November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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IMPORTANT NEWS: blacklist.pl script not working because of infiltrated.net expiration
by Michael Collins 19 Nov '10

19 Nov '10

Hello all, It seems that infiltrated.net expired yesterday. This is the source of the data for our blacklist.pl script. The script pulls two files: http://www.infiltrated.net/voipabuse/addresses.txt http://www.infiltrated.net/voipabuse/netblocks.txt It uses these two files for building a better ACL that rejects known bad guys. Since these files are not currently available the script will receive bogus information and pollute the ACL. If you have run the blacklist.plscript prior to the failure and your system is working properly then please make a backup copy of your freeswitch.xml.fsxml file. This will contain the proper ACL nodes. DO NOT RESTART FREESWITCH OR RELOAD XML BEFORE BACKING UP THIS FILE! If you do then you will lose the information. I am researching this right now and I will update you as soon as I have some information. In the meantime if any of you have a freeswitch.xml.fsxml file with the blacklist info please zip and email me offlist a copy of that file and I will see if I can't extract the IPs and create a temporary workaround. I will also be updating the blacklist.pl script to do some error checking so that it doesn't pollute the ACL. Last item: if you have inside knowledge of what's happening with the VoIP Abuse project please contact me off list. Thanks, Michael S Collins

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transfer_ringback heard by both parties (during att_xfer)
by Fraser Redmond 19 Nov '10

19 Nov '10

I've been exploring att_xfer the last couple of days and am really happy with it, works really well, and it's been interesting looking back at old mailing list emails to see how it's evolved over the years. One thing that isn't quite how I'd expected it to be is that if I set transfer_ringback=$${hold_music} both the original parties hear it, instead of just the person who is being transferred. For example (if we have lines A, B, C: A calls in to B, then B transfers to C and hangs up leaving A & C talking) then during the transfer A hears the hold_music (which is what I'd expect) but when C's phone is ringing B hears it as hold_music instead of a 'normal' ringback. Reading through the wiki, it looked like setting one of these vars before doing the transfer might work, but no change: <action application="set" data="ringback=${us-ring}"/> <action application="set" data="ignore_early_media=true"/> Anything else I should try? Or is this intended behaviour, or a bug? Cheers, Fraser

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dialplan to share skype instances
by Jian Ren 19 Nov '10

19 Nov '10

Hi, Currently I am assigning one instance to each extension like below: <extension name="us.skypeout"> <condition field="username" expression="^10(\d{2})$"> <action application="set" data="skype_instance=$1"/> </condition> <condition field="destination_number" expression="^(\d{11})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="skypopen/interface${skype_instance}/$ </condition> </extension> Are there any other ways like to get unused instances from a pool? I guess it could be something like condition instance1 available? bridge instance1 condition instance2 available? bridge instance2 ... then there could be much more extensions to share skype out. Could anyone help? Thanks! Jian

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