August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[ANN]: Odoo connector-telephony for FreeSWITCH new version
by Trever L. Adams 25 Aug '16

25 Aug '16

My module for OpenERP/Odoo to do click2dial and pull call details out of the system, etc. has been updated. If you find it of interest, please see the three sections below: https://github.com/treveradams/connector-telephony/tree/freeswitch-7.0 This branch is for OpenERP 7.0. I no longer have an installation of 7.0, but this should fix many bugs. If you use it and you have problems, please let me know. I want to keep this working as long as people use it. If it doesn't work for you, please send me a private email. If it works, please comment on https://github.com/OCA/connector-telephony/pull/58 (feel free to comment there if it doesn't work). https://github.com/treveradams/connector-telephony/tree/freeswitch-8.0 This branch is for Odoo 8.0. I never had an Odoo 8.0 installation. However, I believe it should be working. If you use it and you have problems, please let me know. I want to keep this working as long as people use it. If it doesn't work for you, please send me a private email. If it works, please comment on https://github.com/OCA/connector-telephony/pull/59 (feel free to comment there if it doesn't work). https://github.com/treveradams/connector-telephony/tree/freeswitch-9.0 This branch is for Odoo 9.0. It was created over the last few days and is thoroughly tested (to what my installation allows me to). There may be one bug. If you use sip phones internally, you may see this. I believe it is simple to fix if it is there. If it doesn't work for you, please send me a private email. If it works, please comment on https://github.com/OCA/connector-telephony/pull/96 (feel free to comment there if it doesn't work).

2 3

freeswitch-mod-vpx unmet dependencies
by Christoph Blau 25 Aug '16

25 Aug '16

Hey Guys I am trying to install mod_vpx from the official Debian repos but cannot do so... # lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.5 (jessie) Release: 8.5 Codename: jessie # apt install freeswitch-mod-vpx Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: freeswitch-mod-vpx : Depends: libfreeswitch1 (= 1.6.6~13~d2d0b32-1~jessie+1) but 1.6.9~16~d574870-1~jessie+1 is to be installed E: Unable to correct problems, you have held broken packages. # apt show freeswitch-mod-vpx Package: freeswitch-mod-vpx Source: freeswitch Version: 1.6.6~13~d2d0b32-1~jessie+1 Maintainer: Travis Cross <tc(a)traviscross.com> Installed-Size: 259 kB Depends: libc6 (>= 2.2.5), libfreeswitch1 (= 1.6.6~13~d2d0b32-1~jessie+1), libssl1.0.0 (>= 1.0.0), libvpx2 (>= 1.4.0) Suggests: freeswitch-mod-vpx-dbg Homepage: https://freeswitch.org/ Priority: optional Section: comm Download-Size: 134 kB APT-Sources: http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie/main amd64 Packages Description: VP8/VP9 video codec for FreeSWITCH FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. . This package contains mod_vpx for FreeSWITCH. . This module adds the VP8 video codec, also known as WebM. Anyone else having this issue? Kind regards Christoph Blau

3 2

freeswitch server getting INVITE request in ngrep but nothing in fs_cli
by devang nathwani 25 Aug '16

25 Aug '16

I am trying to route the call from opensips server to freeswitch server. My freeswitch server is actually a container which i have created using 'lxc' when I route the call from opensips server to freeswitch server container i got the INVITE requests sip log in 'ngrep' but getting nothing in freeswitch log as in nothing in fs_cli and my call is failing with the disposition 'SIP 408 - Request Timeout' I have tried with routing the call from opensips server to other freeswitch server which is actually having a physical ip and not the container server and when I tried that, i am able to get the logs in freeswitch as in fs_cli. sip profile parameters auth-calls, auth-all-packets, aggressive-nat-detection set to true, there is no firewall or iptable rule between opensips and freeswitch servers. i can access the osips server from freeswitch server and vice versa. So, what am i missing here. please suggest me something.

