November 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Secured media
by Sergey Safarov 03 Nov '15

03 Nov '15

Are is "require-secure-rtp <https://freeswitch.org/confluence/display/FREESWITCH/mod_sofia>" profile attribure is broken in master? I has find that flag "PFLAG_SECURE" is setted, but is not used anywhere. Sergey.

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Bridge call blast scenario
by John Nash 03 Nov '15

03 Nov '15

I need to test a case where I dial two destinations (Lets say destination A and Destination B) simultaneously. I am able to do that using "," as separator with "ignore_early_media" and when I execute it ... Destination A sends media (183) after few seconds (while A still playing) destination B also sends media (183) Now what i need to check is that is there any way to hangup the destination A if anytime we receive 180 or 183 message from destination B. Possible? Regards, John

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ASTPP Version 2.3 is Released
by ASTPP Opensource VOIP Billing 03 Nov '15

03 Nov '15

Hello Everyone, We are glad to release ASTPP new version v2.3 in such a short time with number of bug fixes and changes. *The list goes down here:* 1. Resolved send mail issue 2. Updated Summary reports to display correct statistics 3. Resolved call disposition issue 4. Corrected prefix field validation in ACL module 5. Corrected company name field validation in accounts module 6. Resolved failover issue if primary trunk max channels are used 7. Resolved ACL list security issue in reseller login 8. Allowed to add range in ACL 9. Added one more failover gateway option 10. Limit Max calls / interval 11. Updated invoice and balance script 12. Remove duplicate dialing on trunk from LCR 13. Loader on dashboard 14. Improved CSS of Dashboard for better performance 15. Low credit notification 16. Customer my rates display with markup We offer installation, training, support and customization services for ASTPP. Reach us on sales(a)inextrix.com for details. *To install v2.3:* # wget --no-check-certificate https://goo.gl/HjHXlF -O install.sh # chmod +x install.sh # ./install.sh *Upgrade steps:* # wget --no-check-certificate https://goo.gl/Whc9Vf -O update.sh # chmod +x update.sh # ./update.sh In case if you find any issues, then please feel free to report in our bug tracker below: http://bugs.astppbilling.org/ Thank you everyone for your continuous support in ASTPP journey. -- Regards, ASTPP Team iNextrix Technologies Pvt. Ltd.

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Dial plan transition in freeSWITCH using MOD_XML_CURL
by david.fu＠oocl.com 03 Nov '15

03 Nov '15

Hi FreeSWITCH experts, We would like to use MOD_XML_CURL to send HTTP request to application server which returns dial plan to freeSWTICH. Let's say we have 2 dial plans in the application server. We would like the MOD_XML_CURL to get the first dial plan and then submit request to application server again to get the second dial plan. Is it possible to implement it in freeSWTICH ? Is there any SUMBIT function implemented in dial plan so that the first dial plan could ask MOD_XML_CURL to get the second one ? Thanks. Best Regards, David Disclaimer : This email and all contents are subject to the following disclaimer: http://emaildisclaimer.oocl.com/default.html

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FreeSWITCH behind an Nginx proxy
by Dan Edwards 03 Nov '15

03 Nov '15

We're trying to build a system that allows WebSocket SIP traffic through an Nginx proxy to FreeSWITCH and I'm having trouble getting the proper IP address offered in the SDP. I added ext-rtp-ip and ext-sip-ip to the profile, but if I come in via the Nginx proxy, FS always offers the local address. I made some code changes to FS to catch the X-Forwarded-For header on the initial WebSocket connect, thinking I could substitute the forwarded-for IP address for the Nginx address, but this did not correct the problem. To be clear, if I run Nginx on 172.1.1.1 and FS on 172.1.1.2, when I come in via Nginx, FS sees the SIP IP as 172.1.1.1 and offers 172.1.1.2 as the RTP address. If I come in via a port-forward, FS offers what I have in 'ext-rtp-ip' as the IP address. Is there a way to force FS to always offer ext-rtp-ip, even if the address is local or does this require a code change? If it requires a code change, where does this logic exist? Thank you, Dan

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FreeSWITCH Week in Review (Master Branch) October 24th-October 31st
by krice＠freeswitch.org 02 Nov '15

