- FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_xml_curl performance questions
by Michael Avers 01 Jul '17

01 Jul '17

Hello, I'm planning on moving all our local lua-generated XML to use mod_xml_curl over HTTPS. However, the primary http server is not in the same data center as our FS instances. Ping time is about 50ms. Is this going to be an issue? Is it typical to run xml_curl hosted elsewhere? The web app itself is written in Go and is very fast. Just wondering about network latency aspect of it all. Thanks -Mike

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mod_rad_auth config file not found
by Jospeh Waite 01 Jul '17

01 Jul '17

Hi Guys Following some pointers from the mailing list a couple of weeks back I am attempting to configure and use mod_rad_auth to do radius authentication from the dial plan. I have compiled and installed the module and created the config file as per the confluence page, however the confluence page does not tell me where the config file should be located. I have tried in /usr/local/freeswitch/conf and also /usr/local/freeswitch/conf/autoload_configs However I get the following error 2017-07-01 15:08:27.024695 [ERR] mod_rad_auth.c:762 open of rad_auth.conf failed I have even tried looking at the source of the module but can’t seem to get an idea of where it should be Any pointers would be greatly appreciated. Regards

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Audio issue in Amazon EC2 server
by Ketan Kothari 01 Jul '17

01 Jul '17

Hi All, I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 <https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2>* Once we tried to call outbound call getting *no audio in both side* also FreeSWITCH passing my server local IP as RTP-IP to gateway side, =========================== 2017/06/29 12:57:06.579360 128.22.54.32:5060 -> 88.22.45.123:5060 v=0 o=FreeSWITCH 123456456 1245210017 IN IP4 *192.168.1.101* s=FreeSWITCH c=IN IP4 *192.168.1.101* t=0 0 m=audio 31010 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ============== 128.22.54.32 = Amazon EC2 server public IP *192.168.1.101* = Amazon EC2 server local IP 88.22.45.123 = Gateway provider. Could you please help me on audio issue in local-ip selected as RTP_IP and passing to gateway side?

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Use FreeSWITCH to solve a problem and win a prize!
by Anthony Minessale 30 Jun '17

30 Jun '17

This year at the ClueCon Coder Games there is a category for the most innovative use of FreeSWITCH to solve a problem. The prize is a bar top arcade. Get started now! Check out the other challenges and prizes on our web site: http://cluecon.com/coder-games.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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DTMF 2833 not working when switching to dynamic payload type 96
by John Narduchi 30 Jun '17

30 Jun '17

Having some trouble getting DTMF to work where an INVITE comes into FreeSWITCH offering a=rtpmap:101 telephone-event/16000. FreeSWITCH responds with a=rtpmap:96 telephone-event/8000 but then is unable to change the payload type to 96. This invite is a bit out of the ordinary as it does not specify telephone-event/8000. INVITE SDP *m=audio 63244 RTP/AVP 9 0 101* *b=AS:80* *a=rtpmap:101 telephone-event/16000* *a=fmtp:101 0-15* *a=ptime:20* *a=rtcp-xr:voip-metrics* *a=sendrecv* //fs logs after INVITE *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4192 Set telephone-event payload to 101@16000* *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4596 sofia/external/1231231234(a)X.X.X.X Set 2833 dtmf send payload to 101 recv payload to 101* FreeSWITCH then replies with the following SDP. *m=audio 33734 RTP/AVP 0 96* *a=rtpmap:0 PCMU/8000* *a=rtpmap:96 telephone-event/8000* *a=fmtp:96 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* When the digits come in to FS they are type 96, confirmed via tcpdump however FreeSWITCH doesn't recognize them and this entry shows up in the logs when digits are entered. *2017-06-29 19:29:02.975006 [WARNING] switch_core_media.c:2563 Could not change to payload type 96, ignoring...* *2017-06-29 19:29:02.975006 [DEBUG] switch_core_media.c:2540 alternate payload received (received 96, expecting 0)* Sofia.conf.xml and external.xml are vanilla. Have tested liberal-dtmf=true but no luck. Tested on version 1.6.10 and 1.6.18. Was hoping someone might have a setting to try or any suggestions? Thanks!

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may be fix DNS?
by Sergey Safarov 30 Jun '17

30 Jun '17

[safarov@safarov ~]$ ping files.freeswitch.org PING files.freeswitch.org(2607:f348:1021::7 (2607:f348:1021::7)) 56 data bytes >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No route >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No route

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best way to fork media streams for a bridged call?
by Dave Horton 29 Jun '17

29 Jun '17

If I have a bridged call and want to duplicate the two media streams and send each in a separate (one-way, sendonly) SIP dialog to a remote uri, what would be the best way to do that? It looks like it could possibly be achieved using the eavesdrop application (?), or is there an alternate preferred/tested method?

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FS service issues (systemd)
by Rick Jarvis 29 Jun '17

29 Jun '17

Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install]

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Callcenter module, can I originate call for an agent in uuid-standby mode ?
by Khalil Khamlichi 29 Jun '17

29 Jun '17

Hi, I need to give my agents ability to make manual calls, hopefully without leaving their actually established call (they are in uuid-standby mode and in Idle state so there is no live member on the line ). my questions: Is it possible to originate a new call and bridge with agent uuid-standby session ? would it not break the callcenter establised uuid-standby session ? would the agent return to its uuid-standby session after the originated call is hangup ? ofcourse if this is too complicated, I would just connect the agent thru a second line, while leaving his uuid-standby call on the first line, though it would be so cool to somehow stay on that same uuid-standby session and enjoy both calllcenter module and manual dialing. Thanks in advance, and I appreciate your help. Khalil

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SKYPE: mod_skypopen END-OF-LIFE
by Vladimir Gaitner 29 Jun '17

29 Jun '17

Hello Geowanni, Do you have any planns to rewrite the module?

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