- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH 1.6 && FreeBSD
by Eric Masson 24 Nov '15

24 Nov '15

Hi, Instructions found in Confluence only need to be slightly amended to build Freeswitch 1.6 : https://freeswitch.org/confluence/display/FREESWITCH/FreeBSD pkg install autoconf automake curl git gmake jpeg ldns libedit libtool openssl pcre pkgconf speex sqlite3 wget sudo libsndfile lua52 opus libvpx mkdir ~/src cd ~/src git clone -b v1.6 https://freeswitch.org/stash/scm/fs/freeswitch.git cd freeswitch ./bootstrap.sh -j ## Edit modules.conf to disable ## applications/mod_fsv & endpoints/mod_verto ./configure gmake sudo gmake install cd-sounds-install cd-moh-install Regards. Éric Masson

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mod_skypopen
by danny k 24 Nov '15

24 Nov '15

Hi Everyone, I am having a issue with mod_skypopen. I have compiled freeswitch with mod_skypopen fine with no errors. After compiling I have started and confirmed the skype clients are running. From there I start freeswitch and run load mod_skypopen and here are the results of that : freeswitch@debian> load mod_skypopen 2015-11-24 10:03:34.396527 [INFO] mod_enum.c:880 ENUM Reloaded 2015-11-24 10:03:34.396527 [INFO] switch_time.c:1415 Timezone reloaded 1781 definitions 2015-11-24 10:03:34.436535 [WARNING] mod_skypopen.c:1759 [1.7.0+git~20151122T213812Z~08816959aa~32bit ] [WARNINGA 1759 ][skype101 ][IDLE,IDLE] STARTING interface_id=1 2015-11-24 10:03:34.636545 [NOTICE] mod_skypopen.c:1788 [1.7.0+git~20151122T213812Z~08816959aa~32bit ] [NOTICA 1788 ][skype101 ][IDLE,IDLE] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2015-11-24 10:03:34.636545 [NOTICE] mod_skypopen.c:1797 [1.7.0+git~20151122T213812Z~08816959aa~32bit ] [NOTICA 1797 ][skype101 ][IDLE,IDLE] Found a running Skype client, connected to its SKYPE API for interface_id=1, waiting 60 seconds for CURRENTUSERHANDLE==t3st3d 2015-11-24 10:04:34.736536 [ERR] mod_skypopen.c:1838 [1.7.0+git~20151122T213812Z~08816959aa~32bit ] [ERRORA 1838 ][skype101 ][IDLE,IDLE] The Skype client to which we are connected FAILED to gave us CURRENTUSERHANDLE=t3st3d, interface_id=1 FAILED to start. No Skype client logged in as 't3st3d' has been found. Please (re)launch a Skype client logged in as 't3st3d'. Skypopen exiting now +OK Reloading XML -ERR [module load file routine returned an error] 2015-11-24 10:04:35.736626 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** Please note that I have read through https://freeswitch.org/confluence/display/FREESWITCH/mod_skypopen Also I have ran install.pl

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Version 1.6.5-1 Issue with Mod_flite load
by Jonathan Hunter 23 Nov '15

23 Nov '15

hi Guys, Running on Centos 6.6 I have upgraded to version 1.6.5 and I cant load mod_flite, I now see; load mod_flite +OK Reloading XML -ERR [module load file routine returned an error] 2015-11-23 18:05:25.269356 [CRIT] switch_loadable_module.c:1520 Error Loading module /usr/lib64/freeswitch/mod/mod_flite.so **/usr/lib64/libflite.so.1: undefined symbol: snd_pcm_hw_params_set_channels** This is with flite 2.0.0-1.el6 and flite-devel.x86_64 . Has anyone else had this issue? Running previously on 1.4 was ok. Thanks Jon

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2 FS with shared database, cannot call subscribers on different servers
by Michael Nielsen 23 Nov '15

23 Nov '15

I've got 2 Freeswitch 1.6.2 setup on different servers: fs1.my-domain.com and fs2.my-domain.com. I've got a DNS record with 2 A-records for the servers. When two users ends up on he same server they can call each other. If they land on different servers they cannot. Then they endt up in voicemail. I'm running the servers with a shared database in postgresql. Should I do anything else besides db in the core for FS? They are registrering with the same "main" domain which is fs.my-domain.com.

