April 2020 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using channel variables as values to time of day attributes in conditions
by Chad Phillips 09 Apr '20

09 Apr '20

Will this dialplan XML condition work in newer versions of FreeSWITCH? <condition wday="${REFORWARDING_DAYS}" hour="${REFORWARDING_HOURS}" break="never"> ${REFORWARDING_DAYS} and ${REFORWARDING_HOURS} are dialplan variables I set prior to calling the extension with that condition. I tried this on an older version of FreeSWITCH and it doesn't seem to work, I end up having to hard code the values for the 'wday' and 'hour' attributes.

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Accessing the Identity header in the dialplan
by Alexandru Tripon 09 Apr '20

09 Apr '20

Hello, I wish to kindly ask how can I access the "Identity" header. I set the following variable to true in my sip profile:`<param name="parse-all-invite-headers" value="true"/>` in order to access all invite header but without luck. The SIP invite header is the following: ` INVITE sip:1002@192.168.59.203 SIP/2.0. Record-Route: <sip:192.168.59.203;lr>. Call-ID: dd566f5f1c0c7f746226fdd374fdb2ae@0:0:0:0:0:0:0:0. CSeq: 1 INVITE. From: "1001" <sip:1001@192.168.59.203>;tag=1bcc4483. To: <sip:1002@192.168.59.203>. Via: SIP/2.0/UDP 192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0. Via: SIP/2.0/UDP 192.168.59.213:5060 ;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3. Max-Forwards: 69. Contact: "1001" <sip:1001@192.168.59.213:5060 ;transport=udp;registering_acc=192_168_59_203>. User-Agent: Jitsi2.11.20200408Linux. Content-Type: application/sdp. Content-Length: 897. Date: Thu, 09 Apr 2020 10:11:24 GMT. Identity: eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9jZXJ0LmV4YW1wbGUub3JnL3Bhc3Nwb3J0LmNlciJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyI8bnVsbD4iXX0sImlhdCI6MTU4NjQyNzA4NCwib3JpZyI6eyJ0biI6IjxudWxsPiJ9LCJvcmlnaWQiOiIxMjM0NTYifQ.MEQCIH9KqTnrIW6pAlCqrYbHEFoCF3AsS6crh4ptSEzfnOM_AiAjZrZJ2Ur1csuYvJN8NygYEg1N-3RseLSoNHQOEX_Z5w;info=< https://cert.example.org/passport.cer>;ppt="shaken". Test-Header: mytest. ` And the message log I get is : ` variable_sip_i_from: ["1001" <sip:1001@192.168.59.203>;tag=1bcc4483] variable_sip_i_to: [<sip:1002@192.168.59.203>] variable_sip_i_call_id: [dd566f5f1c0c7f746226fdd374fdb2ae@0:0:0:0:0:0:0:0] variable_sip_i_cseq: [1 INVITE] variable_sip_i_max_forwards: [69] variable_sip_i_contact: ["1001" <sip:1001@192.168.59.213:5060 ;transport=udp;registering_acc=192_168_59_203>] variable_sip_i_user_agent: [Jitsi2.11.20200408Linux] variable_sip_i_date: [Thu, 09 Apr 2020 10:11:24 GMT] variable_sip_i_content_type: [application/sdp] variable_sip_i_content_length: [897] variable_sip_i_via: [ARRAY::SIP/2.0/UDP 192.168.59.203:5060;branch=z9hG4bK2003.33316c51.0;received=192.168.58.201|:SIP/2.0/UDP 192.168.59.213:5060;branch=z9hG4bK-323138-463e2aa4a02196f588595ac028866af3] variable_sip_i_record_route: [<sip:192.168.59.203;lr>] variable_sip_i_test_header: [mytest] ` I tested this in latest release( https://github.com/signalwire/freeswitch/releases/tag/v1.10.2) and on master(commit: ae0444e9cbccdee55a80467d605e1e8c3363a36d) and the same thing happened: the Test-Header is present in the log but the Identity header is not. How can I access the "Identity" header in the dialplan? Thanks for your time, Tripon Alexandru

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execute_on_answer equivalent for A-leg or bridged call?
by Thilo-Alexander Ginkel 09 Apr '20

09 Apr '20

Hello everyone, I am attempting to play a sound on both legs after creating a bridge once the B-leg has answered. execute_on_answer explicitly only works for the newly answered channel, so no sound is playing on the (previously already connected) A-leg. Is there a way to make sure the sound is playing on both legs / the bridged call? ATM, I'm doing this (in Lua): session:execute("bridge", "{origination_callee_id_name='" .. callee_id .. "'}{execute_on_answer='playback tone_stream://%(500,0,500)'}user/jitsi@" .. domain_name) Thanks, Thilo

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Call not connecting
by kumar uppu 07 Apr '20

