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C11 style in libyuv?
by Luis Daniel Lucio Quiroz 21 Apr '16

21 Apr '16

Hello Is it my imagination of I see c11 style in libyuv (the one bundled in fs 1.6.7)? This may discard older compilers. Ld

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mod_http_cache doesn't retrieve any file
by Alexander Mustafin 21 Apr '16

21 Apr '16

Hi there! I am testing mod_http_cache module and unfortunately it doesn’t work for me properly. I loaded module, and trying console commands such as ‘http_prefetch’ and others. A respond of server look like: freeswitch(a)fs.sip3.net> http_prefetch http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… +OK 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:718 Adding http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… to cache index 1 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:1069 opening /var/cache/freeswitch/c9/396ca9-b29f-4a97-b40d-cb1430a7f1b6.wav for URL cache 2016-04-20 12:22:38.810363 [ERR] mod_http_cache.c:1103 open() error: No such file or directory 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:565 Locked cache 2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:675 Failed to download URL http://download.wavetlan.com/SVV/Media/HTTP/WAV/Media-Convert/Media-Convert… 2016-04-20 12:22:38.810363 [DEBUG] mod_http_cache.c:576 Unlocked cache And server does not try to obtain file from the url, because I listen 80 port and I can’t see anything. Possibly the problem here '2016-04-20 12:22:38.810363 [INFO] mod_http_cache.c:653 Cache MISS: size = 1 (0 MB), hit ratio = 0/2’ but I don’t know what it’s about. Any clue for this problem? Best regards, Alexander Mustafin mustafin.aleksandr(a)gmail.com

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Re: [Freeswitch-users] Freeswitch and Ping
by Michael Jerris 21 Apr '16

21 Apr '16

sounds like a broken router to me On Wednesday, April 20, 2016, Clive Lansink <clive(a)lansink.co.nz> wrote: > > Hi everyone. I often use Freeswitch set up as a client to provide a soft > phone on my laptop. > > But recently I noticed a significant problem when at the same time I was > running a standard Windows "ping" in the background. The speech I heard > from the other end would often stutter, and the remote party complained > that my speech sometimes sounded gittery. > > I reasoned that ping doesn't take up much network bandwidth so while in > conversation I did something like copy a large file from my laptop to > another PC on the network and there was no disturbance to the conversation. > It only seems to be ping that causes this. > > Is there something about pinging that might impact on Freeswitch or VOIP? > What would cause this? > > Further information. > > the above refers to the standard Windows ping command. I also have a > Python script that pings a specified host, and running it also caused the > call quality to degrade. I tried various changes to the source code to no > effect, including: > * only creating the socket once and then reusing it instead of it being > created for each ping. > * Only creating the data packet once and reusing it instead of it being > created for each ping. > * Using a very short packet size. > * Eliminating the select statement and just sleeping for a period of time > before reading the response in case the select itself was at fault. > * Eliminating the step of dissecting and decoding the received packet. > None of these made any difference so it seems to me that whatever it is > must be something quite fundamental to something at the lower networking > level. > > Thank you. > > > Clive Lansink > Email: Clive(a)Lansink.Co.NZ <javascript:;> > Phone: +64 9 520-4242 > Mobile: +64 21 663-999 > Fax: +64 21 789-150 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <javascript:;> > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <javascript:;> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Freeswitch and Ping
by Clive Lansink 21 Apr '16

21 Apr '16

Hi everyone. I often use Freeswitch set up as a client to provide a soft phone on my laptop. But recently I noticed a significant problem when at the same time I was running a standard Windows "ping" in the background. The speech I heard from the other end would often stutter, and the remote party complained that my speech sometimes sounded gittery. I reasoned that ping doesn't take up much network bandwidth so while in conversation I did something like copy a large file from my laptop to another PC on the network and there was no disturbance to the conversation. It only seems to be ping that causes this. Is there something about pinging that might impact on Freeswitch or VOIP? What would cause this? Further information. the above refers to the standard Windows ping command. I also have a Python script that pings a specified host, and running it also caused the call quality to degrade. I tried various changes to the source code to no effect, including: * only creating the socket once and then reusing it instead of it being created for each ping. * Only creating the data packet once and reusing it instead of it being created for each ping. * Using a very short packet size. * Eliminating the select statement and just sleeping for a period of time before reading the response in case the select itself was at fault. * Eliminating the step of dissecting and decoding the received packet. None of these made any difference so it seems to me that whatever it is must be something quite fundamental to something at the lower networking level. Thank you. Clive Lansink Email: Clive(a)Lansink.Co.NZ Phone: +64 9 520-4242 Mobile: +64 21 663-999 Fax: +64 21 789-150

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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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