October 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

webrtc with ecdsa
by GM phy 05 Jan '18

05 Jan '18

Hello, did anybody get webrtc working with a ECDSA signed cert? I have fs 1.6.19 compiled with openssl 1.0.2g and wss.pem file is -----BEGIN EC PARAMETERS----- BggqhkjOPQMBBw== -----END EC PARAMETERS----- -----BEGIN EC PRIVATE KEY----- MHcCAQEEIPNZ4YUGKQ2cySfnEp1RSOSDp9MTQ21xyISpBmBPRN1/oAoGCCqGSM49 AwEHoUQDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m +A4huOY2MtbwXtBdg2kqjYbcrkAk5zwa6Q== -----END EC PRIVATE KEY----- -----BEGIN CERTIFICATE----- MIIGGTCCBQGgAwIBAgIMAPaMiX8YXElRs/bJMA0GCSqGSIb3DQEBCwUAMEwxCzAJ BgNVBAYTAkJFMRkwFwYDVQQKExBHbG9iYWxTaWduIG52LXNhMSIwIAYDVQQDExlB bHBoYVNTTCBDQSAtIFNIQTI1NiAtIEcyMB4XDTE3MTAxNzEzNDY1MVoXDTE4MTAx ODEzNDY1MVowQTEhMB8GA1UECxMYRG9tYWluIENvbnRyb2wgVmFsaWRhdGVkMRww GgYDVQQDDBMqLmxhY29udGluZW50YWwuY29tMFkwEwYHKoZIzj0CAQYIKoZIzj0D AQcDQgAEnxYcss4/4yCzBl85psRqTLeQ0FnXqsoDCC/2lKMTsrS7mlPgCo7m+A4h uOY2MtbwXtBdg2kqjYbcrkAk5zwa6aOCA88wggPLMA4GA1UdDwEB/wQEAwIDiDCB iQYIKwYBBQUHAQEEfTB7MEIGCCsGAQUFBzAChjZodHRwOi8vc2VjdXJlMi5hbHBo .... when we try to connect (we are using sipjs and tried from chrome and firefox last versions) we get a handshake failure. any lead or help will be appreciated, thanks in advance.

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[OT] baresip
by Alfred E. Heggestad 23 Dec '17

23 Dec '17

Hi, My apologies for this off-topic post, but I hope that this can be of interest to the nice Freeswitch community. We have released a new version of Baresip which is a open source SIP client that is compatible with Freeswitch. you can find the source code here: https://github.com/alfredh/baresip/releases/tag/v0.5.5 We are doing regular interop-testing with Freeswitch, mainly SIP and RTP related features. From the Baresip project we send the best greetings to the nice people on this list :) /alfred

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WebSocket behind NGINX
by Anton 04 Dec '17

04 Dec '17

Hello All, I have to proxy all websocket requests though a nginx server. Right now I am using next configuration: map $http_upgrade $connection_upgrade { default upgrade; '' close; } server { listen 443; server_name wss.somedomain.com.ua; ssl on; ssl_certificate /etc/nginx/cert.pem; ssl_certificate_key /etc/nginx/private.key; location / { proxy_pass http://127.0.0.1:5066; proxy_http_version 1.1; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; proxy_read_timeout 86400s; } access_log /var/log/nginx/wss_access; error_log /var/log/nginx/wss_error debug; } I dumped traffic from nginx and found out that "switching protocol" phrase was successful but INVITE message from my browser in pending state. Maybe FreeSWITCH wants real IP not loopback? Who have faced with similar problem? BR, Anton

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Embedding Freeswitch
by Naveen Khanna 23 Nov '17

23 Nov '17

Hi, Can someone suggest a low cost standard single board computer, around $ 50, to run 25 concurrent sessions & 100 registrations of SIP clients with limited applications of Freeswitch. Regards, Naveen Khanna

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Video recording
by Freeswitch user 15 Nov '17

15 Nov '17

Hello users, Is there any way in freeswitch to record two way video calls. Please suggest!! Thanks

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Options for initiating call from browser (not WebRTC)
by Oivvio Polite 14 Nov '17

