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Presence api for custom function
by Tóth Róbert 21 Nov '10

21 Nov '10

Hello How can i use the Presence api for my custom application? I would like to set my phone's BLF to set that other phones may or may not be able to pickup the ringing calls. I make a dialplan application for this, and the setting is stored in db. Almost everything is OK, the dialplan application works, BLF's color is changing, but after some seconds it changes back to red. It is obviously because there is not a real phone-ext behind the number of what the phone is subscribing. I thought if i use the presence api, its state does not changes until i send another event for that entity. But it is not true. What can i do? thx

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Call-State in LUA
by Tóth Róbert 21 Nov '10

21 Nov '10

Hi, Is it possible to get a channel's Call-State in LUA? Is there a function for this similar to session:getState()? (I would like to know if a channel is in HELD state. Thanks

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Freeswitch not reply to a RE-INVITE
by Stephen Wilde 21 Nov '10

21 Nov '10

I have a problem with a customer: Freeswitch doesn't reply to all RE-INVITE (and all calls are dropped by customer switch after 180s for this reason). At RE-INVITE reception Freeswitch reply with a 100 Trying and never send a 200 OK. I have compared INVITE and RE-INVITE and to me seem to be ok, for example: INVITE: Call-ID: 26100859-3498991521-863759(a)SBC2.mydomain.com REINVITE: Call-ID: 26100859-3498991521-863759(a)SBC2.mydomain.com INVITE Via: SIP/2.0/UDP 192.168.1.4:5060 ;branch=z9hG4bK44ee87fffd4e657f4514fdad9de39f9e RE-INVITE Via: SIP/2.0/UDP 192.168.1.4:5060 ;branch=z9hG4bK0ed216868c20ca374c8893a4fb9a39b0 I have pastebin the captured packets: http://pastebin.freeswitch.org/14527 Can anyone help me? I'm using Freeswitch GIT commit 97c65a004ee9e9b88dad7ff4ac1b2ec893b8c5ba, Date Nov 15 2010 Thanks in advance, Stephen

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ivr simple question
by budi wibowo 20 Nov '10

20 Nov '10

hi i just try to make ivr, first i use the sample provided by freeswitch, demo-ivr with extension 5000 di default dialplan. i call 5000 but got message forbidden from my sip client. thx and regards budi

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Re: [Freeswitch-users] T.38 and Audiocodes or Patton gateway
by Dave Redmore 20 Nov '10

20 Nov '10

I'm curious about your recommendation to use a long (600 sec) media/RTP timeout value in the sofia profile. Is this a specific need for the Audiocodes device or just generally better for T.38? I'm assuming you are referring to the "rtp-timeout-sec" parameter? I'm using FreeSwitch as a media proxy to a T38 provider and have had decent success, but usually have problems with longer faxes (8+ pages) and am wondering if this setting might help? Thanks, Dave Redmore ----- Original Message ----- From: "Kristian Kielhofner" <kris(a)kriskinc.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Tuesday, November 16, 2010 9:39:22 AM GMT -06:00 US/Canada Central Subject: Re: [Freeswitch-users] T.38 and Audiocodes or Patton gateway Tim, I've have good experiences with the MP-20x series using T.38 with FreeSWITCH. On the AudioCodes: Make sure your T.30 and T.38 redundancy settings are at least "1". Make sure ECM is disabled. Use the latest 2.6 firmware (not 3.x). In FreeSWITCH: Make sure ECM is disabled. Make sure to have a very high (600 seconds seems ideal) media/rtp timeout parameter on the Sofia profile for the AudioCodes device. On the fax machine: Make sure ECM is disabled. Disable Super G3 or anything higher than 14.4. On Mon, Nov 15, 2010 at 1:08 PM, Tim St. Pierre <fs-list(a)communicatefreely.net> wrote: > Hello, > > I am trying to get a reliable fax setup working. > > I have set up a nice fax receiver mechanism with mod_spandsp and it works very well when I send a > fax using a Linksys SPA2102 > > I have poor results with Audiocodes MP202B, and can't get anything to work with a Patton M-ATA > > I have a number of Audiocodes MP202B in service and would rather not replace them if I can avoid it. > > In each case, the fax fails with "communications error" on the sending machine, and "Call hungup > prematurely" on the SpanDSP side. > > There aren't too many knobs on the gateways. Are there any knobs in mod_spandsp that might help > this condition? Anyone else have any experience with these devices? > > Thanks! > > -Tim > > _______________________________________________ > 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 > -- 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(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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Web interface (or a client application) to controlling calls, conferences, fifos, faxes, etc.
by Tóth Róbert 20 Nov '10

