February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Nibblebill round to minute
by Zoltán Szabó 27 Feb '15

27 Feb '15

Hi, Is there any way to configure nibblebill to round up the billing of a call leg to minutes? So for example if the call is just 25 seconds long, I would like to bill one minute. If the call is 80 seconds, 2 minutes, etc. Many thanks, Zoltan

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Feb '15

27 Feb '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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Re- FreeSWITCH CDRs ${billsec} not correct when using Lua scripts and running a hangup handler with any delays inside it
by Andrew Keil 27 Feb '15

27 Feb '15

To FreeSWITCH users, I am currently experimenting with Lua scripts on FreeSWITCH (latest build) and found an interesting issue with FreeSWITCH default CDRs (inside log/cdr-csv/1000.csv). See below for my simple Lua script called from a diaplan as follows: <include>  <extension name="ivr_demo"> <condition field="destination_number" expression="^(9101)$"> <action application="lua" data="myservice.lua"/> </condition> </extension> </include> What I noticed is that placing a call into FreeSWITCH via a SIP phone (calling 9101) and landing on myservice.lua everything processes as expected. If I hangup after 3 seconds then I should expect the CDR to show ${billsec} = 3. However this is not the case since both ${duration} and ${billsec} are actually set once the Lua script completes (not based on HANGUP). So inside 1000.csv I see duration and billsec both having 8 seconds. You will see below that inside my hangup handler (function myHangupHook(s, status, arg)) I fire an event and wait up to 5000ms for an event to be returned (currently I am not returning an event so the 5000ms will take place). This explains the extra 5 seconds after the 3 seconds of the call. My question is since FreeSWITCH knows that the HANGUP has taken place why is the ${billsec} not correct (ie. should be 3 seconds)? I tried adding "session:hangup()" as the first line inside my hangup handler (function myHangupHook(s, status, arg)) and this made no difference. Is there a solution to this issue? Perhaps there is a different CDR variable that will contain the correct duration based on when HANGUP was actually received. Thanks in advance for any assistance with this. Regards, Andrew Keil Simple Lua Script: -- myservice.lua -- To test a simple service function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n") -- answer the call freeswitch.consoleLog("INFO", "ANSWER CALLER\n") session:answer(); -- sleep a second freeswitch.consoleLog("INFO", "Sleep for 1000ms\n") session:sleep(1000); freeswitch.consoleLog("INFO", "Playing 12345\n") session:say("12345", "en", "number", "pronounced"); freeswitch.consoleLog("INFO", "Finished playing 12345\n") if (session:ready()) then -- hangup freeswitch.consoleLog("INFO", "HANGUP\n") session:hangup() end end -------------------------------------------------------------------------------------------- -- Functions section ... function myHangupHook(s, status, arg) freeswitch.consoleLog("INFO", "myHangupHook: " .. status .. "\n") -- send hangup event test (after caller hangs up) local event = freeswitch.Event("CUSTOM", "visytel::HANGUP?") event:addHeader("hi", "there") event:addHeader("channel-uuid", channeluuid) -- Add Event consumer if a response is required! local con = freeswitch.EventConsumer("CUSTOM", "visytel::HANGUP!") -- fire event freeswitch.consoleLog("INFO", "Fire HANGUP event\n") event:fire() -- Wait for 5000ms to test response local retevent = con:pop(1, 5000) if retevent then freeswitch.consoleLog("INFO", "Reply received!\n") freeswitch.consoleLog("INFO", string.format("Reply received: %s\n",retevent:getHeader("Result"))) end end -- Setup handlers here myparam="empty"; session:setHangupHook("myHangupHook", "myparam") -- Source channel-uuid here {used inside myHangupHook handler and for all fired events inside the header channeluuid = session:getVariable("uuid"); MainService() -- End of lua script

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How to measure QOS?
by Andrew V 27 Feb '15

27 Feb '15

How do you measure quality of service across all calls? What are ways to avoid bad quality of service? Say the call volume is in your control. Is it as easy as not letting the call volume get out of control? The attached is a hypothesis of the relationship between QOS and the number of concurrent calls. Does it work that way? What other factors need to be taken into account? [image: Inline image 1]

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server to NAT'd server gateways
by Robert Oldham 26 Feb '15

26 Feb '15

I have two FreeSWITCH servers that I need to gateway calls between. Sip_A is on a public network with a static IP address. Sip_B is on a private network that is NAT'd. I am trying to pass calls from devices connected to Sip_B, through Sip_A and on to another server. I am also trying to pass calls from Sip_A into Sip_B. I don't have access to the firewall at the location of Sip_B so I am not able to forward standard ports into Sip_B. I am fairly new to FreeSWITCH, so I'm not certain how to solve this problem well. Many of the documents I have read online recommend using ACLs and port forwards for this. However, the private network does not have a static IP address that I can give access to and, as mentioned, I cannot make port forwards. Instead, I have created an extension on Sip_A and a gateway on Sip_B that registers using the extension on Sip_A. That works fine for passing calls from Sip_B through Sip_A and on. Sip_B has been maintaining a registered connection to Sip_A through the NAT'd connection for several days now without trouble. Not knowing a better solution, I have been searching for a gateway configuration on Sip_A that would allow it to route calls back over the registered extension from Sip_B. I have not found a way to accomplish it yet. As an alternative, I have also tried without success to create a passive gateway on Sip_A that the gateway on Sip_B could register with. Any help is appreciated! Thanks, Robert

