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mangedESL project
by Hossein Gholizadeh 23 May '18

23 May '18

Hi ... i want to use managed ESL in C# and downloaded the sample project of freeswitch ... in libs\esl\managed i opened the managedESL project ... but here in ESLPINVOKE.cs we have added dll ESL by DllImport ... when i want to compile the project these ESL dlls have error because it cannot find the ESL dll ... is the issue about ESL project which is written by C++ or anything else? if it is i want to add ESL... but after adding it i cannot add esl project a reference ...

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Problem when switching user between different canvases
by Muhammad Umair 23 May '18

23 May '18

Hi Everyone I have set up FreeSWITCH as a video conference bridge so users can connect with Verto. It’s working great, but when I switch a user to a different canvas, using vid-canvas, the first few frames are very “blocky”. Is there a setting that would prevent this from happening, or reduce the effect on the video stream when moving a user from one canvas to another? We have plenty of CPU, so it’s not that. Any tips would be appreciated. Thanks, Umair

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Freeswitch TLS gateway failure with 503
by Aqs Younas 23 May '18

23 May '18

Greetings list, I am trying to make a TLS gateway working with my freeswitch. But it fails with Service Unavailable [503]. Though I am perfectly able to make TLS calls between phones. Only gateway is not working. I am populating gateway configuration using lua. Below is my profile and gateway configuration. <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="configuration"> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> <param name="debug-presence" value="0"/> </global_settings> <profiles> <profile name="external"> <aliases> </aliases> <gateways> <gateway name="ab345b4c-b961-4d3a-97a9-f227b52a7240"> <param name="username" value="admin(a)pbx.primaryline.com.au"/> <param name="password" value="XXXXXXXXXXX"/> <param name="from-domain" value="pbx01.primaryline.com.au"/> <param name="proxy" value="203.134.5.10"/> <param name="expire-seconds" value="800"/> <param name="register" value="true"/> <param name="register-transport" value="tls"/> <param name="contact-params" value="transport=tls"/> <param name="retry-seconds" value="30"/> <param name="context" value="public"/> <variables> </variables> </gateway> </gateways> <domains>     <domain name="all" alias="false" parse="false"/> </domains> <settings> <param name="caller-id-type" value="pid"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5080"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-type" value="rfc2833"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <param name="outbound-codec-prefs" value="PCMU,PCMA,GSM"/> <param name="hold-music" value="local_stream://default"/> <param name="zrtp-passthru" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="local-network-acl" value="localnet.auto"/> <param name="manage-presence" value="false"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-ip" value="124.40.187.231"/> <param name="sip-ip" value="124.40.187.231"/> <param name="ext-rtp-ip" value="124.40.187.231"/> <param name="ext-sip-ip" value="124.40.187.231"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5081"/> <param name="tls-cert-dir" value="/etc/freeswitch/ssl"/> <param name="tls-passphrase" value=""/> <param name="tls-verify-date" value="true"/> <param name="tls-verify-depth" value="2"/> <param name="tls-verify-in-subjects" value=""/> <param name="tls-version" value="tlsv1"/> <param name="debug" value="0"/> <param name="user-agent-string" value="FreeSWITCH"/> </settings> </profile> </profiles> </configuration> </section> </document> Here is my 'sofia status' output. freeswitch(a)pbx.primaryline.com.au> sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[::1]:5080 RUNNING (0) external-ipv6 profile sip:mod_sofia@[::1]:5081 RUNNING (0) (TLS) external profile sip:mod_sofia@124.40.187.231:5080 RUNNING (0) external profile sip:mod_sofia@124.40.187.231:5081 RUNNING (0) (TLS) external::ab345b4c-b961-4d3a-97a9-f227b52a7240 gateway sip:admin@pbx.primax.com.au@203.134.5.10 FAIL_WAIT internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS) internal profile sip:mod_sofia@124.40.187.231:5060 RUNNING (0) internal profile sip:mod_sofia@124.40.187.231:5061 RUNNING (0) (TLS) ================================================================================================= I am really stuck here, google search is leading me nowhere. Any pointer/suggestion is much appreciated. Thanks.

