- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch and twilio
by Ahmed Habiba 07 Oct '15

07 Oct '15

if you have a deep look at inbound gateway it is for receiving calls from Twilio as you can see below, this line informing freeswitch where to route inkling call. <param name="extension" value="1000"/> > > > From: Michael Nielsen <mic.niel84(a)gmail.com> > Date: October 2, 2015 at 11:24:51 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Subject: Re: [Freeswitch-users] Freeswitch and twilio > > > Outbound works fine. > > Without having the inbound gateway calls are still routed to my FS to my public context from Twilio. > Adding the twilio-inbound doesn't seem to change anything...? > > > > On Friday, October 2, 2015, Brian West <brian(a)freeswitch.org <mailto:brian@freeswitch.org>> wrote: > Well you'd send outbound calls out the outbound and the inbound one is just to register with twilio so they know where to send your inbound calls. Nothing more. > > On Fri, Oct 2, 2015 at 1:46 PM, Michael Nielsen <mic.niel84(a)gmail.com <javascript:_e(%7B%7D,'cvml','mic.niel84(a)gmail.com');>> wrote: > I'm trying to setup twilio as gateway following this guide: > https://www.twilio.com/docs/sip-trunking/sample-configuration#freeswitch <https://www.twilio.com/docs/sip-trunking/sample-configuration#freeswitch> > > In their example of twiliotrunk.xml they have twilio-inbound and twilio-outbound. > > What is the difference between the two gateways? I know their names indicate inbound and outbound calling, but how can Freeswitch tell the difference between them. Especially how can Freeswitch now to use twilio-inbound for incoming calls from twilio and not just twilit-outbound as the only difference is register-proxy and expire-seconds between the two? > > Make sense? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <javascript:_e(%7B%7D,'cvml','consulting(a)freeswitch.org');> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <javascript:_e(%7B%7D,'cvml','FreeSWITCH-users(a)lists.freeswitch.org');> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > -- > Brian West > brian(a)freeswitch.org <javascript:_e(%7B%7D,'cvml','brian(a)freeswitch.org');> > > Twitter: @FreeSWITCH , @briankwest > http://www.freeswitchbook.com <http://www.freeswitchbook.com/> > http://www.freeswitchcookbook.com <http://www.freeswitchcookbook.com/> > Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest > > > > _______________________________________________ > 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

7 37

Why only 3 frame sizes in mod_opus?
by Michel Aoun 07 Oct '15

07 Oct '15

I think that only 3 frame sizes (10ms, 20ms and 40ms) are implemented in mod_opus? Is there a reason for not adding the entire list of 2.5, 5, 10, 20, 40, and 60 ms (https://tools.ietf.org/html/rfc6716#section-2.1.4)? Thanks

2 1

FS priority
by Bote Man 07 Oct '15

07 Oct '15

I'm trying to set the priority on a new FreeSWITCH installation built from master on Debian 8 running on bare metal. It is currently running at "very low" priority according to Resource Monitor in the GUI and 'top' reports FS is running at priority = -2 (that's negative two) and nice = 19 So with the way FreeSWITCH is now launched by systemd is it considered a service or a user application that is simply run in the background? This affects how systemd treats its control groups and priority and how I will go about troubleshooting this. Thanks. Bote

7 21

Install Error (libvpx2)
by Yücel ALTUNAY 07 Oct '15

07 Oct '15

Hi, I have an error below while instaling Freeswitch. making all mod_av make[4]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_av' Makefile:833: *** You must install libvpx2-dev to build mod_av. Stop. make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod/applications/mod_av' make[3]: *** [mod_av-all] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [all] Error 2 How can i solve this problem. Thank you.

