June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH 64bit compilation error (Solaris 10)
by email lists 08 Dec '10

08 Dec '10

Forwarding the issue below to see if anyone is familiar with this issue, and/or what our next steps should be. Thanks, Vladimir Looks like a problem with a Makefile not honoring CFLAGS,etc. Perhaps you can report this to the dev team. Other components built fine but this damn spidermonkey is buggering. # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o | head /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o : ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgat eway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_de sc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniso ap.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniss dpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniup npc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwg et.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixm l.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] # file /opt/freeradius-client/sbin/radacct /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not stripped Build output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia.lo Compiling sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) "sofia.c", line 3522: warning: enum type mismatch: arg #2 (E_ENUM_TYPE_MISMATCH_ARG) Creating mod_sofia_la-sofia_glue.lo Compiling sofia_glue.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_presence.lo Compiling sofia_presence.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_reg.lo Compiling sofia_reg.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_sla.lo Compiling sofia_sla.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia.la making all mod_speex Compiling mod_speex.c... mkdir .libs Compiling mod_speex.c ... Creating mod_speex.so... making all mod_spidermonkey cd config; /usr/sfw/bin/gmake -j1 export ld: fatal: file now.o: wrong ELF class: ELFCLASS64 ld: fatal: File processing errors. No output written to now gmake[7]: *** [now] Error 1 gmake[6]: *** [export] Error 2 gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/sfw/bin/gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2

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FreeSWITCH Nagios Plugin
by Kristian Kielhofner 28 Oct '10

28 Oct '10

Hello everyone, I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like to connect to the event socket and run some api commands and return them (as opposed to checking SIP, for example). I haven't found anything and I've started to write one in perl using ESL. I'm sure whatever I come up with is going to be pretty ugly and I'd much rather use something else. Has this been done? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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extension mobility using freeswitch
by babak yakhchali 30 Sep '10

30 Sep '10

Hi Is it possible to implement something like cisco ccm extension mobility using freeswitch and cisco xml objects and services for cisco ip phones? thanx

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Re: [Freeswitch-users] Contact Field
by Kristian Kielhofner 16 Sep '10

16 Sep '10

I assume he is talking about the contact field in the INVITE, not the REGISTER... I don't know how to accomplish this off the top of my head but I'd be interested too (quite a few carriers request this). -- Kristian Kielhofner http://blog.krisk.org ------------------------------ *From*: freeswitch-users-bounces(a)lists.freeswitch.org < freeswitch-users-bounces(a)lists.freeswitch.org> *To*: freeswitch-users(a)lists.freeswitch.org < freeswitch-users(a)lists.freeswitch.org> *Sent*: Tue May 18 11:56:28 2010 *Subject*: Re: [Freeswitch-users] Contact Field HI Ken, Add the following line <param name="extension" value="10 Digit Number"/> Lloyd On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer < kenfulmer(a)icstechnologysolutions.com> wrote: > We are connecting a FS gateway to PaeTec. They need the contact field to > be the same as the calling number. Using the trunk registration 10 digit > number doesn’t work for them. > > > > I added the following parameter to remove “gw + external” in the contact > field to the left of the @ sign: > > > > <param name="extension-in-contact" value="true"/> > > > > Is there something else I should add, so the contact field shows the 10 > digit calling number? > > > > Thanks, > > > > Ken Fulmer > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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[contrib] a new java ESL inbound/outbound client
by david varnes 30 Aug '10

30 Aug '10

Hi all, I had a basic java inbound ESL client kicking around that I have used in a couple of small projects over the last year. I needed something a little more complete for a new project so I dusted it off and made it less incomplete. It still needs more work and testing, but it certainly is usable right now. I have tested it against FS 1.0.4 and latest trunk. I have put it in my contrib area in svn in hopes that some may find it useful: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-cli… I would be interested in any feedback ... Features * Apache License (ASL) version 2 * Standalone Inbound client * Framework classes to easily create an Outbound socket client * based on Netty [1] nio library version 3.1.5.GA (previously was using Apache MINA, but this is easier) * logging via slf4j * only dependencies are slf4j-api and netty (both Apache licensed) * single jar which is a valid OSGi bundle * built using maven * eclipse projects * reasonable level of java docs Still todo * Docs * Simple example apps * .. more in TODO.txt in project root. There is no binary jar available right now since I don't know how/if I can put files up to file.freeswitch.org. In the meantime to build you need maven [2] installed. If you are unfamiliar with maven usage, I can post a simple howto. davidv [1] http://www.jboss.org/netty/downloads.html [2] http://maven.apache.org

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calls ending with MEDIA_TIMEOUT
by Dan 21 Jul '10

21 Jul '10

Hi guys, I have been running two freeswitch boxes (13754M) that answer calls from a cisco 5300 (both on the same network) and records them to disk with a small lua application. This has been working well for the past few months. I decided to upgrade one of them to trunk ( git-3fbd9e2 2010-06-11 11-08-51 -0500 ) and have run into a problem. Some calls will fail with a MEDIA_TIMEOUT after a few minutes, the time it takes to fail ranges from 4 minutes to 10 minutes, I don't have a full sip trace or pcap dump yet, I reverted back to the old freeswitch version (on the same hardware) and have not been able to reproduce it in a test environment yet ( I continue to try). Below are the relevant lines from the log files for one of the calls: 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2257 (sofia/external/nobody(a)192.168.21.4) Callstate Change ACTIVE -> HANGUP 2010-06-23 07:42:19.033466 [NOTICE] mod_sofia.c:884 Hangup sofia/external/nobody(a)192.168.21.4 [CS_EXECUTE] [MEDIA_TIMEOUT] 2010-06-23 07:42:19.033466 [DEBUG] switch_channel.c:2273 Send signal sofia/external/nobody(a)192.168.21.4 [KILL] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_session.c:1023 Send signal sofia/external/nobody(a)192.168.21.4 [BREAK] 2010-06-23 07:42:19.033466 [DEBUG] switch_core_codec.c:146 sofia/external/nobody(a)192.168.21.4 Restore previous codec PCMU:0. My configuration is bone stock, so the rtp timeout value is at 300, but I have some calls that have lasted only 4 minutes. One other piece of information is that on one of the recordings that was hung up after 4 minutes and 17 seconds the recorded file was only 24 seconds long (it stopped recording after the first 24 seconds) , so I'm assuming freeswitch did not think there were any rtp packets to record. Any ideas on where to start debugging this? I have setup a new freeswitch box connected to the same 5300 to reproduce, but have not been able to generate the call volume ( there where around 30 calls being recorded) yet, but I'm working on it. Thanks!

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