August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

link my freeswitch to external database
by Ahmed Alzaeem 02 Sep '16

02 Sep '16

Dear Dev , I’m newbies to freeswitch & fusionpbx i have : ============== CentOS <http://centos.org/> v7 Freeswitch v1.6 FusionPBX v4 mysql external database to be linked my with fusionpbx . ========== I’m really confused with the option from where do i tell fusionpbx to use the external db ? i already added that db from the gui during the installation but it didn’t work or at least wasn’t chosen to be the db of my freeswitch . what i have modified was the file : /etc/freeswitch/autoload_configs/switch.conf.xml i added line <!-- <param name="core-db-dsn" value="odbc://DRIVER=mysql;SERVER=112.32.33.4;UID=lsfusion;PWD=4424243!;DATABASE=fusionpbx;OPTION=67108864"> —> but seems didn’t work . I’m really don’t know where to tell my server to use the external mysql db hope to help cheers

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Hangup hooks in Python
by Trever L. Adams 02 Sep '16

02 Sep '16

I am working on adding functionality to an external project that already supports Asterisk. I have used hook events with lua to capture call data on hangup and hangup complete before. Unfortunately, I need authenticated json rpc or xmlrpc. The original code I am working from is also in Python. Lua xml and json rpc modules do not seem to do authenticated requests. Since I already have Python, it seems to be easier. https://wiki.freeswitch.org/wiki/Mod_python#Hangup_hook_.2B_SQLAlchemy_cras…, it says that hangup hooks are very buggy and should be avoided. Is this still the case in the last year or so (recent versions)? If so, what is the best way to do this? If not, can someone post a simple equivalent of event:getHeader in a simple handler for hangups so that I can start working on this? Do I use def fsapi(session, stream, env, args) or def handler(session, args)? Thank you for help getting started, Trever

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WebRTC/Verto API?
by Fred Pettersson 01 Sep '16

01 Sep '16

Hi guys, I am new to Freeswitch so please go easy on me. :) I've bought the Freeswitch 1.6 Cookbook but I am looking for an API specification on the Verto API, where can I find information on the Verto API, what methods there are and it's parameters, how do I configure WebRTC/Verto in Freeswitch etc.? Any pointers are welcome. /Fred

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Re: [Freeswitch-users] No ringback with loopback channels and bowout_on_execute=false
by Luis Azedo 01 Sep '16

01 Sep '16

Hi, there is a pending pull request trying to fix that. https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/937… try to apply it and see if it works for you.

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Audio/video lags for some participants in Verto videoconference
by Chad Phillips 31 Aug '16

31 Aug '16

I have an occasional issue where a user’s video stream will lag behind their audio stream in a videoconference. This doesn’t happen to all users, usually just one or two, and the lag gets worse as time passes, sometimes upwards of 20-30 seconds behind the audio — quite distracting :) Curious if anyone else has run into this, and any possible remedies. I did see one suggestion on a forum that it could be related to a user’s browser plugins. Chad

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how to disable updates of sip_registrations.ping_expires when I am not doing nat
by Dave Horton 31 Aug '16

31 Aug '16

I’m observing intermittent errors of the type: "Deadlock found when trying to get lock; try restarting transaction..” with a sql update of the ping_expires column in sip_registrations table. I am not using OPTIONS ping or any nat features at all (I’m behind an SBC that does that), so I am not sure if there is any reason why the ping_expires value should be updated at all. In looking at the code (v1.6.8) in sofia.c it appears that the sofia_reg_check_ping_expire function will always be called, and that within that function the update to ping_expires will always be attempted, regardless of my profile settings. Am I reading that correctly, or is there a way to disable it? Dave

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Voicemal ring count
by Ned Ludd 31 Aug '16

31 Aug '16

Is there a way to make a call go to voicemail if it's not answered after a certain number of rings? I can use "call_timeout" to send it after a certain length of time, but I want to do it after a certain number of rings. Extra credit if you can make it a variable, so I can set the number of rings per user. Thanks. -- ----- Ned Ludd

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No ringback with loopback channels and bowout_on_execute=false
by Antonio Silva 31 Aug '16

31 Aug '16

Hi, I've the following scenario: 1. originate to 200 2. 200 answers 3. transfer/bridge call to 2103 <-- missing ringback tone with transfer_ringback set to %(1500,3000,425.0) in 3, without ring tone there is no audio until 2103 answers the call. The command used to trigger the call is: originate {loopback_bowout_on_execute=false,loopback_bowout=false,ignore_early_media=false}loopback/200/local 2103 XML local Is it possible to do this? or i'm missing something.. I file a jira with more information: https://freeswitch.org/jira/browse/FS-9470 Thanks, António

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Question about "terminators" attribute for playback for httapi
by Lin, Xiaoping 31 Aug '16

31 Aug '16

I have a question about "termintaors" attribute for httapi playback tag. I did not see it in the online document and I did explore and plan to use it, but I am not sure it will be supported in the future. The book "FreeSwicth 1.2" did talk about it though. I would like to know that "terminators" will be kept as an attribute in the future? Many Thanks! ************* From online : https://freeswitch.org/confluence/display/FREESWITCH/mod_httapi <work>*ACTIONS*</work> <playback file name error-file action digit-timeout input-timeout loops asr-engine asr-grammar><bind action strip>*EXPR*</bind></playback> : Plays a file and optionally collects input. ATTRS: file : The file name : Parameter name into which to save result error-file : Error file to play on invalid input action : Change the new target url temp-action : Change url to submit to, just for the next loop digit-timeout : Pause to wait for digits after file plays (when input bindings are present) input-timeout : Pause to wait for more digits in a multi-digit input loops : Maximum number of times to play the file when input bindings are present asr-engine : Automatic speech recognition engine to use asr-grammar : ASR grammar to use ======== from the book FreeSwitch 1.2 playback playback plays a file and optionally collects input. It has the following attributes: * file: The path to the file to play * name: Param name to save result * error-file: Error file to play on invalid input * digit-timeout: Timeout waiting for digits after file plays (when input bindings are present) * input-timeout: Timeout waiting for more digits in a multi-digit input * loops: Maximum number of times to play the file (when input bindings are present) * asr-engine: Automated Speech Recognition (ASR) engine to use * asr-grammar: Automated Speech Recognition (ASR) grammar to use * terminators: The keys that you want to use to immediately stop and process the digits collected

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Detecting media when recording
by Gonzalo Gasca Meza 31 Aug '16

31 Aug '16

Im using ESL originate to record active calls: My command: originate{ignore_early_media=false,bridge_early_media=false,originate_timeout=70,bridge_answer_timeout=70,call_timeout=70,[leg_timeout=70],execute_on_answer_1='sched_hangup 18 ALLOTTED_TIMEOUT',RECORD_STEREO=true,execute_on_answer_2='record_session /var/www/html/recordings/HGWHML2RQP_ *44dcfdc0-styq-khxn-qdea-hmms-44dd089c6c*.wav'}sofia/gateway/twilio/+1111111111 handle_calls Is there a way to detect if during active call there is human voice (speech recognition)? If so keep recording file, otherwise discard the file? Using Freeswitch 1.7 Thank you

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