2 1

Opus PLC issue - can this be a bug?
by Thomas 07 Oct '15

07 Oct '15

Hello list, on a connection with an unfortunate amount of packet loss, I'm getting a lot of these: mod_opus.c:289 FEC in CELT_ONLY mode ?! ... with the occasional ... mod_opus.c:801 Decoder Error: corrupted stream fs:4080 plc:false! switch_core_io.c:622 Codec RAW Signed Linear (16 bit) decoder error! [9] ... and ... switch_rtp.c:875 Invalid STUN/ICE packet received 493 bytes That's accompanied by robotic distortion. This sounds more like FS is receiving total junk rather than missing packets, but surely there must be checksums preventing errors inside these packets? Can this really be a bug in FS, or do I need an exorcist for that internet connection? :) Cheers! Thomas P.S.: Last nights master

6 6

Cannot get verto video conf demo working
by Tao Zhang 06 Oct '15

06 Oct '15

Hi, I am new to FreeSwitch, WebRTC and etc.. so please bear with me. :-) I tried to set up the latest FreeSwitch Video Conf demo on my server and could not get it working. I can call extension 3500 and enter the conf, but I only see a black area with no video and no sound. On the other hand, if I access the official video conf demo page https://conference.freeswitch.org/verto/#page-main from FreeSwitch, it works fine. I have been trying to figure out what is wrong with my FS setup. I cannot see any obvious errors from the FS server log and from the JS console. But from the chrome://webrtc-internals/, two ssrc_xxxx_send stats show bytes sent are zero. Also two ssrc_xxxx_recv stats show bytes received also zero. So it seems there is some problem related to the peer connection between the client and the FS server? I also noticed another difference in webrtc-internals outputs between my server setup and the official working one. For my set up, Channel-video-1 shows: >>>>>>>>>>>>>>>>>>>>>>>>>>>>> Channel-video-1 Statistics Channel-video-1timestamp10/6/2015, 5:48:56 PMgoogComponent1 localCertificateId googCertificate_29:3D:26:CC:D2:E0:1D:35:1E:0B:66:EA:B4:08:B7:FC:EE:FC:7B:0A:76:C5:6F:AE:B7:BD:D4:7B:73:10:E6:F7 selectedCandidatePairIdConn-video-1-0 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> For the official setup, Channel-video-1 shows: >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Statistics Channel-video-1timestamp10/6/2015, 5:37:53 PM selectedCandidatePairIdConn-video-1-0googComponent1dtlsCipher TLS_ECDHE_RSA_WITH_AES_256_CBC_SHAlocalCertificateId googCertificate_56:D3:CE:E1:0C:81:05:2A:FB:E7:17:3F:34:FB:21:F3:DD:F8:67:99:65:2D:D6:41:57:76:64:3A:ED:00:96:01 remoteCertificateId googCertificate_EB:11:4A:1F:D3:CE:18:2B:16:CE:6A:F4:71:5D:F6:A6:9E:22:63:8C srtpCipherAES_CM_128_HMAC_SHA1_80 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> I guess the dtls-srtp connection is not set up properly, since remoteCert and other fields are missing in the webrtc-internals dump for my setup. But I cannot find any related error msgs and I have no idea how to debug further. Could anybody help? Thanks so much Tao

