July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

soundtouch with playback_terminators + inbound dtmf
by Yuriy Nasida 31 Jul '12

31 Jul '12

Hello guys, I change my voice via 'soundtouch' and I try to record this via simple 'record' command.<action application="set" data="playback_terminators=#"/> <action application="record" data="/tmp/data.wav 20 200"/> I have to stop recording after '#' but can not make this because I must recieve DTMF inband. I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf.How can I stop recording? Any ideas? Thanks.

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IMPORTANT: Last chance to book at the Hyatt
by Michael Collins 30 Jul '12

30 Jul '12

Greetings! I wanted to let you know that the Hyatt now has less than 10 rooms available for ClueCon 2012 this year! If you plan on booking please do so * immediately*. You can call or click the link on our ClueCon hotels page<http://www.cluecon.com/hotels/> . See you next week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Weekly News and Notes
by Michael Collins 30 Jul '12

30 Jul '12

Greetings from the FreeSWITCH team! Things have been busy with all of the ClueCon<http://www.cluecon.com/?cc12-0730> preparations! We hope to see you in Chicago next week. In the meantime we are still doing our weekly conference calls. This past Wednesday we enjoyed a very nice demonstration by Dave Kompel (IRC: drk__) on how to implement Fail2Ban <http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0730>-like functionality in a Windows environment. Dave's code is simple and elegant. If you missed the presentation then please download it here<http://torrents.freeswitch.org/conf_call_2012-07-25.torrent?cc12-0730>and listen to it. The source code is also available <http://wiki.freeswitch.org/files/wftb.zip?cc12-0730>. This week we will be having a community discussion on several subjects, not the least of which is the interesting news that Google uses FreeSWITCH internally<http://bit.ly/GOOGandFS> . This is the last weekly news and notes before ClueCon. Next Monday I'll be sending this out *from Chicago**!* To the scores of people who have registered and sponsored us this year I would like to give a well-deserved thanks. We are looking forward to seeing everyone next week. If you haven't yet registered <http://www.cluecon.com/register?cc12-0730> there is still time, but please hurry! The sooner we know you're coming, the better prepared we will be to take care of you when you arrive. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0730

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Re: [Freeswitch-users] H264 transcoding
by Yufei Tao 30 Jul '12

30 Jul '12

Hi Seven Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. I'd be very grateful if you could share the code and give some instructions. Thank you very much! Yufei Subject: Re: [Freeswitch-users] H264 transcodin From: Seven Du <dujinfang(a)gmail.com><mailto:dujinfang@gmail.com> Date: 28/07/12 01:45 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org> Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.… , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. 7. On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: Thanks everyone for the responses! If I understand it correctly, if I installed ffmpeg on itself separately from FS, I could write a module for FS, in which I just call the ffmpeg program by running a command line. This way would it be classified as "not combine them into a larger work", thus free from license incompatibility problem? Not sure if that'll work for real-time transcoding of x264 though? Thanks very much for you opinions! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 ForwardedMessage.eml Subject: Re: [Freeswitch-users] H264 transcoding From: Seven Du <dujinfang(a)gmail.com><mailto:dujinfang@gmail.com> Date: 28/07/12 02:10 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org> I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. On Friday, July 27, 2012 at 2:43 PM, William King wrote: libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg The only pieces now may just be the FS side of things for video. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king(a)quentustech.com<mailto:william.king@quentustech.com> On 07/26/2012 08:46 PM, Anthony Minessale wrote: you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 <curriegrad2004(a)gmail.com><mailto:curriegrad2004@gmail.com> wrote: Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice <krice(a)freeswitch.org><mailto:krice@freeswitch.org> wrote: we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum <terry(a)digital-outpost.com><mailto:terry@digital-outpost.com> wrote: Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" <yufei.tao(a)redembedded.com><mailto:yufei.tao@redembedded.com> wrote: Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ

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Re: [Freeswitch-users] CPU Usage
by Peter Olsson 30 Jul '12

30 Jul '12

First of all, make sure you are not using CentOS 6.x – since there are known (CPU) issues with CentOS 6 and FS. The original question doesn’t say much though... I guess another recommendation could be to make sure to bypass media, which will make FS use less CPU. However, it’s kind of impossible to know what to answer on a general question like that. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Mi Ke Skickat: den 30 juli 2012 17:11 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] CPU Usage try running it with -heavy-timer as recommended here: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-td544376… ----- Original Message ----- From: Andrew Paul Sent: 07/30/12 05:01 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] CPU Usage Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0. !DSPAM:5016a5e332761623080383!

