about this? These are the kinds of issues that you should report so we can fix them... sitting on them and NOT reporting them only delays the 1.0.5 release. /b On Dec 8, 2009, at 5:46 AM, Jon Bruel wrote: Changing the core db into a MySQL via ODBC caused some problems even after it seemed to work. For instance, console help caused an error with an error description indicating that a SQL SELECT query including the reserved word key has been fired. It this problem likely to be solved if I used another version of the MySQL? ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through
Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. 08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk i meddelelsen 4b1dfabc02e10...@mail.fribert.dk: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0 to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (sofia/internal/1...@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1...@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen 659603.29094...@web56408.mail.re3.yahoo.com: Question -- If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dk ( http://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk ) reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: -- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question -- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: -- This is correct as long as you have a gateway that is registered called musimi.dk Question -- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: -- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?
Our plan for 1.0.5 is that we will also have rpm and deb packages for many distros on our own repo. Stay tuned. This has been another major reason for the delay in 1.0.5. Mike On Dec 8, 2009, at 1:26 PM, Joseph L. Casale wrote: For those interested, here's how to compile and install Dahdi (which doesn't need Asterisk at all, unlike some docs on the Net seem to imply due to references to /etc/asterisk/*.conf): I understand that Some Debian based distro's have Dahdi in their repo's making it simple, but not many know that Digium runs its own repo for rpm based distros: http://packages.asterisk.org/ Can't get easier than that... jlc ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
Can you copy the address of the pastebin so that people can see it? After you hit the Send button, the address is posted back at the top of your browser, like: http://pastebin.freeswitch.org/11441 From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Tuesday, December 08, 2009 12:35 PM To: 'Michael Jerris'; freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 10:49 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry _ From: Jerry Richards [mailto:jerry.richa...@teotech.com] Sent: Monday, December 07, 2009 7:44 AM To: 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' Subject: RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry _ From: Michael Jerris [mailto:m...@jerris.com] Sent: Saturday, December 05, 2009 7:30 PM To: Jerry Richards Subject: Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org http://pastebin.freeswitch.org/ On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 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_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net/ #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP
are you using more than one profile here? if so you have to repeat the siptrace on command for each one. This trace makes little sense to me because I think half of it is missing. but you can see several packets coming in like 20 times each which means you have some kind of nat or network problem causing the other end of this call to send retries on all the packets. On Tue, Dec 8, 2009 at 2:57 PM, Jerry Richards jerry.richa...@teotech.comwrote: Here is the Pastebin Link: http://pastebin.freeswitch.org/11432 Thanks, Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Tuesday, December 08, 2009 12:35 PM *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Anthony and Michael, I downloaded the latest trunk, rebuilt it, and re-ran the test with the logs that Anthony told me to turn on. I put the results up in the PasteBin. Best Regards, Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Monday, December 07, 2009 10:49 AM *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP When I got the latest trunk the make gets an error. Should I perhaps disable the mod_amr? making all mod_amr make[5]: *** No rule to make target '/mod_amr.c', needed by 'mod_amr.so'. Stop The method I used to get the latest trunk follows: svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk freeswitch Best Regards, Jerry -- *From:* Jerry Richards [mailto:jerry.richa...@teotech.com] *Sent:* Monday, December 07, 2009 7:44 AM *To:* 'Michael Jerris'; 'freeswitch-users@lists.freeswitch.org' *Subject:* RE: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP I am changing the 3pcc setting because one of my gateways sends INVITEs without SDP. I will try to update to the latest trunk today and capture traces as Anthony described. If I can't do it today, it might be at the end of the week. Best Regards, Jerry -- *From:* Michael Jerris [mailto:m...@jerris.com] *Sent:* Saturday, December 05, 2009 7:30 PM *To:* Jerry Richards *Subject:* Re: [Freeswitch-users] FS Machine Sends ICMP DESTINATION UNREACHABLE When Gateway Sends RTP Jerry- Any update on this? Mike On Dec 4, 2009, at 3:59 PM, Anthony Minessale wrote: Why are you changing the 3pcc setting, is this an invite with no sdp? you need to take a trace from FS. 1) update to latest trunk first so line number match up. 2) issue these commands sofia profile internal siptrace on console loglevel debug save the output and put it on pastebin http://pastebin.freeswitch.org On Fri, Dec 4, 2009 at 2:47 PM, Jerry Richards jerry.richa...@teotech.com wrote: I have Mediant 1000 gateway, and for some reason, when I make an outbound call, FS enters the CS_CONSUME_MEDIA state and never connects the call. A Wireshark trace shows that FS is replying to the gateway's inbound RTP packets with ICMP DESTINATION UNREACHABLE. But the gateway is sending RTP packets to the same port that FS specified in the outbound INVITE. It appears in the log that FS is discarding the 200 OK from the gateway. I disabled the Firewall and SELinux on the Freeswitch machine. I tried changing enable-3pcc to true and also proxy, but it has no effect. Anyone know what could be the issue? I posted the Freeswitch log in the pastebin. Best Regards, Jerry ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] FXO PCI card with Pfsense Freeswitch.