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TCMalloc and freeswitch
by Ali Pey 26 Feb '15

26 Feb '15

Hello, Is there a way to switch to TCMalloc without recompiling freeswitch? Can someone please point me to (or provide) some instructions? (Preferably for Centos 6) Thanks, Ali Pey

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How to play file into session not endlessly?
by Борисов, Дмитрий / Dmitriy Borisov 26 Feb '15

26 Feb '15

I have next task: play sound file list to session only once. Callee must to control session by DTMF. Session must to hangup when sound file list are played if no keys a pressed by callee. Session is created and controlled by Lua. How? -- with best regards, Dmitriy Borisov

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updated xml file not seen by freeswitch
by dragoncloud 26 Feb '15

26 Feb '15

Hi, Don't know how I did this, but, I put in the wrong IP address in my xml file for freeswitch. I've edited and corrected it, but, freeswitch is still showing the incorrect one and it's in NOREG status. Can anyone tell me how to get freeswicth to see the new xml file? I've done "sofia profile external rescan reloadxml" but that doesn't do it. I've even tried "sofia profile (profile name) killgw (gateway name)" to get rid of the incorrect one, but, to no avail. Any help would be appreciated! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/updated-xml-file-not-seen-by-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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rtcp on a leg only
by Chris Martineau 26 Feb '15

26 Feb '15

?Hi, I have read in a number of places that you can have rtcp on 1 leg only but can see no examples or description of how to do this. I only want to send rtcp on the incoming a leg is this possible, and if so how do you do it? Thanks Chris

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Testing WebRTC - FS/master
by Victor Medina 26 Feb '15