4 4

mod_verto and https://webrtc.freeswitch.org/verto/index.html#page-main
by Alexander Haugg 25 Aug '16

25 Aug '16

Hi, it is my first time with mod_verto. The certificate is generated like described here: https://webrtc.freeswitch.org/verto/index.html#page-main And the ca.crt is installed on the client pc. But the following error message is again: 2016-08-24 15:03:47.044880 [INFO] mod_verto.c:4025 192.168.241.2:43246 Client Connect. 2016-08-24 15:03:47.044880 [INFO] mod_verto.c:1978 192.168.241.2:43246 Starting client thread. 2016-08-24 15:03:47.084876 [DEBUG] mod_verto.c:1825 192.168.241.2:43246 WS SETUP FAILED [] 2016-08-24 15:03:47.084876 [INFO] mod_verto.c:2005 192.168.241.2:43246 Ending client thread. 2016-08-24 15:03:47.084876 [INFO] mod_verto.c:2013 192.168.241.2:43246 Thread ended What can I do? Is the certificate the problem? The Client PC and the Freeswitch PC are in the same Network Cluster. The Freeswitch PC is a Debian 8.5 system. Thanks a lot Alex

3 3

divert_events on inbound ESL server
by . 25 Aug '16

25 Aug '16

Hi, I am wondering if it is possible to turn on 'divert_events' in order to get additional event notifications for things like gtalk and DETECTED_SPEECH over an inbound ESL connection? I have an outbound ESL application that fields calls for a particular number and runs some simple grammars. Here I send 'divert_events on', and I am able to see these additional events no problem. I also have a 'monitor' application setup in inbound mode, which just listens to 'everything'. However this inbound application does not see additional events like DETECTED_SPEECH, although it can see the application calls to play_and_detect_speech. If I try to send 'divert_events on' from the inbound socket I see an error message "ERR not controlling a session", so I guess this is not correct. Is there a way to listen for these additional messages over the inbound socket, or not? Thanks!

1 1

Case with Freeswitch
by Gonzalo Gasca Meza 25 Aug '16

25 Aug '16

https://freeswitch.org/jira/browse/FS-9460

1 0

Compatibility issues with Error Cause: 88 [INCOMPATIBLE_DESTINATION]
by james 25 Aug '16

25 Aug '16

Hello: I am using portsip webRTC gateway to test the compatibility. I use two phones for webRTC testing. One is normal webRTC client and another one is softphone. FreeSWITCH work as a IPPBX. The call flows are: the problem is that webRTC calls sip phone without any problem, but sip phone calls to webRTC than failed. errors are: 2016-08-23 16:29:39.971775 [DEBUG] switch_core_state_machine.c:569 (sofia/internal/pu7ai86t(a)9t5jv2hi82a5.invalid) State Change CS_REPORTING -> CS_DESTROY freeswitch@iZ23lkvsnwpZ> 2016-08-23 16:29:39.971775 [DEBUG] switch_core_session.c:1647 Session 171 (sofia/internal/pu7ai86t(a)9t5jv2hi82a5.invalid) Locked, Waiting on external entities freeswitch@iZ23lkvsnwpZ> 2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] freeswitch@iZ23lkvsnwpZ> 2016-08-23 16:29:39.981772 [NOTICE] switch_ivr_originate.c:2771 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION] freeswitch@iZ23lkvsnwpZ> 2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] freeswitch@iZ23lkvsnwpZ> 2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401 Originate Failed. Cause: INCOMPATIBLE_DESTINATION freeswitch@iZ23lkvsnwpZ> EXECUTE sofia/internal/1004(a)120.26.130.142 answer() ------------------------- It looks freeswitch send with ICE to webRTC. Does anyone know how i do disable the webRTC negotiation and only send normal sip info to webRTC gateway

5 5

Re: [Freeswitch-users] freeswitch server getting INVITE request in ngrep but nothing in fs_cli
by devang nathwani 24 Aug '16

