- FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL + BYE SIP reason code
by Gonzalo Gasca Meza 26 Aug '16

26 Aug '16

Hi guys, I'm getting the following BYE reason code from my Service Provider: recv 740 bytes from udp/[177.71.206.193]:5060 at 02:57:50.740042: ------------------------------------------------------------------------ BYE sip:gw+twilio@52.1.1.1:5090;transport=udp;gw=twilio SIP/2.0 CSeq: 1 BYE From: <sip:+551121174769@turn2data-telephonist.pstn.br1.twilio.com >;tag=76430350_6772d868_ebbbd5d1-d9b1-49db-981e-ec4b3e238564 To: "" <sip:+55114XXXYYYY@172.31.3.166>;tag=mjZyjF9Ute8DD Call-ID: 7d8a1944-e512-1234-4eaf-02d310a86cb3 Max-Forwards: 68 Via: SIP/2.0/UDP 177.71.206.193:5060;branch=z9hG4bKfaa.6c72ded1.0 Via: SIP/2.0/UDP 172.19.147.73:5060 ;rport=5060;received=177.71.206.215;branch=z9hG4bKebbbd5d1-d9b1-49db-981e-ec4b3e238564_6772d868_205642673073712 * Reason: MVTSLocal;cause=10;text="BYE received",SIP;cause=406;text="Service Unavailable"* User-Agent: Twilio Gateway X-Twilio-CallSid: CA5411d5015d3df04541c6e14e9fb41719 Content-Length: 0 When placing call via ESL I get reported NORMAL_CLEARING vs 406 I don't see 406 Listed in Wiki: https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table According to RFC seems to be valid https://tools.ietf.org/html/rfc3326 Examples are: Reason: SIP ;cause=200 ;text="Call completed elsewhere" Reason: Q.850 ;cause=16 ;text="Terminated" Reason: SIP ;cause=600 ;text="Busy Everywhere" Reason: SIP ;cause=580 ;text="Precondition Failure" Can we add support to handle these type of reasons to a common one ? Like Unknown? I will talk to my SP to see if they can change it to 503. Any suggestions?

3 4

[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