April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_python ESL
by Deepika Yadav 20 Apr '16

20 Apr '16

Hi, I want to change the speaking status of a conference member through a python script as: import ESL freeswitchcon = ESL.ESLconnection(freeswitch_ip, freeswitch_port, freeswitch_name) cmd = conferencename + member_id freeswitchcon.api("conference", str(cmd)) This api doesn't doesn't fired though "originate" command works Another option : import freeswitch new_api_obj = API() new_api_obj.executeString("conference "+conferencename+" "+member_id) but this can't be called, due to module "_freeswitch" import error, instead need to called via the dialplan that calls the handler, doesn't looks a good way to do it I am able to achieve this functionality in mod_java ESL simply using ; org.freeswitch.swig.API a = new API(new JavaSession()); String rt = a.executeString("conference "+conferencename+" mute "+member_id); How can I achieve this in python? Regards, Deepika

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'Call Pickup', replaces headers.. does FS handle them properly?
by Benjamin Cropley 20 Apr '16

20 Apr '16

According to sect. 7.1 of https://www.ietf.org/rfc/rfc3891.txt, when an INVITE is recieved with a replaces header, containing the call-id of another dialog, the receiving party needs to respond immediately with a 200OK containing the new INVITES call-id and cancelling the other call-id. In the below, Alice calls Bobs Desk but he's not there. He's in his lab, and hears the Desk ring, so does a dialoginfo call pickup.. Bob Bob Alice desk lab | | | *1 |-----INVITE----------->| | *2 |<----180---------------| Bob hears desk phone | | | ringing from lab but | | | isn't REGISTERed yet | | | | | |<--fetch dialog state --| | |---response ----------->| *3/4 |<-----INVITE with Replaces/200/ACK--------------| *5/6 |------CANCEL/200------>| | *7 |<-----487--------------| | |------ACK------------->| | | | | | | | I have a setup of OpenSIPS and FreeSWITCH. OpenSIPS loadbalances all INVITES to FreeSWITCH, and then any subsequent packets are just relayed on as a stateful proxy. With that in mind, see what happens if I try do the same.. https://pastebin.freeswitch.org/24660 - sip trace of.. 1. Alice calls bobs desk 2. Bobs desk rings 3. Bobs lab tries to pick up the call, so that bobs lab is the one speaking to alice and bobs desk stops ringing 4. Instead, Alice gets cut off and bobs desk ends up speaking to bob lab https://pastebin.freeswitch.org/24650 - FS logs of this issue As you can see, on line 19, Bobs lab sends an INVITE with a callid matching that of the one Alice established... FS sends a 200OK back.. All good so far. At this point I logically feel FS would then send a 200OK to the a dialog/Alice and bridge the two.. but instead it kills the dialog with alice and takes place directly with bobs desk.. tl;dr is FS handing replaces correctly here? Any advice would be much appreciated. Ben

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Freeswitch and opensips
by Nduwayezu, Joselyne 20 Apr '16

20 Apr '16

I'm using Opensips as load balancer in front of FreeSwitch. To tell Freeswitch that calls come from Opensips proxy, do i have to create a new external profile in sip_profiles directory or add an extension in dialplan/public.xml or both of two? Second question, in this file :/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml , i read that i have to specify the CIDR, is the ip address of Opensips? Thank you NDUWAYEZU Joselyne -- Haute École de Bruxelles

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No audio Freesswitch
by Deepika Yadav 20 Apr '16

20 Apr '16

Hi, I am using Amazon ec2 platfrom to run freeswitch application. On restarting I am not able to hear any audio both sides, I have configured the settings by following the steps given here: https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 I have allowed all traffic for inbound and outbound fro all IPs (Ec2 GUI security group) netstat -an | grep ':5060' gives tcp 0 0 172.31.22.224:5060 0.0.0.0:* LISTEN tcp6 0 0 ::1:5060 :::* LISTEN udp 0 0 172.31.22.224:5060 0.0.0.0:* udp6 0 0 ::1:5060 :::* netstat -an | grep ':5080' tcp 0 0 172.31.22.224:5080 0.0.0.0:* LISTEN tcp6 0 0 ::1:5080 :::* LISTEN udp 0 0 172.31.22.224:5080 0.0.0.0:* udp6 0 0 ::1:5080 :::* Any guidance to proceed would be highly thankful Regards, Deepika

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Conference Setup
by Deepika Yadav 20 Apr '16

