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Freeswitch Week in Review(Master Branch) September 21th- 28th
by Kathleen King 30 Sep '14

30 Sep '14

Hello, again. This week in the FreeSWITCH master branch we had 28 commits. It was a quiet week with mostly miscellaneous work and a few bugs fixed. Some of the stability use cases that were fixed this week include: a leak in lua when a script doesn't execute properly, a possible buffer overrun in websocket URI, a crash caused by trying to duplicate a NULL channel variable in the core, and memory corruption issue and possible leaks when using mod_translate. The following bugs were squashed: 0cc7bc8 Add missing CURLOPT_NOSIGNAL options in mod_snom and mod_v8 7d7223e FS-6842 Added send-uncompressed-header param- set to true for logstash support in mod_graylog2 Jira: https://jira.freeswitch.org/browse/FS-6842 13d9e99 FS-6850 Handle gcc quirk with parameter checking 139b032 FS-4762 Improve regular expression to parse Jerusalem timezone files In terms of stability these were the use cases that were fixed: 1bb0b8e Fix leak in lua when script does not execute properly in xml_binding handler in mod_lua 59e7134 Fix possible buffer overrun in websocket uri and sync the ws.c between sofia and verto 9e72c84 Fix possible buffer overrun in websocket uri and sync the ws.c between sofia and verto (missing code from last commit) c018c28 FS-6851 Fix for a crashed caused by trying to duplicate NULL channel variables in the core. Jira: https://jira.freeswitch.org/browse/FS-6851 f7de058 FS-6854 Fix bug where websockets would stop processing new calls in libsofia and mod_verto Jira: https://jira.freeswitch.org/browse/FS-6854 0767191 FS-6803 Fix for a memory corruption issue and possible leaks when using mod_translate Jira: https://jira.freeswitch.org/browse/FS-6803 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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time_test and timer_test
by karl 30 Sep '14

30 Sep '14

Hi there just wonder when using time_test and timer_test what does the output tell me. Can i see how much jitter i have and in relation to what? Can someone explain this to me, please. would be really nice to understand the output. KR Karl -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/time-test-and-timer-test-tp75… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Docs and BugHunt Conf Call Time Adjustment
by Ken Rice 30 Sep '14

30 Sep '14

Hey guys quick note on this weeks conference calls! The Docs and BugHunts will be at 11AM CDT (that¹s 1600 GMT) Join us on https://conference.freeswitch.org/ and click the Call 888 button! This week on the Docs Call, Book and Docs Review/Update Today Tuesday Sep 30th! The BugHunt on Thursday will be at 11AM CDT/1600GMT also! Join us then! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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RTP flows and device doing a local redirection
by Florent Krieg 30 Sep '14

30 Sep '14

Hi mates, We are encountering a problem with some of our customers in the following situation: * we are running Freeswitch with a public IP address, detecting far-end nat traversal when needed (customers behind NAT) * our customers interconnects their IPBXes to our FS platform (we do only SIP trunking with FS) * if the customer has its IPBX behind a NAT and sets a local redirect, there's no sound (and I completely understand why, I'm just trying to find a solution, if any...) So there are two distinct calls: PSTN_X <===> FS <===> customer1 IPBX customer1 IPBX <===> FS <===> PSTN_Y (redirection) When PSTN_Y answers the call, FS sends 200 OK w/ SDP to customer1 IPBX. The IPBX then sends an empty RTP packet, I reckon to allow the remote platform (FS in this case) to be able to detect the IP/port couple where to send RTP packets. The problem I have is that on the 'caller' side, this single empty RTP packet is sent by the IPBX before sending the 200 OK (like, a few ms before) and then FS seems to not be ready for it and reply ICMP unreachable. To make this whole stuff almost-work, I've set 'rtp_auto_adjust_threshold' to 1, otherwise FS will wait for far more than 1 RTP packet to detect NAT. My idea is that the IPBX should send the 200 OK first and right after this empty RTP packet. The problem is I'm encountering the issue with two different constructors (on one of them we ended using STUN, but the other one has a terrible STUN implementation, hence I'm really looking for a solution). I know it's tricky (SIP, NAT, redirection and so on) but what do you think about this use case? Would you know a way to solve this trouble? If you have any clue please share! Thanks in advance Florent

