May 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to make FS to listen for 2 RTP streams on one call ?
by Evtim Ivanov 23 May '18

23 May '18

Hello all, I'm trying to implement this (shown in the picture) scenario, but to use FS instead of OrekaTR: There are several small bumps on the road, which I successfully overcame, but finally I met a challenge which is bigger, than my skills, so I need the community help. The problem is, that RtpProxy can't mix the media, so instead of one mixed stream on the forked call, I have 2 separate RTP streams. I have no idea how to make FS to listen both of them. Is this possible and how? Any advice is highly appreciated. Best! Evtim

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Adding SIP Headers in Javascript
by Preston Hagar 23 May '18

23 May '18

Quick Version: Is there some special way to set custom SIP headers when you are bridging a call from Javascript instead of XML? Full Details: We am using Voice Operator Panel (VOP) as a receptionist console. We have multiple external numbers that all funnel into a few receptionist desks. If I add a SIP Diversion header, then VOP will use that to show the receptionist which number the caller dialed to reach them, which would help things out a lot. If I add it in an XML dialplan and then do a direct bridge to one of the receptionist numbers, it works great (snip of working direct dialplan): <action application="set" data="sip_h_Diversion=<sip:${ destination_number}(a)pbx.example.org >;privacy=off;reason=unconditional;counter=1;screen=no"/> <action application="bridge" data="user/1000(a)pbx.example.org"/> The problem is I need a bit of advanced logic to figure out which receptionist are online and then ring those, then fail over to other numbers and ultimately IVR if no one answers, so I instead pass calls to them off to javascript, so the bridge line above becomes something link: <action application="javascript" data="${jslib}/frontdesk.js ${dialed_extension} ${domain_name} ${destination_number}" /> I've tried setting the SIP header in the XML above the javascript line and then exporting it <action application="set" data="sip_h_Diversion=<sip:${ destination_number}(a)pbx.example.org>;privacy=off;reason=unconditional;counter=1;screen=no"/> <action application="export" data="sip_h_Diversion" /> and setting it in the Javascript: var diversionHeader = 'sip_h_Diversion=<sip:' + $destination_number + '@ pbx.example.org>;privacy=off;reason=unconditional;counter=1;screen=no'; session.execute('set',diversionHeader); session.execute('export','sip_h_Diversion'); but in both cases (or with both in place), the SIP header is never written. Is there some magic to setting SIP headers in Javascript? I can't seem to find any examples or documentation about it. Thanks!

4 9

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

1 0

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

4 4

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

1 0

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

1 0

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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