- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to achieve MRCP server load balancing from FreeSwitch??
by Jignesh Vasoya 18 May '18

18 May '18

Hi, I was exploring an option to use mod-distributor for using a simple round robin load balancing between different MRCP speech servers. I have tried to fill in "default asr profile" using mod-distributor placeholders(variables). But it is not working?? Is there any way to achieve simple load balancing between different Speech Servers in FreeSwitch? -- *Thanks,* *Jignesh*

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[VAD events][conference call]
by Bilal Abbasi 18 May '18

18 May '18

Hi Users, I want to get the VAD events on a conference call, i want to join a conference call silently and just wanted to get vad events. What i did is to land the conference call in a context and run echo application.(as vad events appears on b leg) But that caused noise in conference, so i droped the rtp packets using iptables rules. Problem: Everything works great, but now i want to traverse in conference(like pin etc) and i send dtmf 2833 which also get droped by iptables. Question: is there a way i can get vad events and be as a silent listener of conference as well. P.S: previously i used originate conference and landed on park(), but that do not generate vad events. So i switched to echo() Regards Abbasi

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xml_curl for directory with verto
by Rick Jarvis 17 May '18

17 May '18

Ok I am now seeing post requests (small steps!), but cannot get an authentication success. Maybe something I’m misunderstanding (most likely!). This is the XML I’m sending back to FS: <document type="freeswitch/xml"> <section name="directory"> <domain name=“default"> <groups> <group name="default"> <users> <user id=“203"> <params> <param name="password" value="password"/> <param name="vm-password" value="982375982735"/> <param name="jsonrpc-allowed-methods" value="verto"/> <param name="jsonrpc-allowed-event-channels" value="demo,conference,presence"/> <param name="jsonrpc-allowed-jsapi" value="all"/> <param name="jsonrpc-allowed-fsapi" value="all"/> <param name="verto-context" value="default"/> <param name="verto-dialplan" value="XML"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value=“203"/> <variable name="user_context" value=“default"/> <variable name="effective_caller_id_name" value=“verto"/> <variable name="effective_caller_id_number" value=“203"/> <variable name="outbound_caller_id_name" value=“verto"/> <variable name="outbound_caller_id_number" value=“203"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> And this is what I’m getting on FS console (with xml_curl debug_on): 2018-05-17 17:09:34.902788 [DEBUG] mod_verto.c:1876 BAD READ -1000 2018-05-17 17:09:34.902788 [DEBUG] mod_verto.c:2019 x.x.x.x:49935 Ending client thread. 2018-05-17 17:09:34.902788 [DEBUG] mod_verto.c:2027 x.x.x.x:49935 Thread ended 2018-05-17 17:10:34.402793 [DEBUG] mod_verto.c:4139 x.x.x.x:50348 Client Connect from x.x.x.x:50348 accepted 2018-05-17 17:10:34.402793 [DEBUG] mod_verto.c:1992 x.x.x.x:50348 Starting client thread. 2018-05-17 17:10:34.482797 [ALERT] mod_verto.c:1384 READ x.x.x.x:50348 [{ "jsonrpc": "2.0", "method": "login", "params": { "sessid": "c674980b-27ca-4c3e-37d1-3421736b990f" }, "id": 2 }] 2018-05-17 17:10:34.482797 [DEBUG] mod_verto.c:1281 x.x.x.x:50348 re-connecting session c674980b-27ca-4c3e-37d1-3421736b990f 2018-05-17 17:10:34.482797 [ALERT] mod_verto.c:604 WRITE x.x.x.x:50348 [{ "jsonrpc": "2.0", "id": 2, "error": { "code": -32000, "message": "Authentication Required" } }] 2018-05-17 17:10:34.482797 [ALERT] mod_verto.c:1384 READ x.x.x.x:50348 [{ "jsonrpc": "2.0", "method": "login", "params": { "sessid": "c674980b-27ca-4c3e-37d1-3421736b990f" }, "id": 1 }] 2018-05-17 17:10:34.482797 [ALERT] mod_verto.c:604 WRITE x.x.x.x:50348 [{ "jsonrpc": "2.0", "id": 1, "error": { "code": -32000, "message": "Authentication Required" } }] 2018-05-17 17:10:34.482797 [ALERT] mod_verto.c:1384 READ x.x.x.x:50348 [{ "jsonrpc": "2.0", "method": "login", "params": { "login": "203", "passwd": "password", "loginParams": { }, "userVariables": { }, "sessid": "c674980b-27ca-4c3e-37d1-3421736b990f" }, "id": 3 }] 2018-05-17 17:10:34.542787 [ALERT] mod_verto.c:604 WRITE x.x.x.x:50348 [{ "jsonrpc": "2.0", "id": 3, "error": { "code": -32001, "message": "Login Incorrect" } }]

