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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

1 0

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

9 22

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

2 3

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 ?

13 28

Freeswitch and twilio
by Michael Nielsen 02 Oct '15

02 Oct '15

I'm trying to setup twilio as gateway following this guide: 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?

2 2

Audio delays and jitter
by Danny Gershman 02 Oct '15

02 Oct '15

We are using the OPUS codec with Chrome and WebRTC to make SIP calls into FS conferences. We have packet captures at the edge of network that sound clear, but when they get to FreeSWITCH they sound distorted via delays and clipping. I've tried several combinations of jitterbuffer settings, disabled/enabling RTP timers, conference. I have a mechanism I use in which I replay the RTP packets using SIPP and try them out. Is there a way to dynamically adjust the jitterbuffer so that the right size is being used? Right now I have it set at 180:540:540. Maybe there is some other setting I could try that would help? Perhaps there's a way to force OPUS to use less bandwidth at the expense of quality? I'm willing to share the PCAPs some examples. Thanks in advance. --Danny Gershman

7 23

Reg exp in tutorial
by Michael Nielsen 02 Oct '15

02 Oct '15

In the doc's, should the 10_gateway_inbound.xml contain: expression="^[a-z\.]" instead of: expression="^[a-z]" So it can catch the domain and context int.example.net.

2 3

Setting loglevel to warning removes log in fs_cli
by Michael Nielsen 02 Oct '15

02 Oct '15

I've set <param name="loglevel" value="warning"/> in switch.conf.xml. This removes a lot of output from the fs_cli. But if I want to try and log things to debug I'm not able to from fs_cli, I need to change the setting in switch.conf.xml and restart FreeSWITCH. How can I easily change debug/log level from fs_cli? I've tried /debug and /log, without any luck.

5 7