December 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH
by Matthew Halos 22 Dec '14

22 Dec '14

Hi Moishe, I am using SCCP. Thanks, MATTHEW HALOS ---------- Forwarded message ---------- From: Moishe Grunstein <max(a)nysolutions.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Cc: Date: Sun, 21 Dec 2014 14:47:26 +0000 Subject: Re: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Are you running sip firmware or SCCP? Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Matthew Halos Sent: Sunday, December 21, 2014 9:14 AM To: freeswitch-users Subject: [Freeswitch-users] CISCO IP PHONE FOR FREESWITCH Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks! MATTHEW HALOS _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)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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newbie question (first build): FreeSWITCH just "hangs" when I run it
by Craig Stevenson 22 Dec '14

22 Dec '14

Hoping someone can help out a newbie figure out what I'm doing wrong. This is my first attempt at building and I keep running into same issue each time. Any pointers would be greatly appreciated. My problem is that FreeSWITHC seems to just get stuck when I try to run the binary. And and if I execute with -ncwait optoin, it keeps spitting out a line saying that it is waiting for background process to be ready This is what happens when I try to run the compiled system on a home ESXi server in my internal home network. # ./freeswitch 2014-06-22 21:23:06.011792 [INFO] switch_event.c:670 Activate Eventing Engine. 2014-06-22 21:23:06.011947 [WARNING] switch_event.c:652 Create additional event dispatch thread 0 <and then just sits there.... > BUILD VERSION: - Debian-7.5.0-amd64-netinst.iso - FreeSWITCH master (git:git clone git://git.*freeswitch.org/freeswitch.git <http://freeswitch.org/freeswitch.git>)* TEST MACHINE: ESXi 5.1 VM - Debian GNU/Linus 6 (64 bit) - 1 CPU - 1 GB RAM - 16GB Memory - 1 NIC = E1000 INSTALLATION STEPS with output piped through 'tee' to capture logs: su (apt-get update 2>&1) | tee ~/apt-get-update.txt (apt-get upgrade 2>&1) | tee ~/apt-get-upgrade.txt (dpkg --get-selections 2>&1) | tee ~/dpkg-get-selections.txt (apt-get install autoconf automake devscripts gawk g++ git-core libjpeg-dev libncurses5-dev libtool make python-dev gawk pkg-config libtiff5-dev libperl-dev libgdbm-dev libdb-dev gettext libssl-dev libcurl4-openssl-dev libpcre3-dev libspeex-dev libspeexdsp-dev libsqlite3-dev libedit-dev libldns.dev libpq-dev 2>&1) | tee ~/apt-get.txt (dpkg --get-selections 2>&1) | tee ~/dpkg-get-selections-after-apt-get.txt cd /usr/src (git clone git://git.*freeswitch.org/freeswitch.git <http://freeswitch.org/freeswitch.git>* 2>&1) | tee ~/git.txt cd /usr/src/freeswitch (./bootstrap.sh 2>&1) | tee ~/bootstrap.txt # same result if I add flag -j (./configure --enable-core-pgsql-support 2>&1) | tee ~/configure.txt (make install 2>&1) | tee ~/make-install.txt (make all cd-sounds-install cd-moh-install 2>&1) | tee ~make-sounds.txt cd /usr/local/freeswitch/bin #./freeswitch

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CISCO IP PHONE FOR FREESWITCH
by Matthew Halos 21 Dec '14

21 Dec '14

Hello to All! I am newbie to FreeSWITCH. I have a Cisco IP Phone with the Model, CP-7911G. Anybody knows how to use the device with FreeSWITCH? Thanks! MATTHEW HALOS

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Job Posting
by Kyle Haefner 20 Dec '14

20 Dec '14

Hi All, I know this treads the line of spam a bit. But we are looking to hire someone who likes open source and is familiar with VoIP and SIP. Colorado State University VoIP Integrator Colorado State University (CSU) seeks applications for a VoIP Integrator who will be responsible for the integration and management of voice communications and unified messaging systems. General Duties and Responsibilities This position will be responsible for:  Managing the unified communications application, sipXecs  Performing Linux system administration on underlying servers  Working closely with Networking and Telecom staff to coordinate the transition to VoIP services  Developing technical and end user documentation  Being on call in a rotation schedule For the full position announcement, including required and preferred qualifications as well as instructions for applying, visit: http://www.acns.colostate.edu/jobs. For full consideration, complete applications must be received by 11:59pm MST, January 12, 2015. Colorado State University is an EO/EA/AA employer. Colorado State University conducts background checks on all final candidates. Thanks! Kyle -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner(a)colostate.edu