1 0

ACL not working
by Charles Bujold 06 Oct '15

06 Oct '15

We are trying to have Freeswitch work for 2 LAN segments (192.168.20.0/24 and 192.168.25.0/24) We have create an ACL for Freeswitch as follows. However, the lan segment 192.168.25.0/24 cannot make calls, their calls are routed via the WAN and not internally ( the Freeswitch server is on the 192.168.20.0/24 segment). What are we missing? Not certain when to use Deny and Allow settings in ACL? Any help would be appreciated. What is the proper configuration for the ACL xml? Internal Profile is listed below if it needs to be changed please let us know. Thanks <configuration name="acl.conf" description="Network Lists"> <network-lists>  <list name="lan" default="deny"> <node type="deny" cidr="192.168.20.0/24"/> <node type="allow" cidr="192.168.25.0/24"/> </list> <list name="localnet.auto" default="allow"> <node type="allow" cidr="192.168.20.0/24"/> <node type="allow" cidr="192.168.25.0/24"/> </list>  <list name="domains" default="deny">  <node type="allow" domain="$${domain}"/>  <node type="allow" cidr="192.168.20.0/24"/> <node type="allow" cidr="192.168.25.0/24"/> </list> </network-lists> </configuration> Our Internal SIP Profile is configured as follows: <profile name="internal">   <aliases>  </aliases>  <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains>     <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="public"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="$${recordings_dir}"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.w av"/> <param name="manage-presence" value="true"/> <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/> <param name="presence-privacy" value="$${presence_privacy}"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="$${internal_ssl_enable}"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${internal_tls_port}"/> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-policy" value="none"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="challenge-realm" value="auto_from"/> <param name="aggressive-nat-detection" value="true"/> </settings> </profile>

3 2

FreeSWITCH Week in Review (Master Branch) September 26th-October 2nd
by krice＠freeswitch.org 05 Oct '15

05 Oct '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1j0UGtV FreeSWITCH Week in Review (Master Branch) September 26th-October 2nd Hello, again. This past week in the FreeSWITCH master branch we had 90 commits! Most of the features for this week went toward the verto communicator and are: created a source map file, created the reset banner action, floor and presenter badges, and locked icon in floorLocked status, added an About screen with version information and links to FS.org and added a link to Confluence with documentation for VC, and made mute/unmute audio/video clickable. Other features this week include: refactoring the local_stream API to be more consistent and add auto complete, compatibility with Solaris 11 process privileges, improvements to the way FEC info is detected within frames by adding support for ptimes higher than 20 ms for FEC detection and improvements to jitter buffer debugging in mod_opus. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8243 [mod_opus] Improve the way FEC info is detected within frames by adding support for ptimes higher than 20 ms for FEC detection FS-8161 [mod_opus] Keep FEC enabled only if loss > 10 ( otherwise PLC is supposed to be better) FS-8179 [mod_opus] Improvement on new jitter buffer debugging (debug lookahead FEC) FS-8254 [verto_communicator] Create a source map file FS-8263 [verto_communicator] Created the reset banner action, floor and presenter badges, and lock icon in floorLocked status FS-8288 [verto_communicator] Added an About screen with version information and links to FS.org and added a link to Confluence with documentation for VC FS-8289 [verto_communicator] Make