- FreeSWITCH-users - FreeSWITCH Mailing Lists

Firefox 39, webrtc
by Igor Olhovskiy 24 Jul '15

24 Jul '15

Hi! I'm using sip.js with FS (tried FS 1.5.15 and 1.7). Upgrading to Firefox 39 is seems to be bad idea, cause call through it fails due to 2015-07-15 10:48:39.168250 [ERR] switch_rtp.c:2907 audio Handshake failure 1 2015-07-15 10:48:39.168250 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL On FF 38 and Chrome all is ok. Am I only one with this issue of it's really something wrong with FF? -- Best regards, Igor

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beep to valer_parked call on bridge
by jaflong jaflong 24 Jul '15

24 Jul '15

On a call that has been valet_parked how can I play a beep/audio to it when a incoming call is brdige to it

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FS 1.6 video dependency build issue
by Peter Stevens 24 Jul '15

24 Jul '15

Hi, I have a clean install of Debian 8 on a 32 bit machine. I'm following information on FS 1.6 video wiki (https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video) to<https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video)%…> build this version, but the installation of the dependencies part doesn't install all of the required dependencies. After running: DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most I had to then: apt-get install libavcodec-extra apt-get install portaudio19-dev Which were both ok, but running: apt-get install libsngtc-dev I received following: Package libsngtc-dev is not available, but is referred to by another package. This may mean that the package is missing, has been obsoleted, or is only available from another source E: Package 'libsngtc-dev' has no installation candidate Re-ran: DEBIAN_FRONTEND=none APT_LISTCHANGES_FRONTEND=none apt-get install -y --force-yes freeswitch-video-deps-most 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-video-deps-most : Depends: libsngtc-dev but it is not installable E: Unable to correct problems, you have held broken packages. I've had a look around amongst the http://files.freeswitch.org/repo/deb/debian/dists/jessie/main directory structure and have discovered that the libsngtc-dev is listed in both the binary-amd64/packages.gz and the binary-amd64/packages.gz files, but is only listed in the Contents-amd64.gz file and not in the Contents-i386.gz file. Don't know whether this is significant or not, but how/where can I obtain the libsngtc-dev file to complete the dependency installation for freeswitch-video-deps-most? Any help appreciated. Peter ---------------------------- http://www.bbc.co.uk This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this. ---------------------

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question about eavesdrop
by Rafael Kaihatu 24 Jul '15

24 Jul '15

Hi All, Let's say I have a 2-party call (leg A and B talking with each other) I would like to connect a 3rd leg to hear only A I understood that I can configure with eavesdrop, if the 3rd leg can speak with A or B or both or neither. is it possible to do that with eavesdrop? is there another command? and eventually, I want to connect a 4th leg to hear only B Thanks Rafael Kaihatu

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FreeSWITCH Friday FreeForAll Reminder!
by krice＠freeswitch.org 24 Jul '15

24 Jul '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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api commands not working in outbound socket
by jaflong jaflong 24 Jul '15

24 Jul '15

Hi list I have a php script run through ivrd. api commands are returned with "ERR command not found" with done through sendRecv or api functions any suggestion why it fails? example 1 --------- code: $command = "api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user"; $conn->sendRecv($command); tcpdump: api uuid_getvar db39795e-31e1-11e5-9af9-cfe09f60c9f7 variable_sip_from_user Content-Type: command/reply Reply-Text: -ERR command not found example 2 --------- code: $command = "api version"; $conn->sendRecv($command); tcpdump: api version Content-Type: command/reply Reply-Text: -ERR command not found example 3 --------- code: $command = "version"; $conn->api($command); tcpdump: api version Content-Type: command/reply Reply-Text: -ERR command not found

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force-register-domain
by Nikolay Zaytsev 24 Jul '15

24 Jul '15

Hello,everyone. Ive tried to use a1-hash instead of username and password.(when i use plain username and password everything works fine). However, freeswitch does not allow users to register,when i use domain which is different from ip_address of the Freeswitch server. I am using domain - nick.sfedu. Domain in vars.xml,directory/default.conf is set as nick.sfedu and force-register-domain in internal profile is set to the same value. When I use plain username and password, i can see the following: Call-ID: NGJhMDU5NTM0MDllZWY2ODllZjFjYzMwYzAzMzQxMTc User: 11007(a)nick.sfedu Contact: "Nick" <sip:11007@10.132.144.157:49610 ;rinstance=fd2236fab37d0ba1;fs_nat=yes;fs_path=sip%3A11007%4010.132.144.157%3A49610%3Brinstance%3Dfd2236fab37d0ba1> Agent: X-Lite 4.7.1 74247-a844961e-W6.1 Status: Registered(UDP-NAT)(unknown) EXP(2015-07-24 14:55:08) EXPSECS(144) Host: fsw1-test IP: 10.132.144.157 Port: 49610 Auth-User: 11007 Auth-Realm: 10.160.1.23 MWI-Account: 11007(a)nick.sfedu where 10.160.1.23 is ip address of the Freeswitch server,but it should be nick.sfedu,as far as i understand. Best regards, Nikolay Zaytsev

1 0

ReINVITE - 488 Not Acceptable Here
by Sergio García 24 Jul '15

24 Jul '15

Hello all, I am using *FreeSwitch* as a *WebRTC gateway* thanks to its Websocket support (in my case WSS), but I'm facing this strange problem. Audio and Video calls are working perfectly fine, but when I try to set a call On Hold, FreeSwitch replies with "488 Not Acceptable Here" error to the ReINVITE I'm sending. The only error I can see in the logs is: * [ERR] sofia.c:7280 Reinvite Codec Error!* The only difference between the original INVITE and this ReINVITE is that I try to set IP address to *0.0.0.0*, port to *0* and media attribute to *inactive*. I don't understand what part of the SDP, FS doesn't "like". Attached you can find a more detailed log file. Thank you very much in advance. Regards, -- *Sergio García Ramos * VoIP Engineer @ Quobis <http://www.quobis.com/> | e: sergio.garcia(a)quobis.com | t: +34902999465

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Record in WAV IEEE float format
by Zoltán Szabó 24 Jul '15

24 Jul '15

Hi, I would like to record in Wave IEEE float signed 32bit, 8000hz 256kbps, mono format. For this I can set the record_stereo to false in the session and tried to modify mod_sndfile.c to configure the format, so far I tried like: if (mode & SFM_WRITE) { context->sfinfo.channels = handle->channels; context->sfinfo.samplerate = handle->samplerate; context->sfinfo.format = SF_FORMAT_WAV | SF_FORMAT_FLOAT; } But I receive some weird data when trying to load up the wav in goldwave. The format looks fine but the data is not. Do you have any suggestion? Many thanks, Zoltan

1 0

Strange mod_sofia logic
by Igor Olhovskiy 24 Jul '15

24 Jul '15

Hi! Found really strange behavior, but may be it's correct. I'm using FusionPBX as a FreeSwitch GUI, but problem more to FS. So, when I'm setting context of profile 'internal' to 'public' (yes, it's incorrect, but possible) this profile stop accept registrations. Answering "SIP/2.0 405 Method Not Allowed" and gives list of methods like INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,PRACK,NOTIFY,PUBLISH,SUBSCRIBE But when changing to 'default, all works correct and list of methods are INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE So the question is: how changing of context can inflict on methods supported? And how to control methods supported in more obvious way? (For ex, disable REFER) FreeSwitch version is 1.4.20 -- Best regards, Igor

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