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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

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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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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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Help detecting voice mail
by Carlos Fererro 20 Oct '15

20 Oct '15

All, in the dial out call to the PSTN/mobile phone - what would be the right way to detect that the call got answered by the voice mail? I was hoping to see some kind of event on the event socket (I am using inbound event socket) but unfortunately didn't see anything. Of course, I might accidentally miss it too ... Once again, greatly appreciate any help on the subject Carlos

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systemd
by rick.jarvis＠magicmail.mooo.com 19 Oct '15

19 Oct '15

This is my first Jessie install, so trying to get my head around systemd. I’ve installed from repos, and it all works fine with ‘service start freeswitch’ as I used to use. However, as update-rc.d doesn’t work for starting on boot, I figured this is due to the startup changes in jessie. So I followed this to the letter: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video> But no joy: root@voice:/etc/systemd/system# systemctl start freeswitch Job for freeswitch.service failed. See 'systemctl status freeswitch.service' and 'journalctl -xn' for details. root@voice:/etc/systemd/system# systemctl status freeswitch.service ● freeswitch.service - freeswitch Loaded: loaded (/lib/systemd/system/freeswitch.service; disabled) Active: failed (Result: start-limit) since Mon 2015-10-19 21:47:53 BST; 26s ago Process: 1215 ExecStart=/usr/bin/freeswitch -u freeswitch -g freeswitch -ncwait -nonat -rp (code=exited, status=1/FAILURE) Oct 19 21:47:53 voice systemd[1]: freeswitch.service: control process exited, code=exited status=1 Oct 19 21:47:53 voice systemd[1]: Failed to start freeswitch. Oct 19 21:47:53 voice systemd[1]: Unit freeswitch.service entered failed state. Oct 19 21:47:53 voice systemd[1]: freeswitch.service start request repeated too quickly, refusing to start. Oct 19 21:47:53 voice systemd[1]: Failed to start freeswitch. Oct 19 21:47:53 voice systemd[1]: Unit freeswitch.service entered failed state. What am I doing wrong?

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FreeSWITCH Week in Review (Master Branch) October 10th-October 17th
by Ken Rice 19 Oct '15

19 Oct '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1NkQ7VN FreeSWITCH Week in Review (Master Branch) October 10th-October 17th Hello, again. This past week in the FreeSWITCH master branch we had 25 commits. Our feature this week in the verto communicator is the addition of ngSanitize as a dependency, vertoFilters module and picturify filter, and a change to the chat image display behavior (break line before rendering). Join this Wednesdays at 12:00 CT for some more FreeSWITCH fun with Daniel-Constantin Mierla! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8030 [verto_communicator] Added ngSanitize as a dependency, vertoFilters module and picturify filter and changed chat image display behavior (break line before rendering). Improvements in build system, cross platform support, and packaging: FS-8194 FS-7910 FS-7937 Various systemd service improvements FS-8350 [build] Fix Windows build errors. FS-7928 FS-7618 [Debian] Systemd and package build improvements The following bugs were squashed: FS-8328 [mod_conference] Fixed missing ‘else’ keyword FS-8306 [mod_amqp] If the exchange doesn’t exist, then create it, else fail. This resolves several error cases. And now command queues can specify the queue to subscribe to. This enables very interesting use cases that would involve single job queue, and multiple consumers. FS-8331 [verto_communicator] Do not show reconnect splash when user has clicked logout FS-8335 [mod_easyroute] Fixed a small error check that results in error message not being displayed FS-6833 FS-6834 [mod_sofia] Found and fixed a few missing content-types in requests/responses with SDP that were outside the norm FS-8338 [core] Fix for ringback not working correctly on stereo channels FS-7834 [mod_sofia] Fixed MOH not working with inbound-bypass-media and resume-media-on-hold FS-8287 [mod_local_stream] Fixed a segfault from refactor And, this past week in the FreeSWITCH 1.4 branch we had 2 new commits merged in from master. And the FreeSWITCH 1.4.23 release is here! Go check it out! The following bugs were squashed: FS-8338 [core] Fix for ringback not working correctly on stereo channels FS-8354 [mod_conference] Reverted a back ported patch for rate change detection because it introduced a regression that caused an audio issue

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link video from one call to the audio of another in a conference
by Hristo Trendev 19 Oct '15

19 Oct '15

Hi, Is there a way to somehow force the video of one call to follow the audio of another call so they act as if they are coming from a single endpoint? Essentially what I'm looking for is something like the "video-no-video-avatar" option, which can take a video stream instead of PNG. What I am trying to achieve is to use an existing Polycom Soundstation with a set of extension microphones and to add a video-only stream from a laptop webcam to the mix. I don't want to use the mic/speakers of the laptop though, because the Soundstation provides much better audio experience. Ideally it would be possible to dial into a conference from a Soundstation (audio only) and from a laptop (video only, with deaf and mute set) and be able to link both calls together so the laptop video follows the Soundstation audio with regards to floor changes. The Soundstation mics and speakers will then be used for audio streaming and playback and the laptop will stream and playback the video only. Best, Hristo

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