- FreeSWITCH-users - FreeSWITCH Mailing Lists

LRN DIPS using 302 redirects
by Aqs Younas 11 Jul '14

11 Jul '14

Does anybody knows how to do this.?I have search a lot but could not find anything worthy. Could someone please provide me with an example.? Any help in this regard would be much appreciated.

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mod_verto and registrar
by Tamas Jalsovszky 11 Jul '14

11 Jul '14

Hello, I'm thinking about to use mod_verto in the future and one question came up regarding registration. Can we make the following scenario with mod_verto? webrtc client registers to an FS instance with mod_verto (FS1) - like a SIP reg. The media however has to bypass it and go between the client and the media gateway (FS2). I'm not sure whether verto or sip should be used between FS1 and FS2. webrtc client----> FS1 mod_verto registrar/proxy --- FS2 mod_verto media gateway --> SIP Sorry if my question was a bit silly :) jalsot

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ClueCon 2013 Reminder: Register Today and Save!
by Ken Rice 10 Jul '14

10 Jul '14

This is just a friendly reminder that Ticket Prices for ClueCon 2014 go up $200 on July 18th! That¹s a week and a half away! Don¹t wait! Register Today! Also, Keep in mind that tomorrow those chances to win in the drawings get cut in 1/2! Visit ClueCon.com Today and Register! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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LRN DIPS using 302 redirects
by Aqs Younas 10 Jul '14

10 Jul '14

Does anybody knows how to do this.?I have search a lot but could not find anything worthy. Any help in this regard would be much appreciated.

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INVITE without client IP or port for RTP
by Kees Jan Koster 09 Jul '14

09 Jul '14

Dear All, We are currently investigating the possibility to forego STUN checks on the client and to avoid using RTP relays. By not using these we simplify the client call code, reduce the number of components to maintain and avoid the relay latency. Ideally, the client does not send any transport information as part of the call setup. Instead, we'd like call to be established without FreeSwitch knowing the IP address or media ports for the client. Once a call is answered, FreeSwitch tells the client what IP and port it should use for the media stream. The client sends the RTP stream to the specified IP address and UDP port on FreeSwitch. FreeSwitch uses the incoming stream to determine the IP address and UDP port of the client and sends its RTP stream there. 1) Is this possible? 2) What do I have to do to do this. -- Kees Jan http://java-monitor.com/ kjkoster(a)kjkoster.org +31651838192 The secret of success lies in the stability of the goal. -- Benjamin Disraeli

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Freeswitch Week in Review(Master Branch) June 29th - July 5th
by Kathleen King 09 Jul '14

09 Jul '14

Hello, again. This week in the FreeSWITCH master branch we had 46 commits. We saw a few improvements and tweaks to mod_verto, the addition of 44.1khz codec definitions to the FreeSWITCH core, added PUT support to mod_curl `curl` API and application, updated to use PocketSphinx 0.8, and, for endpoints using stereo, there is now 3D positioning support in mod_conference using openal. FreeSWITCH now deployed Stash from Atlassian! If you have a Jira account, login to Stash with the same credentials and check it out. The following bugs were squashed: 1518240 FS-6637 fix for transfer_after_bridge in bypass_media mode in switch_ivr_bridge Jira: http://jira.freeswitch.org/browse/FS-6637 9ea93c4 Avoid buffer over-read on null cipher AEAD in srtp 3bf2b9a Prevent buffer overflow from untrusted RTP/SRTP lengths in srtp 9b14633 fix compiler error on CentOS 6 caused by recent short uuid logging change in fs_cli b9da514 Set the algorithm member on cipher_t when allocating AES crypto instance. Apply same fix to NULL cipher. aa4261d Avoid buffer-overflow on short RTCP/SRTCP packets in srtp 5d70df9 FS-6637 fix for transfer_after_bridge in bypass_media mode in switch_ivr_bridge Jira: http://jira.freeswitch.org/browse/FS-6637 2423fc7 FS-6540 This should proxy all refers now, the param has changed to proxy-refer in mod_sofia Jira: http://jira.freeswitch.org/browse/FS-6540 New features that were added: 41a7d1c some refactoring to make new stereo stuff more stable in mod_conference 1a56298 add 44.1khz codec defs to FS core 0ac78ca add openal code to mod_conference 0bb7f16 check for member channel in mod_conference d833b9d expand table width a bit in mod_verto 024162c Add support for 16-byte auth tag for AES GCM mode in srtp. c9d0f89 update favicon.ico 9e0d1d6 only count unmuted channels in positioning in mod_conference c17d58b Refactor the curl PUT read callback in mod_curl 7f9348d fix volume adjustments in stereo conference in mod_conference 39200cd increase memory for buffers to hold new decoded stereo codecs in the FS core 831d589 preserve timer name for recovery in the FS core 1ef5d11 Adjustments made to ringing var for mod_verto 3b629c8 updated directory using asr/tts in lua 6fad1bf clean up demo a little in mod_verto cdbed1f Update to use PocketSphinx 0.8 in mod_pocketsphinx f0d16d0 expand positioning api usage in mod_conference a84b76b Add PUT support to mod_curl `curl` api and app in mod_curl 2fea352 Advance the data_stream data pointer in mod_curl 99f1566 fix jitterbuffer endless loop of missing packets in switch_stfu 0df352c update image in mod_verto 2cbea04 clean up demo a little Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (253) 617-3210 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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ClueCon Weekly and FreeSWITCH Weekly News and Notes
by Ken Rice 09 Jul '14

