October 2015 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Cannot install or make libvpx2-dev on MacBook Pro running OS X 10.10
by Sanjoy Ghosh 02 Mar '16

02 Mar '16

Hello, Need help. I have been trying to build FREESwitch on a MacBook Pro running 10.10.5. I can’t find an install for libvpx2-dev, nor can I download sources to build. How do I get around this? Thanks, Sanjoy

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[Freeswitch-dev] Soo you might have noticed that we got hit a burst of spam on the mailing list
by Ken Rice 22 Oct '15

22 Oct '15

Hey Guys, As you might have noticed, we just got hit with a burst of spam on the mailing list. This is something the FreeSWITCH Team takes very seriously. We have plugged the hole that let this happen. Please keep in mind that we do moderate the list to try and avoid things like this. 1) If you are a new user on the mailing lists your messages will be automatically moderated for a period of time until the list mods can reasonably determine that you are not a spam bot 2) If you spam the list you will be remoderated and all posts to the list reviewed before being allowed to go to the list. 3) Repeated offenders will be banned from the list. Thanks Ken

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[Freeswitch-dev] testers wanted for low latency voip/video conferencing qdisc
by Dave Taht 18 Oct '15

18 Oct '15

Dear Freeswitch folk: It is generally my hope that many of you have been using things like fq_codel (now the default in fedora, and many other linux derived OSes) and qos systems like the "sqm"[1] system now in openwrt, and streamboost, etc in order to improve voip and videoconferencing behavior along the edge, and beat bufferbloat. For the past 3 years we have been working on a successor to fq_codel, called "cake" - which among many other improvements like 8way set associative hashing, and less cpu use, also obeys what we know of the defined diffserv markings for voip and video conferencing - not that we think they are widely used as yet. We are finally at the point where cake can be tested more widely with an out of tree kernel module, and patches to tc. A complete technical description of cake, links the github repos, and how to install are at: http://www.bufferbloat.net/projects/codel/wiki/CakeTechnical And we hope to submit to kernel mainline after more testing gets done. We have many tests (see flent.org for the flent tool) for voip but we have yet to develop good ones for videoconferencing, and in addition to any testing of cake you can do, I would love to know how we too can add tests for that to our test suites. I had a chance last friday to come on the vuc show and talk about fq_codel, cake, bloat, etc on the latter half, and the first half was dominated by our controversial proposal to fix the edge of the internet. - but my heart is in seeing the technical fixes for better voip and videoconferencing get more widely used so we can finally deploy them! https://www.youtube.com/watch?v=yGYZ83nq4Js http://www.businesswire.com/news/home/20151014005564/en (for the letter we were talking about) And my group of bufferbloat folk plan to be on the cluecon show nov 18th. More freeswitch data to toss around - logs - packet captures - empirical evidence - anything would be great to have before then. TIA. [1] http://wiki.openwrt.org/doc/howto/sqm -- Dave Täht Do you want faster, better, wifi? https://www.patreon.com/dtaht

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[Freeswitch-dev] Full Video/Video Conference support on Windows?
by Alexander Haugg 16 Oct '15

16 Oct '15

Hi All I'm working signs some weeks with the actual FreeSwitch (1.7...) and the new video features on Ubuntu Linux (well done, thanks!). But (unfortunately) I need this on windows too. Yesterday I was building the newest git version on windows with VS 2015 as x64 release version without problems, but without the video relevant projects (mod_vpx and other). I'd found this jira ticket. video conference block in windows<https://freeswitch.org/jira/browse/FS-8159> I there a plan or a roadmap for a presumptive do date? Thanks a lot!

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[Freeswitch-dev] TALK NOTALK Events not Fired
by Ronnie Beck 13 Oct '15

13 Oct '15

Hi All, I have been looking into the events TALK and NOTALK. I don't see them firing and I am not sure why. I have added the following variables to the dialplan: <action application="export" data="fire_talk_events=true"/> <action application="export" data="fire_not_talk_events=true"/> In my module, I am trying to catch the events by binding an event handler here like so: switch_event_bind("CALLER TALKING", SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, vad_start_talking , NULL); switch_event_bind("CALLER NOT TALKING", SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, vad_stop_talking , NULL); I read a suggestion in the mailing list archives that one should use the event type "SWITCH_EVENT_CUSTOM" with a subtype of "conference::maintenance". I tried this also but I don't think my problem is that I am not catching the event but rather the event is simply never fired. Has anyone a working example of TALK and NOTALK event handling? Many thanks, Ronnie

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[Freeswitch-dev] Freeswitch ignoring BYE sent by LegA when ignore_early_media=True
by Sushil Tripathi 13 Oct '15

13 Oct '15

I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number. If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB, but if ignore_early_media is true in that case BYE sent by Leg A is ignored by freeswitch causing call to keep ringing at LegB. <extension name="SIP-TO-SIP"> <condition field="destination_number" expression="^91(\d+)$"> <action application="answer"/> <action application="set" data="hold_music=/srv/sounds/123.mp3"/> <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/> <action application="hangup"/> </condition> </extension> While debugging for the root cause of this issue we found that this bug is because of side-effect of commit-id "3e00d270d7477a63b3c2e845b5473e6e7e3f1cfc" which was added to fix bug https://freeswitch.org/jira/browse/FS-6006 . Now we are not sure whether to remove this patch or not. Can you please suggest how to proceed further. Thanks & Regards, Sushil Tripathi +91-9910700668 -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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[Freeswitch-dev] [mod_callcenter] dynamic queues
by Tomas Boros 13 Oct '15