20 Nov '10

Hello I am finding a web interface (or a client application) to: - controlling existing calls (kill, spy, intercept, transfer to another extension, etc.) - controlling conferences (throw out sb, invite sb, volume control, lock, pin code, etc) - controlling fifos (listing, pick up sb, ordering, hang up sb, etc) - send faxes, managing received faxes - view call detail records and play recorded sessions (if recorded) As i see, FreePBX and FusionPBX are good to configuring FS, they are for system administrators. Do you know an interface for everyday system users? Is there a system for this? thx Rob

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Problem compiling FreeSWITCH on OpenBSD 4.8
by Comète 20 Nov '10

20 Nov '10

Hello, i'm trying to compile freeswitch with sources from git repository on a fresh OpenBSD 4.8 (i386) install and i get this error during the "gmake" part of the installation gmake install ... gmake "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/lefieux-m/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/lefieux-m/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/lefieux-m/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /home/lefieux-m/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /home/lefieux-m/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /home/lefieux-m/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /home/lefieux-m/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /home/lefieux-m/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` install-recursive Making all in xml/expat mkdir .libs Compiling src/switch_apr.c ... cc1: warnings being treated as errors In file included from ./src/include/switch_platform.h:120, from ./src/include/switch.h:109, from src/switch_apr.c:33: /usr/include/arpa/inet.h:74: warning: 'struct in_addr' declared inside parameter list /usr/include/arpa/inet.h:74: warning: its scope is only this definition or declaration, which is probably not what you want /usr/include/arpa/inet.h:75: warning: 'struct in_addr' declared inside parameter list gmake[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1 gmake: *** [install] Error 2 this is the list of packages i've installed: autoconf-2.59p3 automatically configure source code on many Un*x platforms automake-1.9.6p4 GNU standards-compliant Makefile generator bzip2-1.0.5 block-sorting file compressor, unencumbered colorls-4.6p0 ls that can use color to display file attributes curl-7.20.0 get files from FTP, Gopher, HTTP or HTTPS servers cvsps-2.1 generate patchsets from CVS repositories gettext-0.18.1 GNU gettext git-1.7.1p0 GIT - Tree History Storage Tool gmake-3.81p1 GNU make jpeg-8b IJG's JPEG compression utilities libiconv-1.13p1 character set conversion library libidn-1.11 internationalized string handling libltdl-1.5.26p0 GNU libtool system independent dlopen wrapper libogg-1.2.0 Ogg bitstream library libtool-1.5.26p4 generic shared library support script libvorbis-1.3.1 audio compression codec library metaauto-0.9 wrapper for gnu auto* p5-Error-0.17015 error/exception handling in an OO-ish way pcre-8.02p1 perl-compatible regular expression library python-2.6.5 interpreted object-oriented programming language rsync-3.0.7 mirroring/synchronization over low bandwidth links sqlite3-3.6.23p0 embedded SQL implementation tiff-3.9.4 tools and library routines for working with TIFF images vim-7.2.444-no_x11 vi clone, many additional features zsh-4.3.10p1 Z shell, Bourne shell-compatible the ./bootstrap.sh and ./configure parts produce no error and work well any idea ? Thanks Comete

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mod_stress
by mashudi72 - 20 Nov '10

20 Nov '10

Dear all, on the freeswitch git, there is mod_stress module, how to use mod_stress module with javascripts? thank in advanced, regards Mashudi

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Dynamic Conference creation
by Farhan Husain 20 Nov '10

20 Nov '10

Hello, I could successfully create audio conference by adding extension in dialplan. However, how can I create conferences dynamically? Can I use mod_event_socket for that? Is there any command to create a conference dynamically? Thanks, Farhan

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Simulate a Collect Call behaviour
by Listas Devel-IT 19 Nov '10

19 Nov '10

Hello All, I am in need to simulate a Collect Call behaviour for a particular case and I have tried Asterisk to do that but without any luck and I am wondering if FS can do that. Scenario: A calls B when B answers A hear an announcement (pre-recorded) and B hears another announcement. If B stay in the line after heard all the announcement the call is connected. I have found this ( http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation) but I am not sure if it can accomplish this scenario. I really appreciate any tips. Best regards Rodrigo Telles

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