09 Jul '14

Greetings FreeSWITCHers! Some news and Notes for you! ClueCon Weekly, today 1PM Eastern. Todays Topic, Updates on Infrastructure changes and where we are headed. Update on ClueCon Prep! OpenSIPS Announcement! And Community Scrum! Next week (July 16) Bogdan of OpenSIPS fame will be with us on ClueCon Weekly! And July 2rd Fred Dixon of Big Blue Button will be joining us! IF you have not already registered for ClueCon now is the time to do so! Visit ClueCon.com!! Time is running out for those extra chances to win! Register by July 18th and get 2 extra tickets for the Door Prize Drawings! What are the give-a-ways this year? We have everything from Bawls Caffeine Mints to $100 ThingGeek Gift Certificates to who knows what else the sponsors are bringing! I¹m sure there will be a pile of hardware also! Hear you at Noon! And See you in August! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Less than one month to go before OpenSIPS Summit!
by Alex Goulis 09 Jul '14

09 Jul '14

*Less than one month to go before the OpenSIPS Summit!* Ever wonder how you can: * load balance your FreeSWITCH servers * provide fail over protection for your FreeSWITCH servers * distribute services to your FreeSWITCH server clusters If you have, then you need to attend the OpenSIPS Summit! For all of you attending ClueCon, we'll be holding an OpenSIPS summit on Aug 4 running most of the day before ClueCon officially kicks off. Receive a 25% discount for the summit by using "clucon2014" as your discount code. Last year's summit was a tremendous success. Everyone that attended enjoyed interesting topics and discussions. Some of the topics this year will include: * Redundancy and high availability for Cloud Solutions using gateways such FreeSWITCH * Advanced SIP Routing using OpenSIPS * Using OpenSIPS as a single entry point for a SIP network You can get more information about the summit at: http://www.opensips.org/Community/Summit-2014Chicago Registration is still open: http://www.opensips.org/Community/Summit-Registration Last year's summit highlights: http://www.opensips.org/Community/Summit-2013Chicago

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configure dialplan gsmopen
by D- Bamboo 09 Jul '14

09 Jul '14

Hi thanks for your answer when i try this command - make a call from the fs_cli and park it: originate gsmopen/RR/+6281313888001 &park() - make a call from the fs_cli and make sound test: originate gsmopen/RR/+6281313888001 &echo() i receive the call to my cell phone. but when i answer there is no sound to my server. and for this instruction - I can see from your logs that on incoming calls you use: bridge(user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}) - which later is translated to: EXECUTE gsmopen/gsm01 bridge(user/(a)192.168.1.3) - basically the variable 'user_data' isn't set, which causes the problems on incoming calls sorry sir i don't know how to configure or setting this instruction. what i try to make is i can call to my cell phone from x-lite and i can received a call from cell phone to my x-lite. sorry again i really newbie. i just setting inbound route and outbound route from fusionpbx and i dont know how to set expression input. may you can help me with example xml file and where directory i must put in thank you very much

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configure dialplan gsmopen
by D- Bamboo 09 Jul '14

09 Jul '14

hi please help me to configure my dialplan outbound and inbound call. i use ms windows 7, freeswitch source that compile with VS 2010, mod_gsmopen, huawei e153u, x-lite 4. already follow this step https://wiki.freeswitch.org/wiki/Gsmopen but still can to establish calling to my cellular phone. this is my log when make a call from x-lite to my cell phone and other log when receive call from my cell phone to my modem number. https://jira.freeswitch.org/secure/attachment/21491/freeswitch%28calling%29… https://jira.freeswitch.org/secure/attachment/21492/freeswitch%28receive%20… thanks for your help

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