October 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

Ring group and call-forwards.
by Igor Olhovskiy 05 Oct '15

05 Oct '15

Hi! I'm using Fusion to configure FreeSwitch and encountered a bug, that MIGHT be related to FreeSwitch. Idea - I have an extension with ring group on it made this way bridge({ignore_early_media=true}[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=0,dialed_extension=108,extension_uuid=c728abd0-6107-460d-b1b6-4861c6b6998f]user/ 108(a)pbx.securenetvox.net,[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=15,dialed_extension=107,extension_uuid=4bd03430-767b-41da-8b78-0902d590cdc2]user/ 107(a)pbx.securenetvox.net) And on extension 108 i have a custom dialstring: {presence_id=0038091908XXXX(a)pbx.securenetvox.net ,instant_ringback=true,domain_uuid=32ade86a-9d28-4468-9fc9-2a39fe9ffe50,sip_invite_domain= pbx.securenetvox.net,domain_name=pbx.securenetvox.net,domain= pbx.securenetvox.net ,extension_uuid=c728abd0-6107-460d-b1b6-4861c6b6998f,toll_allow='',sip_h_X-accountcode= pbx.securenetvox.net,accountcode=pbx.securenetvox.net }loopback/0038091908XXXX So, I dial from extension 107 to ring group extension and got this (part of log): ............. 2015-10-05 14:10:50.051507 [DEBUG] sofia.c:6658 Channel sofia/internal/ 107(a)pbx.securenetvox.net entering state [received][100] 2015-10-05 14:10:50.051507 [DEBUG] sofia.c:6668 Remote SDP: v=0 o=107-jitsi.org 0 0 IN IP4 212.232.26.242 s=- c=IN IP4 212.232.26.242 t=0 0 m=audio 25732 RTP/AVP 96 97 98 9 100 102 0 8 103 3 104 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 usedtx=1 a=rtpmap:97 SILK/24000 a=rtpmap:98 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:100 speex/32000 a=rtpmap:102 speex/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 iLBC/8000 a=rtpmap:3 GSM/8000 a=rtpmap:104 speex/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-xr:voip-metrics m=video 28234 RTP/AVP 105 99 a=rtpmap:105 H264/90000 a=fmtp:105 profile-level-id=4DE01f;packetization-mode=1 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=4DE01f a=recvonly a=imageattr:105 send * recv [x=[0-1440],y=[0-900]] a=imageattr:99 send * recv [x=[0-1440],y=[0-900]] a=nortpproxy:yes .............. 2015-10-05 14:10:50.131508 [NOTICE] switch_cpp.cpp:1341 [ring group] app_data: {ignore_early_media=true}[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=0,dialed_extension=108,extension_uuid=c728abd0-6107-460d-b1b6-4861c6b6998f]user/ 108(a)pbx.securenetvox.net,[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=15,dialed_extension=107,extension_uuid=4bd03430-767b-41da-8b78-0902d590cdc2]user/ 107(a)pbx.securenetvox.net EXECUTE sofia/internal/107(a)pbx.securenetvox.net bridge({ignore_early_media=true}[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=0,dialed_extension=108,extension_uuid=c728abd0-6107-460d-b1b6-4861c6b6998f]user/ 108(a)pbx.securenetvox.net,[sip_invite_domain=pbx.securenetvox.net ,confirm=false,leg_timeout=30,leg_delay_start=15,dialed_extension=107,extension_uuid=4bd03430-767b-41da-8b78-0902d590cdc2]user/ 107(a)pbx.securenetvox.net) 2015-10-05 14:10:50.131508 [DEBUG] switch_channel.c:1221 sofia/internal/ 107(a)pbx.securenetvox.net EXPORTING[export_vars] [domain_name]=[ pbx.securenetvox.net] to event 2015-10-05 14:10:50.131508 [DEBUG] switch_channel.c:1221 sofia/internal/ 107(a)pbx.securenetvox.net EXPORTING[export_vars] [origination_callee_id_name]=[96] to event 2015-10-05 14:10:50.131508 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [ignore_early_media]=[true] 2015-10-05 14:10:50.131508 [DEBUG] switch_ivr_originate.c:2554 Parsing session specific variables 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_domain]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [confirm]=[false] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [leg_timeout]=[30] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [leg_delay_start]=[0] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [dialed_extension]=[108] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [extension_uuid]=[c728abd0-6107-460d-b1b6-4861c6b6998f] 2015-10-05 14:10:50.131508 [DEBUG] switch_channel.c:1221 sofia/internal/ 107(a)pbx.securenetvox.net EXPORTING[export_vars] [domain_name]=[ pbx.securenetvox.net] to event 2015-10-05 14:10:50.131508 [DEBUG] switch_channel.c:1221 sofia/internal/ 107(a)pbx.securenetvox.net EXPORTING[export_vars] [origination_callee_id_name]=[96] to event 2015-10-05 14:10:50.131508 [DEBUG] switch_ivr_originate.c:2101 Parsing global variables 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [presence_id]=[00380919082171(a)pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [instant_ringback]=[true] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [domain_uuid]=[32ade86a-9d28-4468-9fc9-2a39fe9ffe50] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [sip_invite_domain]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [domain_name]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [domain]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [extension_uuid]=[c728abd0-6107-460d-b1b6-4861c6b6998f] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [sip_h_X-accountcode]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [DEBUG] switch_event.c:1698 Parsing variable [accountcode]=[pbx.securenetvox.net] 2015-10-05 14:10:50.131508 [INFO] mod_loopback.c:1064 Sending early media 2015-10-05 14:10:50.131508 [DEBUG] mod_opus.c:289 Opus encoder set bitrate to local settings [-1000bps] 2015-10-05 14:10:50.131508 [DEBUG] mod_opus.c:289 Opus encoder set bitrate to local settings [-1000bps] 2015-10-05 14:10:50.131508 [DEBUG] switch_core_media.c:2506 Set Codec sofia/internal/107(a)pbx.securenetvox.net opus/0 0 ms 960 samples 0 bits 2 channels 2015-10-05 14:10:50.131508 [DEBUG] switch_core_codec.c:111 sofia/internal/ 107(a)pbx.securenetvox.net Original read codec set to opus:116 2015-10-05 14:10:50.131508 [DEBUG] switch_core_media.c:5178 AUDIO RTP *[sofia/internal/107(a)pbx.securenetvox.net <107(a)pbx.securenetvox.net>] 10.0.20.60 port 17392 -> (null) port 0 codec: 0 ms: 20* *2015-10-05 14:10:50.131508 [ERR] switch_core_media.c:5843 AUDIO RTP REPORTS ERROR: [Missing remote host]* 2015-10-05 14:10:50.131508 [NOTICE] switch_core_media.c:5844 Hangup sofia/internal/107(a)pbx.securenetvox.net [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] ........ So, at some state original channel info is lost.... Is this a bug or something another? -- Best regards, Igor

