November 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Issue with cc_warning_tone in mod_callcenter module
by Samir Doshi 16 Nov '16

16 Nov '16

Hi, We are doing impementation using mod_callcenter module by taking reference from documentation ( https://freeswitch.org/confluence/display/FREESWITCH/mod_callcenter) So far everything works well except cc_warning_tone playback. We are never getting tone when some connect to agent. Debug Logs : FS cli Log : https://pastebin.freeswitch.org/view/f9be26b9 Log with sip debug : https://pastebin.freeswitch.org/view/aab8ea85 Log with sip debug + loglevel 9 : https://pastebin.freeswitch.org/view/ff793521 Any hint? Best Regards -- Samir Doshi *iNextrix Technologie**s Pvt. Ltd*. http://www.inextrix.com *Disclaimer:* The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful.

2 5

Problems with diaplan var settings
by Tim Smith 16 Nov '16

16 Nov '16

Debian GNU/Linux 8 (jessie) Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) Can anyone explain to my why this works : <action application="bridge" data="{origination_privacy=screen+hide_name+hide_number,sip_h_Privacy=id,sip_cid_type=pid,sip_from_display='Anonymous',sip_invite_from_uri='<sip:anonymous@anonymous.invalid>'}sofia/gateway/example/$1$2"/> And yet this doesn't (nothing happens, no From rewrite, no privacy settings, nothing ... I have tried swapping "set" for "export" but that makes no difference either) : <action application="set" data="sip_from_display=Anonymous"/> <action application="set" data="sip_invite_from_uri=<sip:anonymous@anonymous.invalid>"/> <action application="set" data="sip_cid_type=pid"/> <action application="set" data="origination_caller_id_name=${outbound_caller_id_number}"/> <action application="set" data="origination_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="origination_privacy=hide_name+hide_number+screen"/> <anti-action application="set" data="sip_cid_type=none"/> <anti-action application="set" data="sip_invite_domain=my.example.com"/> <anti-action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <anti-action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> </condition> <action application="set" data="sip_h_X-example-tag=${usr_example_id}"/> <action application="set" data="sip_h_X-example-param1=${usr_example_param1}”/> <action application="set" data="sip_h_X-example-param2=${usr_example_param2}”/> <action application="bridge" data="sofia/gateway/example/$1$2"/>

1 0

problems with playback between v1.6.9 and last 1.6 version
by Caragea Silviu 16 Nov '16

16 Nov '16

Hello, I have an app that's playing several files once the call is answered: <action application="playback" data="/usr/local/woow-vm/sounds/en/pinconf.wav"/> <action application="playback" data="/usr/local/woow-vm/sounds/en/1.wav"/> <action application="playback" data="/usr/local/woow-vm/sounds/en/2.wav"/> <action application="playback" data="/usr/local/woow-vm/sounds/en/3.wav"/> <action application="playback" data="/usr/local/woow-vm/sounds/en/4.wav"/> On v1.6.9 or older works perfect. I tried on last 1.6.x master form today and only first file I can hear being played. All other are not played and in logs I can see very fast: switch_ivr_play_say.c:1910 done playing file /usr/local/woow-vm/sounds/en/1.wav On the v1.6.9 this message is triggered at the end of the file. I will try to find the commit that broken this but help is appreciated. Silviu

5 7

Can't get SRTP working with Freeswitch
by Tim Smith 16 Nov '16

16 Nov '16

Debian GNU/Linux 8 (jessie) Linux vxf 3.16.0-4-amd64 #1 SMP Debian 3.16.36-1+deb8u2 (2016-10-19) x86_64 GNU/Linux FreeSWITCH Version 1.6.12-20-b91a0a6~64bit (-20-b91a0a6 64bit) I'm trying to get a simple freeswitch setup working with a softphone. The problem I am having is when I dial the FS local talking clock (9170), I can hear the time when I just have TLS enabled, but when I have TLS transport and call-encryption (SRTP) enabled, I can't hear the time any more ? This is the fs_cli log when using just TLS : http://pastebin.com/yUuS6rWX This is the fs_cli log with TLS and SRTP: http://pastebin.com/ydi6Rkyx

2 2

sofia_glue.c:329 Invalid tls-verify-policy value: none
by Tim Smith 15 Nov '16

15 Nov '16

I am running the latest version of freeswitch-stable from the deb repository on Debian 8 .... just like you recommend in the docs. ;-) The default config it installs seems to take issue with the following files : sip_profiles/internal.xml sip_profiles/external-ipv6.xml sip_profiles/external.xml More specifically the line <param name="tls-verify-policy" value="none"/> Someone seems to have asked this question before http://lists.freeswitch.org/pipermail/freeswitch-users/2015-July/114881.html But nobody seems to have given an answer at the time !