13 Oct '15

Hello fs developers I am working on a feature for mod_callcenter, which will enable creation of queues via fs_cli. Queues right now can be created via xml or DB backend, but for a clustered environment I think it would be usable to create and destroy queues live. I am new to the code, so I need some help: 1. I found, that number of arguments for fs_cli is limited to 6. Is it possible to override this limit? 2. Do the freeswitch code has builtin variable validators? I found something like queue_set_config in the code, which looks like a it, but it is used in context with function switch_xml_config_parse_event, which loads and validates the data from XML files. I need to validate the values coming from arguments of the function. 3. Do you think this is simply a correct approach of creating queues, do this make sense :) ? Thank you, Thomas Boros

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[Freeswitch-dev] FreeSWITCH and Twilio
by Michael Nielsen 13 Oct '15

13 Oct '15

Hello everyone, I'm trying to setup FreeSWITCH with Twilio - to place outbound calls. I've created a SIP trunk in Twilio and are now trying to terminate calls via Twilio - all 10 digit calls should go this way (2 digit country code and 8 digit number - Danish numbers). This is my twilio.xml in sip_profiles/external: <include> <gateway name="Twilio-outbound"> <param name="username" value="MY-USER"/> <param name="password" value="MY-PASSWORD" /> <param name="proxy" value="MY-SIP.pstn.twilio.com"/> <param name="register" value="false"/> </gateway> </include> This is my 02_twilio.xml in dialplan/default/: <include> <extension name="international.twilio"> <condition field="${toll_allow}" expression="international"/> <condition field="destination_number" expression="^(\d{10})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/Twilio-outbound/$1"/> </condition> </extension> </include> When I try and make a call for a Danish number (for example 4512345678) I get a "Bad Request 400". This is my FS log: freeswitch@MY-SIP> 2015-07-23 07:19:45.745254 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/ 4561912233(a)MY-SIP.MY-DOMAIN.com [4af1a50e-5145-4ad8-8b9a-499a27e9e3ec] 2015-07-23 07:19:45.905238 [INFO] mod_dialplan_xml.c:635 Processing 4561912233 <4561912233>->4512345678 in context default 2015-07-23 07:19:45.905238 [NOTICE] switch_channel.c:1075 New Channel sofia/external/4512345678 [54267cad-615c-478a-9ecc-61950a995068] 2015-07-23 07:19:45.905238 [NOTICE] sofia.c:7543 Hangup sofia/external/4512345678 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2015-07-23 07:19:45.925245 [INFO] mod_dptools.c:3276 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2015-07-23 07:19:45.925245 [NOTICE] switch_channel.c:4747 Hangup sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1641 Session 47 (sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com) Ended 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/4561912233(a)MY-SIP.MY-DOMAIN.com [CS_DESTROY] 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1641 Session 48 (sofia/external/4512345678) Ended 2015-07-23 07:19:45.925245 [NOTICE] switch_core_session.c:1645 Close Channel sofia/external/4512345678 [CS_DESTROY]

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[Freeswitch-dev] Strange mod_portaudio permissions issue
by Mark Haun 13 Oct '15

13 Oct '15

Hi, I posted this to freeswitch-users but did not receive any replies. Probably no one but the developers have a chance to know what is happening, so I'm posting it here in the hope that it jogs someone's memory. It would be nice to understand it properly. --- I built freeswitch version 1.7.0+git~20150730T192909Z~ab7f83c654~32bit for my ARM-based home server (an NVIDIA Jetson board), with the intention of using mod_portaudio with a wireless USB headset for my main home phone. It works well when running freeswitch directly as root, but when I configured freeswitch to run at system startup, mod_portaudio could no longer find any audio devices. No problem, I thought, it's a simple permissions issue. I am using the example debian init script from the wiki (this is a Jessie install with SysV init rather than systemd). It runs freeswitch as user freeswitch, group daemon using the -u and -g command-line options. So I added "freeswitch" to the audio group, then used sudo to verify that I could play sound files with "aplay" as the freeswitch user. That worked, but it did not solve the mod_portaudio failure. Thinking there must be an issue with portaudio separate from ALSA, I fetched and built the "pa_devs" utility which enumerates and prints the available audio devices. (This is basically what mod_portaudio is doing on startup.) Running as the freeswitch user, that worked too. Hmmmm. I tried running freeswitch from the command line using sudo rather than the -u and -g options, i.e. $ sudo -u freeswitch /usr/local/freeswitch/bin/freeswitch -nc and that worked as well! Then I started experimenting with -u and -g and discovered that $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch allows mod_portaudio to load correctly, whereas $ /usr/local/freeswitch/bin/freeswitch -nc -u freeswitch -g daemon causes mod_portaudio to fail. Can anyone explain what's happening here? In both cases the process UID, GID, EUID, EGID, FUID, and FGID are the same (freeswitch:daemon) as displayed with "ps xao pid,uid,gid,euid,egid,fuid,fgid". Could this be a bug in freeswitch? There are a couple of threads from 2-3 years ago in the archives, where mod_portaudio permissions issues were never resolved and the solution remained a mystery. For the time being I guess this (not using -g) is a workaround, but I'd sure like to understand what is going on. Mark

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[Freeswitch-dev] Upper or Pass thru Registration
by Abraham Celik 12 Oct '15

12 Oct '15

Hi All, Ive googled and searched rough the forum, and i can’t see that anyone has successfully implemented Upper registration on Freeswitch. I see that Sangoma has implemented the feature and mentioned that they will upload it back in a change window, but that was a few years ago. Has anyone been successful? Regards, Abraham

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