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Announcing the ClueCon Coder Games!
by Ken Rice 22 Jul '15

22 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1CUbP1a Announcing the ClueCon Coder Games! So You Think You Can Code? You’ve seen the presentations, you’ve asked your questions, you have the resources, now it is your time to shine by using the sponsor APIs to create something exciting! We want to see what you can do! Bonus points for each API you can incorporate! Go check out the APIs now to get a head start on the competition and get those creative juices flowing! You have less than two weeks to prepare! Sponsor APIs: FreeSWITCH, Tropo, Kandy, Twilio, Plivo, and more… IPv6 Round Table IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the board at ARIN. Flowroute – Jeopardy Think you know about SIP? Do you know enough to beat the competition? Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the test and come see how much you really know! DTMF-u Do you like all things games? Well, then this is the game for you! But there is a twist! Before you can win the game you must build it! Using your choice of language or API you must build a game! There will be three categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle. You could build something that plays a random DTMF sequence, receives player input, and then either continues or fails the player. Or maybe a WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it! Creative ways to fail a player may give you bonus points. The top three games will be played by everyone and the winner of each will take home a prize! All gaming bots will be screened via a Turing test to ensure no unintended apocalyptic consequences. Show and Tell Alright, now is your chance! You have been playing with the code all day and this is your chance to show off! We want to know what you’ve done and how you did it. Use your creativity and skills as a programmer to impress the judges and win a prize! This is a no holds barred all out free for all! Any language doing anything! Knock our socks off and take home a fabulous prize and a year’s supply of bragging rights! Raffle Grand Prize! The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13″ Retina MacBook Pro!

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ClueCon Coder Games Monday Aug. 3rd 2015
by Brian West 21 Jul '15

21 Jul '15

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg] *August 3rd 2015* August 3rd – August 6th 2015 877-742-2583 • marketing(a)cluecon.com Register NOW! · $1299 Staying at the Hotel · $1599 Staying Elsewhere *Contact Us* https://cluecon.com marketing(a)cluecon.com *So You Think You Can Code?* You’ve seen the presentations, you’ve asked your questions, you have the resources, now it is your time to shine by using the sponsor APIs to create something exciting! We want to see what you can do! Bonus points for each API you can incorporate! Go check out the APIs now to get a head start on the competition and get those creative juices flowing! You have less than two weeks to prepare! Sponsor APIs: FreeSWITCH <http://www.freeswitch.org> - Corvisa <https://www.corvisa.com> - Tropo <https://www.tropo.com> - Kandy <https://www.kandy.io> - Twilio <https://www.twilio.com> - Plivo <https://www.plivo.com> - and more... IPv6 Round Table IPv6 and why you should deploy it ASAP: John Brzozowski, Fellow and Chief Architect, IPv6 at Comcast, Bill Sandiford President of CNOC, Member of the board at ARIN. Flowroute - Jeopardy Think you know about SIP? Do you know enough to beat the competition? Flowroute is hosting a SIP themed game of Jeopardy! Put your brain to the test and come see how much you really know! DTMF-u Do you like all things games? Well, then this is the game for you! But there is a twist! Before you can win the game you must build it! Using your choice of language or API you must build a game! There will be three categories of DTMF-u; Tic-tac-toe, DTMF pattern recognition, or Freestyle. You could build something that plays a random DTMF sequence, receives player input, and then either continues or fails the player. Or maybe a WebRTC based game of tic-tac-toe? Or surprise us! Have fun with it! Creative ways to fail a player may give you bonus points. The top three games will be played by everyone and the winner of each will take home a prize! All gaming bots will be screened via a Turing test to ensure no unintended apocalyptic consequences. Show and Tell Alright, now is your chance! You have been playing with the code all day and this is your chance to show off! We want to know what you've done and how you did it. Use your creativity and skills as a programmer to impress the judges and win a prize! This is a no holds barred all out free for all! Any language doing anything! Knock our socks off and take home a fabulous prize and a year’s supply of bragging rights! Grand Prize [image: Description: mac1]The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro! See the Important Dates Section for Registration details! Why I Think You MUST COME To ClueCon! [image: Description: kk]Hi, I’m Kathleen. I’m the FreeSWITCH and ClueCon Social Media Correspondent. I’ve been working hard all year keeping you all up to date on what’s going on with FreeSWITCH. Today I’m here to let you know more about the upcoming ClueCon 2015 Conference! This year we are adding an optional day on Monday with an all-day Hack-A-Thon with great coding contests, game shows and kick-off fun! If you are interested in WebRTC, Voice over IP or Open Source projects like FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure to the most knowledge and technology in one place. Also, it’s the most fun you can possibly have while still getting a ton of work done! I really look forward to seeing you all there and enjoying the amazing talks, the Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more. Make sure you register today so you can reserve your place among the attendees! Be sure to follow us on Facebook and Twitter to get my latest updates in info! [image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image 2] <https://twitter.com/cluecon> [image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA> [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s>

