February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to convert a json string returned from a webservice to a json object in lua
by Ashwin Rath 17 Feb '16

17 Feb '16

I am using LUA to POST to a webservice and i receive a json string . How do i convert this string to a json object ? -- Ashwin Kumar Rath

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how to sip uri call
by jungle Boogie 17 Feb '16

17 Feb '16

Hello All, I'd like my freeswitch server to make sip uri calls to various URI's. I see this in dialplan/default.xml:   If I enable this, I can call any sip URI but then I can't call an extension local to the FS instance nor out on the PSTN. I see in In the same file, I do see this: <action application="bridge" data="sofia/${use_profile}/$1(a)conference.freeswitch.org"/> But I think there must be a method to enable URI calling for any domain--not just explicitly listed URI. If there's a confluence page that already covers this, feel free to send the article to me so I can try it out. Thanks for any assistance! -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

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AMD
by Abaci B 16 Feb '16

16 Feb '16

Hi all, I know that in general Intel Xeon processors are recommended over AMD for FreeSWITCH, my question is if AMD is really bad or just not as good as Intel, I can get now a server with 4x AMD 6174 (48 Core total) for about the same price I would pay for a Dual Xeon 6 Core (12 core total), Should I go for the AMD which will give me 4x the amount of cores or stick to Xeon, my goal is to get the maximum amount of channels/calls (doing lua IVR with no transcoding) and conferencing. Any help, feedback, benchmarks or personal experience would be appreciated Thanks

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repeating: sofia_presence_sub_callback: endpt is internal
by Cj B 16 Feb '16

16 Feb '16

Hi everyone, I just recently moved a setup from 1.4.26 to 1.6.6 and I keep seeing the following repeat over and over in the logs. 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal 2016-02-16 14:03:38.940698 [DEBUG] sofia_presence.c:2908 sofia_presence_sub_callback: endpt is internal http://pastebin.com/nxi9bH0N <http://pastebin.com/nxi9bH0N> Calls appear to be completing properly and phones are registered. I was just wondering what this means and if there’s a way to stop it? uname -a Linux my.domainname.com <http://my.domainname.com/> 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u1 (2015-12-14) x86_64 GNU/Linux /usr/local/freeswitch/bin/freeswitch -version FreeSWITCH version: 1.6.6+git~20160111T201612Z~d2d0b3283a~64bit (git d2d0b32 2016-01-11 20:16:12Z 64bit) Thanks, Cj B

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Re: [Freeswitch-users] problems with bridging a call, looks like transcoding is disabled
by Roman Kudinov 16 Feb '16