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Call Conference Time out issue
by jobin dcruz 20 Dec '14

20 Dec '14

Hi, I had set call conference hang up time,but it is working after 4 seconds as per attendees count. This is dial plan example <action application="set" data="execute_on_answer_2=sched_hangup +${time} alloted_timeout"/> Ex: i had set one min(60 sec) hangup time,but call conference ended with 64 sec

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display incoming line
by Paul Klenk 20 Dec '14

20 Dec '14

Hi, I use several incoming SIP lines connected to freeswitch. How yould I have the incoming line be displayed on the SIP-telephones (Grandstream) besides Caller-ID? e.g.: ------------------ Tom's-Line +17778889999 -----------------

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Re: [Freeswitch-users] Hoot get variables in a LUA Hook Event.
by Jaime 19 Dec '14

19 Dec '14

Hi David, Many Thanks for the tip, I¹m tested it and work fine in 1.4. Merry Christmas to you and everyone. Jaime From: David Witham <david.witham(a)netsip.com.au> Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Date: martes, 16 de diciembre de 2014, 4:43 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Hi Jamie, What we have done is add these lines in the dialplan: <action application="set" data="session_in_hangup_hook=true"/> <action application="set" data="api_hangup_hook=lua myscript.lua"/> Then in <FSROOT>/scripts/myscript.lua: cdr_uuid = session:getVariable("cdr_uuid"); This is on a FreeSWITCH 1.2 box. Hope this helps. regards, David From: freeswitch-users-bounces(a)lists.freeswitch.org on behalf of Jaime Sent: Tuesday, 16 December 2014 13:04 To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] Hoot get variables in a LUA Hook Event. Hello, I¹m trying to use the Hook event¹s with LUA script¹s when the channel Hangup a call, but I¹m unable to get the variables, for example uuid, ani or billed seconds for the call, I¹m using freeswitch 1.4.13 on Centos 6.6 and my configuration is: In /autoload_configs/lua.conf.xml [] <hook event="CHANNEL_HANGUP_COMPLETE" script="catch-event-cdr1.lua"/> In my catch-event-cdr1.lua script I wrote: ses = freeswitch.Session(); my_uuid = ses:getVariable("uuid"); freeswitch.consoleLog("notice"," uuid=("..my_uuid..")\n") But I¹m get the following error: 2014-12-15 23:51:46.100755 [ERR] switch_cpp.cpp:724 session is not initalized 2014-12-15 23:51:46.100755 [NOTICE] switch_cpp.cpp:1328 uuid=() Also, in the wiki doc I see a reference for the ³env² Object, (https://wiki.freeswitch.org/wiki/Mod_lua#Special_Case:_env_object) but if I call it as the example in the same lua script my result is: How I call the env variable: dat = env:serialize() freeswitch.consoleLog("INFO","Here's everything:\n" .. dat .. "\n²) And I got the following error: 2014-12-16 00:00:17.440782 [ERR] mod_lua.cpp:203 /usr/share/freeswitch/scripts/catch-event-reg6.lua:27: attempt to index global 'env' (a nil value) Your help will be appreciated. Jamie Jaimecm(a)gmail.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)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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Re: [Freeswitch-users] Moving Voicemails
by Kyle Haefner 19 Dec '14

