May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

MS Lync / Skype For Business - MOH hangup
by jay binks 05 May '16

05 May '16

Hi All, Im hoping that the collective experience of the FS Community might be able to help with some interop fun with Skype For Business, specifically where the SFB end sends a bye at around 3 min, after the call has been on hold for ( for most of that time ) I dont see anything obvious in the signalling, but a colleague who is quite experienced with Lync feels it may be related to how we are handling "a=inactive" when the call is on hold. I cant at this point share any of these SIP traces ( mainly because im yet to have this entire scenario tested in our lab ). I guess what Im asking is, does anyone have SFB working well with Freeswitch, and have you tested long periods of silence ( pressing mute in the SFB client ) and long periods of the call on hold. if so, can you potentially share any settings you had to tweak in order to make this work. Sincerely Jay

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Message is less than minimum record length
by Brian May 05 May '16

05 May '16

Some reason my freeswitch system has suddenly stopped listening for messages. Instead it immediately cuts the talker off and says the message is too short. Any ideas? How do I debug this? i am going to conduct an audio test later on to make sure this is working, last I checked phone calls were working fine however. 2016-04-07 10:09:31.172293 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:1910 done playing file /var/lib/freeswitch/storage/voicemail/default/microcomaustralia.com.au/2000/greeting_1.wav 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2016-04-07 10:09:40.192379 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-record_message.wav] (en:en) 2016-04-07 10:09:40.232300 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL 2016-04-07 10:09:40.232300 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:44.792345 [DEBUG] switch_ivr_play_say.c:1910 done playing file /usr/share/freeswitch/sounds/en/us/callie/voicemail/vm-record_message.wav 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:559 Raw Codec Activated, ready to waste resources! 2016-04-07 10:09:45.912334 [DEBUG] switch_ivr_play_say.c:673 Raw Codec Activated 2016-04-07 10:09:45.912334 [DEBUG] switch_core_codec.c:221 loopback/voicemail-b Push codec L16:100 2016-04-07 10:09:47.912303 [DEBUG] switch_core_codec.c:246 loopback/voicemail-b Restore previous codec PCMA:8. 2016-04-07 10:09:47.932384 [DEBUG] mod_voicemail.c:1245 Message is less than minimum record length: 3, discarding it. 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [en] 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:250 Handle play-file:[voicemail/vm-too-small.wav] (en:en) 2016-04-07 10:09:47.932384 [DEBUG] mod_loopback.c:601 loopback/voicemail-b CHANNEL KILL 2016-04-07 10:09:47.932384 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms 2016-04-07 10:09:51.492336 [DEBUG] sofia.c:6858 Channel sofia/external/61419503953(a)203.31.79.18 entering state [terminated][487] 2016-04-07 10:09:51.492336 [NOTICE] sofia.c:7879 Hangup sofia/external/61419503953(a)203.31.79.18 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:707 sofia/external/61419503953(a)203.31.79.18 ending bridge by request from read function 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:780 BRIDGE THREAD DONE [sofia/external/61419503953(a)203.31.79.18] 2016-04-07 10:09:51.492336 [DEBUG] mod_loopback.c:601 loopback/voicemail-a CHANNEL KILL 2016-04-07 10:09:51.492336 [DEBUG] switch_ivr_bridge.c:701 sofia/external/61419503953(a)203.31.79.18 ending bridge by request from write function -- Brian May <brian(a)linuxpenguins.xyz> https://linuxpenguins.xyz/brian/

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How to do a conference that most parties cannot hear each other?
by Yehavi Bourvine 05 May '16

05 May '16

Hi, One of our users has a special request: He wants to make a conference (or a multi-participants phone call) which has the following behaviour: - Everyone hears him, - He hears everyone - All the other participants cannot hear each other. He needs this for some kind of a quick real time bid. Any idea whether this can be done and how? Today he simply holds 5 handsets and juggling among them. Thanks! __Yehavi:

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[ANN] RoCKSwitch ALPHA 0.99.2
by Richard Mace 04 May '16

