- FreeSWITCH-users - FreeSWITCH Mailing Lists

Verto re-INVITE equivalent
by Adam Ben-Ayoun 16 Jan '18

16 Jan '18

Hi, We have mobile apps (iOS/Android) and a website that uses FreeSwitch for audio/video conferencing using mod_conference. We are currently using SIP and considering to move to Verto for signaling. In our app users are connecting without video (not sending video) that implies a=recvonly and later on they can add video dynamically using SIP re-INVITE. How can we implement this using Verto? I tried verto.attach but it doesn't really generate a new local SDP on fs.. Any idea? Thanks, Adam

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WebRTC using RTP/AVP instead of RTP/SAVPF
by Alexander Haugg 16 Jan '18

16 Jan '18

Hi, in the dialplan I am using "<action application="bridge" data="{media_webrtc=true}user/123(a)my.sip.domain" />" The media description header have ever the RTP/SAVPF as set. The most Clients wich support webrtc and ICE have the possibiliti to work without SRTP (that's nice vor debugging problems). Is it possible to set RTP/AVP for webrtc calls? Thanks a lot.

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(no subject)
by Ali Haider 16 Jan '18

16 Jan '18

HIii where from get this package freeswitch-meta-all : Depends: freeswitch-mod-flite (= 1.9.0~628~7f9e6f3-1~xenial+1)

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No MWI - SIP msg Messages-Waiting: no
by Sean Devoy 16 Jan '18

16 Jan '18

I have captured the SIP packets for my VTech VSP 735. After the subscribe succeeds, I get a NOTIFY from FreeSwitch to the Phone that includes " Messages-Waiting: no". Can anyone tell me why? Or better still tell me how to enable MWI for this terminal. I went to the docs the parameter suggested. Below is a partial sip dump and the Directory entry for the phone. Any help would really be greatly appreciated. ------------------------------------------------------------------------ recv 476 bytes from udp/[71.121.214.88]:1026 at 16:33:39.074823: ------------------------------------------------------------------------ SUBSCRIBE sip:223@66.241.102.41:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.121.214.88:1026;rport;branch=z9hG4bK673363934 From: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=1645429613 To: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=N3se6oVe4v7B Call-ID: 153343976 CSeq: 25 SUBSCRIBE Contact: <sip:223@71.121.214.88:1026> Accept: application/simple-message-summary Max-Forwards: 70 User-Agent: Vtech Vesa VSP735 1.0.3-0 Expires: 600 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ send 727 bytes to udp/[71.121.214.88]:1026 at 16:33:39.076436: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 71.121.214.88:1026;rport=1026;branch=z9hG4bK673363934 From: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=1645429613 To: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=N3se6oVe4v7B Call-ID: 153343976 CSeq: 25 SUBSCRIBE Contact: <sip:223@66.241.102.41:5060> Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=600 Content-Length: 0 ------------------------------------------------------------------------ send 913 bytes to udp/[71.121.214.88]:1026 at 16:33:39.103860: ------------------------------------------------------------------------ NOTIFY sip:223@71.121.214.88:1026 SIP/2.0 Via: SIP/2.0/UDP 66.241.102.41;rport;branch=z9hG4bKB1H0peQSN33gH Max-Forwards: 70 From: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=N3se6oVe4v7B To: <sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com>>;tag=1645429613 Call-ID: 153343976 CSeq: 2027115047 NOTIFY Contact: <sip:223@66.241.102.41:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces Event: message-summary Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=600 Content-Type: application/simple-message-summary Content-Length: 73 Messages-Waiting: no Message-Account: sip:223@66.241.102.41<sip:223@fs_bfis.bizfocused.com> Directory Entry: <user id="223">  <variables> <variable name="outbound_caller_id_name" value="Sean"/> <variable name="outbound_caller_id_number" value="4439219427"/> <variable name="internal_caller_id_name" value="Sean"/> <variable name="internal_caller_id_number" value="223"/> <variable name="user_context" value="from-internal-BFIS"/> <variable name="user_originated" value="true"/> <variable name="toll_allow" value="domestic"/> <variable name="accountcode" value="BFIS"/> <variable name="mwi-account" value="223@voicemail_BFIS"/> <variable name="callgroup" value="1"/> </variables> <params> <param name="password" value="KEPTSECRET"/> <param name="manage-presence" value="true"/> <param name="manage-shared-appearance" value-="true" /> <param name="mwi-account" value="223@voicemail_BFIS"/> </params> </user>

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Originate Retry Call After I Answer
by Raimundo Pérez Nieves 16 Jan '18

16 Jan '18

Hi guys, Why originate retry call when I answer? $con->api("originate","{originate_retries=0,origination_uuid=$uuid}user/1001); If I answer the call made by originate, it goes and repeat again Thanks

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Re: [Freeswitch-users] Received an unsupported RTCP packet version 3
by Madovsky 16 Jan '18

