October 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

rtp_use_dtls vs. webrtc_enable_dtls
by Johannes Singler 22 Oct '15

22 Oct '15

There are two channel variables whose names are very similar: rtp_use_dtls (only regarded when "true") and webrtc_enable_dtls (only regarded when "false") They are not mentioned in the Confluence wiki, both are used switch_core_media.c. So what do they mean exactly, and when should either of them be used (or could they be merged maybe)? Johannes

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Break unaswered bridge
by Alexey Melnichuk 21 Oct '15

21 Oct '15

This is simple dialplan that I try to implement <extension> <condition field="destination_number" expression="^\d+$" break="on-true"> <action application="bind_digit_action" data="conf,*1,exec:execute_extension,xfer"/> <action application="digit_action_set_realm" data="conf"/> <action application="conference" data="100@default"/> </condition> <condition field="destination_number" expression="^xfer$" break="on-true"> <action application="bind_digit_action" data="xfer,*1,?????????"/> <action application="digit_action_set_realm" data="xfer"/> <action application="digit_action_set_realm" data="xfer"/> <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' target_num 30000 #"/> <action application="bridge" data="user/${target_num}@${domain_name}"/> </condition> </extension> I have two problem with this dialplan 1. in secon `bind_digit_action` I can not use `exec` only `api`. e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does not work. Application executes only when I return from first `exec:execute_extension` 2. I can not cancel bridge before answer I can not use transfer because I do not whant leave conference (e.g. user has `end_conference=true` flag). I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}" -- kill_legs.lua local uuid = argv[1] local sql = ("select uuid from channels where call_uuid='%s' and uuid<>'%s'") :format(uuid, uuid) dbh:query(sql, function(row) local res = api:executeString("uuid_kill " .. row.uuid) end) But this is does not work because it kills only active channels, but for dial-string like `loopback/111/my.domain,[leg_delay_start=20]loopback/222/my.domain` second channels creates only after 20 sec. And because I may have more then one channels I can not use `originate_uuid` In documentation I found `break` application but as I say I can not use it. `uuid_break` api break only play audio.

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record_session - dtmf and silence
by Leonardo Lima Ribeiro 21 Oct '15

21 Oct '15

I have a lua script creates a session and do a lot of stuff… And I record this session using the record_session api command… The problem is, sometimes I have session:execute(‘send_dtmf’,’1234@250’)… And in the log, I can see that too, but not at the session wav file. Is it possible to have this dtmf’s at the wav file? The second problem is, sometimes I have a big silence, like 10 seconds of no sound, and at the session wav file I don’t have this silence interval, but it’s important to me… Could I have this silence at the session wav too? Thank you, Leonardo Ribeiro

1 0

In B2BUA mode, how can I delay answering first call
by Dan Edwards 21 Oct '15

21 Oct '15

I am using FreeSWITCH as a B2BUA to handle WebRTC calls and link them to a PBX that does not have WebRTC support. When the PBX calls a WebRTC user, FreeSWITCH answers the initial call, then initiates the call to the WebRTC user. This causes the first 2-4 seconds of the audio to be lost. Is there a way to have the first call wait until the second call is answered? Thank you, Dan

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Fwd: [SR-Users] Kamailio and FreeSwitch - ClueCon Weekly Conference Call
by Giovanni Maruzzelli 21 Oct '15

21 Oct '15

---------- Forwarded message ---------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Tue, Oct 20, 2015 at 8:32 PM Subject: [SR-Users] Kamailio and FreeSwitch - ClueCon Weekly Conference Call To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org>, " business(a)lists.kamailio.org" <business(a)lists.kamailio.org> On Wednesday, October 21, 2015, at 17:00GMT (12:00CT, 18:00 London, 19:00 Berlin), the Cluecon weekly conference call will focus on Kamailio and FreeSwitch. I will be joining the call, answering the questions about Kamailio and its options to integrate with FreeSwitch. Expect the FreeSwitch core developers to be around to handle the questions about their project. Participation is open for anyone, you can dial in for audio or video sessions using a SIP phone or webrtc capable browser: - sip:888@conference.freeswitch.org - https://cantina.freeswitch.org/vc More dial in options (e.g., PSTN) are presented at: - https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Kazoo changes on Freeswitch 1.6
by Luciano Augusto Rezende 21 Oct '15

21 Oct '15

Hello guys, We are building a platform using Kazoo, which uses Freeswitch in its core. With the recent release of Freeswitch 1.6 we started to study how to use it (and its neat video features) with Kazoo, but we are still uncertain because Kazoo uses (and recommends) a Freeswitch 1.4 fork with some changes they made. I found this task on Jira: https://freeswitch.org/jira/browse/FS-7776 And if I understood Luis Azedo's comment correctly the Kazoo's changes are already included in FS 1.6. Can anyone confirm that? Thanks Luciano Augusto Rezende lrezende(a)daitangroup.com

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Dialout problem
by stephen＠picardogroup.com 20 Oct '15

20 Oct '15

2 1

Re: [Freeswitch-users] record_session wav files are corrupted
by Ian de Sousa 20 Oct '15