07 Apr '20

HI All, Freeswitch sending INVITE with following details INVITE sip:709@49.204.176.126:40067;ob SIP/2.0 Via: SIP/2.0/UDP 52.205.194.64:5080;rport;branch=z9hG4bK0KeS91cvXUjtm But some how while sending INVITE Registration contact is updating with Contact: <sip:709@49.204.176.126:40241;ob> Can any please tell me what i am missing? Thanks, Kumar -- Kumar

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SIP 302 Redirect
by Giri 07 Apr '20

07 Apr '20

Good Day, I have a scenario where the Outbound GW sends SIP 302 and all I need is to send the SIP 302 message to the Inbound. No Routing needs to take place, is this possible? With Best Regards, Girish Dharmaraj

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Problems with TLS after upgrading to Buster
by Walter Behrend 06 Apr '20

06 Apr '20

Hello there, hope someone else also had the problem - and found a solution for it. My "internal" profile has TLS enabled with tlsv1, 1.1 and 1.2 - this worked like a charm on stretch. I'm using the freeswitch-repos. I upgraded to buster and here my problems started. Seems the gentls_cert only creates SHA1 (CA)Certificates - so freeswitch started with openssl error messages "md too weak". Tried at first to bypass this error by setting the tls_ciphers to "DEFAULT:@SECLEVEL=0" but this error still occured. So as a consequence, I modified the gentls_cert script and replaced everywhere the parameter -sha1 with -sha256. This error disappeared now, but the next one is coming up. It seems it does not matter which value I set for "tls_version" - in every case, my TLS enabled port only accepts TLS 1.3 connections. I have the problem that we're also using older phones which only support TLS 1.0. Error message is: tport_tls.c:157 tls_log_errors() TLS setup failed: 14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol I tried with openssl s_client and the parameters -tls1 -tls1_1 and so on - it really only worked for -tls1_3 Any idea about this? setting tls_version to tlsv1,tlsv1.1,tlsv1.2 does not change anything. Also setting the value just to tlsv1 does not help, I verified this with the phones AND with openssl s_client. Still only TLS 1.3 gives results here. Thanks in advance... BR Walter

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MySQL setup
by Sean Devoy 06 Apr '20

06 Apr '20

Hi, I am running fs 1.10 on Debian 10 64bit. I have searched for help configuring FreeSwitch to use MySQL. I have configured db.conf.xml for Mysql: <param name="core-db-dsn" value="odbc://DRIVER=mysql;SERVER=serveriphere;UID=userhere;PWD=passwordhere;DATABASE=freeswitch;OPTION=67108864" /> I have reloaded, restarted and rebooted. FreeSwitch continues to use sqlite. Is there more to configure? Do I need to install unixODBC and/or MySql driver? Here is the error: 2020-04-05 20:57:48.159620 [ERR] switch_odbc.c:368 STATE: 01000 CODE 0 ERROR: [unixODBC][Driver Manager]Can't open lib 'mysql' : file not found 2020-04-05 20:57:48.159658 [CRIT] switch_core_sqldb.c:646 Failure to connect to ODBC DRIVER=mysql;SERVER=127.0.0.1;UID=xxxxxx;PWD=XXXXXXXXXX;DATABASE=freeswitch;OPTION=67108864! 2020-04-05 20:57:48.160130 [CONSOLE] switch_loadable_module.c:1804 Successfully Loaded [mod_db] 2020-04-05 20:57:48.160146 [NOTICE] switch_loadable_module.c:350 Adding Application 'db'

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How to connect SIP call to rtsp stream
by Victor Chukalovskiy 06 Apr '20

06 Apr '20

Good day, I'm trying connect incoming video call into rtsp stream (e.g. security camera h.264 video only or h.264 video + PCMU audio). I tried these two approaches: <action application="playback" data="av://rtsp://192.168.2.7/streaming/channels/103"/> or <action application="playback" data="rtsp://192.168.2.7/streaming/channels/103"/> And FS doesn't like it: [ERR] switch_core_file.c:306 Invalid file format [av] for [rtsp://192.168.2.7/streaming/channels/103]! or [ERR] switch_core_file.c:306 Invalid file format [rtsp] for [192.168.2.7/streaming/channels/103]! What is the right syntax to accomplish this? Also, which modules need to be compiled & loaded for this to work? Many thanks, -victor

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Re: [Freeswitch-users] Voice Conference Voice
by Ahmed habiba 03 Apr '20

