- FreeSWITCH-users - FreeSWITCH Mailing Lists

Missed Calls with Hunt Group
by soapee01.fs＠stubbornroses.com 16 Jun '17

16 Jun '17

Hi, Here's the command that I'm running. originate user/102@domain &bridge(user/100@domain,user/101@domain) On FS version 1.2.22, if the user 100 answers the call, user 101 will not see a missed call notification. On FS version FS 1.6.5 if the user 100 answers the call, user 101 will show a missed call. Is there something in the docs I've missed? I'd really like to set it back to the old way, but it would be really cool if there's a variable somewhere I've missed that lets you choose the behavior. Thanks! James

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Re: [Freeswitch-users] Build Problem in VS2015
by krice＠tollfreegateway.com 16 Jun '17

16 Jun '17

Not sure whats you are doing incorrect here, but I have just built master, I use the built in git bits with VS2015, and then drop to a command prompt (via the team explorer tab, select branches, right click the repo and select open command prompt) Then git pull, git clean -fdx, git reset -hard origin/master , git pull >From here back to the solution explorer open the FreeSWITCH.2015 solution file and build as normal. I think you have something skewed there old ssl vs new ssl bits From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Adrian Worutowicz Sent: Thursday, June 15, 2017 4:44 AM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Build Problem in VS2015 Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ <https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/Free Switch/> I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian.

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Fax receive issue with t30 codec
by Hardik Patel 16 Jun '17

16 Jun '17

Hello, I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). Now I am trying to receive fax, my issue is when i try to send fax in softphone(Zoiper) from the log i am seeing that it is sending fax using t30 codec. and i am not receiving the fax at destination, is it because of codec, should it only work with t38 codec? if that is the issue than how am i be able to send the fax using t38 from zoiper? Here i am attaching the fs log with loglevel 9 and sip trace is also enabled. 127.0.0.2 => carrier/provider IP 123456789 => Fax number test(a)gamil.com => Email Address 127.0.0.4 =>UI IP Pastebin link:https://pastebin.freeswitch.org/view/9ec52715

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FreeSWITCH Week in Review (Master Branch) December 5th – December 12th
by Ken Rice 15 Jun '17

15 Jun '17

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1P8dyo3 FreeSWITCH Week in Review (Master Branch) December 5th – December 12th This week the verto communicator had some new updates to the administrator menu and the core added a new origination_audio_mode variable. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we have Italo Rossi and the Evolux call center team! And, head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8616 [verto_communicator] A new menu for moderator, added gain buttons, and removed the 3-dot-button, moving its behavior to member-name div FS-8632 [core] Add origination_audio_mode originate variable with options for sendonly, recvonly or sendrecv Improvements in build system, cross platform support, and packaging: FS-8293 [verto] Add quality level 0 to conference (default is 1) and fix some logic in auto bandwidth The following bugs were squashed: FS-8625 [core] Fixed a segfault caused by an external incoming call from Google Voice. FS-8642 [core] Fixed CF_VIDEO_READY being set on non-video calls FS-8603 [verto_communicator] Added device validation to prevent lost microphones after reset FS-8640 [verto_communicator] Don’t clear conference member reservation id on members that don’t have a reservation ID FS-8633 [mod_verto] Fix for the first verto to join a conference does not get “conference-livearray-join” event FS-8621 [mod_av] Fixed H264 HD1080P video quality issues FS-8631 [mod_db] Updated the regex to allow DSN to match the rest of FS code FS-8643 [mod_sofia] Fixed some memory leaks

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Build Problem in VS2015
by Adrian Worutowicz 15 Jun '17

15 Jun '17

Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ <https://stash.freeswitch.org/scm/fs/freeswitch.git%20/c/ESI/Components/Free Switch/> I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian.

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AMR Codec for Audio File Playback
by Ron 15 Jun '17