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CPU Usage
by Andrew Paul 30 Jul '12

30 Jul '12

Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0.

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Re: [Freeswitch-users] CPU Usage
by Mi Ke 30 Jul '12

30 Jul '12

try running it with -heavy-timer as recommended here: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-td544376… ----- Original Message ----- From: Andrew Paul Sent: 07/30/12 05:01 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] CPU Usage Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0.

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Re: [Freeswitch-users] Setting effecting_caller_id_name
by Phil Quesinberry 30 Jul '12

30 Jul '12

If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won't work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. <extension name="fix_currently_running" continue="true"> <condition field="caller_id_name" expression="^Currently running a lookup"> <action application="set" data="effective_caller_id_name=Wireless/Unknown"/> </condition> </extension> If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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H264 transcoding
by yufei.tao 30 Jul '12

30 Jul '12

Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei

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Trouble registering Cisco 7942
by Andre Sveen 30 Jul '12

30 Jul '12

Hi. I am using firmware 8.5(4) SIP. I see the register message on the FreeSwitch machine using ngrep. Everything seems ok, but I get unauthorized. Then I thought to myseld that this might be NAT issue. So I tried setting up a Asterisk on the same subnet as the telephone is sitting on. Same result. I have taken some info out of security reasons. Theese are marked in red text. The ngrep log below is from the FreeSwitch machine. I have also read from zero to hero here. http://www.freeswitch.org/node/401 The two first link give description of how to setup a TFTP server and soforth. I already have this. Important to mention that the Cisco 7942 is a newer phone than in theese first two links so to get the relevant config look at the third link. Below is the third link from this page. Have used lots of info from this page. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configurat… Best regards Andre U public_ip_office_network:3197 -> ip_of_the_pbx_public_ip:5060 REGISTER sip:dns_name_of_pbx SIP/2.0. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed. From: <sip:3000@dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5. To: <sip:3000@dns_name_of_pbx>. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d(a)192.168.1.240. Max-Forwards: 70. Date: Wed, 16 Dec 2009 08:44:00 GMT. CSeq: 102 REGISTER. User-Agent: Cisco-CP7942G/8.5.3. Contact: <sip:3000@192.168.1.240:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-58bfea206ae0>";+u.sip!model.ccm.cisco.com="434". Supported: (null),X-cisco-xsi-7.0.1. Content-Length: 0. Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP[mac address of phone] Load=term42.default Last=cm-reset-tcp". Expires: 3600. . U ip_of_the_public_ip:5060 -> public_ip_office_network:38398 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed;received=public_ip_of_office_network;rport=38398. From: <sip:3000@dns_name_of_pbx>;tag=58bfea206ae00003efc5fd41-f09664b5. To: <sip:3000@dns_name_of_pbx>;tag=02KZmrmjpDyHD. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d(a)192.168.1.240. CSeq: 102 REGISTER. User-Agent: XXXX internal. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. WWW-Authenticate: Digest realm="centrex.iplink.no", nonce="b91fe311-c1d7-e111-bd36-001e0bc2ead2", algorithm=MD5, qop="auth". Content-Length: 0. Then it repeats itself. Config file phone SEP[macaddress].cnf.xml this is downloaded to phone via TFTP. <device> <deviceProtocol>SIP</deviceProtocol> <sshUserId>username</secret> <sshPassword>password</secret> <devicePool> <dateTimeSetting> <dateTemplate>D-M-Y</dateTemplate> <timeZone>Central Europe Standard/Daylight