03 Apr '20

Hello, Any one can help? "I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS”? Thanks, Ahmed Habiba. > > > From: Ahmed habiba <ahabiba(a)gmail.com> > Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba) > Date: 2 April 2020 at 5:04:27 PM GMT+3 > To: freeswitch-users(a)lists.freeswitch.org > > > Hi, > > Here is the Jitter calculated from my machine against the server using perf - the server is baremetal > > [ ID] Interval Transfer Bitrate Jitter Lost/Total Datagrams > [ 5] 0.00-20.00 sec 2.38 MBytes 1.00 Mbits/sec 0.000 ms 0/1786 (0%) sender > [ 5] 0.00-20.00 sec 2.38 MBytes 1000 Kbits/sec 0.321 ms 1/1786 (0.056%) receiver > > Perf result from server side: > [ ID] Interval Transfer Bandwidth Jitter Lost/Total Datagrams > [ 5] 0.00-20.09 sec 2.38 MBytes 996 Kbits/sec 0.321 ms 1/1786 (0.056%) > > > Client iperf command: > iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20 > > > Thanks, > Ahmed Habiba. > >> >> 1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli) >> >> From: Giovanni Maruzzelli <gmaruzz(a)gmail.com <mailto:gmaruzz@gmail.com>> >> Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy >> Date: 2 April 2020 at 1:23:29 PM GMT+3 >> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> >> Reply-To: gmaruzz(a)opentelecom.it <mailto:gmaruzz@opentelecom.it> >> >> >> If you are running in baremetal (eg, no VMs), then check packet loss/jitter... >> >> >> >> On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: >> Hello, >> >> I’m trying to use FreeSWITCH voice conference, however hear some cuts is the voice from time to time some minor distortions, I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS? Kindly advice how to overcome this issue. >> >> To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m using it for 3 to 4 users only, nothing else ring on this machine. >> >> Thanks, >> Ahmed Habiba. >> _________________________________________________________________________ >> >> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> >> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. >> Build your next product on our scalable cloud platform. >> >> Join our online community to chat in real time https://signalwire.community <https://signalwire.community/> >> >> Professional FreeSWITCH Services >> sales(a)freeswitch.com <mailto:sales@freeswitch.com> >> https://freeswitch.com <https://freeswitch.com/> >> >> Official FreeSWITCH Sites >> https://freeswitch.com/oss <https://freeswitch.com/oss> >> https://freeswitch.org/confluence <https://freeswitch.org/confluence> >> https://cluecon.com <https://cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org <mailto: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> >> https://freeswitch.com <https://freeswitch.com/> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> > >

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Re: [Freeswitch-users] Voice Conference Voice Choppy(Ahabiba)
by Ahmed habiba 02 Apr '20

02 Apr '20

Hi, Here is the Jitter calculated from my machine against the server using perf - the server is baremetal [ ID] Interval Transfer Bitrate Jitter Lost/Total Datagrams [ 5] 0.00-20.00 sec 2.38 MBytes 1.00 Mbits/sec 0.000 ms 0/1786 (0%) sender [ 5] 0.00-20.00 sec 2.38 MBytes 1000 Kbits/sec 0.321 ms 1/1786 (0.056%) receiver Perf result from server side: [ ID] Interval Transfer Bandwidth Jitter Lost/Total Datagrams [ 5] 0.00-20.09 sec 2.38 MBytes 996 Kbits/sec 0.321 ms 1/1786 (0.056%) Client iperf command: iperf3 -u -c 176.31.123.52 -b 1M -i 1 -t 20 Thanks, Ahmed Habiba. > > 1. Re: Voice Conference Voice Choppy (Giovanni Maruzzelli) > > From: Giovanni Maruzzelli <gmaruzz(a)gmail.com> > Subject: Re: [Freeswitch-users] Voice Conference Voice Choppy > Date: 2 April 2020 at 1:23:29 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: gmaruzz(a)opentelecom.it > > > If you are running in baremetal (eg, no VMs), then check packet loss/jitter... > > > > On Thu, Apr 2, 2020 at 11:53 AM Ahmed habiba <ahabiba(a)gmail.com <mailto:ahabiba@gmail.com>> wrote: > Hello, > > I’m trying to use FreeSWITCH voice conference, however hear some cuts is the voice from time to time some minor distortions, I look into code and I see something not sure if I’m right or wrong, when the conference loop is not able to read from input it sleep for 100 MS? Kindly advice how to overcome this issue. > > To Note, I’m using server with "Xeon E3-1245v2 (4c/8th) - 32GB” and I’m using it for 3 to 4 users only, nothing else ring on this machine. > > Thanks, > Ahmed Habiba. > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com <https://signalwire.com/> > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time https://signalwire.community <https://signalwire.community/> > > Professional FreeSWITCH Services > sales(a)freeswitch.com <mailto:sales@freeswitch.com> > https://freeswitch.com <https://freeswitch.com/> > > Official FreeSWITCH Sites > https://freeswitch.com/oss <https://freeswitch.com/oss> > https://freeswitch.org/confluence <https://freeswitch.org/confluence> > https://cluecon.com <https://cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto: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> > https://freeswitch.com <https://freeswitch.com/> > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > >

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