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How to remove recorded files with silence
by Eugene Prokopiev 23 May '18

23 May '18

Hi, Is it possible to automatically remove recorded files which contains nothing except silence? Can mod_dptools:record do this just after recoding attempt? -- WBR, Eugene Prokopiev

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Troubleshooting REGISTER
by Lars Paulsen 23 May '18

23 May '18

Hi All, I am having trouble with a gateway configuration. The REGISTER coming from the gateway to FreeSWITCH is being rejected ("Forbidden"). Now I noticed that I do not see any output in fs_cli on an incoming REGISTER message. The debug level is already set to 7. What else do I need to do to enable some output for REGISTER messages? Thanks in advance for any help. Best Regards, Lars Paulsen

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Achieving TLS + SRTP for inbound calls
by David P 23 May '18

23 May '18

We use conferences to allow a verto user to call and connect with an Asterisk channel. We would like to secure both signalling and media via TLS + SRTP, and I've read https://freeswitch.org/confluence/display/FREESWITCH/SIP+TLS a few times to understand how to do this. Note that that page has a broken link: https://wiki.freeswitch.org/wiki/Secure_RTP First, is it still true that FS doesn't offer prebuilt installs (for Ubuntu) to support this kind of security? Assuming that it must be compiled, I began to follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie with the additional first step of: apt-get install libssl-dev I soon ran into "Unable to locate package freeswitch-video-deps-most". What should I try next? Cheers, David

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Closing a conference from verto
by David P 22 May '18

22 May '18

I have a collection of extensions set up as conferences, each of which have one channel from verto and one to an Asterisk server. I would like each conference to close shortly after the verto user hangs up. I've been following this as a guide: https://freeswitch.org/confluence/display/FREESWITCH/mod_conference I believe my version of FS is recent. I tried to make it work by setting each extension in my dialplan to have entries like this: <action application="conference_set_auto_outcall" data="[conference_member_flags=endconf]sofia/gateway/asterisk/${as_ext}"/> <action application="conference" data="${destination_number}@myconference+flags{endconf}"/> In particular, I added "+flags{endconf}" to the verto channel, and only the verto channel, so that its hangup would be treated as a trigger to close the conference. Also, in the "myconference" profile, I try to make the closing of the conference happen 5 seconds after the verto user hangs up: <param name="endconf-grace-time" value="5"/> But when I make test calls, there's no evidence that the conference ends even a minute after my hangup. In particular, the Asterisk channel remains active (when I check "core show channels" in its commandline interface). I can't make another call to any of my extensions, unless I kill the conference via Asterisk. Is there anything else I need to configure in FreeSwitch? Cheers, David

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javascript choose external channel
by Francesco Facco de Lagarda 22 May '18

22 May '18

I'm developing a call app using javascript I have 2 possibile gateways for outgoing calls: one is a RealTone FXO box on analogue line one is a MessageNet SIP account.. in JS I THOUGHT I could use them as follows: for the realtone FXO var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); for messageNet SIP ACCOUNT var sessOut = new Session("sofia/gateway/messagenet/" + dialedNum + "@192.168.0.216:5060"); But FS ALWAYS dials out on the FXO, in fs_cli I see: switch_channel.c:1104 New Channel sofia/external/069087651@192.168.0.216:5060 <mailto:sofia/external/069087651@192.168.0.216:5060> why not New Channel sofia/gateway/Messagenet/sofia/external/069087651@192.168.0.216:5060 <mailto:sofia/external/069087651@192.168.0.216:5060> As I'm specifying n the "new Session" code ???

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Fax by javascript
by Hossein Gholizadeh 22 May '18

22 May '18

Hi ... I'm using faxlib for fax in freeswitch but as in api docs is mentioned, it's out-of-date . is that good to use it ? if not what can i do ? is there any sample for fax by javascript ? and last question : is it simple to use freeswitch by javascript?

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Does verto supports live broadcasting?
by Xiemin Chen 22 May '18

22 May '18

Hi all, I want to achieve a live broadcasting system with verto and webrtc, Is it able to? It seems that if I don't enable video in verto session then I cannot receive other's video too, Anything I missed? Sincerely Xiemin

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