26 Feb '15

Hi guys! I am currently testing WebRTC peers in FS/master. I am seeing some very strage messages in the console messages... 2015-02-24 15:08:48.982146 [INFO] switch_core_media.c:5742 Activating VIDEO RTCP ICE 2015-02-24 15:08:48.982146 [NOTICE] switch_rtp.c:4012 Activating RTCP video ICE: a53ce8c8:e5uaURjkW7HYp2q9 192.168.0.22:60544 2015-02-24 15:08:48.982146 [INFO] switch_rtp.c:3104 Activate RTP/RTCP video DTLS client 2015-02-24 15:08:48.982146 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/test001(a)cibersys.me! 2015-02-24 15:08:55.242147 [WARNING] switch_core_media.c:2757 NO candidate ACL defined, Defaulting to wan.auto 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2790 Choose audio Candidate cid: 1 proto: udp type: host addr: 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57539 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2795 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.56.1:57540 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2795 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.0.19:57541 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2884 No audio RTCP candidate found; defaulting to the same as RTP [2001::9d38:6abd:1cdf:dd44:2fab:aebf:57539] 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2919 setting remote audio ice addr to 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57539 based on candidate 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2944 setting remote rtcp audio addr to 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57539 based on candidate 2015-02-24 15:08:55.242147 [WARNING] switch_core_media.c:2757 NO candidate ACL defined, Defaulting to wan.auto 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2790 Choose video Candidate cid: 1 proto: udp type: host addr: 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57542 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2795 Save video Candidate cid: 1 proto: udp type: host addr: 192.168.56.1:57543 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2795 Save video Candidate cid: 1 proto: udp type: host addr: 192.168.0.19:57544 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2884 No video RTCP candidate found; defaulting to the same as RTP [2001::9d38:6abd:1cdf:dd44:2fab:aebf:57542] 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2919 setting remote video ice addr to 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57542 based on candidate 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:2944 setting remote rtcp video addr to 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57542 based on candidate 2015-02-24 15:08:55.242147 [INFO] switch_core_media.c:5321 Activating Audio ICE 2015-02-24 15:08:55.242147 [NOTICE] switch_rtp.c:4012 Activating RTP audio ICE: FdnOx4U9rnHRTmEB:9QvMJrEjEGezGFdk 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57539 2015-02-24 15:08:55.242147 [INFO] switch_core_media.c:5364 Activating RTCP PORT 57539 2015-02-24 15:08:55.242147 [INFO] switch_core_media.c:5372 Skipping RTCP ICE (Same as RTP) 2015-02-24 15:08:55.242147 [INFO] switch_rtp.c:3104 Activate RTP/RTCP audio DTLS server 2015-02-24 15:08:55.242147 [NOTICE] switch_core_media.c:4411 sofia/internal/7um30ccl(a)su3passsu9im.invalid Starting Video thread 2015-02-24 15:08:55.242147 [INFO] switch_core_media.c:5691 Activating Video ICE 2015-02-24 15:08:55.242147 [NOTICE] switch_rtp.c:4012 Activating RTP video ICE: HafKsDarj6JzQCem:D5srzbzvDWkbATuh 2001::9d38:6abd:1cdf:dd44:2fab:aebf:57542 2015-02-24 15:08:55.242147 [INFO] switch_core_media.c:5729 Activating VIDEO RTCP PORT 57542 mux 1 2015-02-24 15:08:55.262155 [INFO] switch_core_media.c:5739 Skipping VIDEO RTCP ICE (Same as VIDEO RTP) 2015-02-24 15:08:55.262155 [INFO] switch_rtp.c:3104 Activate RTP/RTCP video DTLS server 2015-02-24 15:08:55.262155 [NOTICE] sofia.c:7484 Channel [sofia/internal/7um30ccl(a)su3passsu9im.invalid] has been answered 2015-02-24 15:08:55.262155 [NOTICE] switch_ivr_originate.c:3519 Channel [sofia/internal/test001(a)cibersys.me] has been answered 2015-02-24 15:08:55.302144 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 2001:0:9d38:6abd:1cdf:dd44:2fab:aebf:57542 to 208.84.81.64:9312 2015-02-24 15:08:55.302144 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 2001:0:9d38:6abd:1cdf:dd44:2fab:aebf:57539 to 208.84.81.64:9311 2015-02-24 15:08:55.482147 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 192.168.0.22:60538 to 208.84.81.64:9293 2015-02-24 15:08:55.662144 [NOTICE] switch_rtp.c:1151 Auto Changing stun/rtp/dtls port from 192.168.0.22:60542 to 208.84.81.64:9295 2015-02-24 15:08:55.862149 [INFO] switch_rtp.c:2927 Changing audio DTLS state from HANDSHAKE to SETUP 2015-02-24 15:08:55.862149 [INFO] switch_rtp.c:2835 audio Fingerprint Verified. 2015-02-24 15:08:55.862149 [INFO] switch_rtp.c:3377 Activating Audio Secure RTP SEND 2015-02-24 15:08:55.862149 [INFO] switch_rtp.c:3355 Activating Audio Secure RTP RECV 2015-02-24 15:08:55.862149 [INFO] switch_rtp.c:2875 Changing audio DTLS state from SETUP to READY 2015-02-24 15:08:56.022146 [INFO] switch_rtp.c:2927 Changing video DTLS state from HANDSHAKE to SETUP 2015-02-24 15:08:56.022146 [INFO] switch_rtp.c:2835 video Fingerprint Verified. 2015-02-24 15:08:56.022146 [INFO] switch_rtp.c:3377 Activating Video Secure RTP SEND 2015-02-24 15:08:56.022146 [INFO] switch_rtp.c:3355 Activating Video Secure RTP RECV 2015-02-24 15:08:56.022146 [INFO] switch_rtp.c:2875 Changing video DTLS state from SETUP to READY 2015-02-24 15:08:56.502150 [ERR] switch_rtp.c:2976 video DTLS packet not written 2015-02-24 15:08:56.502150 [ERR] switch_rtp.c:2976 audio DTLS packet not written 2015-02-24 15:08:57.502148 [ERR] switch_rtp.c:2976 video DTLS packet not written 2015-02-24 15:08:57.502148 [ERR] switch_rtp.c:2976 audio DTLS packet not written 2015-02-24 15:08:59.502152 [ERR] switch_rtp.c:2976 audio DTLS packet not written 2015-02-24 15:08:59.522149 [ERR] switch_rtp.c:2976 video DTLS packet not written 2015-02-24 15:09:03.502146 [ERR] switch_rtp.c:2976 audio DTLS packet not written 2015-02-24 15:09:03.582144 [ERR] switch_rtp.c:2976 video DTLS packet not written 2015-02-24 15:09:03.882145 [NOTICE] sofia.c:952 Hangup sofia/internal/7um30ccl(a)su3passsu9im.invalid [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2015-02-24 15:09:03.902145 [NOTICE] switch_ivr_bridge.c:1608 Hangup sofia/internal/test001(a)cibersys.me [CS_EXECUTE] [NORMAL_CLEARING] 2015-02-24 15:09:03.902145 [NOTICE] switch_core_session.c:1641 Session 13 (sofia/internal/7um30ccl(a)su3passsu9im.invalid) Ended 2015-02-24 15:09:03.902145 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/7um30ccl(a)su3passsu9im.invalid [CS_DESTROY] 2015-02-24 15:09:03.902145 [NOTICE] switch_core_session.c:1641 Session 12 (sofia/internal/test001(a)cibersys.me) Ended 2015-02-24 15:09:03.902145 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/test001(a)cibersys.me [CS_DESTROY] Testing JsTry, Chrome & Firefox _LATEST_ 1.- I can call from WebRTC enpoint to WebRTC endpoint, WebRTC/Softphone(Zoiper Biz/Latest) & Softphone(Zoiper Biz/Latest)/WebRTC , no video, no audio, but calls are placed 2.- Testing the ECHO doesn't work neither I am suspecting some sort of NAT problem. Any advice? -- Víctor E. Medina M. Software [image: Zoiper Click2Dial]+58424 291 4561 <zoiper://+584242914561>[image: ve] BB #79A8AFA2 /@VMCibersys

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