July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

execute_on_fax_success/failure
by Darcy Primrose 07 Oct '13

07 Oct '13

I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. <condition field="destination_number" expression="^1(\d{10})$|(\d{10})$"> <action application="set" data="faxtempdirectory=/faxfiles/${caller_id_number}"/> <action application="system" data="mkdir ${faxtempdirectory}"/> <action application="set" data="faxDirection=outbound"/> <action application="set" data="emailAddress=$1$2@${fax2net_server}"/> <action application="set" data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> <action application="set" data="execute_on_fax_success=system /./var/mail/sentfaxes/fax2mail.sh ${faxFile} $1$2@${fax2net_server} ${fax_security_code} ${fromRc} fax2net_orig=${fax_email}= ${fax_transfer_rate}"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="${faxFile}.tif"/> <action application="hangup"/> </condition> Thanks Darcy Primrose

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Error Creating SIP UA for profile: X
by Henrik Aagaard Sørensen 04 Oct '13

04 Oct '13

I've had a server running only FreeSWITCH for about 6 months now. I have had it turned off for some while. Now, when trying to start FreeSWTICH, I get the errors: Error Creating SIP UA for profile: X where X differs between internal, external and others. When starting my CentOS 6 server I run: service iptables stop service ip6tables stop just to make sure that they aren't blocking anything. I have nothing else running on CentOS. I've tried to log everything from console, warning etc. but there's nothing indicating what the problem is. Can someone help me out?

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Cordless phones
by Jonas Gauffin 19 Sep '13

19 Sep '13

Hi, I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time). All suggestions are welcome. Regards, Jonas

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Unreliable DTMF
by Alex Lake 13 Sep '13

13 Sep '13

I've got an application that doesn't seem to work well with DTMF coming in (supposedly using rfc2833) from some phones - notably Samsung S3s. I have told my carrier to use rfc2833. I have put dtmf-type="rfc2833" into my SIP profile. I see "a=fmtp:101 0-16" in the session progress message I was wondering what this message "Activate Buggy RFC2833 Mode" means! When a normal user hits a DTMF key, we get this in the log: 2013-07-25 13:26:19.308267 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:744 2013-07-25 13:26:19.308267 [DEBUG] switch_channel.c:471 RECV DTMF 1:744 Does this all indicate that rfc2833 is working?! Anyone else had trouble with S3s?

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IM messaging on FreeSwitch
by mehroz 10 Sep '13

10 Sep '13

Hi, Anyone with an experience of SIP SIMPLE IM Messaging implemention? I dont see the stuff well documented on http://wiki.freeswitch.org/wiki/Mod_sms <http://wiki.freeswitch.org/wiki/Mod_sms> I followed simple steps to test it with reply back as in default chatplan. <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="reply" data="Hello, you said: ${_body}"/> </condition> </extension> </context> </include> I tested it with eyeBeam, Jitsi, it doesnt work, and i do not see reply back. and i see in logs: 2013-03-07 14:58:49.090936 [INFO] mod_sms.c:334 Processing text message 9999->9999 in context default Chatplan: 9999 parsing [default->demo] continue=false Chatplan: 9999(a)198.84.61.52 Regex (PASS) [demo] to(9999(a)198.84.61.52) =~ /^(.*)$/ break=on-false Chatplan: 9999(a)198.84.61.52 Action reply(Hello, you said: ${_body}) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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eavesdrop not sending video
by D D 08 Sep '13

08 Sep '13

Hi, I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. I have followed the directions described at http://wiki.freeswitch.org/wiki/Video-recording (including a rebuild after doing the git checkout of video-media-bug), and I can successfully use record_fsv to record the video to a file. Any ideas why the eavesdrop does not send the video? Thanks, David

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mod_opus codec on Windows
by Jeff Leung 05 Sep '13

05 Sep '13

As I heard on last Wednesday's conference call, mod_opus isn't "available" on Windows, basically meaning that FreeSWITCH users can't fully implement WebRTC on Windows Platforms just yet. I have taken the steps to have the codec successfully build on this platform; However it's a pain in the neck to build as the opus library generates multiple libs which includes CELT and SILK codecs (Go figure). Shoot me a message off-list if you would like to use the binary I've built for Win64 platforms. Before anybody asks for a 32-bit binary, the short answer is going to be no. I won't be providing a 32-bit binary as I intend this binary to be used in a production environment.

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Have someone zrtp with correct sas ?
by J.B. BERLIN 29 Aug '13

29 Aug '13

Dear users, I try run FS last git build from today 19. June 2011 with ZRTP. Zfone show me green but the sas are everytime different! Tested with 2x Notebooks on Fedora 14 Twinkle(via yum) & with two desktops with Win7 x64 / Zfone 0.92.build.219 and X-Lite on both sides. FS is build with libzrtp-0.81.514 zrtp.lua is enabled ! global_setvar zrtp_secure_media=true" is set Or have someone a real good step by step guide ? Many thanks and best regards Jacek

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Error message " [ERR] switch_cpp.cpp:48 Cannot queue any more events“
by tangdu 22 Aug '13

22 Aug '13

I use Mod event to fill channel cariables in my mysql table。When freeswitch runing some time，I got a lot of the following error messages: [ERR] switch_cpp.cpp:48 Cannot queue any more events。 This messages telling me that i have consumed too many events in my event consumer object. it has reached its max. Then the channel cariables can not be filled in mysql。 How can I solve this? Thanks！ -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Can't call from Firefox 22 to Freeswitch using sipml5
by Gustavo Salazar 22 Aug '13

22 Aug '13

Hi, I want to make calls using sipml5 and freeswitch. I have installed freeswitch from the git repository in an ec2 instance with elastic ip address, configured sip_profiles/internal.xml with the EIP and here the results: Using Google Chrome and the sipml5 client I register as user 1001 with password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a conference I created, and hear myself in an echo test dialing 9196. No problem with that. Using Firefox 22 and the same sipml5 client I register as user 1002 and password 1234, then dial 5000, I see the audio files are played in the freeswitch cli but I can not hear anything. I guess the problem is not in my sipml5 client neither Firefox because when I use them to register to webrtc.freeswitch.org there is not any problem, I hear the voice after dialing 5000. I think I am missing some configuration in my Freeswitch server, where I also loaded the opus module and set OPUS in the file vars.xml , in the parameters global_codec_prefs and outbound_codec_prefs. I attach the console output from Freeswitch and the console output from Firefox Here the freeswitch output http://pastebin.com/S0yJxT1J Here the Firefox console output http://pastebin.com/8kBGdY3J Thanks in advance for the help -- Gustavo Salazar

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