- FreeSWITCH-dev - FreeSWITCH Mailing Lists

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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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 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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[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 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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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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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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Freeswitch IAX2 Support
by Bharat Lalcheta 12 Oct '15

12 Oct '15

Hello All, I have one DIDWW DID configured with IAX2 protocol and routed to this server. When i am dialling this DID, i am getting calls on server on 4569 port which i checked in ngrep. However, there is no event on freeswitch. Just for your information, I configured mod_opal on both latest 1.2 and 1.4 version and found same result for both. Can anyone please guide me what i need to do additionally to get IAX2 calls on freeswitch? Or can anyone guide me how to troubleshoot this issue ? Do i need to configure anything for IAX2 ? Is there any thi ng i can test to check it is configured and working properly ? Thanks & Regards, Bharat Lalcheta

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About mod_java Console output (stdout/stderr)
by Pong Lam 12 Oct '15

12 Oct '15

Hi all, By default , System.out.println will output to stdout and throwing Exception will output to stderr. But I cannot find any output in the freeswitch.log, please advice? Regards Pong

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Safely Traversing call lists
by Ronnie Beck 12 Oct '15

12 Oct '15

Hi All, I have just a small question about the thread safety of the following code: switch_console_callback_match_t *callList = switch_core_session_findall(); for( node = callList->head; node; node = node->next ) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "(%s) Checking session '%s'\n", __FUNCTION__, node->val); } I just want to check some specific channel variables on all the active calls on a Freeswitch server. However it seems to me that while traversing the list, it is possible that "node" could be deleted by freeswitch while my code is still using it. Is there some sort of protection I should be doing in order to safely traverse the list and investigate each session? I thought I could lock each session as I interrogate the session however this doesn't protect against the situation where the "next" node is deleted as my code switches to that node. In C++, if you traverse a list with an iterator, changes to the list invalidate the iterator. So one must obtain a new iterator each time the list is changed. But I don't see how to do this with the list provided by switch_core_session_findall(). Any tips? Cheers, Ronnie Beck

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