24 Aug '16

Yeah it's a possibility, but no ipatables or firewall rule affecting here, that is sure thing On Aug 24, 2016 11:05 AM, "Stanislav Sinyagin" <ssinyagin(a)gmail.com> wrote: or maybe iptables are coming in the way On Tue, Aug 23, 2016 at 4:12 PM, Michael Jerris <mike(a)jerris.com> wrote: > confirm what ip freeswitch is bound to and the ports it is listening on and that you have basic IP connectivity where necessary. > >> On Aug 23, 2016, at 6:35 AM, devang nathwani <devang.nathwani31589@gmail. com> wrote: >> >> I am trying to route the call from opensips server to freeswitch server. My freeswitch server is actually a container which i have created using 'lxc' >> >> when I route the call from opensips server to freeswitch server container i got the INVITE requests sip log in 'ngrep' but getting nothing in freeswitch log as in nothing in fs_cli and my call is failing with the disposition 'SIP 408 - Request Timeout' >> >> I have tried with routing the call from opensips server to other freeswitch server which is actually having a physical ip and not the container server and when I tried that, i am able to get the logs in freeswitch as in fs_cli. >> >> sip profile parameters auth-calls, auth-all-packets, aggressive-nat-detection set to true, there is no firewall or iptable rule between opensips and freeswitch servers. i can access the osips server from freeswitch server and vice versa. >> >> So, what am i missing here. please suggest me something. > > > _________________________________________________________________________ > 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 _________________________________________________________________________ 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

3 2

Odd issues with WebRTC and WSS
by Luca Pradovera 24 Aug '16

24 Aug '16

Hello, I am running two servers, one (A) with ref 6c0f18d960, the other (B) with ref 0074589311, and the configuration is copied over verbatim. Certificates have been built using Letsencrypt, with the same script on both boxes. On server A, any WebRTC client that tries to initiate a call gets a "switch_core_cert.c:209 FP FILE ERR!" and the call immediately goes down. The issue is present whether I test with Verto Communicator or my Verto-based custom client. Where could I start looking? The logs are not very helpful, it seems. Thanks! Luca Pradovera