16 Jan '18

As the warning says, FS does not support RTCP version 3 message, maybe a patch would be welcome. On 1/16/2018 3:53 AM, Alexander Haugg wrote: > Thanks, but no, I need the event socket event "RECV_RTCP_MESSAGE". > If the message will written, then the event will not fired. > Switch_rtp.c line 6650 > } else { > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), > SWITCH_LOG_WARNING, "Received an unsupported RTCP packet version %d\n", msg->header.version); > return SWITCH_STATUS_FALSE; > > -----Ursprüngliche Nachricht----- > Von: Madovsky [mailto:infos@madovsky.org] > Gesendet: Dienstag, 16. Januar 2018 12:17 > An: Alexander Haugg <Alexander.Haugg(a)c4b.de> > Betreff: Re: [Freeswitch-users] Received an unsupported RTCP packet version 3 > > I think it's safe to ignore unless you need > > any management with RTCP > > On 1/15/2018 11:16 PM, Alexander Haugg wrote: >> I have the same Problem with the Bria 4 Client. >> My FS Verison is "Version 1.9.0+git~20170608T004121Z~00ab2b40b4~64bit (git 00ab2b4 2017-06-08 00:41:21Z 64bit)" >> >> -----Ursprüngliche Nachricht----- >> Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Mikhail Demekhov >> Gesendet: Donnerstag, 10. November 2016 09:27 >> An: freeswitch-users(a)lists.freeswitch.org >> Betreff: [Freeswitch-users] Received an unsupported RTCP packet version 3 >> >> Hello! >> >> I have FreeSWITCH Version 1.6.12. >> After connecting partner FS-1.7 I had a lot of warning: >> >> [WARNING] switch_rtp.c:6207 Received an unsupported RTCP packet version 3 >> >> How to fix it? >> >

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mod_fifo / mod_callcenter question
by Abaci B 16 Jan '18

16 Jan '18

Hi all, I was wondering if anyone would know a way to implement the option for a caller waiting in a queue to be able to hang up while waiting and not loose their position in the queue and get a call from the system when they are next in line to speak to an agent. Thanks

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Freeswitch don't send a RTCP package
by Alexander Haugg 16 Jan '18

16 Jan '18

Hi, I am using the Bria 4 client with the Freeswitch Version 1.9.0+git~20170608T004121Z~00ab2b40b4~64bit (git 00ab2b4 2017-06-08 00:41:21Z 64bit) The Parameter "<param name="rtcp-audio-interval-msec" value="5000" />" is set in the SIP profile. The Client send the SDP: v=0 o=- 13160559761493634 2 IN IP4 my_ip s=Bria 4 release 4.8.1 stamp 84929 c=IN IP4 my_ip t=0 0 a=ice-ufrag:6ffe8a a=ice-pwd:943efb7d3a5ae8f3bbd86bfa8ef89e05 m=audio 59196 RTP/AVP 0 101 c=IN IP4 my_ip a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ssrc:1606096547 cname:tId74Q/00rktO5VD a=sendrecv a=rtcp:59197 IN IP4 my_ip a=candidate:1 1 UDP 659136 my_ip 59196 typ host a=candidate:1 2 UDP 659134 my_ip 59197 typ host a=remote-candidates:1 fs_ip 30992 2 fs_ip 30993 a=ssrc:1606096547 cname:tId74Q/00rktO5VD what can i do?

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FreeSWITCH mod_watson in beta
by Brian West 16 Jan '18

16 Jan '18

Requirements: 1. Debian 8 2. FreeSWITCH installed from packages 3. You can purchase licenses here https://freeswitch.com/cart.php?gid=2 (Use discount code GETWATSON for 25% off, expires 01/31/2018) Each license comes with one port of TTS and one port of ASR. You'll receive an email with the .deb attached that you can install via dpkg -i freeswitch-mod-watson.deb To install your license run /usr/sbin/validator Follow the prompts, Unzip the resulting .zip file into /etc/freeswitch/ Then as root start the license server: /usr/sbin/freeswitch_license_server Then fs_cli -x 'load mod_watson' You should now visit https://www.ibm.com/watson/ and sign up for an account. Once you have your api token update /etc/freeswitch/autoload_configs/watson.xml with the appropriate details for your Watson account. If you have questions or need assistance please email support(a)freeswitch.com Thanks, -- Brian West | Co-founder and Developer Need Commercial support? email sales(a)freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&e…> Email: brian(a)freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwg…>

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Re: [Freeswitch-users] Originate doubt
by Raimundo Pérez Nieves 16 Jan '18

16 Jan '18

I tried but it does not work. Iam not trying to bridge. Just originate an outbound call Enviado desde mi iPhone El 16/01/2018, a la(s) 13:03, Madovsky <infos(a)madovsky.org> escribió: > you should use "expand originate" > > since you use variables into your bridge > >> On 1/16/2018 3:46 AM, Raimundo Pérez Nieves wrote: >> Hi guys >> Why this works? >> <action application="bridge" data="sofia/external/550075372035172(a)64.34.75.23"/> >> And this not? >> $con->bgapi("originate","{origination_uuid=$uuid}sofia/external/55007537203517%64.34.75.23 5004"); >> And I exchange % for @. >> Get this error >> 2018-01-16 11:18:39.410205 [WARNING] mod_sofia.c:4652 Cannot locate registered user 550075372035172(a)64.34.75.23 >> _________________________________________________________________________ >> 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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