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Disable 401 and 407 for internal profile
by Admin 21 Jul '15

21 Jul '15

Thanks Brian. That seems to work partially and strangely too. All calls made to extensions above '1019' stopped working now. When a call is destined to any extension above '1019' comes back with '480 Temporarily Unavailable'. Below is a pasted copy of the SIP trace for a call from 1001 to 1019 (works), and 1001 to 1020 (doesn't work). Registration output is also pasted below. ========================================================================== recv 726 bytes from udp/[192.168.1.100]:5062 at 11:18:28.880410: ------------------------------------------------------------------------ INVITE sip:1020@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100> Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE, SUBSCRIBE Supported: 100rel, replaces, from-change Timestamp: 1 Content-Type: application/sdp Content-Length: 238 v=0 o=1001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6001 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6001 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 333 bytes to udp/[192.168.1.100]:5062 at 11:18:28.880719: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100> Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE Timestamp: 1 0.000152 User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:18:28.863966 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001(a)192.168.1.100 [d4084b55-8351-48fa-87ff-01d4b2355d4e] 2015-07-21 11:18:28.863966 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1020 in context public 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:315 sofia/internal/1001(a)192.168.1.100 has executed the last dialplan instruction, hanging up. 2015-07-21 11:18:28.863966 [NOTICE] switch_core_state_machine.c:317 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING] send 836 bytes to udp/[192.168.1.100]:5062 at 11:18:28.885154: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 Max-Forwards: 70 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100>;tag=mrKveDD123K3K Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 Remote-Party-ID: "1020" <sip:1020@192.168.1.100>;party=calling;privacy=off;screen=no ------------------------------------------------------------------------ 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1641 Session 20 (sofia/internal/1001(a)192.168.1.100) Ended 2015-07-21 11:18:28.883950 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001(a)192.168.1.100 [CS_DESTROY] recv 361 bytes from udp/[192.168.1.100]:5062 at 11:18:28.887876: ------------------------------------------------------------------------ ACK sip:1020@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3263-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32631 To: 1020 <sip:1020@192.168.1.100>;tag=mrKveDD123K3K Call-ID: 1-3263(a)192.168.1.100 CSeq: 1 ACK Contact: <sip:sipp@192.168.1.100:5062;transport=UDP> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ >>>>>>>>>>> 1001 - 1019 - Working >>>>>>>>>>>>>>>> recv 726 bytes from udp/[192.168.1.100]:5062 at 11:20:10.977891: ------------------------------------------------------------------------ INVITE sip:1019@192.168.1.100 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100> Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE, SUBSCRIBE Supported: 100rel, replaces, from-change Timestamp: 1 Content-Type: application/sdp Content-Length: 238 v=0 o=1001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6001 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6001 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 333 bytes to udp/[192.168.1.100]:5062 at 11:20:10.978324: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100> Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Timestamp: 1 0.000208 User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:10.963951 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1001(a)192.168.1.100 [3ea757fa-c80f-471f-909c-21d6b35ccd94] 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1019 in context public 2015-07-21 11:20:10.963951 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/1001(a)192.168.1.100 to XML[1019@default] 2015-07-21 11:20:10.963951 [INFO] mod_dialplan_xml.c:635 Processing 1001 <1001>->1019 in context default 2015-07-21 11:20:10.983953 [NOTICE] switch_channel.c:1075 New Channel sofia/internal/1019@192.168.1.100:5061 [2bb31faa-8cc2-400a-8997-3d5126acf148] send 1189 bytes to udp/[192.168.1.100]:5061 at 11:20:10.990948: ------------------------------------------------------------------------ INVITE sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr Max-Forwards: 68 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061> Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: <sip:mod_sofia@192.168.1.100:5060> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 249 X-FS-Support: update_display,send_info Remote-Party-ID: "1001" <sip:1001@192.168.1.100>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1437471870 1437471871 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 23740 RTP/AVP 0 8 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 325 bytes from udp/[192.168.1.100]:5061 at 11:20:10.994691: ------------------------------------------------------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:10.983953 [NOTICE] sofia.c:6729 Ring-Ready sofia/internal/1019@192.168.1.100:5061! 