- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Is there a way for FS not to send back any response to INVITE
by Dave Horton 01 Jun '16

01 Jun '16

I do this with my proxy that fronts freeswitch (https://www.npmjs.com/package/drachtio-fs-load-balancing-proxy <https://www.npmjs.com/package/drachtio-fs-load-balancing-proxy>) by evaluating SIP headers for the common sip scanner signatures and dynamically adding the senders to an iptable black list. This does result in a 100 Trying going back to the sender (since that is sent by my sip stack before proxy processing) but no final response, and every subsequent message from that address is silently dropped.

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Is there a way for FS not to send back any response to INVITE
by Oleg Stolyar 01 Jun '16

01 Jun '16

Hi, In order to protect against scanning attacks I'd like for FS to not respond to INVITES unless they match certain conditions. I understand that currently FS always responds with 100 Trying right away before processing the call and then, if the call does not match anything in the dialplan, responds with a 302 Moved Temporarily. The 302 can be replaced with another response code (for example 403 Forbidden which is what I am doing now) using the *respond* dialplan app. However, that might encourage the scanner to keep trying. So I guess there are two questions: 1. Is there a way not to send back 100 Trying at all? 2. Is there a way to not send any final response?

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Webrtc Verto jsapi
by stevem 01 Jun '16

01 Jun '16

I was not able to find any documentation on this. Basically all I want to do from my frontend. Is post an event to the backend. Basically run the command: uuid_broadcast 69d58330-7f7f-4e52-bb2c-9e25d9f5238b event::Event-Name=HOLD. I'm using FreeSWITCH Version 1.4.21~64bit I got the verto/webrtc working using the demo located here: https://webrtc.freeswitch.org/verto/index.html#page-main I'm able to connect to the dialplan and drop the caller (PSTN/ITSPs) and verto client into a conference and the audio is flowing. I just want to send a message to the backend to tell it to do things. In my example above I want to set the caller on hold. My verto connection in javascript looks like this: $.verto.init({}, bootstrap); function bootstrap(status) { that.vertoHandle = new jQuery.verto({ login: userId + '@' + host, passwd: password, tag: "vertoAudio", socketUrl: 'wss://' + host + ':8082', audioParams: { googAutoGainControl: false, googNoiseSuppression: false, googHighpassFilter: false }, iceServers: true }, vertoCallbacks); } var vertoCallbacks = { onWSLogin: onWSLogin, onWSClose: onWSClose, onDialogState: onDialogState }; On successful login i make a call: that.call= that.vertoHandle.newCall({ destination_number: that.liveCall.ani, caller_id_name: user.firstName + ' ' + user.lastName, caller_id_number: userId, useVideo: false, useStereo: false, userVariables: {} }); Thanks alot. This is my first post and I've only been doing Freeswitch for a few months. I have alot of experience in Asterisk though. Let me know if more data is needed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Webrtc-Verto-jsapi-tp7596220.… Sent from the freeswitch-users mailing list archive at Nabble.com.

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SIP gateways, context
by Yu Boot 01 Jun '16

01 Jun '16

I use single profile both for SIP gateways and for password-protected clients. Why FS ignores context parameter in SIP gateway settings? I can route inbound calls from these SIP trunks only in "default" context by inbound IP address. Password-protected clients (from directory) don't suffer from this, all inbound calls are routed via respective contexts. Sent from my ASUS

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Freeswitch Inbound Calls No sound
by Deepika Yadav 01 Jun '16

01 Jun '16

Hi, I need to make calls from a mobile number that can be answered by Freeswitch extension playing an IVR. To achieve this, I have configured a VOIP-GSM gateway having inserted a mobile sim that upon receiving the incoming calls forwards it to a SIP URI. Gateway Settings: PBX IP : ec2 public ip Protocol : SIP Port : 5080 Target: 9811806537 Freesswitch resides at Amazon ec2 cloud, Dialplan for public context is : <extension name="mydid"> <condition field="destination_number" expression="^(9811806537)$"> <action application="transfer" data="4444 XML default"/> </condition> </extension> Default Context : <extension name="ivr_demo"> <condition field="destination_number" expression="^4444$"> <action application="answer"/> <action application="start_dtmf" /> <action application="sleep" data="2000"/> <action application="ivr" data="rhd_ivr"/> </condition> </extension> where Server IP - XX.XX.XX.XX Gateway IP- YY.YY.YY.YY Freeswitch logs can be found here : http://pastebin.com/7BbM5dWD <http://pastebin.com/i4gs9uVJ> Files sip_profiles/external.xml sip_profiles/internal.xml <param name="ext-rtp-ip" value="XX.XX.XX.XX"/> <param name="ext-sip-ip" value="XX.XX.XX.XX"/> Any pointer to move forward would be highly appreciated Reagrds, Deepika

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Loop in the dialplan
by Agustí Ubalde 01 Jun '16

01 Jun '16

Hi all, How to simulate a loop in the dialplan? Like goto in Asterisk. Thanks! *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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SIP Message question
by Nagorny, Dimitry 01 Jun '16

01 Jun '16

Hello List, AFAIK, all INVITE SIP messages (INVITE sip:100@mydomain.com) are going through the set up dialplans. Are all MESSAGE SIP messages (MESSAGE sip:100@mydomain.com) also go through these dialplans? If no, how can I handle them? This might be a stupid question but any help is still appreciated. Best Regards Dimitry Nagorny Trainee