16 Feb '16

Hi Brian, I saw this variable but looks like used it the wrong way. Thanks for the help, the codecs negotiation works properly now! 16.02.2016 19:16, Brian West: > This topic was talked about on the list in the past week: > https://freeswitch.org/jira/browse/FS-8321 > > "BEHAVIOR CHANGE Add variable media_mix_inbound_outbound_codecs to mix > inbound and outbound codecs" > > /b > > On Tue, Feb 16, 2016 at 6:40 AM, Roman Kudinov > <roman.kudinov(a)novelapp.com <mailto:roman.kudinov@novelapp.com>> wrote: > > Hi all, > > I have a problem with bridging a call. My FS 1.6.6 is setup to > bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway. > I have two branches in the dial plan. > > 1) One works through mod_conference which calls an outbound number > using conference_set_auto_outcall > > 2) Another works by the direct bridging of incoming rtmp call into > outbound SIP call. > > Whilst the first branch works just fine, the second one does not. > They both use the same sofia profiles, SIP gateways and outbound > SIP numbers. > They both are called from the same RTMP source. Here are the > snippet of codes. > > ================== This one works ==================================== > > <extension name="conference_set_auto_outcall"> > > <condition field="destination_number" expression="123"> > > <action application="answer"/> > > <action application="set" > data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> > > <action application="set" > data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> > > <action application="set" > data="conference_auto_outcall_profile=default"/> > > <action application="conference_set_auto_outcall" > data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/> > > <action application="conference" > data="$1+flags{moderator|endconf|mute}"/> > > </condition> > > </extension> > =================================================== > > ================ This one does not work ================ > > <extension name="phone_only_session"> > > <condition field="destination_number" expression="456"> > > <action application="set" data="ignore_early_media=true"/> > > <action application="set" > data="absolute_codec_string=PCMU,PCMA,opus"/> > > <action application="bridge" > data="sofia/gateway/sip_profile/number"/> > > </condition> > > </extension> > ======================== > > I'd like to outline that they use the same SIP profiles, they are > called from the same RTMP-source (they differs by the > destination_number), they > call the same SIP number. > I turned on SIP tracing on and found that the call that is > initiated by mod_conference offers the codecs according to > outbound_codec_prefs set > in vars.xml, here is the piece of log: > > m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:102 SPEEX/8000 > > a=rtpmap:103 SPEEX/16000 > > a=rtpmap:104 SPEEX/32000 > > a=rtpmap:105 opus/48000/2 > > a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; > maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-16 > > a=rtpmap:106 telephone-event/16000 > > a=fmtp:106 0-16 > > a=rtpmap:108 telephone-event/32000 > > a=fmtp:108 0-16 > > a=rtpmap:110 telephone-event/48000 > > a=fmtp:110 0-16 > > a=ptime:20 > > But the directly bridged call offers incoming codec only, e.g. speex > > m=audio 24972 RTP/AVP 102 101 > > a=rtpmap:102 SPEEX/16000 > > a=rtpmap:101 telephone-event/16000 > > a=fmtp:101 0-16 > > a=ptime:20 > > I tried everything I could imagine. I set absolute_codec_string in > the dialplan (you can see it in the above snippet). > I explicitly set > > <X-PRE-PROCESS cmd="set" > data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/> > in vars.xml > > I tried to change > > <param name="inbound-codec-negotiation" value="generous"/> > from generous to greedy > > I tried with true/false in the following parameters in > internal.xml and > external.xml SOFIA profiles > > <param name="inbound-late-negotiation" value="false"/> > > <param name="inbound-zrtp-passthru" value="false"/> > Nothing changes. > > Moreover the direct bridging worked fine on FS 1.4.7, it offered > PCMU and Speex codecs to SIP. > I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I > missed an important setting that makes "bridge" application to work in > proxy media mode. > I checked I don't have neither bypass or proxy words in vars.xml, > sofia.conf.xml, internal.xml, external.xml, public.xml or they are > commented. > > Does anybody have any ideas about the reason for such behavior? > > > Thanks, > Roman > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > 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 > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > > */Brian West/* > brian(a)freeswitch.org <mailto:brian@freeswitch.org> > > > */Twitter: @FreeSWITCH , @briankwest/* > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: > /r/freeswitch <https://www.reddit.com/r/freeswitch> > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest >

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verto jsonrpc settings?
by Nathan Neulinger 16 Feb '16

16 Feb '16

What does the verto/directory setting: <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/>' expose? It's very unclear from documentation. Does this make ALL conference/presence events available to that user that logged in via verto or only those related to their particular registration/call? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Answered/abandoned calls mod_callcenter
by Rodrigo Ramírez Norambuena 16 Feb '16

16 Feb '16

Hello everyone!, I'm testing a mod_callcenter for make a FreeSWITCH version of a open source software (https://github.com/roramirez/qpanel/tree/fs) Now i using ESL to send command and get information from the module. I looking for a way to know the data of answered/abandoned call from a queue and agents. Somebody can give a tips or a light? Regards, -- Rodrigo Ramírez Norambuena http://www.rodrigoramirez.com

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problems with bridging a call, looks like transcoding is disabled
by Roman Kudinov 16 Feb '16