mute/unmute audio/video clickable FS-8287 [mod_local_stream] Refactor local_stream API to be more consistent and add auto complete FS-8195 [core] Compatibility with Solaris 11 process privileges Improvements in build system, cross platform support, and packaging: FS-8236 Fixed build without libyuv on compilers that error on unused static function and fixed ifdefs for building without libyuv FS-8239 [mod_av] Fixed the default value to avoid failed build on CentOS 7 FS-8255 [Debian] Fixed codename changes since Jessie was released as stable FS-8271 [Debian] Simplify package building for the default case FS-8270 [Debian] Fix for package installation failing if /etc/freeswitch/tls is missing FS-8285 [Debian] Removed heart attack inducing warning message when updating deb packages FS-7817 Removed use of _NONSTD for Windows builds of spandsp, so (hopefully) eliminate compatibility problem The following bugs were squashed: FS-8221 [verto_communicator] Fix number in call history FS-8223 [verto_communicator] Fixing members list layout when callerid is too long FS-8225 [verto_communicator] Avoid duplicate members when recovering calls FS-8214 [verto_communicator] Better handling calls in VC, answering them respecting useVideo param FS-8291 [verto_communicator] Fixed contributors url FS-8229 [verto_communicator] Changing moderator actions bullet menu color to #333 FS-8219 [verto_communicator] Fix for camera not deactivating after init or after hangup FS-8245 [verto_communicator] Fix for Video Resolutions available in “Video Quality” drop down not always correct FS-8251 [verto_communicator] Factory reset now clears all local storage FS-8257 [verto_communicator] Fixed configuration provision url because configuration doesn’t work with `grunt serve` and non pathname urls FS-8273 [verto] [verto_communicator] Clear the CF_RECOVERING flag in a spot that was missed FS-8260 [verto_communicator] Prompt for banner text FS-8232 [mod_conference] Conference sending too many video refresh requests FS-8241 [mod_conference] Fix for conference stops playing video when local_stream changes source FS-8261 [mod_conference] Fixed the conference segfaulting when trying to reset the banner FS-8220 [core] Fix for DTMF not working between telephone-event/48000 A leg and telephone-event/8000 B leg FS-8166 [core] Mute/unmute while shout is playing audio fails because the channel “has a media bug, hard mute not allowed” FS-8252 [core] Fixed a crash in rtp stack on dtls pointer FS-8283 [core] Handle RTP Contributing Source Identifiers (CSRC) FS-8275 [core] Fix for broken DTMF FS-8282 [core] Fix for sleep is not allowing interruption by uuid_transfer FS-8240 [mod_local_stream] Fixed a/v getting out of sync when running in the background and added video profile parameter for recording 264 and made it default FS-8216 [mod_av] Fixed a regression in hup_local_stream from last commit FS-8274 [mod_av] Fixed a memory leak caused by images not being freed in video_thread_run FS-7989 [fixbug.pl] Escape double quotes from summary and added more debugging data FS-8246 [mod_json_cdr] Use seconds as default value for delay parameter FS-8256 [mod_opus] More FMTP cleanup FS-8284 [mod_opus] Use use-dtx setting from config in request to callee. FS-8234 [mod_opus] Send correct (configured) fmtp ptime,minptime,maxptime when originating call And, this past week in the FreeSWITCH 1.4 branch we had 6 new commits merged in from master. And the FreeSWITCH 1.4.23 release is here! Go check it out! The following bugs were squashed: FS-8190 [mod_event_socket] When using nixevent, freeswitch stops sending us certain custom event that were NOT part of the nixevent command FS-7673 [mod_v8] ODBC NULL value incorrectly evaluated FS-8215 Fixed the accuracy of MacOSX nanosleep FS-8269 [mod_sms] Fixed a build issue FS-8244 [mod_dptools] Fixed a compilation issue