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FreeSWITCH Week in Review (Master Branch) November 14th – November 21st
by Ken Rice 23 Nov '15

23 Nov '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1PVLSC6 FreeSWITCH Week in Review (Master Branch) November 14th – November 21st This week we had a number of features and a very important security fix listed below. It is highly recommended that you upgrade as soon as possible to avoid this vulnerability and you can find out more about the 1.6.5 release here. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! This week we have Martin O’Shield from Windy City SDR! And head over to freeswitch.com to learn more about FreeSWITCH support. Security issues: A bug allowing for a remotely exploited DoS attack through custom crafted network traffic via JSON has been fixed. We classify this issue as High Severity. A patch was added by Anthony Minessale in commit 4bdca81 to resolve this issue. All versions from 1.4.4 through the previous release are vulnerable. We highly recommend updating to the current release version as soon as possible.http://ift.tt/1S9TUXf New features that were added: FS-8543 [mod_conference] Improve mute handling on conference and WebRTC FS-8546 [mod_conference][mod_verto] Make original video demo backward compatible with livearray-json-status FS-8529 [mod_conference] Added video-floor to personal canvas mode FS-8401 [verto_communicator] Added Speaker selection in settings model and video page and fixed model to modal FS-8545 [verto_communicator] Improve controls for screen share, fixed a read lock regression, do not allow video floor on a member with a reservation id set, and add missing code to deal with screen share part FS-8549 [mod_http_cache] Add support for AWS_ACCESS_KEY_ID and AWS_SECRET_ACCESS_KEY environment variables in S3 profiles FS-8547 [core] Add error log into stats to log when quality impacting events begin and end FS-8559 [mod_shout] Add “mpga” to the list of supported extensions Improvements in build system, cross platform support, and packaging: FS-8333 [build][Debian] Added mod_hiredis.deb The following bugs were squashed: FS-8537 [mod_lua] Fixed a segfault caused by passing nil to various lua functions FS-8527 [mod_conference] Do not send the video of last_video_floor_holder to video_floor_holder if the videos are related FS-8542 [verto_communicator] Fixed the tooltips of video controls FS-8053 [mod_conference][mod_sofia] Fix for WebRTC’s SDP containing a=sendonly for video, but the client still receiving the video stream FS-8553 [config] Include verto_contact into the dial-string in the samples FS-8556 [mod_verto] Screen shares are not recoverable so do not try FS-8293 [mod_verto] Fixed some regressions where speed test caused excessive downlink bandwidth The FreeSWITCH 1.4 branch had this week’s previously mentioned security fix and a bug fix back ported as well as the release of 1.4.26. And again, keep in mind that 1.4 is quickly moving toward end of life and won’t be supported any longer except for high level security issues. FS-8537 [mod_lua] Fixed a segfault caused by passing nil to various lua functions

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FreeSWITCH, HipChat, Packages, Communications and more!!
by Ken Rice 23 Nov '15

23 Nov '15

Hey FreeSWITCHers, There has been a lot of discussion over the past several weeks reguarding FreeSWITCH and 1.4 going end of life, and packaging for 1.6 on various platforms. To help facilitate communications, the FreeSWITCH Team uses HipChat at https://hipchat.freeswitch.org/ You too can join us on the FreeSWITCH Hipchat server, just use the email address and password you created for your Jira account, and set your hipchat server for hipchat.freeswitch.org (if you are using hipchat desktop client they now support multiple profiles and servers so you can use both!) There are several rooms on the HipChat server that may be of interested to you. 1) FreeSWITCH 2) FreeSWITCH Packaging 3) ClueCon channels for the weekly call and many more! Use this rooms for topic specific discussions to interact with the community and the Core FreeSWITCH Devs and let's work together to address some of these packages! Want to join via the web without signing up for an account, but want to chime in on package discussion? Check in at https://hipchat.freeswitch.org/gSPuUxI0F Come join us and help keep things moving forward! K

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[USER_NOT_REGISTERED] after Answer
by Peter Steinbach 23 Nov '15