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Re: [Freeswitch-users] how to make FS to listen for 2 RTP streams on one call ?
by Evtim Ivanov 16 May '18

16 May '18

Yes for the IP address , no for the port. On 16 May 2018 at 14:12, kaiduan xie via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---------- Forwarded message ---------- > From: kaiduan xie <kaiduanx(a)yahoo.ca> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Wed, 16 May 2018 13:11:35 +0000 (UTC) > Subject: Re: [Freeswitch-users] how to make FS to listen for 2 RTP streams > on one call ? > Does RTP proxy use same IP address/port to send/receive the two RTP > streams? > > /Kaiduan > > On Wednesday, May 16, 2018, 8:37:29 a.m. EDT, Evtim Ivanov < > ivanov.evtim(a)gmail.com> wrote: > > > Hello all, > > I'm trying to implement this (shown in the picture) scenario, but to use > FS instead of OrekaTR: > > > There are several small bumps on the road, which I successfully overcame, > but finally I met a challenge which is bigger, than my skills, so I need > the community help. The problem is, that RtpProxy can't mix the media, so > instead of one mixed stream on the forked call, I have 2 separate RTP > streams. I have no idea how to make FS to listen both of them. Is this > possible and how? > Any advice is highly appreciated. > > Best! > Evtim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)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 > 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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External authentication
by Rick Jarvis 16 May '18

16 May '18

Now, I’m sure I’ve read about this somewhere, and now I need to do it, I can’t find it of course! Basically need to hand the user authentication off to another server, via https. Got this kind of thing working nicely in the dialplan using httapi, but can’t figure out how to do it for user registration? Needs to be real time, so exporting directory XML *to* FS isn’t going to do it. Any ideas?

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Answering to sip2sip.info INVITE Freeswitch sends strange SDP: 'm=audio 0 RTP/AVP 19'
by Xenia Obolenskaya 16 May '18