1 0

Presence isolaiton/alias
by Артур Краев 05 Oct '15

05 Oct '15

Is there any way to achieve presence isolation for arbitrary users in one domain? I'm look fuctionality which works like asterisk hints: https://wiki.asterisk.org/wiki/display/AST/Extension+State+and+Hints It should work like: User 1000 may have internal number 100 User 1001 may have internal number 101 both use context ctx1, so user 1000 may subscribe to 101 presence but not 1001 and user 1001 may subscribe to 100 presence but not 1000 and users in other context may not subscribe nowhere 100, 101 and 1000, 1001 How can this may be achieved? -- With best regards, Artur Kraev

1 0

ESL Reporting of RTCP / Jitter is incorrect... does not match RTP originally reported
by David S. Dee 05 Oct '15

05 Oct '15

I am trying to do some external actions based on the RTCP reporting of Jitter that comes in. My problem is that the numbers that show up in the freeswitch logs no longer match what gets reported via Event'ing eg, the freeswitch logs show: 2015-09-23 08:22:24.221437 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:24.241238] SSRC[626145724]RTT[0.092102] A[874528193] - DLSR[38600] - LSR[874483557] 2015-09-23 08:22:29.622369 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:29.640323] SSRC[626145724]RTT[0.106003] A[874882028] - DLSR[63832] - LSR[874811249] 2015-09-23 08:22:35.041676 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:35.60396] SSRC[626145724]RTT[0.142044] A[875237238] - DLSR[88997] - LSR[875138932] 2015-09-23 08:22:40.461243 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:40.480321] SSRC[626145724]RTT[0.181702] A[875592438] - DLSR[113836] - LSR[875466694] 2015-09-23 08:22:45.882464 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:45.900353] SSRC[626145724]RTT[0.215042] A[875947645] - DLSR[139264] - LSR[875794288] 2015-09-23 08:22:51.301265 [NOTICE] switch_rtp.c:5206 Receiving an RTCP packet[2015-23-08 14:22:51.320380] SSRC[626145724]RTT[0.249069] A[876302852] - DLSR[164560] - LSR[876121969] And the (short form) of what I receive via the ESL are: Sep 23 09:33:55: RTCP message not sent, too soon: SSRC: 255239bc, RTP-Timestamp: 0, Sender-Packet-Count: 0, Octect-Packet-Count: 874483557, Last-RTP-Timestamp: 2018933311, RTP-Rate: 8000, Capture-Time: 1443018144241365, Source0-Fraction: 129, Source0-Lost: 3388998144, Source0-Highest-Sequence-Number-Received: 626145724, Source0-Jitter: 16777216, Source0-LSR: 788957780, Source0-DLSR: 17891 Sep 23 09:34:00: RTCP message not sent, too soon: SSRC: 255239bc, RTP-Timestamp: 0, Sender-Packet-Count: 0, Octect-Packet-Count: 874811249, Last-RTP-Timestamp: 2018973151, RTP-Rate: 8000, Capture-Time: 1443018149640379, Source0-Fraction: 129, Source0-Lost: 3388998144, Source0-Highest-Sequence-Number-Received: 626145724, Source0-Jitter: 16777216, Source0-LSR: 788957780, Source0-DLSR: 17891 Sep 23 09:34:05: RTCP message not sent, too soon: SSRC: 255239bc, RTP-Timestamp: 0, Sender-Packet-Count: 0, Octect-Packet-Count: 875138932, Last-RTP-Timestamp: 2019013151, RTP-Rate: 8000, Capture-Time: 1443018155060479, Source0-Fraction: 129, Source0-Lost: 3388998144, Source0-Highest-Sequence-Number-Received: 626145724, Source0-Jitter: 16777216, Source0-LSR: 788957780, Source0-DLSR: 17891 Sep 23 09:34:15: RTCP message not sent, too soon: SSRC: 255239bc, RTP-Timestamp: 0, Sender-Packet-Count: 0, Octect-Packet-Count: 875794288, Last-RTP-Timestamp: 2019093311, RTP-Rate: 8000, Capture-Time: 1443018165900409, Source0-Fraction: 129, Source0-Lost: 3388998144, Source0-Highest-Sequence-Number-Received: 626145724, Source0-Jitter: 16777216, Source0-LSR: 788957780, Source0-DLSR: 17891 Sep 23 09:34:20: RTCP message not sent, too soon: SSRC: 255239bc, RTP-Timestamp: 0, Sender-Packet-Count: 0, Octect-Packet-Count: 876121969, Last-RTP-Timestamp: 2019133311, RTP-Rate: 8000, Capture-Time: 1443018171320579, Source0-Fraction: 129, Source0-Lost: 3388998144, Source0-Highest-Sequence-Number-Received: 626145724, Source0-Jitter: 16777216, Source0-LSR: 788957780, Source0-DLSR: 17891 (yes, the timestamps are off right now, that's actually fine). This used to work in an older version of Freeswitch, but when I rebuilt my system with an updated FS version last month, this broke. You'll note that the Sender-Packet-Count is (obviously) wrong, as well as the RTP-Timestamp, and the DLSR and LSR... none of them advance, while you can see them advancing in the freeswitch log. I suspect it is alignment of the values in the structure that gets populated for the ESL, but my C is rusty enough that I can't trace it anymore. Is anyone else able to view the Event data for RTCP packets? Thanks, David

3 4