20 Oct '15

Hi All, Just wondering if there has been a fix for this issue. Im running 1.4.19 and occasionally we are getting calls playing very quickly when played back.. then the last bit of the recording / call it switches to normal speed at about 1/2 way. testing so far (100 calls) seems to indicate unless we set an “answer” beforehand we get this issue 50% of the time. setting answer isn’t good as caller will get incoming call charge due to specific NGN numbers coming in. Ian DS ####### Do you have mod_sndfile loaded? Steve On 4 Apr 2013, at 16:06, fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> wrote: > On 04-02-2013 05:17, Steven Schoch wrote: >> On Tue, Apr 2, 2013 at 10:28 AM, <fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >> >>> If I try to import the >>> audio as raw format in audacity you can hear some high pitched >>> talking >>> that plays back very quickly. >> >> When you import as raw data, try changing the sample rate to 8000. >> >> The reason Audacity can't open the file is because the header, which >> specifies the encoding, channels, sample rate, etc. is damaged or >> missing. >> >> -- >> Steve >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> >> <http://lists.freeswitch.org/pipermail/freeswitch-users/2013-April/094369.ht…> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> 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> >> http://www.freeswitch.org <http://www.freeswitch.org/> > > Well why would the header to damaged or missing. I have updated the > config to the following and it still doesnt work. If I look at the > header info with windows by going to file properties it doesnt list > anything in the header info. The file is just a bunch of high pitched > noise if I dont import as raw. I can hear the audio fine when importing > raw and 8000hz rate. I am not sure what else I need to do to make this > work. > > I followed the steps listed on: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session> > > <extension name="bandwidth.10d" > > <condition field="destination_number" expression="^(\d{10})$" > > <action application="set" > data="sip_h_X-accountcode=${accountcode}" /> > <action application="set" data="sip_h_X-Tag=" /> > <action application="set" data="call_direction=outbound" /> > <action application="set" data="hangup_after_bridge=true" /> > <action application="set" > data="effective_caller_id_name=${outbound_caller_id_name}" /> > <action application="set" > data="effective_caller_id_number=${outbound_caller_id_number}" /> > <action application="set" data="inherit_codec=true" /> > <action application="set" data="record_sample_rate=8000" /> > <action application="set" data="RECORD_TITLE=Recording > ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}" > /> > <action application="set" data="RECORD_SOFTWARE=FreeSWITCH" /> > <action application="set" data="RECORD_STEREO=true" /> > <action application="set" data="RECORD_DATE=${strftime(%m-%d-%Y > %H:%M)}" /> > <action application="set" data="RECORD_ARTIST=FreeSWITCH" /> > <action application="set" data="RECORD_COPYRIGHT=(c) 2013" /> > <action application="set" data="RECORD_COMMENT=FreeSWITCH" /> > <action application="record_session" > data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" > /> > <action application="bridge" > data="sofia/gateway/customer1-bandwidth/$1" /> > </condition> > </extension> >

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Re: [Freeswitch-users] record_session wav files are corrupted
by Ian de Sousa 20 Oct '15

20 Oct '15

Hi All, Just wondering if there has been a fix for this issue. Im running 1.4.19 and occasionally we are getting calls playing very quickly when played back.. then the last bit of the recording / call it switches to normal speed at about 1/2 way. testing so far (100 calls) seems to indicate unless we set an “answer” beforehand we get this issue 50% of the time. setting answer isn’t good as caller will get incoming call charge due to specific NGN numbers coming in. Ian DS ####### Do you have mod_sndfile loaded? Steve On 4 Apr 2013, at 16:06, fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> wrote: > On 04-02-2013 05:17, Steven Schoch wrote: >> On Tue, Apr 2, 2013 at 10:28 AM, <fs at tcowan.net <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >> >>> If I try to import the >>> audio as raw format in audacity you can hear some high pitched >>> talking >>> that plays back very quickly. >> >> When you import as raw data, try changing the sample rate to 8000. >> >> The reason Audacity can't open the file is because the header, which >> specifies the encoding, channels, sample rate, etc. is damaged or >> missing. >> >> -- >> Steve >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> >> >> >> <http://lists.freeswitch.org/pipermail/freeswitch-users/2013-April/094369.ht…> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org <http://www.freeswitch.org/> >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/> >> http://www.cluecon.com <http://www.cluecon.com/> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> >> 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> >> http://www.freeswitch.org <http://www.freeswitch.org/> > > Well why would the header to damaged or missing. I have updated the > config to the following and it still doesnt work. If I look at the > header info with windows by going to file properties it doesnt list > anything in the header info. The file is just a bunch of high pitched > noise if I dont import as raw. I can hear the audio fine when importing > raw and 8000hz rate. I am not sure what else I need to do to make this > work. > > I followed the steps listed on: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session> > > <extension name="bandwidth.10d" > > <condition field="destination_number" expression="^(\d{10})$" > > <action application="set" > data="sip_h_X-accountcode=${accountcode}" /> > <action application="set" data="sip_h_X-Tag=" /> > <action application="set" data="call_direction=outbound" /> > <action application="set" data="hangup_after_bridge=true" /> > <action application="set" > data="effective_caller_id_name=${outbound_caller_id_name}" /> > <action application="set" > data="effective_caller_id_number=${outbound_caller_id_number}" /> > <action application="set" data="inherit_codec=true" /> > <action application="set" data="record_sample_rate=8000" /> > <action application="set" data="RECORD_TITLE=Recording > ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}" > /> > <action application="set" data="RECORD_SOFTWARE=FreeSWITCH" /> > <action application="set" data="RECORD_STEREO=true" /> > <action application="set" data="RECORD_DATE=${strftime(%m-%d-%Y > %H:%M)}" /> > <action application="set" data="RECORD_ARTIST=FreeSWITCH" /> > <action application="set" data="RECORD_COPYRIGHT=(c) 2013" /> > <action application="set" data="RECORD_COMMENT=FreeSWITCH" /> > <action application="record_session" > data="$${base_dir}/recordings/${domain_name}/outbound10d-${strftime(%m-%d-%Y-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav" > /> > <action application="bridge" > data="sofia/gateway/customer1-bandwidth/$1" /> > </condition> > </extension> >

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OS X Wiki Updates October 2015
by Mario G 20 Oct '15

20 Oct '15

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