16 Feb '16

Hi all, I have a problem with bridging a call. My FS 1.6.6 is setup to bridge calls from RTMP-based source (using mod_rtmp) to a SIP gateway. I have two branches in the dial plan. 1) One works through mod_conference which calls an outbound number using conference_set_auto_outcall 2) Another works by the direct bridging of incoming rtmp call into outbound SIP call. Whilst the first branch works just fine, the second one does not. They both use the same sofia profiles, SIP gateways and outbound SIP numbers. They both are called from the same RTMP source. Here are the snippet of codes. ================== This one works ==================================== > <extension name="conference_set_auto_outcall"> > <condition field="destination_number" expression="123"> > <action application="answer"/> > <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/> > <action application="set" data="conference_auto_outcall_caller_id_number=$${effective_caller_id_number}"/> > <action application="set" data="conference_auto_outcall_profile=default"/> > <action application="conference_set_auto_outcall" data="{ignore_early_media=true}sofia/gateway/sip_profile/number"/> > <action application="conference" data="$1+flags{moderator|endconf|mute}"/> > </condition> > </extension> =================================================== ================ This one does not work ================ > <extension name="phone_only_session"> > <condition field="destination_number" expression="456"> > <action application="set" data="ignore_early_media=true"/> > <action application="set" data="absolute_codec_string=PCMU,PCMA,opus"/> > <action application="bridge" data="sofia/gateway/sip_profile/number"/> > </condition> > </extension> ======================== I'd like to outline that they use the same SIP profiles, they are called from the same RTMP-source (they differs by the destination_number), they call the same SIP number. I turned on SIP tracing on and found that the call that is initiated by mod_conference offers the codecs according to outbound_codec_prefs set in vars.xml, here is the piece of log: > m=audio 18684 RTP/AVP 0 102 103 104 105 8 101 106 108 110 > a=rtpmap:0 PCMU/8000 > a=rtpmap:102 SPEEX/8000 > a=rtpmap:103 SPEEX/16000 > a=rtpmap:104 SPEEX/32000 > a=rtpmap:105 opus/48000/2 > a=fmtp:105 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:106 telephone-event/16000 > a=fmtp:106 0-16 > a=rtpmap:108 telephone-event/32000 > a=fmtp:108 0-16 > a=rtpmap:110 telephone-event/48000 > a=fmtp:110 0-16 > a=ptime:20 But the directly bridged call offers incoming codec only, e.g. speex > m=audio 24972 RTP/AVP 102 101 > a=rtpmap:102 SPEEX/16000 > a=rtpmap:101 telephone-event/16000 > a=fmtp:101 0-16 > a=ptime:20 I tried everything I could imagine. I set absolute_codec_string in the dialplan (you can see it in the above snippet). I explicitly set > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,G722,OPUS,PCMA"/> in vars.xml I tried to change > <param name="inbound-codec-negotiation" value="generous"/> from generous to greedy I tried with true/false in the following parameters in internal.xml and external.xml SOFIA profiles > <param name="inbound-late-negotiation" value="false"/> > <param name="inbound-zrtp-passthru" value="false"/> Nothing changes. Moreover the direct bridging worked fine on FS 1.4.7, it offered PCMU and Speex codecs to SIP. I've upgraded to FS 1.6.6 and now it doesn't work. It looks like I missed an important setting that makes "bridge" application to work in proxy media mode. I checked I don't have neither bypass or proxy words in vars.xml, sofia.conf.xml, internal.xml, external.xml, public.xml or they are commented. Does anybody have any ideas about the reason for such behavior? Thanks, Roman

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400 Bad To Header.
by Aqs Younas 16 Feb '16

16 Feb '16

Hi users, My freeswitch is replying with me 400 Bad To Header whenever It receives Bye containing this To Header. To: 1111#15122031234 <sip:1111#15122031234@192.168.10.50:5060 >;tag=NKU2jrZvFaK0j But I wonder in the case of Invite containing, this header everything is fine. Is there something wrong with this header? (#?) Best regards.

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WebRTC Core Dump for Video Conference.
by Veerabhadrarao Kankatala 16 Feb '16

16 Feb '16

Hello, I am using freeswitch-1.6.5 version for webRTC I am using Chrome to intiate call (WebRTC Client) to another Video device (Grand stream), and i am using Default Dialplan to make call both are registered in freeswitch WebRTC Client: 1001 Default user GrandStream device: 1009 Default user When i make audio calling it is working fine, but when i make Video call from WebRTC Client ----> GrandStream device, my freeswitch is crashing Please find the attachment for Call Log to results the coreDump Please tell me how to resolve this issue and let me know still anything i am missing. -- Thanks & Regards Veerabhadrarao Kankatala Software Developer (C & Unix ) PANAMAX INFOTECH LIMITED Mobile: +91-8401231249 Messenger Id: Skype: veerabhadrarao.kankatala E-mail: veerabhadrarao.kankatala(a)panamaxil.com

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