2 4

Call Lost
by Leonardo Lima Ribeiro 24 Aug '16

24 Aug '16

Hello guys, I have a lua script that does an outbound call through my external sip gateways… The problem is that sometimes the call is lost and I don’t know why, this is the complete log of that call: 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.349227 [NOTICE] switch_channel.c:1091 New Channel sofia/external/8008000800 [813d53c0-0733-40af-9562-922ea19a7f5b] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.349227 [DEBUG] mod_sofia.c:4776 (sofia/external/8008000800) State Change CS_NEW -> CS_INIT 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8008000800) Running State Change CS_INIT 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:516 (sofia/external/8008000800) State INIT 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] mod_sofia.c:88 sofia/external/8008000800 SOFIA INIT 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] sofia_glue.c:1257 sofia/external/8008000800 sending invite version: 1.7.0 git 222eff1 2015-12-17 22:32:55Z 64bit 813d53c0-0733-40af-9562-922ea19a7f5b Local SDP: 813d53c0-0733-40af-9562-922ea19a7f5b v=0 813d53c0-0733-40af-9562-922ea19a7f5b o=FreeSWITCH 1471968528 1471968529 IN IP4 208.115.55.11 813d53c0-0733-40af-9562-922ea19a7f5b s=FreeSWITCH 813d53c0-0733-40af-9562-922ea19a7f5b c=IN IP4 208.115.55.11 813d53c0-0733-40af-9562-922ea19a7f5b t=0 0 813d53c0-0733-40af-9562-922ea19a7f5b m=audio 22422 RTP/AVP 102 9 0 8 104 101 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:102 opus/48000/2 813d53c0-0733-40af-9562-922ea19a7f5b a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:9 G722/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:0 PCMU/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:8 PCMA/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:104 telephone-event/48000 813d53c0-0733-40af-9562-922ea19a7f5b a=fmtp:104 0-16 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:101 telephone-event/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=fmtp:101 0-16 813d53c0-0733-40af-9562-922ea19a7f5b a=ptime:20 813d53c0-0733-40af-9562-922ea19a7f5b a=sendrecv 813d53c0-0733-40af-9562-922ea19a7f5b m=video 26230 RTP/AVP 103 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:103 VP8/90000 813d53c0-0733-40af-9562-922ea19a7f5b b=AS:1024 813d53c0-0733-40af-9562-922ea19a7f5b a=rtcp-fb:103 ccm fir 813d53c0-0733-40af-9562-922ea19a7f5b a=rtcp-fb:103 ccm tmmbr 813d53c0-0733-40af-9562-922ea19a7f5b a=rtcp-fb:103 nack 813d53c0-0733-40af-9562-922ea19a7f5b a=rtcp-fb:103 nack pli 813d53c0-0733-40af-9562-922ea19a7f5b 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:40 sofia/external/8008000800 Standard INIT 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:48 (sofia/external/8008000800) State Change CS_INIT -> CS_ROUTING 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:516 (sofia/external/8008000800) State INIT going to sleep 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8008000800) Running State Change CS_ROUTING 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8008000800) State ROUTING 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] mod_sofia.c:141 sofia/external/8008000800 SOFIA ROUTING 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_ivr_originate.c:67 (sofia/external/8008000800) State Change CS_ROUTING -> CS_CONSUME_MEDIA 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:532 (sofia/external/8008000800) State ROUTING going to sleep 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.390583 [DEBUG] switch_core_state_machine.c:473 (sofia/external/8008000800) Running State Change CS_CONSUME_MEDIA 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.410024 [DEBUG] switch_core_state_machine.c:551 (sofia/external/8008000800) State CONSUME_MEDIA 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.410024 [DEBUG] switch_core_state_machine.c:551 (sofia/external/8008000800) State CONSUME_MEDIA going to sleep 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:30.429866 [DEBUG] sofia.c:6756 Channel sofia/external/8008000800 entering state [calling][0] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] sofia.c:6756 Channel sofia/external/8008000800 entering state [proceeding][183] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] sofia.c:6766 Remote SDP: 813d53c0-0733-40af-9562-922ea19a7f5b v=0 813d53c0-0733-40af-9562-922ea19a7f5b o=BroadWorks 931492917 1 IN IP4 208.73.144.71 813d53c0-0733-40af-9562-922ea19a7f5b s=- 813d53c0-0733-40af-9562-922ea19a7f5b c=IN IP4 208.73.144.71 813d53c0-0733-40af-9562-922ea19a7f5b t=0 0 813d53c0-0733-40af-9562-922ea19a7f5b m=audio 17916 RTP/AVP 0 101 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:0 PCMU/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=rtpmap:101 telephone-event/8000 813d53c0-0733-40af-9562-922ea19a7f5b a=fmtp:101 0-15 813d53c0-0733-40af-9562-922ea19a7f5b a=maxptime:20 813d53c0-0733-40af-9562-922ea19a7f5b m=video 0 RTP/AVP 103 813d53c0-0733-40af-9562-922ea19a7f5b b=AS:1024 813d53c0-0733-40af-9562-922ea19a7f5b 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4208 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4153 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4069 Set telephone-event payload to 101@8000 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:2898 Set Codec sofia/external/8008000800 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_codec.c:111 sofia/external/8008000800 Original read codec set to PCMU:0 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4417 Set telephone-event payload to 101@8000 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:4473 sofia/external/8008000800 Set 2833 dtmf send payload to 101 recv payload to 101 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:6027 AUDIO RTP [sofia/external/8008000800] 208.115.55.11 port 22422 -> 208.73.144.71 port 17916 codec: 0 ms: 20 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_rtp.c:3793 Starting timer [soft] 160 bytes per 20ms 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:6326 sofia/external/8008000800 Set 2833 dtmf send payload to 101 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:6333 sofia/external/8008000800 Set 2833 dtmf receive payload to 101 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_core_media.c:6356 sofia/external/8008000800 Set rtp dtmf delay to 40 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/8008000800! 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.788096 [DEBUG] switch_channel.c:3461 (sofia/external/8008000800) Callstate Change DOWN -> EARLY 813d53c0-0733-40af-9562-922ea19a7f5b 2016-08-23 18:22:33.828095 [DEBUG] switch_cpp.cpp:625 (sofia/external/8008000800) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE After that nothing happens and this is my Lua command to start this call: local new_session = freeswitch.Session("[origination_caller_id_name=‘9009000900',origination_caller_id_number=‘9009000900']sofia/gateway/14_RINGCENTRAL/8008000800", session); Do you know what I can do to avoid this problem? Thank you, Leo

1 0