I dont think there are any supported hw for bsd, there are legacy sangoma and zaptel drivers floating around but they are not supported by the vendors. On Tue, Dec 8, 2009 at 3:25 PM, Orien Love or...@tx.rr.com wrote: I am looking for a 4 port FXO card to use with my PfSense installation of freeswitch. does anybody know if the Asterisk Trixbox 4 FXO Voip Digium TDM410 will work on pfsense? or could somebody recommend one that would. Thank You Orien ___ FreeSWITCH-users mailing list FreeSWITCH-users@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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Rhino Cards for sale R2T1-EC and R24FXX-EC
I have 2 Rhino cards for sale if anyone needs one. They are both Best Offer. I have a R2T1-EC and a R24FXX-EC with 12 dual FXS modules. Both have never been used more than a few times for testing purposes. Both cards work fine and are guaranteed not to be DOA. Reece Savage Information Technology Manager King Ballow Law Offices 315 Union Street Suite 1100 Nashville, TN 37201 Phone (615) 726-5525 Fax (615) 254-7907 rsav...@kingballow.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through
have you created Extension 1002? -nandy On Wed, Dec 9, 2009 at 3:20 AM, mailinglist mailingl...@fribert.dk wrote: Hi All Ok, after reading a bit more I think I see what I've done wrong, but I don't know how to fix it properly. Looking in the Dialplan directory I see the following: default (dir) default.xml features.xml public (dir) public.xml Under the default dir the webinterface has created the 001_musimi.dk.xml file that I've created. But as I understand it, it doesn't use it. How do I make it use it, I would very much like to keep the webinterface editor, and not have to do it via ssh and vi all the time. 08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk i meddelelsen 4b1dfabc02e10...@mail.fribert.dk: Hi Mark Ok, thanks. Yes I have a gateway placed in external called musimi.dk (or should it be in public?), and I'll just create the empty XML's in lan to get rid of that error. I'll remove the second part of the dialplan, my idea was that it was needed for calls between sip phones hooked up to the freeswitch. Now the remaining problem: When I call ext 1002 from ext 1001 I see this message and get an error, the same goes for dialing 0 to get an external number: 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 in context default 2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb] 2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: NO_ROUTE_DESTINATION 2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 (sofia/external/$1) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/external/$1 [CS_DESTROY] 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 ( sofia/internal/1...@10.11.12.25) Ended 2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/1...@10.11.12.25 [CS_DESTROY] I don't see any mention of the statements in the Dialplan, so for me it looks like it haven't registered the Dialplan? Best regards Kenneth 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen 659603.29094...@web56408.mail.re3.yahoo.com: Question -- If I do a reloadxml it gives me this output on the console: freeswi...@firewall.fribert.dkhttp://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk reloadxml 2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) Error including /usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such file or directory) I'm not sure if it's a genuine problem,as I can see it, it just complains that I haven't created any sip_profiles in /lan, but is that necessary? Response: -- This isn't really a problem. To get rid of the error simply put a blank xml file into each folder as in the internal and external directories. Dump the lan directory and lan profile as mentioned earlier. Question -- Extension Name musimi.dk Enabled true Order 001 Description ... condition ^0(.\d+)$ action bridge sofia/gateway/musimi.dk/$1 Response: -- This is correct as long as you have a gateway that is registered called musimi.dk Question -- Extension Name 10.11.12.25 Enabled true Order 002 Description ... action bridge sofia/internal/$ Response: -- No idea what this is for its not needed as far as I can tell. Now please summarize what you still need help on. Mark J Crane http://fusionpbx.com pfSense FreeSWITCH package developer ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Lua and database access to core_db
On Wed, Dec 9, 2009 at 7:56 AM, Kendall Stauffer k...@ksac.com wrote: Hey you guys, I know this isn’t the right place for this, but I have been working with asterisk for 5 years now, and just got freeswitch working (on windows, not os x yet). All I can say is AWESOME --- thanks so much Out of curiosity. Did you choose freeswitch because it runs on windows and asterisk doesn't? I find some people choose freeswitch because they don't know or want to use linux (obviously this doesn't apply to you) and some people choose it because they want a windows solution and asterisk doesn't run on windows. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
