November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

And the Weddings Back on!
by Ken Rice 01 Nov '13

01 Nov '13

Its my pleasure to announce Brian West and Greg Dunn will resume their wedding on Monday! After a good delay due to a health issue, Greg is now able to travel! So you guys send the congrats to Brian and Greg! The happy couple has posted a list of items they can use @ http://www.amazon.com/gp/registry/wedding/11UO8CGGBUU59/ref=cm_sw_em_r_wr_Mo 8Csb1EXEX9W_wb Again Congrats Brian & Greg! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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Scalability and Bridge
by Valter Cardoso 01 Nov '13

01 Nov '13

I'm thinking in using multiple FreeSwitch instances to support high traffic load, but I don't know how to deal with bridges in this scenario. If I had two incoming calls, each one created in a different FS instance, it's not possible to create a bridge with these calls, right? How should I deal with bridge using multiple FS instances? Thanks, Valter

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RPS vulnerability, Yealink responds.
by Ken Rice 01 Nov '13

01 Nov '13

Hey Guys, As you know theres been a big buzz about Remote Provisioning Services and vulnerabilities that are possible. We have been in touch with Yealink, and they have released the following statement. ----- Snip ------- Recently we received reports stating there were security vulnerabilities in Yealink phones¹ RPS feature. We have investigated in these reports and found out that this problem was already solved in our V71 version of firmware. Correction and Precaution Solutions for RPS Security Vulnerability Issues Yealink V71 version firmware could completely guard the Yealink IP Phones from the security risks. How to operate: 1) To upgrade the Yealink IP Phones to V71 firmware; 2) To encrypt the configuration files with Yealink cfg encryption tool. The Yealink cfg encryption tool could be downloaded per below link, http://www.yealink.com/Upload/T2X/T2X-V71/Yealink%20Configuration%20Encrypti on%20Tool.zip Yealink take our customers¹ security very seriously and are constantly working to improve the security features of our products. Yealink therefore also highly recommend our customers regularly update firmware to the latest version to benefit from our new features and facilities. ----- Snip ------ So if you are using Yealink and RPS encrypt those configs and upgrade to atleast the V71 firmware. -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org G+ ClueCon : http://fs0.us/cluecon-gplus FB ClueCon : http://fs0.us/cluecon-fb G+ FreeSwitch : http://fs0.us/freeswitch-gplus FB FreeSWITCH : http://fs0.us/freeswitch-fb Twitter : @FreeSWITCH_WIRE irc.freenode.net #freeswitch

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ACK disappeared from SIP session, Help
by "Гришин В.С." 01 Nov '13

01 Nov '13

the scheme is PBX---E1--Cisco 2811-----VoIP SIP-----FreeSWITCH-----VoIP SIP---asterisk(1.8.20)---IP Phone FreeSWITCH IP is 191.21.19.12 Asterisk IP is 162.48.13.26 Call comes to FreeSWITCH from a Cisco 2811. SIP session between freeswitch and asterisk(1.8.20) |Time | 191.21.19.12 | | | | 162.48.13.26 | |21.048754| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.048758| INVITE SDP (g711A CN) |SIP From: "91295" <sip:9195@191.21.19.12 To:<sip:9647@162.48.13.26 | |(5065) ------------------> (5060) | |21.065794| 100 Trying| |SIP Status | |(5065) <------------------ (5060) | |21.122406| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |21.143580| RTP (g711A) |RTP Num packets:50 Duration:0.922s SSRC:0xB5320A6 | |(21866) <------------------ (12772) | |22.282684| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |22.296762| 181 Call is being forwarded |SIP Status | |(5065) <------------------ (5060) | |27.558268| 180 Ringing |SIP Status | |(5065) <------------------ (5060) | |35.039883| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.307222| RTP (g711A) |RTP Num packets:1490 Duration:29.773s SSRC:0x530CAB37 | |(21866) <------------------ (12772) | |35.476614| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |35.532995| RTP (g711A) |RTP Num packets:2966 Duration:29.680s SSRC:0x308FC816 | |(21866) ------------------> (12772) | |36.436846| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |38.282686| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |42.025399| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |45.637280| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |49.343115| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |53.119940| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |56.907800| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |60.791453| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |64.606730| 200 OK SDP (g711A) |SIP Status | |(5065) <------------------ (5060) | |65.219392| BYE | |SIP Request | |(5065) <------------------ (5060) | |65.234354| 200 OK | |SIP Status | |(5065) ------------------> (5060) | |65.234359| 200 OK | |SIP Status | |(5065) ------------------> (5060) | It is visible that there is no package with ACK confirmation from a FreeSWITCH to Asterisk. Asterisk sends several times SIP OK on a freeswitch, and without having waited confirmations (ACK) after 23 seconds sends SIP BY on a freeswitch and on IP Phone. After that call stops. Before disconnection subscribers hear each other of 29 (+-) seconds In tcpdump between Cisco and FreeSWITCH it is visible that the SIP ACK packet from Cisco arrives on a freeswitch. In what place the SIP ACK in a FreeSWITCH disappeared? What to me to analyse in SIP session? Maybe someone solved a similar problem? please, Help! Vladislav Grishin

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Ring timer
by Andre 01 Nov '13

01 Nov '13

Is the call_timeout a timeout for ring timeout? I see this note: WARNING: Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received. I am not setting Early Media. What timeout should I use to let the call ring for x seconds than hang up? Thanks

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