3 5

freeswitch.org down?
by Joel Serrano 15 Nov '16

15 Nov '16

Getting: Service Temporarily Unavailable The server is temporarily unable to service your request due to maintenance downtime or capacity problems. Please try again later. ------------------------------ Apache/2.2.22 (Debian) Server at freeswitch.org Port 443

2 2

ClueCon Weekly - November 16th 2016 @ 1PM Eastern - Join me on the call.
by Brian West 15 Nov '16

15 Nov '16

FreeSWITCHers, Tomorrow I'll be presenting on how I fight back against the card member services scams using FreeSWITCH and how effective my methods were and how I uncovered the real company behind it all. https://youtu.be/-gj5kZbUXRI Dial 888 @ https://conference.freeswitch.org/vc/ <https://t.co/APylI6MpJE> to join the conversation! -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com (50% Discount using code FreeSwitch50) http://www.freeswitchcookbook.com (50% Discount using code FreeSwitch50) https://www.gofundme.com/freeswitch_ubuntu 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

1 0

Help with FreeTDM and GSM Interface
by Antonio Modesto 14 Nov '16

14 Nov '16

Hi, I'm having problems to connect calls using a digium tdm800p card. The problem only happens with the ports that are connected to a intelbras 4100 gsm interface, ports connected to the POTS are not affected. I'll try the best I can to express the situation: 1) Cold restart the gsm interface and wait for it to boot up 2) Try to place a call using it (freetdm/<port>/<number>) 3) The call stays silent, after a while the carrier reports that a invalid number has been informed 4) At this point, all following calls always return busy. If I connect an analog phone to the gsm interface and start dialing digits, it gives a busy tone right away, before I finish dialing the number 5) Cold restart the gsm interface again 6) Try to place a call using it (freetdm/<port>/<number>) 7) The calls stays silent, but this time I type the same number again. 8) The call connects. This problem does not happen with an analog phone or with chan_dahdi (asterisk), just with freetdm. My assumption is that when freetdm opens the channel for the first time and starting sending the dtmf digits, the gsm interface is not able to read them and sits waiting, after a while it timeouts and stops working, even for further calls. Here is my freetdm.conf: [span zt dahdi_span_1] trunk_type => FXO debugdtmf => yes analog-start-type => kewl fxo-channel => 1:1-6 zt.cont: [defaults] codec_ms => 20 wink_ms => 150 flash_ms => 750 echo_cancel_level => 64 rxgain => 0.0 txgain => 0.0 dahdi/system.conf: loadzone=br defaultzone=br fxsks=1-6 echocanceller=mg2,1-6 If anybody could help me with this I would really aprecciate. Regards. -- Atenciosamente, Antônio Modesto

2 1

Error starting profile with WSS-binding enables
by Cj B 14 Nov '16

14 Nov '16

Hi All, I have a strange issue with starting the internal profile with wss-binding enabled. If I start freeswitch, the profiles won’t start even if I try to manually start them: http://pastebin.com/bLB8XnxS <http://pastebin.com/bLB8XnxS> Then If I go and remove the wss-binding, start the profile, then add wss-binding back and just do a rescan, it starts and the wss-binding works. This is on FreeSWITCH version: 1.6.12+git~20161111T161001Z~873c5ada12~64bit (git 873c5ad 2016-11-11 16:10:01Z 64bit) Any hints as to how to fix this? I’d really like to be able to get my profiles to start without intervention. This was not happening when I was running 1.6.8, it only started after the upgrade to 1.6.12. Thanks. Cj B

3 4

RTCP in passthrough mode
by Александр Дружилов 14 Nov '16

14 Nov '16

Could somebody explain me how Freeswitch works with a RTCP packets in video passthrough mode? I ask because I found a problem when one user starts sending video to another client through Freeswitch. The problem in depth: I have two clients and whant to make WebRTC video conference between them (but use only passthrough mode because it's just a screensharing when one user sends his screen to anybody other in conference). So first user starts conference and sends his screen to Freeswitch (and he can see that Freeswitch sends him back his screen properly). When another user attaches to this conference he does not see any video, but at the same time I see in chrome://webrtc-internals that he receives exactly the same video stream that first user sends to freeswitch. I started to look deeper and found that second client sends a lot of PLI requests (about 5 per second) to Freeswitch. I tend to think that it's because second client attaches to conference in the middle of RTP stream and he didn't receive a full intra-picture and he can't start encode the video stream. Also I found that no RTCP packets was sent from Freeswitch to the first client to get this intra-picture. So returning to my first question: what if Freeswitch just ignores this RTCP packets in passthrough mode?

2 2