1 0

MC8704 on mod-gsmopen
by Francis 05 Oct '15

05 Oct '15

Hi all, I've been looking into this card as a replacement for my current usb dongle setup, since 2g services are soon to shut down in rural areas here. The MC8704 is quad band WCDMA, so it would give access across all networks: http://www.m2mconnectivity.com.au/brands/sierra-wireless/sierra-wireless-ai… .. and possibly running it through a USB adapter like this: http://www.ebay.com.au/itm/271455696096?ssPageName=STRK:MESINDXX:IT&_trksid… ... though I'm not sure if this would work, with the card using the I2S interface. Is there any possibility this might work? I don't mind doing some testing, as long as it's feasible. Cheers, Francis

4 12

Freeswitch Videoconference mux and mp4 not working
by Enrico Grillo 04 Oct '15

04 Oct '15

Hi everybody, I'm new to freeswitch and I'm experimenting with the 1.6.2 distro installed on a Virtualbox VM with Debian 8.2. I have installed by "apt-get install freeswitch-all". My goal is to set up a private server to handle video conferences with 2-4 participants using PCs (Jitsy), iPad (Polycom) and Verto WebRtc. I have a mandatory requirement to record audio+broadcasted video on an mp4 file. Adding and configuring h26x codec I'm been succesful in doing direct video calls from Polycom to Jitsy and even using the default 3000 video conference room where every participant sees the speaker video. All this calls use video passthru h264. If I try to use the default 3500 room the audio works fine but the broadcasted video is black. I suspect that I have missed some required support for mux/transcode encoding and sending. I have also done some experiments on video recording adding mod-vlc with the aim to obtain mp4 video recordings but I can't figure how to make it work. If I do a video call to the preconfigured 9192 I can record and playback a .fsv video but unfortunately I'm unable to convert it to a standard mp4 file. If I edit default.conf replacing .fsv wuth .mp4 when I call 9192 I don't see any error message (even after log 7) but mod-vlc creates and empty mp4 file. What I' missing? Thank you for any help or hints Enrico Grillo Torino Italy

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FreeSWITCH Unstable Repo
by Brian West 03 Oct '15

03 Oct '15

FreeSWITCHers, To help speed the testing and release cycle up, Team FreeSWITCH has created a FreeSWITCH Unstable Debian repo, This repo is updated after every commit to master and the debs lag behind about an hour from each commit. WARNING this repo is untested and should NOT be used in production. How to use this repo: 1. New Debian Jessie system 2. Run These Commands: echo "deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main" > /etc/apt/sources.list.d/99FreeSWITCH.list echo "deb http://files.freeswitch.org/repo/deb/debian/ jessie main" >> /etc/apt/sources.list.d/99FreeSWITCH.list wget -O - https://files.freeswitch.org/repo/deb/debian/key.gpg |apt-key add - apt-get update apt-get install freeswitch-all 3. To update: apt-get update apt-get dist-upgrade Any commit that goes in while the current build is running will be included in the next build that starts. Questions? Mailing List / IRC / Hipchat Comments? Mailing List / IRC / Hipchat Bugs? https://freeswitch.org/jira/ Thanks, and happy testing! -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | 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

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Freeswitch to ignore 302 redirect in bridge app
by Igor Olhovskiy 03 Oct '15

03 Oct '15

Hi! Is there any option for freeswitch to ignore redirect turned on ip-phone? For example, I have bridge to multiple destinations and one of destination can simply ruin the logic turning on redirect on it's phone. -- Best regards, Igor

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unable to clone master " git clone https://freeswitch.org/stash/scm/fs/freeswitch.git"
by Shabbir abbasi 03 Oct '15

03 Oct '15

git clone https://freeswitch.org/stash/scm/fs/freeswitch.git Initialized empty Git repository in /usr/src/freeswitch/.git/ error: SSL write: error -5938 while accessing https://freeswitch.org/stash/scm/fs/freeswitch.git/info/refs fatal: HTTP request failed any solution ?

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