15 Jun '17

Hi All, Is it possible to execute a command playback for an audio file using AMR Codec? We tried the following configuration in dialplan and gave different errors: Configuration with “pre-answer”: <extension name="AMR_Testing"> <condition field="destination_number" expression="^\d{7}$"> <action application="pre_answer"/> <action application="sleep" data="2000"/> <action application="playback" data="/usr/local/freeswitch/sounds/aaaaa.wav"/> </condition> </extension> Error Log: EXECUTE sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 pre_answer() 2017-06-15 15:39:45.285763 [INFO] mod_dptools.c:1355 Sending early media 2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 AMR/0 0 ms 160 samples 12200 bits 1 channels 2017-06-15 15:39:45.285763 [DEBUG] switch_core_codec.c:111 sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 Original read codec set to AMR:96 2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:6927 PROXY AUDIO RTP [sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062] 192.168.1.129:62020->192.168.1.129:62020 codec: 98 ms: 20 2017-06-15 15:39:45.285763 [ERR] switch_core_media.c:7549 AUDIO RTP REPORTS ERROR: [Missing local host] 2017-06-15 15:39:45.285763 [NOTICE] switch_core_media.c:7550 Hangup sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Configuration with “answer”: <extension name="AMR_Testing"> <condition field="destination_number" expression="^\d{7}$"> <action application="answer"/> <action application="sleep" data="2000"/> <action application="playback" data="/usr/local/freeswitch/sounds/aaaaa.wav"/> </condition> </extension> Error Log: 2017-06-15 15:41:49.965732 [NOTICE] mod_dptools.c:1312 Channel [sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062] has been answered 2017-06-15 15:41:49.965732 [DEBUG] switch_channel.c:3772 (sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062) Callstate Change EARLY -> ACTIVE 2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 entering state [completed][200] 2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 entering state [ready][200] EXECUTE sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 sleep(2000) 2017-06-15 15:41:49.985728 [DEBUG] switch_rtp.c:7247 Correct audio ip/port confirmed. EXECUTE sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 playback(/usr/local/freeswitch/sounds/aaaaa.wav) 2017-06-15 15:41:51.985732 [DEBUG] switch_core_file.c:342 File /usr/local/freeswitch/sounds/aaaaa.wav sample rate 44100 doesn't match requested rate 8000 2017-06-15 15:41:51.985732 [WARNING] switch_core_file.c:360 File has 2 channels, muxing to 1 channel will occur. 2017-06-15 15:41:51.985732 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16@8000hz 1 channels 20ms 2017-06-15 15:41:52.005728 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode! 2017-06-15 15:41:52.005728 [ERR] switch_core_io.c:1434 Codec AMR encoder error! 2017-06-15 15:41:52.005728 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/aaaaa.wav 2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:385 sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 has executed the last dialplan instruction, hanging up. 2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/09570000001(a)192.168.1.129<mailto:sofia/internal/09570000001@192.168.1.129>:5062 [CS_EXECUTE] [NORMAL_CLEARING] We also tried to use the “fs_encode" and tried AMR codec to encode the wav file but we received the error below: Opening file aaaaa.wav Opening file aaaaa.AMR 2017-05-25 16:18:32.078353 [INFO] mod_native_file.c:101 Opening File [aaaaa.AMR] 8000hz Frame size is 160 2017-05-25 16:18:32.078366 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode! Codec encoder error 2017-05-25 16:18:32.078372 [WARNING] switch_core_codec.c:920 Codec is not initialized! We tried using the native file configuration and absolute codec but same errors shown for “pre_answer” and “answer” configuration. May we request your help if there is another way to run playback for audio files using AMR codec. Thank you. Best Regard, Ron Menez ron.menez(a)entropysolution.com<mailto:ron.menez@entropysolution.com>

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Paging
by Rick Jarvis 14 Jun '17

14 Jun '17

Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…?

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Re: [Freeswitch-users] RPORT still being sent in TCP calls
by Josh Nankin 14 Jun '17

14 Jun '17

Opened https://freeswitch.org/jira/browse/FS-10392 with regards to this On Wed, Jun 14, 2017 at 1:22 PM, Daniel Greenwald <dig1234(a)gmail.com> wrote: > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > Date: Mon, Jun 12, 2017 at 4:42 PM > Subject: Re: [Freeswitch-users] RPORT still being sent in TCP calls > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > Change is still in there, confirmed. The packages you are using are very > old (but should also have that patch in it)… I see that you are supplying > modified sip traces. If you can reproduce this on master code, please > create a Jira with configuration and full debug logs with sip trace > (unmodified) attached. > > Thanks > Mike > > On Jun 12, 2017, at 4:32 PM, Daniel Greenwald <dig1234(a)gmail.com> wrote: > > We have upgraded to latest STABLE from Ubuntu packages but we are still > seeing rport in TCP calls: > > > INVITE sip:xxxxxxx@sip.freeswitch.com;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH > Max-Forwards: 70 > From: "user" <sip:xxxxxx@sip.freeswitch.com>;tag=Z1pQm54mB22De > To: <sip:xxxxxx@sip.freeswitch.com;transport=tcp> > Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6 > CSeq: 108312748 INVITE > Contact: <sip:user@x.x.x.x:5060;transport=tcp;gw=provider> > User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit > > On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> Odd number releases are not stable releases they are dev releases. >> The version you quoted is from may 2015 2 years ago. >> >> The latest stable release is 1.6.17 and latest build release changes >> every day. >> >> >> >> On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald <dig1234(a)gmail.com> >> wrote: >> >>> We have noticed that FS is sending RPORT in TCP calls to a gateway. It >>> was reported as fixed in this bug: >>> https://freeswitch.org/jira/browse/FS-6612 >>> >>> We are running: >>> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit >>> >>> I am not sure why RPORT is still being sent. Is this there a config >>> parameter which needs to be set to suppress the RPORT? Or was this change >>> reverted in later versions for some reason. Provider is telling us we >>> should not be sending RPORT in TCP... >>> >>> Any info would be greatly appreciated. >>> >>> Thanks! >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.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 >>> >> >> >> >> -- >> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> <http://freeswitch.org/g+>* >> >> ClueCon Weekly Development Call >> ☎ sip:888@conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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 > >

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hello
by Josh Nankin 14 Jun '17

14 Jun '17

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B-Leg Early Media - Normal Clearing vs Originator Cancelled
by Sp Pho 14 Jun '17

14 Jun '17

Hi, I'm creating a bridge to terminate call arriving inbound from a DID. Everything works fine, however the termination point plays a recorded message - which as rightfully so by the switch is seen as early media (with pre-answer). However, when the originator hangs up while the early media is playing, the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the media does not play and ORIGINATOR_CANCELis the hang-up cause. Is there any way to receive the early media and from the CDR know whether or not the call was subsequently answered or not by the agent (i.e. if the originator canceled the call during the early media or while on hold)? Thanks in advance, SP

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