20 Apr '16

Hi All, I want to set up a conference initiated from a python esl script. My Dialplans are: <extension name="conf_demo"> <condition field="destination_number" expression="^9099$"> <action application="conference" data="radioHealth_${strftime(%Y-%m-%d)}+flags{endconf}"/> </condition> <extension name="conf_demo"> <condition field="destination_number" expression="^9098$"> <action application="conference" data="radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/> </condition> Python script snippets: Calling the first person in unmute mode : freeswitchcon = ESL.ESLconnection('127.0.0.2', '8021', 'ClueCon') freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+" &conference(9099)" Calling the second persion freeswitchcon.api("originate","sofia/gateway/MySIP/91XXXXXXXXXX+" &conference(radioHealth_${strftime(%Y-%m-%d)}+flags{mute})" Two outbound calls are created but these are not bridged, I can't hear the voice of first person I tried bridge flags <extension name="conf_demo"> <condition field="destination_number" expression="^9098$"> <action application="conference" data="bridge :radioHealth_${strftime(%Y-%m-%d)}+flags{mute}"/> </condition> not sure if it is the correct way. I also tried calling from SIP internal account : freeswitchcon.api("originate","sofia/internal/1004@X>X.X.X.X:5080 4446") where 4446 transfers a call to the conference dialplan and called other members from the script to add them in the conference but in this case, I get logs as: switch_channel.c:1055 New Channel loopback/app=voicemail:default X.X.X.X 1004- loopback/app=voicemail:default X.X.X.X 1004-a setup codec opus/ and conference is not setup Any help would be appreciated Regards, Deepiak

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FS Logs show ERR msgs: not enough buffer space for required resample operation!
by Joel Serrano 20 Apr '16

20 Apr '16

Hi, We are occasionally seeing in our logs the following error: 7115ef69-3964-469c-b577-3127a9972bf6 2016-04-12 04:54:55.851240 [CRIT] switch_core_io.c:1255 sofia/-profile_name-/-number-@-carrier- not enough buffer space for required resample operation! 0b59a195-525e-49d4-8310-877d05c06849 2016-04-12 08:01:54.571182 [CRIT] switch_core_io.c:1255 sofia/-profile_name-/-number-@-carrier- not enough buffer space for required resample operation! c1dc692d-95b9-409d-b51e-75cc85c30e1f 2016-04-13 01:07:39.491176 [CRIT] switch_core_io.c:1255 sofia/-profile_name-/-number-@-carrier- not enough buffer space for required resample operation! 44ead897-9b00-48c6-825a-f267c8ac6fbd 2016-04-13 03:35:52.471176 [CRIT] switch_core_io.c:1255 sofia/-profile_name-/-number-@-carrier- not enough buffer space for required resample operation! I have searched and only found this thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2015-October/116298.… Does anyone know what can cause these type of errors? What buffer exactly is the one that is out-of-space? Can we increase such buffer size? FYI: FreeSWITCH (Version 1.6.6 -13-d2d0b32 64bit) Thanks, Joel.

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Verto demo STUN and TURN configs
by Arsen Manukyan 19 Apr '16

19 Apr '16

Guys please help with Verto demo STUN and TURN configs edit I cant found any helpful information

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mod_com_amd not "hearing" audio
by Michael Jepson 19 Apr '16

19 Apr '16

When placing a call and checking for an answering machine using mod_com_amd, it always sees *silent_state*. session:recordFile does not record any audio either, unless I set record_waste_resources=true on the session. I assume mod_com_amd is having the same problem as recordFile is, but there is no option to have mod_com_amd send RTP packages when listening for audio and determining whether we have a person or a machine on the line. How can I get mod_com_amd to listen to the audio? Any help would be greatly appreciated. Best regards, MICHAEL JEPSON LEAD DEVELOPER Michael.Jepon(a)CM.NL<mailto:Michael.Jepon@CM.NL> +31 (0)76 2 012 732 [cid:5BB267AD-C5CE-4941-B84C-466564FD1DF6@cm.local] KONIJNENBERG 30 4825 BD BREDA THE NETHERLANDS WWW.CM.NL<http://www.cm.nl/>

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mp4 Conference recording is always unplayable 352 bytes file
by David Wafula 18 Apr '16

18 Apr '16

I have verto-client connecting to a conference. I have setup auto-record pointing to an mp4 file format for mod_conference. I have installed libavcodec-extra, enabled mod_av. The recording seems to start and stop just fine, except the file is just an unplayable 352 bytes. I must have missed something, just can't to figure out what. Am running FreeSWITCH Version 1.6.7-14-d38d065~64bit (-14-d38d065 64bit) on Debian 8.1 Jesse, x64. Please anyone with pointers help. -- David W

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help, with Verto - verto-min.js configuration
by Arsen Manukyan 18 Apr '16

18 Apr '16

Please help, with Verto - verto-min.js configuration After changing username, password for STUN and TURN server looks like my TURN server doesn't receive any messages form Webrtc phone here is my config: STUN={url:!moz?'stun:turn.mydomain.com:3478':'stun:204.174.104.38'};var TURN={url:'turn:ars@tu rn.mydomain.com:3478 <http://rn.mydomain.com:3478>',credential:'ars123'};var iceServers=null;if(options.iceServers){var tmp=options.iceServers;if(typeof(tmp)== ="boolean"){tmp=null;} if(tmp&&!(typeof(tmp)=="object"&&tmp.constructor===Array)){console.warn("iceServers must be an array, reverting to default ice serve rs");tmp=null;} iceServers={iceServers:tmp||[STUN]};if(!moz&&!tmp){if(parseInt(navigator.userAgent.match(/Chrom(e|ium)\/([0-9]+)\./)[2])>=28)TURN={u rl:'turn:turn.mydomain.com:3478 <http://turn.mydomain.com:3478>',credential:'ars123',username:'ars'};

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