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Overriding domain variables
by Iskren Hadzhinedev 30 Sep '14

30 Sep '14

Howdy all, Is it possible to override a variable set in the domain by setting it again further down in the XML directory? I have the following scenario: <domain name=”sip.test.com”> <variables> <variable name=”call_timeout” value=”30”> </variables> ... <user id=”iskren”> <variables> <variable name=”call_timeout” value=”15”/> </variable> </user> .... </domain> However, calling that user still times out after 30 seconds. Can this be done in the directory, or I must set it in the dialplan? Thanks! Kind regards, -- *Iskren Hadzhinedev*

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Re: [Freeswitch-users] Problem in getting digits value in Playback application
by Manish Talwar 29 Sep '14

29 Sep '14

Hi, Thanks for your response, I will try to use "bind_digit_action" soon. Meanwhile, I am trying to install FreeSwitch setup on our QA Server now. I have followed all neccessary steps for installing it on Server "Ubuntu 10.04.4 LTS (AWS EC2)" . But after installing, while running a command "load mod_portaudio" on "fs_cli" its throwing an error as: 2014-09-24 13:12:11.771250 [ERR] mod_portaudio.c:980 Cannot find an input device 2014-09-24 13:12:11.771250 [ERR] mod_portaudio.c:990 Cannot find an input device 2014-09-24 13:12:11.771250 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** After that, I have tried to install "make cd-sounds-install" and "make cd-moh-install" on server but its still throwing same error after installing it. Please let me know, what step's I need to follow or any Server configuration required for resolving this error on Server. Thanks, Regards, Manish Talwar ________________________________ From: Michael Collins <msc(a)freeswitch.org> Sent: 23 September 2014 02:36 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Problem in getting digits value in Playback application If you want to react immediately when the caller presses a single digit then you should probably be investigating bind_digit_action: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bind_digit_action -MC On Sun, Sep 21, 2014 at 11:53 PM, Manish Talwar <manish.talwar(a)zaristo.com<mailto:manish.talwar@zaristo.com>> wrote: Hi, I am able to get the channel variable value in a request now by "play_and_get_digits" command. But now, I need some suggestion for below query: As per my requirement, I want to play multiple audio file by "playback", run TTS by "Speak", and get digit by "Play_and_get_digits" in a single dialplan as a batch. I am sending below request to freeswitch for running multiple commands as a batch in a request: <document type="xml/freeswitch-httapi"><params></params><variables><myDigit>-1</myDigit></variables><work><getVariable name="myDigit" permanent="1"/><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><speak name="extenSpeak" text="Welcome to Q Pay" engine="flite" voice="slt" ></speak><execute action="http://localhost:9000/ivr/fswelcomeresponse" application="play_and_get_digits" data="0 1 1 2000 # ivr/MP3/dial_menu_index_five.mp3 '' myDigit ''" /><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><playback name="exten" file="ivr/MP3/dial_menu_index_one.mp3" action="http://localhost:9000/ivr/fswelcomeresponse" digit-timeout="1000" input-timeout="1000" ><bind>~\d{1,3}#{0,1}</bind></playback><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><speak name="extenSpeak1" text="Press two for english." engine="flite" voice="slt" ></speak><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><playback name="exten" file="ivr/MP3/dial_menu_index_four.mp3" action="http://localhost:9000/ivr/fswelcomeresponse" digit-timeout="1000" input-timeout="1000" ><bind>~\d{1,3}#{0,1}</bind></playback><execute action="http://localhost:9000/ivr/fswelcomeresponse" application="play_and_get_digits" data="0 1 1 2000 # ivr/MP3/dial_menu_index_five.mp3 '' myDigit ''" /><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><execute application="log" data="INFO my_chan_var contains ${myDigit}"/><speak name="extenSpeak2" text="Press star to repeat." engine="flite" voice="slt" ></speak></work></document> Currently, Its playing all commands as mentioned in dialplan and then move to Action URL after completing all command in a request. But I need to send my request to Action URL immidieatly when user press any digit. I have studied something "break [UDDI]" for skipping current batch request and move to other request in httapi application but I am not getting more details about it. Please let me know, how to I implement my requirement and send my request to Action URL immidieatly when user press any digit? Thanks, Regards, Manish Talwar ________________________________ From: Manish Talwar Sent: 18 