2015-07-21 11:20:11.003950 [INFO] switch_ivr_originate.c:1193 Sending early media 2015-07-21 11:20:11.003950 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1001(a)192.168.1.100! send 1069 bytes to udp/[192.168.1.100]:5062 at 11:20:11.015428: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: <sip:1019@192.168.1.100:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 188 Remote-Party-ID: "1019" <sip:1019@192.168.1.100>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 24500 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 0 RTP/AVP 19 ------------------------------------------------------------------------ recv 593 bytes from udp/[192.168.1.100]:5061 at 11:20:11.995950: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK8829733UK6ttr From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 INVITE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Type: application/sdp Content-Length: 238 v=0 o=5001 53655765 2353687637 IN IP4 192.168.1.100 s=Talk c=IN IP4 192.168.1.100 t=0 0 m=audio 6000 RTP/AVP 0 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 m=video 6000 RTP/AVP 31 a=rtpmap:31 H261/90000 ------------------------------------------------------------------------ send 360 bytes to udp/[192.168.1.100]:5061 at 11:20:11.998449: ------------------------------------------------------------------------ ACK sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bK9Hv29ymZgFHDm Max-Forwards: 70 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411741 ACK Contact: <sip:mod_sofia@192.168.1.100:5060> Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:11.983975 [NOTICE] sofia.c:7488 Channel [sofia/internal/1019@192.168.1.100:5061] has been answered send 1039 bytes to udp/[192.168.1.100]:5062 at 11:20:12.013969: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 INVITE Contact: <sip:1019@192.168.1.100:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 188 Remote-Party-ID: "Outbound Call" <sip:1019@192.168.1.100>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1437471111 1437471112 IN IP4 192.168.1.100 s=FreeSWITCH c=IN IP4 192.168.1.100 t=0 0 m=audio 24500 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 m=video 0 RTP/AVP 19 2015-07-21 11:20:11.983975 [NOTICE] switch_ivr_originate.c:3523 Channel [sofia/internal/1001(a)192.168.1.100] has been answered ------------------------------------------------------------------------ recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:12.018300: ------------------------------------------------------------------------ ACK sip:1019@192.168.1.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 1 ACK Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ recv 371 bytes from udp/[192.168.1.100]:5062 at 11:20:17.020325: ------------------------------------------------------------------------ BYE sip:1019@192.168.1.100:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 2 BYE Contact: 1001 <sip:1001@192.168.1.100:5062> Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] sofia.c:952 Hangup sofia/internal/1001(a)192.168.1.100 [CS_EXECUTE] [NORMAL_CLEARING] send 465 bytes to udp/[192.168.1.100]:5062 at 11:20:17.035053: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5062;branch=z9hG4bK-3281-1-0 From: 1001 <sip:1001@192.168.1.100>;tag=32811 To: 1019 <sip:1019@192.168.1.100>;tag=N1cNg8X4ZcapF Call-ID: 1-3281(a)192.168.1.100 CSeq: 2 BYE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] switch_ivr_bridge.c:758 Hangup sofia/internal/1019@192.168.1.100:5061 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] send 585 bytes to udp/[192.168.1.100]:5061 at 11:20:17.042705: ------------------------------------------------------------------------ BYE sip:1019@192.168.1.100:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF Max-Forwards: 70 From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411742 BYE User-Agent: FreeSWITCH-mod_sofia/1.4.20 +git~20150703T164215Z~b95362f965~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 ------------------------------------------------------------------------ 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session 21 (sofia/internal/1001(a)192.168.1.100) Ended 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1001(a)192.168.1.100 [CS_DESTROY] 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1641 Session 22 (sofia/internal/1019@192.168.1.100:5061) Ended 2015-07-21 11:20:17.023972 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1019@192.168.1.100:5061 [CS_DESTROY] recv 317 bytes from udp/[192.168.1.100]:5061 at 11:20:17.051862: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100;rport;branch=z9hG4bKaUNUBt52Dr7ZF From: "1001" <sip:1001@192.168.1.100>;tag=pa6Dj3e8vN08a To: <sip:1019@192.168.1.100:5061>;tag=32791 Call-ID: 337749d6-aa67-1233-6ca8-705ab6872cf3 CSeq: 78411742 BYE Contact: 1019 <sip:1019@192.168.1.100:5061> Content-Length: 0 ================================================================== Call-ID: 1-2629(a)192.168.1.100 User: 1001(a)192.168.1.100 Contact: "" <sip:1001@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 12:25:01) EXPSECS(3112) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1001(a)192.168.1.100 -------------------------------------------------------- Call-ID: 1-3203(a)192.168.1.100 User: 1020(a)192.168.1.100 Contact: "" <sip:1020@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 13:11:07) EXPSECS(5878) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1020(a)192.168.1.100 -------------------------------------------------------- Call-ID: 1-3216(a)192.168.1.100 User: 1019(a)192.168.1.100 Contact: "" <sip:1019@192.168.1.100:5061> Agent: SIPp Status: Registered(UDP)(unknown) EXP(2015-07-21 13:11:51) EXPSECS(5922) Ping-Status: Reachable Host: everest IP: 192.168.1.100 Port: 5061 Auth-User: unknown Auth-Realm: 192.168.1.100 MWI-Account: 1019(a)192.168.1.100