Time</timeZone> <ntps> <ntp> <ntpMode>Unicast</ntpMode> </ntp> </ntps> </dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>dns_name_of_pbx</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool> <sipProfile> <sipProxies> <backupProxy></backupProxy> <backupProxyPort></backupProxyPort> <emergencyProxy></emergencyProxy> <emergencyProxyPort></emergencyProxyPort> <outboundProxy></outboundProxy> <outboundProxyPort></outboundProxyPort> <registerWithProxy>true</registerWithProxy> </sipProxies> <sipCallFeatures> <cnfJoinEnabled>true</cnfJoinEnabled> <callForwardURI>x--serviceuri-cfwdall</callForwardURI> <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> <rfc2543Hold>false</rfc2543Hold> <callHoldRingback>2</callHoldRingback> <localCfwdEnable>true</localCfwdEnable> <semiAttendedTransfer>true</semiAttendedTransfer> <anonymousCallBlock>2</anonymousCallBlock> <callerIdBlocking>2</callerIdBlocking> <dndControl>0</dndControl> <remoteCcEnable>true</remoteCcEnable> </sipCallFeatures> <sipStack> <sipInviteRetx>6</sipInviteRetx> <sipRetx>10</sipRetx> <timerInviteExpires>180</timerInviteExpires> <timerRegisterExpires>3600</timerRegisterExpires> <timerRegisterDelta>5</timerRegisterDelta> <timerKeepAliveExpires>120</timerKeepAliveExpires> <timerSubscribeExpires>120</timerSubscribeExpires> <timerSubscribeDelta>5</timerSubscribeDelta> <timerT1>500</timerT1> <timerT2>4000</timerT2> <maxRedirects>70</maxRedirects> <remotePartyID>false</remotePartyID> <userInfo>None</userInfo> </sipStack> <autoAnswerTimer>1</autoAnswerTimer> <autoAnswerAltBehavior>false</autoAnswerAltBehavior> <autoAnswerOverride>true</autoAnswerOverride> <transferOnhookEnabled>false</transferOnhookEnabled> <enableVad>false</enableVad> g711a <dtmfAvtPayload>101</dtmfAvtPayload> <dtmfDbLevel>3</dtmfDbLevel> <dtmfOutofBand>avt</dtmfOutofBand> <alwaysUsePrimeLine>false</alwaysUsePrimeLine> <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> <kpml>3</kpml> <natEnabled>false</natEnabled> <natAddress>external IP address</natAddress> <phoneLabel>Bilutleie</phoneLabel> <stutterMsgWaiting>0</stutterMsgWaiting> <callStats>false</callStats> <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> <startMediaPort>16384</startMediaPort> <stopMediaPort>32766</stopMediaPort> <sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>3000</featureLabel> <name>3000</name> <displayName>3000</displayName> <contact>3000</contact> <proxy>dns_name_pbx</proxy> <port>5060</port> <autoAnswer> <autoAnswerEnabled>2</autoAnswerEnabled> </autoAnswer> <callWaiting>3</callWaiting> <authName>3000</authName> <authPassword>secret</authPassword> <sharedLine>false</sharedLine> <messageWaitingLampPolicy>1</messageWaitingLampPolicy> <messagesNumber>*97</messagesNumber> <ringSettingIdle>4</ringSettingIdle> <ringSettingActive>5</ringSettingActive> <forwardCallInfoDisplay> <callerName>true</callerName> <callerNumber>false</callerNumber> <redirectedNumber>false</redirectedNumber> <dialedNumber>true</dialedNumber> </forwardCallInfoDisplay> </line> </sipLines> <voipControlPort>5060</voipControlPort> <dscpForAudio>184</dscpForAudio> <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> <dialTemplate>dialplan.xml</dialTemplate> </sipProfile> <commonProfile> <phonePassword></phonePassword> <backgroundImageAccess>true</backgroundImageAccess> <callLogBlfEnabled>2</callLogBlfEnabled> </commonProfile> <loadInformation>SIP42.8-5-4S</loadInformation> <vendorConfig> <disableSpeaker>false</disableSpeaker> <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset> <pcPort>0</pcPort> <settingsAccess>1</settingsAccess> <garp>0</garp> <voiceVlanAccess>0</voiceVlanAccess> <videoCapability>0</videoCapability> <autoSelectLineEnable>0</autoSelectLineEnable> <webAccess>0</webAccess> <daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive> <displayOnTime>00:00</displayOnTime> <displayOnDuration>00:00</displayOnDuration> <displayIdleTimeout>00:00</displayIdleTimeout> <spanToPCPort>1</spanToPCPort> <loggingDisplay>1</loggingDisplay> <loadServer></loadServer> </vendorConfig> <dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> <dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> <dscpForCm2Dvce>96</dscpForCm2Dvce> </device>

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