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Trying to determine why bind_digit_action does not work in a conference created by outbound socket
by Simon Ditner 11 Nov '13

11 Nov '13

I am trying to use outbound socket to set up a conference, unfortunately I can not get my bind_digit_action calls to work in-conference. My setup is pretty boiler-plate: autoload_configs/conference.conf.xml <configuration name="conference.conf" description="Audio Conference"> ... <profiles> <profile name="simonconf"> <param name="rate" value="8000"/> <param name="interval" value="20"/> <param name="energy-level" value="300"/> <param name="caller-controls" value="none"/> </profile> </profiles> </configuration> dialplan/default.xml <include> <context name="default> <extension name="simon1"> <condition field="destination_number" expression="^800$"> <action application="bind_digit_action" data="random_context,*3,exec:playback,ivr/ivr-welcome_to_freeswitch.wav" /> <action application="digit_action_set_realm" data="random_context"/> <action application="answer"/> <action application="conference" data="random_conference@simonconf "/> </condition> </extension> <extension name="simon2"> <condition field="destination_number" expression="^900$"> <action application="bind_digit_action" data="random_context,*3,exec:playback,ivr/ivr-welcome_to_freeswitch.wav" /> <action application="digit_action_set_realm" data="random_context"/> <action application="socket" data="127.0.0.1:8888 async full"/> </condition> </extension> </context> </include> Listening on port 8888, I have a server which issues: connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: conference execute-app-arg: random_conference@simonconf event-lock: true Calling into simon1, the *3 will work. Calling in to simon2, *3 will not work. If I do not join the user into the conference, *3 works in both setups. Is there any particular reason that *3 would not work in the simon2 setup? Thanks, spd

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enabling tls cause sip_profile error
by adahary 10 Nov '13

10 Nov '13

I have enabled tls/ssl in vars.xml and restarted freeswitch. when reloading mod_sofia all profiles got into errors. I have read that openssl-devel should be install before compiling FS. I did that before building FS without adding anything special for the ssl on the ./configure command (like --enable-zrtp). I'm about to repeat the build procedure but before that I would like to ask if there is something elese to be done beside installing openssl-devel (done) and # ./configure. regards assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/enabling-tls-cause-sip-profil… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Build Problem.
by dotnetdub 10 Nov '13

10 Nov '13

Hey All I am following the instructions to install from the wiki. Running Squeeze 32 bit for testing. git clone git://git.freeswitch.org/freeswitch.git cd freeswitch/ ./bootstrap.sh ./configure this step fails because of this line in configure AM_INIT_AUTOMAKE(libfreeswitch,0.1) If I remove that it fails at: AM_PROG_CC_C_O AC_PROG_LIBTOOL I cloned the repo and did the exact same install two weeks ago on Debian Squeeze and it worked so I copied the source tree from that machine to this machine and its compiling now as I type this.. I think there may be an issue at the moment with the git repo? Brian

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FS with SSL/TLS issues!
by mehroz 10 Nov '13

10 Nov '13

Hi, it seems like i am unable to get it done. Here is what i have done 1. created CA (Root) certificates: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org 2. created Server Certificate: ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in which i am listening.* 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in vars.xml . 4. When i try to Register 3CX on TLS over port 5061, all i see is : tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): tls/59.103.76.82:56863/sips tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): new secondary tport 0x7fbd94f53c70 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): new connection from tls/59.103.76.82:56864/sips. PLease help!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Diagnosing locked calls
by Alex Lake 10 Nov '13

10 Nov '13

Got a few calls that are seeming to last indefinitely. The last entries in the log are 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:92 sofia/internal/002347067124224(a)87.238.73.160 Standard REPORTING, cause: NORMAL_CLEARING 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:761 (sofia/internal/002347067124224(a)87.238.73.160) State REPORTING going to sleep 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/002347067124224(a)87.238.73.160) State Change CS_REPORTING -> CS_DESTROY 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_session.c:1338 Send signal sofia/internal/002347067124224(a)87.238.73.160 [BREAK] 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_session.c:1546 Session 4 (sofia/internal/002347067124224(a)87.238.73.160) Locked, Waiting on external entities I need a strategy to work out what it's waiting for!

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Taking audio samples from bug
by Anton 10 Nov '13

10 Nov '13

When I check the audio sample contained in switch_frame::data array, it always has the same value -1, regardless of the position in the array and sound level. In order to get these samples I use different applications containing the following line: frame = switch_core_media_bug_get_read_replace_frame(bug); and immediately after insert this code: switch_log_printf( SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "data[0]=%d\n", ((int16_t *)(frame->data))[0] ); then been rebuilt with the appropriate module and initiating its launch. After that, expect to see on the console different sample values, but it is always the same -1. Someone explain to me why?