2 1

mod_fifo in a database shared between multiple freswitches
by Mehdi Shirazi 21 Jul '15

21 Jul '15

HiIs it possible to use mode_fifo in a shared database to have a big call center on multi server freeswitch environment ? RegardsMehdi

2 1

problem with attended transfer
by Tomáš Boros 21 Jul '15

21 Jul '15

Hello all, I think I have found a bug in version 1.5.final+git~20150528T173517Z~6a2fc5e0f7~64bit My scenario is the following: A ----> calls ---> B B ----> calls ---> C B ---> sends REFER to join A and C (attended transfer via SIP) The two calls are connected A is able to talk to C, but after a while, the call is disconnected. When the REFER message arrives, FS sends BYE message to B with cause "NORMAL_CLEARING" and after a while (about 10 sec) FS sends a BYE message to C with cause "ATTENDED_TRANSFER" and the transferred call is hung up. I assume, that those causes should be swapped. BYE with attended transfer to B and no BYE at all for C until the call is ended by the endpoints. Do someone have this issue, how you fixed it? Should I upgrade my freeswitch to a newer version? -- Tomáš Boros

3 2

script answers one session, creates another, need to hand off one to another lua without hanging up
by Trever L. Adams 21 Jul '15

21 Jul '15

Hello Everyone, I am stuck. I have a lua script that gets run from a bind digit setup. The only way to make it work was to have the bind digit call another lua script that contains: theret = api:executeString("uuid_dual_transfer " .. my_uuid .. " 'm:^:set:number=" .. call_id_num .. "^ set:name=" .. call_id_name .. "^lua:lua_script.lua/inline' " .. "ext/XML/context") The incoming call goes to the ext the lua script is a menu for the internal user. I have another script that answers some incoming calls. It then creates a new session and gives the internal user a menu. One option is to connect, another to voicemail. I have these working. The problem is that one option is to call the lua script above and to send the incoming call to the same ext. At one point I had it to where it would do one and then the other. I need a way to make them happen simultaneously. The lua script in the executeString above uses session, but the session is created as internal_session in the other script. i have tried bridging with execute_on_answer calling the script with the dual transfer above, but it executes AFTER hangup. Has anyone found a way to solve this problem? session:executeString doesn't seem to exist anymore for lua. Thank you, Trever