02 Nov '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1GItGdR FreeSWITCH Week in Review (Master Branch) October 24th-October 31st FreeSWITCH got some neat improvements this week with work going into improving the handling of vw and vh core file parameters in mod_av to avoid video cropping and crashing, the addition of a new configuration setting in mod_opus to show the decoder stats at the end of the call, and exposing SRTP and SRTCP crypto keys as channel variables to help with debugging. Join us Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-8281 [core] Expose SRTP and SRTCP crypto keys as channel variables to aid with debugging FS-8313 [mod_opus] Introduced new configuration setting ‘decoder-stats’ to show decoder stats at end of call (how many times it did PLC or FEC) FS-8380 [mod_av] Improve the handling of vw and vh core file parameters to avoid video cropping and crashing Improvements in build system, cross platform support, and packaging: FS-8389 [build] Fixed msvc 2015 build warnings FS-8398 [Ubuntu] Added event_handlers/mod_amqp to avoided modules for Ubuntu 14.04 Trusty The following bugs were squashed: FS-8222 [verto_communicator] Updated getScreenId.js in order to detect plugin issues and attached an ‘ended’ event to screenshare stream in order to detect ‘stop sharing’ click FS-8392 [mod_av] Fixed rtpmap to allow both H263 and H263+ codecs to be offered FS-8373 [mod_av] Fix for bad recording quality when using fast encoding FS-8397 [core] Fixed a race condition incrementing the event-sequence number FS-8154 [core] Fixed a segmentation fault occurring while eavesdropping on video call FS-8391 [core] Fixed a SDP parsing error for rtcp-fb FS-8319 [mod_opus] Fixed and cleaned up switch_opus_has_fec() and switch_opus_info() to avoid FALSE positives for packets with FEC at high frame sizes. FS-8344 [mod_opus] Toggle FEC ON only on the last frame which is to be packed The FreeSWITCH 1.4 branch had a few bug fixes added this week. The following bugs were squashed: FS-8338 [core] Fixed an issue when setting the ringback variable with an outbound call via the bridge app, if the inbound leg is stereo the ringback tone is still rendered as mono causing the resulting ringback to be higher pitched and incorrect. FS-8378 [mod_esf] [core] Fixed a crash when using esf_page over loopback when transcoding and added tests for esf over loopback. Also refactor a bit to clarify code and get better debug in gdb FS-8370 [mod_rayo] Fixed another place in where a message was freed after being queued for delivery. This resulted in a freed object being serialized, crashing FS

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DTMF in SIP and pstn
by Vishal Sharma 02 Nov '15

02 Nov '15

Hi, In pstn when I play a sound file which contains dtmf digits, FS produce dtmf digits which can be detected on other end. I use sangoma card. but when I play same file, on sip. no dtmf is generated or detected on other end. it simply play a beep sound ... Can I play a sound file and generate DTMF digits on SIP. Regards, Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

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We're one week away from OpenSIPS Week in Austin!
by Alex Goulis 02 Nov '15

02 Nov '15

We're almost there everyone! The OpenSIPS Summit in Austin is 1 week away. There's still some room left for all you procrastinators! We also have a few spots left for the OpenSIPS LIVE Bootcamp following the summit. Enjoy a rare opportunity to train with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! Help us "Keep Austin Wierd" by celebrating OpenSIPS Week in Austin, TX and become an OpenSIPS Certified Professional in the process! As part of OpenSIPS week in Austin, we'll be holding a LIVE training following the conclusion of the OpenSIPS Summit on Nov 10th. If you've ever wanted to become an OpenSIPS Certified Professional, this is you're chance! You'll have the rare opportunity to train live with the founder of the OpenSIPS Project Bogdan-Andrei Iancu! It's not a secret that there is a growing demand for OpenSIPS Certified Professionals all over the world. This training is intended to offer you a 50/50 view between theoretical and practical applications of OpenSIPS. It covers advanced topics that are critical to telecom network operators in regards to security, scaling, and integration. For participants that would like an introduction to OpenSIPS basics and to qualify for prerequisites to the advanced topics, there will be a 2 day online primer prior to the live training. On November 11th -13th 2015 the live portion of the training will be held at the OMNI Downtown Austin in Austin, TX. This is our preferred hotel for the summit. If you're attending the summit and staying there, you'll enjoy the convenience of not having to leave the hotel. Prior to the live training, we'll be offering the first 2 days of the training online. To register and get more information go to: http://ebootcamp.opensips.org OpenSIPS Summit in Austin November 9th-10th To register and get more information go to: http://www.opensips.org/Community/Summit-2015Austin HOTEL Hotel information can be found at: http://www.opensips.org/Community/Summit-2015Austin#toc5 TRAVEL INFORMATION Airport: Austin=Bergstrom International Airport Website: http://www.austintexas.gov/airport This "can't miss" training is coming to you via a joint effort between OpenSIPS Solutions and V.Office Networks. Don't miss this rare opportunity to train live with the founder of the OpenSIPS Project and become part of the growing group of OpenSIPS Certified Professionals all over the world!

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