- FreeSWITCH-users - FreeSWITCH Mailing Lists

Popup on incoming call on FS
by Abhik Biswas 05 Sep '16

05 Sep '16

I am a recent adopter of FS and sorry if this has been addressed many times before. I would like the incoming call on FS pop up screen or update a page on my desktop. I need to know how this incoming call event can be captured and the Caller-id be picked up from the same. Based on the above I am assuming a notification can be sent to the webpage. Abhik

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Bottleneck between mod_xml_radius and freeradius client?
by Juan Pablo Godoy 05 Sep '16

05 Sep '16

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sofia patch to filter sip messages
by David Villasmil 05 Sep '16

05 Sep '16

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week day and time condition possible bug
by Bipin Patel 05 Sep '16

05 Sep '16

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bgapi with two commands one by one
by Dmitry Mordovin 04 Sep '16

04 Sep '16

Hello I want to execute two commands with bgapi, but can't find solution in FreeSWITCH manual or google. bgapi expand originate <leg_timeout=30,continue_on_fail=true>[origination_caller_id_name='test']sofia/gateway/first_supplier/12345@first_supplier|[origination_caller_id_name='test']sofia/gateway/second_supplier/12345@second_supplier 00294539; curl http://127.0.0.1/cgi?e=originate_done I want command 'curl' to be executed after 'originate' done (call disconnected). Like linux shell can run commands one by one, just separate with ';' Is it possible? If so, how can I do it?

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Building an IVR - Macro not found error
by William Colls 03 Sep '16

03 Sep '16

Freeswitch 1.6.9 debian Jessie 64 bit platform. New to Freeswitch and trying to build a small IVR. I am getting an error when it tries to execute a phrase macro. From looking at freeswitch.xml.fsxml it appears that the macro is loaded (reloadxml produces no errors). When I call into the switch, I can see the call being answered, and my IVR is picked, up and executes with out error until switch_ivr_play tries to play the macro. It then generates the error message "Can't find macro ourhouse_greet_long". Just before the error occurs, I get a debug message "No language specified - Using [en]". If I substitute one of the Demo IVR macros for ourhouse_greet_long, it will be executed with out error. The file containing macro is located in /usr/local/freeswitch/conf/lang/en/ivr. So what have I done wrong? Any help appreciated. I can provide more detail if needed. Thanks for your time. William.

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SDP from public to private network, no audio
by devang nathwani 03 Sep '16

03 Sep '16

I know it may be out of scope but nevertheless I want to route the call from opensips to freeswitch but the opensips is on public Network and freeswitch servers are on private network so SDP is not directly travelling from opensips public network to freeswitch private network and there is no audio during the call To achieve this I am using RTP proxy I want to know how exactly bind the public IP to private IP using rtp proxy, I am missing some configuration I have tried many links but no luck yet If anybody would suggest any opensips configurations, links to refer, other way out of this to achieve this; it would be helpful

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Re: [Freeswitch-users] Popup on incoming call on FS
by Abhik Biswas 03 Sep '16

03 Sep '16

Also looking for guidance on thoughts on how to handle this for SIP and WebRTC calls and text messages. Abhik From: Abhik Biswas Sent: Saturday, September 3, 2016 5:26 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Popup on incoming call on FS I am a recent adopter of FS and sorry if this has been addressed many times before. I would like the incoming call on FS pop up screen or update a page on my desktop. I need to know how this incoming call event can be captured and the Caller-id be picked up from the same. Based on the above I am assuming a notification can be sent to the webpage. Abhik

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Looking for Linux tc wizard
by Chad Phillips 03 Sep '16

03 Sep '16

Anybody here a master of the Linux Traffic Control program? http://lartc.org/manpages/tc.txt Asking because I have a need for a script that will allow me to simulate different types of internet connections (shitty 3G, DSL, wifi, etc), for the purposes of quality testing a WebRTC implementation. I’d like to use tc (and perhaps iptables if that would help) to as closely simulate the various conditions (asymmetric bandwidth, packet loss, etc.) as possible. I’m willing to pay for the help, and would be happy to donate the work back as open source.

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console siptrace filtering
by David Villasmil 02 Sep '16

02 Sep '16

Hello guys, You can get all SIP messages coming and going to a specific profile by typing on the console: “sofia profile [PROFILE] siptrace on” But if you have traffic on the box, this is a NIGHTMARE, as you get ALL messages… So I made a modification to mod_console.c to add a filtering for siptraces. By enabling the filter like so: “console filter [expression]” Where “expression” can be anything present in the packets you want to get, be it username, dialed number, whatever. Only those messages on which the expression was found will be shown: freeswitch@MacBookPro-David> console USAGE: -------------------------------------------------------------------------------- console help console loglevel [[0-7] | <loglevel_string>] console filter <filter-expression> <<<——— NEW functionality console colorize [on|off|toggle] ———————————————————————————————————————— freeswitch@MacBookPro-David> console filter USAGE: -------------------------------------------------------------------------------- console filter <filter-expression|off> Enable, disable filtering, Only messages containing the 'filer-expression' will be printed out. set 'filter-expression' to 'off' to stop filtering ———————————————————————————————————————— Typing: console filter 1001 Would result in only seeing messages which include 1001 EXPRESSION (1001) FOUND IN: <<<recv 1255 bytes from tcp/[127.0.0.1]:52917 at 21:43:11.739980: ------------------------------------------------------------------------ INVITE sip:1001@127.0.0.1 SIP/2.0 Call-ID: 6f9cdc511446251e254e9e617d26647c@0:0:0:0:0:0:0:0 CSeq: 2 INVITE From: "1000" <sip:1000@127.0.0.1>;tag=45dd865b To: <sip:1001@127.0.0.1> Max-Forwards: 70 Contact: "1000" <sip:1000@127.0.0.1:52917;transport=tcp;registering_acc=127_0_0_1> User-Agent: Jitsi2.8.5426Mac OS X Content-Type: application/sdp Via: SIP/2.0/TCP 127.0.0.1:52917;branch=z9hG4bK-343435-1dc76531757a26482fa8ff1bac9acf3b Proxy-Authorization: Digest username="1000",realm="127.0.0.1",nonce="1a2be9b9-fa12-4763-b00c-dfeacc79f9fd",uri="sip:1001@127.0.0.1",response="fce7bf74772a149fe39ec7e50ec61c69",algorithm=MD5,qop=auth,cnonce="xyz",nc=00000001 Content-Length: 566 v=0 o=1000-jitsi.org 0 0 IN IP4 127.0.0.1 s=- c=IN IP4 127.0.0.1 t=0 0 m=audio 5119 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=fmtp:96 usedtx=1 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-xr:voip-metrics m=video 5121 RTP/AVP 97 99 a=recvonly a=rtpmap:97 H264/90000 a=fmtp:97 profile-level-id=4DE01f;packetization-mode=1 a=imageattr:97 send * recv [x=[0-1920],y=[0-1200]] a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=imageattr:99 send * recv [x=[0-1920],y=[0-1200]] ------------------------------------------------------------------------ >>> My goal is to extend this to also filter other messages, like call processing, etc. (ERR messages would not be filtered out) I’m wondering whether it is possible to do a pull request on this? Regards, David ᐧ

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