04 May '16

Hi everyone, I'd like to announce that our app RoCKSwitch is now ready for further testing. RoCKSwitch is a Windows GUI application that creates .xml and uploads them to a FreeSWITCH box (tested on Debian), effectively, aiming to be a GUI for FreeSWITCH. It is fairly basic at the moment, but it supports multiple domains and should allow you to do the following: Add users Add Groups Add Dialplan entries Add Trunks (Gateways) Add Inbound Routes Add Outbound Routes And configure global settings. Please note that RoCKSwitch overwrites existing FreeSWITCH xml files with the ones that it creates itself, so please DO NOT use on production machines. We're really interested in all feedback (both good and bad) Please feel free to head over to our website www.rocksoftware.co.uk and download a copy. Many thanks in advance Richard Mace RoCKSoftware

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freeswitch-mod-esl removes freewtich-all
by Cj B 04 May '16

04 May '16

Hi, I just installed a new debian development server and I want to test mod-esl. When I try to do apt-get install freeswitch-mod-esl it warns me that freeswitch-all will be removed. Am I doing something wrong or does the package not updated yet? apt-get install freeswitch-mod-esl Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libfreeswitch1 Suggested packages: freeswitch-mod-esl-dbg libfreeswitch1-dbg The following packages will be REMOVED: freeswitch-all The following NEW packages will be installed: freeswitch-mod-esl libfreeswitch1 0 upgraded, 2 newly installed, 1 to remove and 0 not upgraded. uname -a Linux my.domain.com 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2 (2016-04-08) x86_64 GNU/Linux freeswitch -version FreeSWITCH version: 1.6.7-14-d38d065~64bit (-14-d38d065 64bit) Thanks, Cj B

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Freeswitch crashes
by Jurijs Ivolga 04 May '16

04 May '16

Hi, Not sure if this is a proper place, but. Today I updated my Freeswitch box from FreeSWITCH Version 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) to FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git f0c3870 2016-04-01 13:40:07Z 64bit). After today update Freeswitch started to crash when I was trying to make VP8 video call: May 4 14:20:39 FsDev kernel: [1737488.755526] freeswitch[10581]: segfault at 100000001 ip 00007f1d2ea15490 sp 00007f1d2d378790 error 4 in libvpx2.so.2.0.0[7f1d2ea0d000+1fa000] Audio call is working perfectly. I'm running Freeswitch on Debian 8. Any help will be appreciated. :) Jurijs

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dial-string pickup possible after group call?
by Jonathan Hunter 04 May '16

04 May '16

Hi Guys, Just wondering if its possible to use the user dial-string command to do call pickup after a call has been sent to a hunt group using the bridge and group_call function? So basically a call to group 5000, sends calls to extension 1000 and 2000 which are in the group. This all works fine but I want to be able to pickup the call by dialling *8 then either 1000 or 2000. I can do this if I am not using the group_call function, and its just a normal inbound call to the extension I want to pickup, so just wondered if its possible to do a pickup of one of the extensions in the group? To start the group_call Im using in my dialplan; Dialing 5000 results in; <action application="bridge" data="${group_call(huntgroup1(a)test.com+A}"/> Which then rings both extension 1000 and 2000 which have the dial-string defined; <param name="dial-string" value="sofia/internal/${dialed_user}@test.com,pickup/${dialed_user}@test.com"/> If I add pickup after group_call, this allows be to pickup extension 5000, but I need to be able to create the pickup channel using the dial-string, am I doing this wrong or does anyone have an idea how to achieve this? And just to be clear pickup works fine for normal inbound calls, its just when implementing group_call. Many thanks Jon

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Join us at ClueCon 2016!
by Kathleen King 03 May '16

03 May '16

[image: Inline image 1] [image: Inline image 3]

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Starting Freeswitch during system boot
by William Colls 03 May '16

03 May '16

Freeswitch 1.6.7 built from source on debian 8 64 bit system. I have set up the script freeswitch/debian/freeswitch-sysvinit.freeswitch.init in /etc/init.d. When I run the script, I get the following message [ ok ] Starting freeswitch (via systemctl): freeswitch.service. but freeswitch doesn't start. There are no error messages in that I can find in any of the expected places (/var/log, /usr/local/freeswitch/log). Any suggestions will be greatfully received. Thanks for your time. William.

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Separate audio and video on WebRTC
by Agustí Ubalde 03 May '16

03 May '16

Hi all, Any idea to separate audio and video on WebRTC connections without vertio module? Thanks, *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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