1 0

FreeSWITCH Week in Review (Master Branch) July 11th-July 17th
by Ken Rice 20 Jul '15

20 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1JtgsMJ FreeSWITCH Week in Review (Master Branch) July 11th-July 17th Hello, again. This passed week in the FreeSWITCH master branch we had 43 commits. We had a number of cool new features this week including: added functionality for capturing screenshots from both legs to uuid_write_png, the addition of new multi-canvas and telepresence features in mod_conference, the addition of vmute member flag to mod_conference, and an API for removing an active ladspa effect on a channel. Join us on 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-7769 [mod_conference] Add new multi-canvas and telepresence features FS-7847 [mod_conference] Add layers that do not match the aspect ration of conference by using the new hscale layer param for horizontal scale, and add zoom=true param to crop layer instead of letterbox, add grid-zoom layout group that demonstrates these layouts, and fix logo ratios and add borders too. FS-7813 [mod_conference] Add vmute member flag. FS-7846 [mod_dptools] Add eavesdrop_whisper_aleg=true and eavesdrop_whisper_bleg=true channel variables to allow you to start eavesdrop in whisper mode of specific call leg FS-7760 [mod_sofia] Revise channel fetch on nightmare transfer and add dial-prefix and absolute-dial-string to the nightmare xml FS-7829 [mod_opus] Add sprop-stereo fmtp param to specify if a sender is likely to send stereo or not so the receiver can safely downmix to mono to avoid wasting receiver resources FS-7830 [mod_opus] Added use-dtx param in config file (enables DTX on the encoder, announces in fmtp) FS-7824 [mod_png] Add functionality for capturing screenshots from both legs to uuid_write_png FS-7549 [mod_ladspa] Added an API for removing an active ladspa effect on a channel. For conformance reasons, the uuid_ladspa command now accepts ‘stop’ and ‘start’, while the previous functionality (without any verb) which will simply add ladspa remains intact. Improvements in build system, cross platform support, and packaging: FS-7845 [mod_conference] Break up mod_conference into multiple source files to improve build performance FS-7769 [mod_conference] Fixed a build issue FS-7820 Fix build system typo. Don’t assign the same variable twice. FS-7043 Fixed apr1 unresolved symbols in libfreeswitch.so.1.0.0 FS-7130 Make /run/freeswitch persistent in the Debian packages, so it will start under systemd The following bugs were squashed: FS-7849 [verto] Remove extra div breaking full screen in html FS-7832 [mod_opus] Fixes when comparing local and remote fmtp params FS-7731 [mod_xml_cdr] Fixed a curl default connection timeout FS-7844 Fix packet loss fraction when calculating loss average

1 0

Mod_verto vs WebRTC(ws/wss)
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, Any one can help me what is difference between mod_verto and WebRTC(ws/wss) mod.

5 12

mod_managed vs eventsocket
by Prashant Choudhary 20 Jul '15

20 Jul '15

Hello All, I have worked with LUA and freeswitch. Now I want to migrate to .Net API as I want to experiment with machine learning using sample calls. I am trying to control around some 2-300 calls directly through my API. This may require the call audio stream to be forwarded in real time to .NET I a few questions that might be very silly. 1. Which one is better to achieve this, mod_managed or ESL. 2. I was able to find some examples related to eventsocket, but same for mod_Managed was rare. Can you suggest me any doc(of-course except the wiki link) or any email chain which explains this or anything else. This will be really a big help for me. As currently I am stuck. And I still find the LUA way easier. Thanks

4 7

Freeswitch WS+SIPJS+sip2sip
by Murugan Pandian 20 Jul '15

20 Jul '15

Hi, I try to register sip2sip.info account from browser using ws signalling,but i am getting SIP/2.0 403 Forbidden error in my console. The same sip2sip.info account work in all soft client

4 5