19 Dec '14

Hi All, I know this treads the line of spam a bit. But we are looking to hire someone who likes open source and is familiar with VoIP and SIP. Colorado State University VoIP Integrator Colorado State University (CSU) seeks applications for a VoIP Integrator who will be responsible for the integration and management of voice communications and unified messaging systems. General Duties and Responsibilities This position will be responsible for:  Managing the unified communications application, sipXecs  Performing Linux system administration on underlying servers  Working closely with Networking and Telecom staff to coordinate the transition to VoIP services  Developing technical and end user documentation  Being on call in a rotation schedule For the full position announcement, including required and preferred qualifications as well as instructions for applying, visit: http://www.acns.colostate.edu/jobs. For full consideration, complete applications must be received by 11:59pm MST, January 12, 2015. Colorado State University is an EO/EA/AA employer. Colorado State University conducts background checks on all final candidates. Thanks! Kyle On Mon, Nov 17, 2014 at 1:35 AM, Joel White <joelewhite(a)gmail.com> wrote: > > I initially configured my FreeSWITCH with an IP address, but now for > scalability and high availability I am changing to a domain name. I > successfully changed the system over, but was wondering how do I move the > voicemails to the new domain folder under voicemail? > > The obvious answer is to move the files, I already did that and restarted > the system. The existing voicemails and greetings are not visible to > FreeSWITCH when I call voicemail. > > How do I transfer them over? > > Thank you in advance for any light you may shed on this predicament > -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: kyle.haefner(a)colostate.edu

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FAX tone is not detected - sometimes
by Ali Pey 19 Dec '14

19 Dec '14

Hello, I have this line in my dial plan: <action application="set" data="execute_on_answer=spandsp_start_fax_detect transfer 'FAX_DETECTED XML default' 10"/> However, the fax is not detected at times. There is one particular fax machine that freeswitch never detects the fax tone. When I listen to the pcap, I do hear the fax tone. Is there anything I can do to improve the fax tone detection? Thanks, Ali Pey

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Fwd: webrtc chrome->sip bad media jssip error
by rentmycoder rentmycoder 19 Dec '14

19 Dec '14

Hi, Maybe it's a know issue, please direct me to the right direction. Setup: FS 1.4.latest, -nonat, Debian64 7.8, http://tryit.jssip.net/ web sip client, Issue: Using Firefox, calls are working great, but with Chrome 39.0.2171.95m calling from browser to SIPendpint fails. The sip client rings, but after answer, the jssip client shows bad media description error.... Ring SDP: v=0 o=FreeSWITCH 1418885182 1418885183 IN IP4 192.168.101.43 s=FreeSWITCH c=IN IP4 192.168.101.43 t=0 0 a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ m=audio 27200 RTP/SAVPF 0 126 106 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 a=rtcp-mux a=rtcp:27200 IN IP4 192.168.101.43 a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 a=ice-ufrag:lO4wBcSvCchx5t6B a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT a=candidate:9761529334 1 udp 659136 192.168.101.43 27200 typ host generation 0 2014-12-18 09:19:46.047277 [DEBUG] sofia.c:6624 Remote SDP: v=0 o=1001 8000 8000 IN IP4 192.168.101.39 s=SIP Call c=IN IP4 192.168.101.39 t=0 0 m=audio 5004 RTP/AVP 0 13 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 Remote SDP: 2014-12-18 09:19:46.067273 [DEBUG] mod_sofia.c:780 Local SDP sofia/internal/ 1002(a)192.168.101.43: v=0 o=FreeSWITCH 1418885182 1418885184 IN IP4 192.168.101.43 s=FreeSWITCH c=IN IP4 192.168.101.43 t=0 0 a=msid-semantic: WMS PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ m=audio 27200 RTP/SAVPF 0 126 106 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=sendrecv a=fingerprint:sha-256 E2:56:BC:E3:1E:CA:AD:55:04:E5:94:8F:D6:AD:1B:CA:E0:B8:90:A9:62:35:95:F0:F1:C0:D7:A7:92:35:D9:C8 a=rtcp-mux a=rtcp:27200 IN IP4 192.168.101.43 a=ssrc:1443731118 cname:FQsNBmyVzeI5MT0k a=ssrc:1443731118 msid:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a0 a=ssrc:1443731118 mslabel:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJ a=ssrc:1443731118 label:PLEThV3VtmQWGIk4ICYTkWiv3AtV43mJa0 a=ice-ufrag:lO4wBcSvCchx5t6B a=ice-pwd:9peBBCn0GWWiFMtRrndSqVeT a=candidate:7802155548 1 udp 659136 192.168.101.43 27200 typ host generation 0

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