September 2014 13:44 To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] Problem in getting digits value in Playback application Hi, As you suggested, "Play_and_get_digits" command is running fine now, But I am still not able to get the value of Channel variable "myChannelVar" in a call as Its returning me blank value. I am sending request as: <document type="xml/freeswitch-httapi"> <params></params> <work> <getVariable name="myChannelVar" permanent="1"/> <execute application="play_and_get_digits" data="1 4 1 5000 # ivr/MP3/card_activate_birth_date.mp3 '' myChannelVar \d+" /> <execute application="log" data="INFO my_chan_var contains ${myChannelVar}"/> <execute application="hangup"/> </work> </document> As a output, its play a audio file and I pressed a key 2222 digits in a call but its returning me blank value in channel variable "myChannelVar". I got a response in a console log as: 2014-09-18 13:28:49.739010 [DEBUG] mod_httapi.c:1217 Process Tag: [execute] EXECUTE portaudio/5000 log(INFO my_chan_var contains ) 2014-09-18 13:28:49.739010 [INFO] mod_dptools.c:1628 my_chan_var contains I have set the permission in my HTTAPI config file to "set-vars", "get-vars" & "expand-vars" tags and its seem like working fine. For reference, please find my HTTAPI config file as an attachment. Please suggest me how do I got a channel variable value and pass it to other POST request. Thanks, Regards, Manish Talwar ________________________________ From: Manish Talwar <manish.talwar(a)zaristo.com<mailto:manish.talwar@zaristo.com>> Sent: 18 September 2014 11:54 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Problem in getting digits value in Playback application Its working fine now, Thanks a lot. Regards, Manish Talwar ________________________________ From: Michael Collins <msc(a)freeswitch.org<mailto:msc@freeswitch.org>> Sent: 18 September 2014 01:00 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Problem in getting digits value in Playback application I think what you did was set the default to allow and everything you listed underneath became denied. Try setting the default back to deny and see if it works as expected. -MC On Wed, Sep 17, 2014 at 7:28 AM, Manish Talwar <manish.talwar(a)zaristo.com<mailto:manish.talwar@zaristo.com>> wrote: Hi, Please revert me about Permission setting ASAP. Regards, Manish Talwar ________________________________ From: Manish Talwar Sent: 17 September 2014 10:59 To: FreeSWITCH Users Help Subject: RE: [Freeswitch-users] Problem in getting digits value in Playback application Hi, I have tried by giving below Permission with my HTTAPI config file. <permission name="execute-apps" value="true">  <application-list default="allow">  <application name="info"/> <application name="play_and_get_digits"/> <application name="hangup"/> </application-list> </permission> and from Web Application, I am sending request as: <document type="xml/freeswitch-httapi"> <params></params> <work> <execute application="play_and_get_digits" data="1 4 1 5000 # /ivr/8000/ivr-enter_number_to_remove_from_blacklist.wav '' foobar \d+" /> </work> </document> but still its giving me error as "[ERR] mod_httapi.c:840 Permission Denied!" However, its work fine with hangup application: <document type="xml/freeswitch-httapi"> <params></params> <work> <execute application="hangup"/> </work> </document> I feel, it must be some setting problem in config file only. Please find my HTTAPI config file as an attachment and let me know if any changes required in this file. Thanks, Regards, Manish Talwar ________________________________ From: Michael Collins <msc(a)freeswitch.org<mailto:msc@freeswitch.org>> Sent: 16 September 2014 23:09 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Problem in getting digits value in Playback application On Tue, Sep 16, 2014 at 9:46 AM, Manish Talwar <manish.talwar(a)zaristo.com<mailto:manish.talwar@zaristo.com>> wrote: Hi, Thanks for quick response, I have already try to use this "play_and_get_digits" command. Its working fine when I use this as a dialplan <extension> tag in FreeSwitch file but while using it in Web Application with "<execute>" httapi command its throwing error as "[ERR] mod_httapi.c:840 Permission Denied!" I have tried to reset Permission of "execute-apps" in httapi-conf.xml file, but still got same error. <http://www.freeswitchsolutions.com> This is almost certainly a permissions issue with your HTTAPI config file. What do you have in your "execute-apps" section? The default is to deny and then explicitly allow only the items you specify. FYI, this is covered quite thoroughly in the FreeSWITCH 1.2 book on pages 277-280. -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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FreeSWITCH 1.4.9 Released!
by Ken Rice 29 Sep '14