14 Nov '17

I have a simple call recording solution set up with a Twilio-like PaaS. The user interacts with a web app where she enters a destination number. My web app then instructs the PaaS to 1. Call the users phone 2. Wait for the user to answer 3. Start a recording. 4. Connect the destination number. The resulting wav is available for the user to download in the web app. I now want to replace the PaaS with FreeSwitch. I've done a lot of experimenting with FS during the last year, and managed to get callrecording of PSTN and WebRTC calls to work. With those calls origination was from PSTN or WebRTC. Now I want to do origination with a HTTP request. When leafing through the FreeSwitch book from PACKT I see that I have multiple options for doing this. 1. fs_cli Have the browser talk to a separate web app (written in node or python or whatever) that the initiates the call with fs_cli. 2. mod_httapi Have the browser talk to a webapp that talks to FS via mod_httapi. In this scenario I can't find any info on how to originate calls in mod_httapi though. 3. mod_event_socket Have a the browser talk to a webapp that talks to FS via mod_event_socket. In all of these scenarios I figure that I'll have the webapp running on the same box as FS and that I'll do all authentication/authorization of the user in the webapp. But which option should I go with? regards Oivvio

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180/183 not passed through sip profiles
by Grant Bagdasarian 08 Nov '17

08 Nov '17

Hi, We're using a freeswitch which bridges calls between two networks: private and public. The server has two interfaces. The calls originate from the private network and are bridged using the public sip profile. For some reason I cannot get the 180 and 183 messages passed back from the outbound leg to the inbound leg when sending calls over the public sip profile. Both sip profiles have been set with: <param name="inbound-bypass-media" value="false"/> <param name="inbound-proxy-media" value="true"/> The below dialplan is hit when a call arrives on the LAN(private) interface of the freeswitch. <?xml version="1.0" encoding="utf-8"?> <include> <context name="local"> <extension name="route_to_public"> <condition field="destination_number" expression="(\d+)$" require-nested="false"> <action application="set" data="bridge_early_media=true" /> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=false"/> <action application="set" data="ignore_early_media=false"/> <action application="set" data="sip_copy_custom_headers=false" /> <action application="set" data="sip_ignore_reinvites=false"/> <action application="set" data="effective_caller_id_number=${translate(${caller_id_number} To_International)}" /> <action application="set" data="effective_caller_id_name=${translate(${caller_id_number} To_International)}" /> <action application="set" data="destination_number_translated=${translate(${destination_number} To_International)}"/> <action application="bridge" data="sofia/gateway/${distributor(public_gws ${sofia(profile public_eth1 gwlist down)})}/${destination_number_translated}" /> </condition> </extension> </context> </include> Any ideas why the 180/183 are not relayed from outbound leg back to inbound leg? A trace shows these messages do arrive on the outbound leg. Grant Bagdasarian Developer +31765727054<tel:+31765727054> cm.com<https://cm.com> [cid:image001.png@01D34F08.7CB33B90]

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SRV and sip timers
by Daniel Greenwald 07 Nov '17

07 Nov '17

When the preferred server in an SRV record is down. FS waits a full 30 seconds before trying the next lower priority server. Each subsequent call also delays 30 seconds. Any tricks to speed up the failover?

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OutGoing Sip Call
by Supun Madushanka 07 Nov '17

07 Nov '17

Hi All I am setup freeswitch in aws instance. i can get locally call successfully. now i had an issue i have a sip address "user(a)sip.linphone.org" i want to get call to that address from locally "1003@ip". can you help me to config freeswitch for achieve that call Thank you -- Best Regards, *Supun Madushanka* [Undergraduate] University of Moratuwa. http://www.mrt.ac.lk Department of Computer Science and Engineering. http://cse.mrt.ac.lk Mobile: +94 71 1135012 <%2B94%280%29%20711135012> E-mail: supun.12(a)cse.mrt.ac.lk

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Temporary Equivalent of fs_path
by Colin Morelli 06 Nov '17

06 Nov '17

Hey all, Trying to figure out how to get the equivalent behavior of fs_path, but only for a single transaction. In other words, I want to start a SIP request hitting a particular proxy, and then simply let Record-Route headers determine where the call should be routed after that. I might be completely overthinking this, but everything I've tried so far (sip_route_uri, fs_path), have resulted in Freeswitch continuing to use the given route for all subsequent requests, rather than simply falling back to the session route. Am I missing something? Best, Colin

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