April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch video conference (broadcast)
by Adam Ben-Ayoun 13 Apr '15

13 Apr '15

Hi guys, I have a conference configured with conference-flags and member-flags set to video-bridge and I am using SIPjs as a WebRTC client to test the setup. My goal is to broadcast from one client to all the rest. This seems to work fine with 2 clients, meaning, one client gets video from the other client, now, when I am adding a 3rd client, it doesn't get video (verified with chrome://webrtc-internals). Any idea? Am I missing something? Thanks! Adam

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Freeswitch basic VP8 passthrough
by Adam Ben-Ayoun 13 Apr '15

13 Apr '15

Hi guys, I am trying to setup a simple WebRTC video conference using VP8 with mod_conference, while audio conferencing works fine, I am not able to setup a video conference using clients like SIPjs, basically, I am opening 2 clients and expect to see video streaming to each side. I did the following: 1. Added the following config to my conference profile in conference.conf.xml: <param name="conference-flags" value="video-bridge"/> <param name="member-flags" value="video-bridge"/> 2. Added VP8 to vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=iSAC@16000h ,VP8,OPUS,G722,PCMU,PCMA,GSM"/> Any idea? P.S The client seem to pass the video params correctly in the SDP, but Freeswitch still responses with the following SDP: v=0 o=FreeSWITCH 1428032845 1428032846 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 52.1.148.244 t=0 0 a=msid-semantic: WMS QyrjMP4wDDWQeK1hU3mBsJvovhMMWjKD m=audio 23118 UDP/TLS/RTP/SAVPF 111 126 106 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1; minptime=10 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 83:F4:57:6D:F9:40:C7:E8:28:6D:59:AE:5F:08:23:3E:9E:17:1E:2F:D8:A9:D5:E7:DB:13:92:B5:DE:4A:66:CA a=rtcp-mux a=rtcp:23118 IN IP4 X.X.X.X a=ssrc:3173617819 cname:I1MkHHGaasQf3InN a=ssrc:3173617819 msid:QyrjMP4wDDWQeK1hU3mBsJvovhMMWjKD a0 a=ssrc:3173617819 mslabel:QyrjMP4wDDWQeK1hU3mBsJvovhMMWjKD a=ssrc:3173617819 label:QyrjMP4wDDWQeK1hU3mBsJvovhMMWjKDa0 a=ice-ufrag:CzeU8A9uHL6uVHXu a=ice-pwd:zzF67b78uOeomr1PR6zQwzb4 a=candidate:8368394426 1 udp 659136 X.X.X.X 23118 typ host generation 0 m=video 0 UDP/TLS/RTP/SAVPF 19 Thanks! Adam

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Diaplan_auth to authenticate calls
by Gustavo Silva 13 Apr '15

13 Apr '15

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transfer on fail in originate string for A Leg
by Jayesh Nambiar 12 Apr '15

12 Apr '15

Hi, I am looking at parking the A Leg of the call when the B Leg in the originate string fails for any reason. In the current scenario the A Leg just hangs up even if it has the park_after_bridge variable set to true. Basically, the park_after_bridge comes into the picture when the B leg is answered and then B Leg hangs up. I tried the following and it worked: originate {transfer_on_fail='USER_BUSY BRIDGE_FAIL XML bridge_fail',park_after_bridge=true}sofia/abc/1234@dest_ip &bridge(sofia/abc/5678@dest_ip) Then I have a dialplan as: <context name=bridge_fail> <extension name="bridge_fail"> <condition field="destination_number" expression="BRIDGE_FAIL"> <action application="park"/> </condition> </extension> </context> This works perfectly when the B-Leg fails with USER_BUSY response and I figured that if I comma separate the hangup causes it will work for the causes that I have defined. But I want the A Leg to get parked if the B Leg fails in any scenario so that I dont miss out on some weird Hangup Cause. I read about auto_cause which can be used in place of the HANGUP Causes but that doesn't seem to work as expected. Is there a straight forward way where I can replace the USER_BUSY with a string which will catch all hangup cause and allow the call to be taken to destination and context defined?? Any assistance will be of great help. Thanks, - Jayesh

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Video with mod_conference
by Adam Ben-Ayoun 12 Apr '15

