February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

2708 anno create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
by Oguzhan Tuncer 06 Feb '15

06 Feb '15

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 06 Feb '15

06 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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Play recording
by Lucky 06 Feb '15

06 Feb '15

in cdr tool not appears the possibility to listen the recordings, (v3.6.3 v3.7.1) i have read this http://wiki.fusionpbx.com/index.php?title=Call_Detail_Records but not solved.. can you help me? thanks

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CDR using lua script
by John Nash 06 Feb '15

06 Feb '15

I am trying to insert CDR using a lua script. <action application="export" data="nolocal:api_hangup_hook=luarun cdr2.lua ${start_stamp} ${created_time} ${progress_time} ${progress_media_time} ${answer_stamp} ${end_stamp} ${duration} ${billsec}....." But when this executes many of the parameters passed to script as nil like billsec , How can I get all parameters related to CDR (Like other CDR modules)

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487 originator cancelled when using loopback channel in multiple endpoint bridge
by Emrah 05 Feb '15

05 Feb '15

Hi, I am trying to accomplish the following and cannot get it to work. On incoming calls, start playing an audio file and execute a bridge at the 7th second of the file without stopping playback. I tried a bunch of combination, but this one is the one I feel the most certain about. The problem is that even though the scenario works, overtime I pick up the channel is hung up with a 487 originator cancelled. Not on the loopback channel, but on the one I picked up. Here is my dialplan: <condition field="destination_number" expression="^.*$"> <action application="set" data="ringback=file_string:///usr/local/freeswitch/sounds/cl-welcome-moh.wav" /> <action application="set" data="transfer_ringback=file_string:///usr/local/freeswitch/sounds/cl-welcome-moh.wav" /> <action application="export" data="rtp_secure_media=true"/> <action application="export" data="absolute_codec_string=PCMU,PCMA" /> <action application="ring_ready"/> <action application="sleep" data="1500" /> <action application="answer" /> <action application="sleep" data="200" /> <action application="set" data="ignore_early_media=true" /> <action application="bridge" data="{originate_delay_start=7000}user/emrah@domain.com:_:loopback/wait/local/xml" /> </condition> I'm running the latest v1.4 stable. Any idea would be much appreciated. Thanks

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conference factory URI
by Sergey Safarov 05 Feb '15

05 Feb '15

I trying configure extension that is generating unique conference URI for different users as described in section 5.4 of RFC4579 <https://tools.ietf.org/html/rfc4579#section-5.4>. This extension will be bonded to "create new conference" button on a SIP phone. Conferences created from different phones will not conflict each other I do not know how do it. Please help me.

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OT: SIP Analysis at Linux CLI
by Michael Collins 05 Feb '15

05 Feb '15

Hey Folks, I'm curious to know tools you guys use at the Linux command line. I'm finding that I have many pcap files with SIP/RTP in them and slurping them over to a laptop w/ Wireshark is frequently cumbersome. My ideal tool here is something that prints out a ladder diagram or even a quick and dirty report on the SIP dialog(s). I would have to say that the closest thing I've found so far is Alexandr Dubovikov's awesome Sipgrep2. I'd like to hear about some of your favorite tools and tricks. -MC

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DTMF relay over bridge and the delay
by Ali Pey 05 Feb '15

05 Feb '15

Hello, Caller A calls in through a SIP trunk and is bridge to Destination B on another SIP Trunk. When A dials a DTMF digits, there is a delay of ~1.5 seconds before it's played on Leg B. When B Leg dials a DTMF digit, there is only a short delay of ~0.3 before it's relayed on Leg A. Is there anyway to reduce this delay or remove it? It causes problem with Auto Attendant menus on Leg B. Any help is very much appreciated. Thanks, Ali Pey

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Using Freeswitch with TLS and without TLS together
by Yücel ALTUNAY 05 Feb '15

05 Feb '15

Hi, I want to use freeswitch with TLS on my mobile phones and without TLS on my GSM gateways together. I see some examples but i coudn't do it. Is someone has any example to do this? Thank you.

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freeswitch+ovz - poor sound, terrible timings
by Денис 05 Feb '15

05 Feb '15

timer_test 20 200 ... 2015-01-29 15:26:32.717637 [CONSOLE] mod_commands.c:846 Timer Test: 96 sleep 20 19975 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 97 sleep 20 35287 2015-01-29 15:26:32.738371 [CONSOLE] mod_commands.c:846 Timer Test: 98 sleep 20 4676 2015-01-29 15:26:32.777674 [CONSOLE] mod_commands.c:846 Timer Test: 99 sleep 20 20051 timer_test 10 200 ... 2015-01-30 20:17:50.717723 [CONSOLE] mod_commands.c:846 Timer Test: 184 sleep 10 10084 2015-01-30 20:17:50.754812 [CONSOLE] mod_commands.c:846 Timer Test: 185 sleep 10 37157 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 186 sleep 10 6206 2015-01-30 20:17:50.760749 [CONSOLE] mod_commands.c:846 Timer Test: 187 sleep 10 9680 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 188 sleep 10 6890 2015-01-30 20:17:50.777621 [CONSOLE] mod_commands.c:846 Timer Test: 189 sleep 10 10005 On host node - all ok. Timers - with posix, soft, timerfd... What can i do?

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