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ClueCon Speaker Registration Deadline 2 Weeks Away!!!!
by Anthony Minessale 31 May '16

31 May '16

Don't forget to get your proposal in. E-mail speakers(a)cluecon.com -- 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=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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SDP with double m=audio lines and with the same port
by Ján Füri 31 May '16

31 May '16

Hello, My freeswitch's INVITE sdp offers two m=audio lines. (RTP/SAVP and RTP/AVP). That's OK, because I want it so ;) But both use the same audio port number. ** Is that possible to change this behavior ? To use different audio port numbers for each m=audio line ... Please check my example below. In SDP from my freeswitch, both m-lines have *m=audio **22284*. The reason, why I want to change this is WebRTC and RTPengine. RTPengine changes this INVITEs to RTP/SAVPF. And the changed RTP/SAVPF is then not acceptable for chrome browsers. Firefox works well (so far). I already reported this to the RTPengine but they say the problem is with INVITE from freeswitch because the m-audio lines use the same port. My example : incoming calls -> Media Server (freeswitch) -> Kamailio (sip and websocket proxy) -> WebRTP and SIP clients I have found a feature *media_webrtc=true* so I could avoid using rtpengine for webrtc clients (that would be really awesome), but then classic sip clients are offered only with RTP/SAVPF and most sip phones do not know RTP/SAVPF so the calls are rejected :( INVITE SDP from freeswitch: v=0. o=SBC 1464589775 1464589776 IN IP4 <freeswitch-public-ip>. s=SBC. c=IN IP4 <freeswitch-public-ip>. t=0 0. *m=audio 22284 RTP/SAVP 8 0 9 3 18.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtcp:22285 IN IP4 <freeswitch-public-ip>. a=crypto:1 AEAD_AES_256_GCM_8 inline:Q/O+iUzZ9AZLcrADA7w/XcpkmuW6yArr1vsaWc0coTyWUIRLX2qfCA7XDAs. a=crypto:2 AEAD_AES_128_GCM_8 inline:mBb/SmBuSQNSm8lC5giUfZMCv0ZYINvsQiX2sw. a=crypto:3 AES_CM_256_HMAC_SHA1_80 inline:ahOTNSsdmHLIOBqvUyGyNjd+gDfAE/+jA6w7XwzGWxBMjtzf5akXNvM/OGy0jQ. a=crypto:4 AES_CM_192_HMAC_SHA1_80 inline:k7002pXV/SUT7JHhZTYaMV5keUTp1EP57M4rcNZnAoZZzsceTXA. a=crypto:5 AES_CM_128_HMAC_SHA1_80 inline:bHcV4E/OnzMkNeaFplPWt4RELILYZeGlifnJNlRV. a=crypto:6 AES_CM_256_HMAC_SHA1_32 inline:zJkowU1tc5rQR5BPpg2m3eE97ZqXLFFJc1Agh89XuZynPFMrXhO266+eMZCd2A. a=crypto:7 AES_CM_192_HMAC_SHA1_32 inline:2DwGI3UAPsLFBS5sJBsc+pzEsITQwDHCvB0u7pK/XEE3G8swsrw. a=crypto:8 AES_CM_128_HMAC_SHA1_32 inline:Jen6Gm56Z+4RTRmfpXTpSpoG5bduyythl1j21a15. a=crypto:9 AES_CM_128_NULL_AUTH inline:Nn83liGQY7eY8LGs9qz/7EoPiHOpRgwY8H7Ts/b5. a=ptime:20. *m=audio 22284 RTP/AVP 8 0 9 3 18.* a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtcp:22285 IN IP4 <freeswitch-public-ip>. a=ptime:20. INVITE SDP from RTPengine : v=0 o=SBC 1464589775 1464589776 IN IP4 212.232.17.66 s=SBC c=IN IP4 <rtpengine-public-ip> t=0 0 *m=audio 31836 RTP/SAVPF 8 0 9 3 18* a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:31837 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F a=ice-ufrag:NauDcjVU a=ice-pwd:AiHk6LF4tE0GElyFMWtLeod1wH a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 31836 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 31837 typ host *m=audio 31866 RTP/SAVPF 8 0 9 3 18* a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=rtcp:31867 a=setup:actpass a=fingerprint:sha-1 11:76:2D:2A:F7:0D:5A:23:9D:F6:0C:E7:4C:DF:1E:CB:BF:5D:76:4F a=ice-ufrag:YxCyPhQV a=ice-pwd:2vYt8YaMhIp2DSLPeYOKDqqBX0 a=candidate:cE0FGbXWIfwj6OGD 1 UDP 2130706431 <rtpengine-public-ip> 31866 typ host a=candidate:cE0FGbXWIfwj6OGD 2 UDP 2130706430 <rtpengine-public-ip> 31867 typ host

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Multi-user video conference
by happy.neko 31 May '16

31 May '16

Hello, I have following requirements for multi-user video conference: 1. One Teacher and up to 10 Students are in the same room. 2. Teacher hears everyone in the room and sees video streams from every student (total bandwidth = 1 audios stream + n video streams). 3. Student hears everyone in the room and sees video stream from Teacher only (total bandwidth = 1 audio stream + 1 video stream). 4. There is no CPU intensive video encoding process on the server. Teacher receives video streams as-is plus muxed audio stream. 5. Conference is auto-recorded on the server. (I don't know exactly how, mp4 with multiple video streams?) Is it possible to implement this with FreeSwitch? If no what requirements should be relaxed? -- Best regards, I. Sokolov

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