12 Apr '15

Hi guys, I am experimenting with mod_conference and video, I have a conference (not in video-bridge mode), I am connecting with 2 clients using SIPjs, my goal is to have video from client1 to client2. Here is what I'v tried: Assume client1 is the broadcaster while client2 is the receiver. Client1 connects to the conference first, he is floor at this point: freeswitch@internal> conference list Conference 1234-172.30.3.212 (2 members rate: 16000 flags: running|answered|enforce_min|dynamic|exit_sound|enter_sound) 1;sofia/internal/anonymous.2po4ni(a)anonymous.invalid ;7c8e7494-decc-11e4-93cd-2391a0614a29;anonymous.2po4ni;anonymous.2po4ni;video|floor;0;0;0;0 Now, client2 joins in, as expected, client1 is still floor: freeswitch@internal> conference list Conference 1234-172.30.3.212 (2 members rate: 16000 flags: running|answered|enforce_min|dynamic|exit_sound|enter_sound) 2;sofia/internal/anonymous.hgbmm4(a)anonymous.invalid ;80a2f38e-decc-11e4-93d3-2391a0614a29;anonymous.hgbmm4;anonymous.hgbmm4;video;0;0;0;0 1;sofia/internal/anonymous.2po4ni(a)anonymous.invalid ;7c8e7494-decc-11e4-93cd-2391a0614a29;anonymous.2po4ni;anonymous.2po4ni;video|floor;0;0;0;0 No video goes through to client2. Now, if I am setting client2 as floor and then setting back client1 as floor, video start to come in.. And just before the video starts, the log shows that client2's video DTLS handshake becomes READY: 2015-04-09 15:24:40.391590 [DEBUG] switch_core_media.c:4361 sofia/internal/anonymous.2po4ni(a)anonymous.invalid Video thread resumed Echo is off 2015-04-09 15:24:41.851825 [INFO] switch_rtp.c:2923 Changing video DTLS state from HANDSHAKE to SETUP 2015-04-09 15:24:41.851825 [INFO] switch_rtp.c:2831 video Fingerprint Verified. 2015-04-09 15:24:41.851825 [INFO] switch_rtp.c:3373 Activating Video Secure RTP SEND 2015-04-09 15:24:41.851825 [INFO] switch_rtp.c:3351 Activating Video Secure RTP RECV 2015-04-09 15:24:41.851825 [INFO] switch_rtp.c:2871 Changing video DTLS state from SETUP to READY In addition, even before doing the floor toggle to get the video going, I see mod_conference's video thread doing: switch_core_session_write_video_frame(imember->session, vid_frame, SWITCH_IO_FLAG_NONE, 0); on line 2991. Something in the video thread seems to delay/lock the video DTLS handshake.... Any idea? Thanks! Adam

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Freeswitch / FusionPBX faxback <-> faxback extensions T.38 Fax FAIL: Ack Time Out
by Bill Ross 12 Apr '15

12 Apr '15

Hi; Faxback Support - SUP-00019595 I am attempting to get T38 Fax working between two FS/FusionPBX installations using faxback VOIP plugin extensions. The fax TX (FS#1) endpoint drops the call due to "Ack Timeout" after two pages. Faxback VoIP client plugin: http://www.faxback.com/Products/FreeMSFaxDownload.aspx I am able to successfully receive a fax using this faxback endpoint if transmitted by FS/spandsp. Note that zrtp is disabled for fax calls, due to T38 udpl (not rtp media) incompatibility Thanks Brian West!: "T.38 Fax takes place over UDPTL, There for it can NOT be encrypted as there was never really a standard for doing so." Details (both FS's): SIP Provider: freephoneline.ca (claims T38 capable) freeswitch - 1.4.18 x86_64 Topology: FS#1: rossco.org, using faxback plugin (x1019) to send fax FS#2:rossco.duckdns.org, using faxback plugin (x1019) to receive fax (faxback x1019)FS#1 -> cloud -> FS#2(faxback x1019) Key variables for fax sender (FS#1), during call: $domain= rossco.org Faxback Fax extension: 1019 t38_passthru = true zrtp_secure_media = false fax_enable_t38 = false (not set) fax_enable_t38_request = false (not set) fax_use_ecm = off (not set) inbound-late-negotiation = true ignore_early_media = true Key variables for fax receiver (FS#2), during call: $domain= rossco.duckdns.org Faxback Fax extension: 1019 t38_passthru = true zrtp_secure_media = false fax_enable_t38 = false (not set) fax_enable_t38_request = false (not set) fax_use_ecm = false (not set) inbound-late-negotiation = true ignore_early_media = false Full Logs w/siptrace: Sender(rossco.org): https://pastebin.freeswitch.org/24117 Receiver(rossco.duckdns.org): https://pastebin.freeswitch.org/24118 Note that the SDP's are correct for T38: v=0 o=Sippy 177245968 2 IN IP4 208.65.240.165 s=FreeSWITCH t=0 0 m=image 49714 udptl t38 c=IN IP4 208.65.240.165 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:200 a=T38FaxMaxDatagram:72 a=T38FaxUdpEC:t38UDPRedundancy Questions: 1 - According to my interpretation of logs, T38 and t38_passthru is negotiated 2 - I do not have the expertise (yet) to determine the reason for "Ack Timeout" 2 - What am I missing? Thanks; Bill Ross

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mod_callcenter
by Ali Jibran 11 Apr '15

11 Apr '15

Is it possible to stop the MOH and let the caller hear a ringing sound as soon as the agent is in ring ready state?

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ARM servers at Scaleway.com
by Stanislav Sinyagin 10 Apr '15

10 Apr '15

I tested a dedicated ARM server offered at Scaleway, and it runs smoothly. The only issue, the initial kernel does not support high-resolution timers, so you need to change to another kernel and reboot. Details here: https://txlab.wordpress.com/2015/04/08/installing-freeswitch-on-scaleway-c1/ a pretty decent offer for a small hosted PBX. Their datacenters are located near Paris.

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voicemail: customize
by Денис 10 Apr '15

10 Apr '15

Hello. We need to change the behavior of voicemail module, ie disable phrases after message. how can this be done? module voicemail_ivr not earned as needed.

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how to increase/decrease stream volume.
by Aqs Younas 10 Apr '15

10 Apr '15

Hi, users. I am playing streams with mod_vlc. Some streams are higher in volume and other lower. I want to control stream volume by pressing some dtmf. Is there any way to do so, like in mod_conference we can increase/decrease volume by pressing some dtmf. Thanks.

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