29 Sep '14

Hey FreeSWITCHers! The FreeSWITCH Dev Team is happy to announce FreeSWITCH 1.4.9 has been released. This release is a bug fix release. Source Tarball is available at http://files.freeswitch.org/ and packages are available via the Yum and Debian repos. K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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API Execute uuid_transfer
by andretodd＠verizon.net 29 Sep '14

29 Sep '14

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Monstrous Sound Delay, WebRTC, sip-0.6.2, mod_verto
by Алексей kob.kz Целищев 29 Sep '14

29 Sep '14

Hello. I work with browser with WebRTC with FreeSWITCH Version 1.5.14b+git~20140730T044231Z~b1f382c67e~64bit (git b1f382c 2014-07-30 04:42:31Z 64bit). Since answer call until the sound is appeared (ICE connection state changed to "connected" in the log) takes ~ 1216ms, which is monstrous for my problems. Have any ideas on this? STUN, as you can see, is disabled. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | new listener added to event connecting sip-0.6.2.js:2655 >>> ||| <<< since answer call : 0 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMediaRequest sip-0.6.2.js:2655 >>> ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMediaRequest sip-0.6.2.js:2655 >>> ||| <<< since answer call : 41 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.eventemitter | emitting event userMedia sip-0.6.2.js:2655 >>> ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event userMedia sip-0.6.2.js:2655 >>> ||| <<< since answer call : 43 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | acquired local media stream sip-0.6.2.js:2655 >>> ||| <<< since answer call : 44 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event connecting sip-0.6.2.js:2655 >>> ||| <<< since answer call : 44 ms. Ожидание соединения: 47 ms. WebRTCCallCenter.js?r=1:212 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "checking" sip-0.6.2.js:2655 >>> ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | ICE Gathering Completed sip-0.6.2.js:2655 >>> ||| <<< since answer call : 47 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event iceComplete sip-0.6.2.js:2655 >>> ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.invitecontext.mediahandler | emitting event getDescription sip-0.6.2.js:2655 >>> ||| <<< since answer call : 48 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.transaction.ist | emitting event stateChanged sip-0.6.2.js:2655 >>> ||| <<< since answer call : 50 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | emitting event accepted sip-0.6.2.js:2655 >>> ||| <<< since answer call : 51 ms. >>>>> Звонок начат Operator.js?r=3:1903 Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 >>> ||| <<< since answer call : 60 ms. Wed Sep 17 2014 16:31:21 GMT+0600 | sip.inviteservercontext | no ACK received, attempting to retransmit OK sip-0.6.2.js:2655 >>> ||| <<< since answer call : 80 ms. Wed Sep 17 2014 16:31:22 GMT+0600 | sip.invitecontext.mediahandler | ICE connection state changed to "connected" sip-0.6.2.js:2655 >>> ||| <<< since answer call : 1216 ms. Nuance: if the operator and the server are in the local network and connected to an internal address, then until the sound appeared goes ~ 280ms, which is a stretch of more or less acceptable. But if I connect to the browser through external network (router with port forwarding), then the delay increases by 5 times. In that demo with Verto https://webrtc.freeswitch.org/verto/index.html >>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:14 GMT+0600 ICE connection state changed to "checking" verto-min.js:41 >>> ||| <<< since answer call: 476 ms. verto-min.js:42 Dialog c46b6eb8-73de-62ce-df07-fb3d002e07eb: state change from trying to active verto-min.js:206 >>>>>>>>>>>>>>>>Thu Sep 25 2014 16:13:15 GMT+0600 ICE connection state changed to "connected" verto-min.js:41 >>> ||| <<< since answer call: 2014 ms. How do I configure Freeswitch and javascript app to sound appeared within 200 ms?

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tel: uri freeswitch
by joe 29 Sep '14

29 Sep '14

2014-09-27 05:52:40.377783 [ERR] sofia_glue.c:983 URL Error! tel: uri's not supported at this time i am not able to set tel uri's in sip_invite_req_uri= during originate pstn outbound call from freeswitch. awaiting positive answer asap. tx

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