16 May '18

recv 1815 bytes from tls/[81.23.228.150]:443 at 07:59:52.249297: ------------------------------------------------------------------------ INVITE sip:gw+sip2sip@<my_IP:5061;transport=tls;gw=sip2sip SIP/2.0 Record-Route: <sip:81.23.228.150:443;transport=tls;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> Record-Route: <sip:81.23.228.150;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> Record-Route: <sip:81.23.228.129;lr;ftag=KKQaHvjDvKecF;did=f94.5418aa17> Record-Route: <sip:81.23.228.150;lr;ftag=KKQaHvjDvKecF> Record-Route: <sip:85.17.186.7;r2=on;lr;ftag=KKQaHvjDvKecF> Record-Route: <sip:85.17.186.7;transport=tcp;r2=on;lr;ftag=KKQaHvjDvKecF> Via: SIP/2.0/TLS 81.23.228.150:443;branch=z9hG4bK9a88.2adb9c55.0 Via: SIP/2.0/UDP 81.23.228.129:5060;branch=z9hG4bK9a88.40d983a7.0 Via: SIP/2.0/UDP 81.23.228.150:5060;branch=z9hG4bK9a88.1adb9c55.0 Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK9a88.80ff70a4.0;i=62a32 Via: SIP/2.0/TCP 81.23.228.160:57446;rport=55026;received=81.23.228.160;branch=z9hG4bKKBNcH3r9B48Xr Max-Forwards: 66 From: "Victor" <sip:rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info>;tag=KKQaHvjDvKecF To: <sip:13579@sip2sip.info> Call-ID: f45c2a4a-d381-1236-1298-047d7b87609e CSeq: 933861599 INVITE Contact: <sip:rCodzvTEQK+uupE5QSBudQ@81.23.228.160:55026> User-Agent: SylkRTC (Firefox 60.0 on Windows 8.1) Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, UPDATE Content-Type: application/sdp Content-Disposition: session Content-Length: 474 v=0 o=mozilla...THIS_IS_SDPARTA-60.0 3887687583368393798 5225707675845380188 IN IP4 81.23.228.160 s=- t=0 0 a=sendrecv m=audio 54672 RTP/AVP 109 9 0 8 101 c=IN IP4 81.23.228.129 a=rtpmap:109 opus/48000/2 a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:WIsk0UoOzvahjfdFEDDgAVd5hXFmUnUpvBoPK1eY In Freeswitch debug I see the correct local SDP: [DEBUG] mod_sofia.c:850 Local SDP sofia/sip2sip/rCodzvTEQK+uupE5QSBudQ(a)guest.sip2sip.info: v=0 o=FreeSWITCH 1526429009 1526429010 IN IP4 <my_IP> s=FreeSWITCH c=IN IP4 <my_IP> t=0 0 m=audio 28590 RTP/SAVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7MKm6yBppNSDw1IJDgyl0a2xkkvmashJVyTACq89 but in siptrace I see that really Freeswitch sends this strange SDP (and nothing else): send 1529 bytes to tls/[81.23.228.150]:443 at 07:59:59.082373: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/TLS 81.23.228.150:443;branch=z9hG4bK9a88.2adb9c55.0;rport=443 Via: SIP/2.0/UDP 81.23.228.129:5060;branch=z9hG4bK9a88.40d983a7.0 Via: SIP/2.0/UDP 81.23.228.150:5060;branch=z9hG4bK9a88.1adb9c55.0 Via: SIP/2.0/UDP 85.17.186.7:5060;branch=z9hG4bK9a88.80ff70a4.0;i=62a32 Via: SIP/2.0/TCP 81.23.228.160:57446;rport=55026;received=81.23.228.160;branch=z9hG4bKKBNcH3r9B48Xr Record-Route: <sip:81.23.228.150:443;transport=tls;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> Record-Route: <sip:81.23.228.150;r2=on;lr;ftag=KKQaHvjDvKecF;did=f94.1a4c1615> Record-Route: <sip:81.23.228.129;lr;ftag=KKQaHvjDvKecF;did=f94.5418aa17> Record-Route: <sip:81.23.228.150;lr;ftag=KKQaHvjDvKecF> Record-Route: <sip:85.17.186.7;r2=on;lr;ftag=KKQaHvjDvKecF> Record-Route: <sip:85.17.186.7;transport=tcp;r2=on;lr;ftag=KKQaHvjDvKecF> From: "Victor" <sip:rCodzvTEQK+uupE5QSBudQ@guest.sip2sip.info>;tag=KKQaHvjDvKecF To: <sip:13579@sip2sip.info>;tag=p8SyjjgjH5HZB Call-ID: f45c2a4a-d381-1236-1298-047d7b87609e CSeq: 933861599 INVITE Contact: <sip:13579@<my_IP>:5061;transport=tls> User-Agent: Freeswitch Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, outbound, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 129 v=0 o=FreeSWITCH 1526429009 1526429010 IN IP4 <my_IP> s=FreeSWITCH c=IN IP4 <my_IP> t=0 0 m=audio 0 RTP/AVP 19 Bearing in mind that sip2sip includes a=crypto attribute in RTP/AVP I added this string in dialplan: <action application="set" data="rtp_allow_crypto_in_avp=true"/> othewise I always received [ERR] switch_core_media.c:4316 a=crypto in RTP/AVP, refer to rfc3711 [NOTICE] switch_channel.c:3812 Hangup sofia/sip2sip/wFxwmegJTLu4n6kK8DpXjw(a)guest.sip2sip.info [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] But now Freeswitch doesn't send the correct SDP with RTP/SAVP (I see it just in debug). FreeSWITCH Version 1.6.19+git~20171120T163416Z~b1b21d0695~64bit (git b1b21d0 2017-11-20 16:34:16Z 64bit)

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caching dialplan in xml_curl
by Akshaya j 16 May '18

16 May '18

Hi All, I just need to cache the dialplan response in xml_curl module. The wiki has an example of caching the directory response setting "cacheable=true" in user-id tag which perfectly works. I'm just curious to know is there a similar way to cache the dialplan as in setting same tag in <condition> tag? I worked around placing cacheable in <section>,<context>,<extension> <condition> nothing seems recognising it. I feel having the cacheable in <condition> tag is meaningful but even tat doesn't work. How do u people make this caching work in dialplans? Any thoughts on this would be helpful.! Regards, Akshaya J

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Re: [Freeswitch-users] file.exists in Freeswitch 1.8/1.9
by Andrey Wolk 15 May '18

15 May '18

You're not doing anything wrong, Nick. As I can see here: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/src/mod/la… the whole File class is dummy and wasn't implemented yet in neither v1.6 nor v1.8. > From: Nick Giannak III <nick(a)hkcradio.com> > To: freeswitch-users(a)lists.freeswitch.org > Greetings. > In Freeswitch 1.6, the file.exist module does not appear to be > implemented in V8. The console shows it as "not implemented". This, > however, is a function listed in the documentation. Is it implemented for > Freeswitch 1.8? Or, alternatively, are we doing something wrong? > Thanks, > Nick >

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file.exists in Freeswitch 1.8/1.9
by Nick Giannak III 15 May '18

15 May '18

Greetings. In Freeswitch 1.6, the file.exist module does not appear to be implemented in V8. The console shows it as "not implemented". This, however, is a function listed in the documentation. Is it implemented for Freeswitch 1.8? Or, alternatively, are we doing something wrong? Thanks, Nick

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Use GStreamer instead of FreeSwitch?
by Chandramouli P 14 May '18

14 May '18

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

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