23 Nov '15

Hello, I have a strange phenomenon with a phone (snomM700 DECT) behind NAT: * phone 43 is ringing (local IP is 192.168.0.56, remote IP is 195.71.aaa.bbb (anonymized) * phone 43 is answering the call * we immediately get [USER_NOT_REGISTERED] Register string is as follows <sip:43@192.168.0.56:1506 0;line=7045;fs_nat=yes;fs_path=sip:43@195.71.aaa.bbb:5060;line=7045> Here is the Freeswitch log starting with phone's SDP: 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] sofia.c:6760 Remote SDP: 3c74476e-1ce3-46d0-874d-e75bbebce8ce v=0 3c74476e-1ce3-46d0-874d-e75bbebce8ce o=43 4408 4408 IN IP4 195.71.aaa.bbb 3c74476e-1ce3-46d0-874d-e75bbebce8ce s=- 3c74476e-1ce3-46d0-874d-e75bbebce8ce c=IN IP4 195.71.aaa.bbb 3c74476e-1ce3-46d0-874d-e75bbebce8ce t=0 0 3c74476e-1ce3-46d0-874d-e75bbebce8ce a=sendrecv 3c74476e-1ce3-46d0-874d-e75bbebce8ce m=audio 50030 RTP/AVP 8 3c74476e-1ce3-46d0-874d-e75bbebce8ce a=rtpmap:8 PCMA/8000 3c74476e-1ce3-46d0-874d-e75bbebce8ce a=ptime:20 3c74476e-1ce3-46d0-874d-e75bbebce8ce a=maxptime:80 3c74476e-1ce3-46d0-874d-e75bbebce8ce a=rtcp:50031 3c74476e-1ce3-46d0-874d-e75bbebce8ce 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] sofia.c:6750 Channel sofia/internal/43@192.168.0.56:15060 entering state [ready][200] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4150 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4205 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4150 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4150 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[SPEEX:99:8000:20:24600:1] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4150 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4150 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G729:18:8000:20:8000:1] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:2897 Set Codec sofia/internal/43@192.168.0.56:15060 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_codec.c:111 sofia/internal/43@192.168.0.56:15060 Original read codec set to PCMA:8 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:4472 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:6019 AUDIO RTP [sofia/internal/43@192.168.0.56:15060] 144.76.47.185 port 12664 -> 195.71.aaa.bbb port 50030 codec: 8 ms: 20 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_rtp.c:3753 Starting timer [soft] 160 bytes per 20ms 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:6237 Activating RTCP PORT 50031 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_rtp.c:4132 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 50031 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_rtp.c:2443 Setting RTCP remote addr to 195.71.aaa.bbb:50031 2 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [DEBUG] switch_core_media.c:6348 sofia/internal/43@192.168.0.56:15060 Set rtp dtmf delay to 40 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.585575 [NOTICE] sofia.c:7714 Channel [sofia/internal/43@192.168.0.56:15060] has been answered 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.605579 [DEBUG] switch_channel.c:3757 (sofia/internal/43@192.168.0.56:15060) Callstate Change RINGING -> ACTIVE 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.625571 [NOTICE] switch_ivr_bridge.c:1416 Hangup sofia/internal/43@192.168.0.56:15060 [CS_CONSUME_MEDIA] [USER_NOT_REGISTERED] 3c74476e-1ce3-46d0-874d-e75bbebce8ce 2015-11-19 15:46:38.625571 [DEBUG] switch_ivr_bridge.c:1689 sofia/internal/43@192.168.0.56:15060 skip receive message [UNBRIDGE] (channel is hungup already) Anybody has a clue where this may come from? Freeswitch is rather new (from 03-Nov-2015). -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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AUDIO RTP REPORTS ERROR: [Missing local host]
by georgek 23 Nov '15

23 Nov '15

Hi , I have read the posts about this error and I can not solve the problem. I have installed FS on Windows Server 2008 R2 STD. Inbound calls are from Cisco Call Manager. I attach configuration files and log file with "sofia global siptrace on" and " console loglevel debug". Can somebody help me to resolve this this error ?. confFiles.zip <http://freeswitch-users.2379917.n2.nabble.com/file/n7596190/confFiles.zip> vars.xml <http://freeswitch-users.2379917.n2.nabble.com/file/n7596190/vars.xml> -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/AUDIO-RTP-REPORTS-ERROR-Missi… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How to enable Screen sharing on Verto Conf?
by Victor Medina 23 Nov '15

23 Nov '15

Hi guys! I been playing around with the verto conference demo on 1.6. How do I enable the screen sharing feature? -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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FS LTS version
by ik 23 Nov '15

23 Nov '15

Hello, It seems that every single year there is a new EOL (end of life) of FS version (e.g. 1.4 at this time) and a new version series that replaces it (e.g. 1.6). The problem is that it's not a simple act to update production servers. When you have a lot of servers (in different clients and locations) it's even harder. Many products, including Linux, and even Asterisk have LTS versions, and newer releases versions. The new releases has new features and way to do stuff, but the LTS get bug fixes and are stable for long period of time (usually 5 years). What does it take to have LTS version with FS, that the new releases will not stop, but EOL will be much longer for some of it's versions? Are there any plans for such things? How we as a community can help? Ido

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