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Spandsp tone_detect segment frequencies
by Nandy Dagondon 09 Nov '13

09 Nov '13

Hello everyone, I'm trying to fine tune call progress tone using tone_detect_start. I set spandsp.conf.xml: <descriptions log-level="2"> Log shows these lines: Tone segment f1=-1, f2=-1, duration=336ms How can I tell which frequency represents -1, 4 or 5? What is the frequency assignment of silence? Let's use North America as an example. /Nandy  <descriptor name="1"> <tone name="CED_TONE" description="ANS / ANSam"> <element freq1="2100" freq2="0" min="700" max="0"/> </tone> <tone name="SIT" description="Special Information Tone"> <element freq1="950" freq2="0" min="256" max="400"/> <element freq1="1400" freq2="0" min="256" max="400"/> <element freq1="1800" freq2="0" min="256" max="400"/> </tone> <tone name="RING_TONE" description="North America ring"> <element freq1="440" freq2="480" min="1200" max="0"/> </tone> <tone name="REORDER_TONE" description="North America reorder"> <element freq1="480" freq2="620" min="224" max="316"/> <element freq1="0" freq2="0" min="168" max="352"/> <element freq1="480" freq2="620" min="224" max="316"/> </tone> <tone name="BUSY_TONE" description="North America busy"> <element freq1="480" freq2="620" min="464" max="536"/> <element freq1="0" freq2="0" min="464" max="572"/> <element freq1="480" freq2="620" min="464" max="536"/> </tone> </descriptor>

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Re: [Freeswitch-users] Modify INVITE
by Aziz Bayd 09 Nov '13

09 Nov '13

2013/11/9, Aziz Bayd <bayd80(a)gmail.com>: > Hello. > > I have a question about FreeSWITCH configuration. > Can I edit or modify the INVITE message content? For example, remove > "To" or "Via" fields in the INVITE content. > If yes, how can I do? > Thanks. >

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Re: [Freeswitch-users] Errors during compiling from today's 1.2.stable build
by bedgar＠vseinc.com 09 Nov '13

09 Nov '13

Ken and Anthony, This is on a physical server that was already successfully running 1.2.14. It is CentOS 6.4. It was a fresh git to an empty directory. Only special thing was the configure requirement that I always do for CentOS. Brian Brian Sent from my Galaxy S®III -------- Original message -------- From: Ken Rice <krice(a)freeswitch.org> Date: 11/08/2013 6:29 PM (GMT-05:00) To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Errors during compiling from today's 1.2.stable build What platform is this on? On 11/8/13 5:57 PM, "bedgar(a)vseinc.com" <bedgar(a)vseinc.com> wrote: I had trouble building from a git of 1.2.stable this afternoon. bootstrap successful configure successful make – not so good /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable /bin/sh: fork: retry: Resource temporarily unavailable Had to kill the make with Ctrl-C make[392]: *** [/home/fsbuild/src/freeswitch/libs/apr-util/libaprutil-1.la] Interrupt make[391]: *** [core] Interrupt rsync’d from another server from a build that retrieved this morning and successfully installed after I did a git clean -d -f –x Successful - FreeSWITCH Version 1.2.14+git~20131108T143841Z~0339e80780~64bit (git 0339e80 2013-11-08 14:38:41Z 64bit) Did another git of the stable build and still had the same error. Did something get borked in stable today? Brian ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Audio Quality of recordings
by Scott Ross 09 Nov '13

09 Nov '13

I've developed a tool which leverages freeswitch event_socket to originate 2 calls (sofia/external/number@ip) whom then navigate an IVR, connect to the same conference bridge then take turns playing back a .wav while the partner call records channel input. The goal is to run the two recorded files against a PESQ library to monitor whether our conference bridges are performing as expected. Problem: I'm hearing poor channel quality (static, degradation) from the recordings and when I listen in on the calls while the test is running. My PESQ scores are coming out much lower than expected. Out of curiosity I tried asterisk and scores were about a point higher (2.9 vs 3.9) so I should be able to rule out the hardware/environment/destination lines I'm running it on. I need help improving freeswitch audio quality for my use-case. I know this is asked alot - I've read all the threads/wiki articles I can find and tried: loading only necessary modules, running a single codec (ulaw), I've tried ensuring my playback reference is also ulaw format without headers, I've tried running on several different OSs: Ubuntu 11.04, RHEL 6.8, RHEL 5.6. Even originating on two separate sofia profiles (although I don't think this should have mattered regardless). I prefer freeswitch over asterisk in all other aspects but need to solve this (important) quality issue. Let me know what outputs I can provide.

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